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authorLinus Torvalds <torvalds@linux-foundation.org>2014-12-11 13:20:50 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2014-12-11 13:20:50 -0800
commitbae41e45b7400496b9bf0c70c6004419d9987819 (patch)
treecf22a65d119da1c414dbc79518857800fbe7a24b /sound/soc
parent7ef58b32f571bffb7763c6252ad7527562081f34 (diff)
parent6e1d7a51392f06899bd7b693f28ac60fa1e00032 (diff)
downloadkernel-bae41e45b7400496b9bf0c70c6004419d9987819.tar.gz
Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This became a fairly large pull request. In addition to the usual driver updates / fixes, there have been a high amount of cleanups in ASoC area, as well as control API helpers and kernel documentations fixes touching through the whole tree. In the driver side, the biggest changes are the support for new Intel SoC found on new x86 machines, and the updates of FireWire dice and oxfw drivers. Some remarkable items are below: ALSA core: - PCM mmap code cleanup, removal of arch-dependent codes - PCM xrun injection support - PCM hwptr tracepoint support - Refactoring of snd_pcm_action(), simplification of PCM locking - Robustified sequecner auto-load functionality - New control API helpers and lots of cleanups along with them - Lots of kerneldoc fixes and cleanups USB-audio: - The mixer resume code was largely rewritten, and the devices with quirks are resumed properly. - New hardware support: Focusrite Scarlett, Digidesign Mbox1, Denon/Marantz DACs, Zoom R16/24 FireWire: - DICE driver updates with better duplex and sync support, including MIDI support - New OXFW driver for Oxford Semiconductor FW970/971 chipset, including the previous LaCie Speakers device. Fullduplex and MIDI support included as well as DICE driver. HD-audio: - Refactoring the driver-caps quirk handling in snd-hda-intel - More consistent control names representing the topology better - Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic fix, ASUS Z99He laptop EAPD ASoC: - Conversion of AC'97 drivers to use regmap, bringing us closer to the removal of the ASoC level I/O code - Clean up a lot of old drivers that were open coding things that have subsequently been implemented in the core - Some DAPM performance improvements - Removal of the now seldom used CODEC mutex - Lots of updates for the newer Intel SoC support, including support for the DSP and some Cherrytrail and Braswell machine drivers - Support for Samsung boards using rt5631 as the CODEC - Removal of the obsolete AFEB9260 machine driver - Driver support for the TI TS3A227E headset driver used in some Chrombeooks Others: - ASIHPI driver update and cleanups - Lots of dev_*() printk conversions - Lots of trivial cleanups for the codes spotted by Coccinelle" * tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits) ALSA: pcxhr: NULL dereference on probe failure ALSA: lola: NULL dereference on probe failure ALSA: hda - Add "eapd" model string for AD1986A codec ALSA: hda - Add EAPD fixup for ASUS Z99He laptop ALSA: oxfw: Add hwdep interface ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to name card ALSA: dice: Add support for MIDI capture/playback ALSA: dice: Add support for capturing PCM samples ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Change the way to start stream ALSA: jack: Add dummy snd_jack_set_key() definition ...
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/Makefile6
-rw-r--r--sound/soc/atmel/Kconfig9
-rw-r--r--sound/soc/atmel/Makefile1
-rw-r--r--sound/soc/atmel/atmel-pcm-dma.c4
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c5
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c151
-rw-r--r--sound/soc/au1x/ac97c.c2
-rw-r--r--sound/soc/au1x/psc-ac97.c2
-rw-r--r--sound/soc/blackfin/bf5xx-ac97.c2
-rw-r--r--sound/soc/blackfin/bf5xx-ad1980.c2
-rw-r--r--sound/soc/cirrus/Kconfig3
-rw-r--r--sound/soc/cirrus/ep93xx-ac97.c2
-rw-r--r--sound/soc/codecs/Kconfig47
-rw-r--r--sound/soc/codecs/Makefile10
-rw-r--r--sound/soc/codecs/ab8500-codec.c32
-rw-r--r--sound/soc/codecs/ac97.c18
-rw-r--r--sound/soc/codecs/ad193x.c14
-rw-r--r--sound/soc/codecs/ad1980.c212
-rw-r--r--sound/soc/codecs/ad1980.h26
-rw-r--r--sound/soc/codecs/adau1373.c6
-rw-r--r--sound/soc/codecs/adau1701.c86
-rw-r--r--sound/soc/codecs/adau1761.c25
-rw-r--r--sound/soc/codecs/adau1781.c33
-rw-r--r--sound/soc/codecs/adau17x1.c71
-rw-r--r--sound/soc/codecs/adau17x1.h10
-rw-r--r--sound/soc/codecs/adav80x.c4
-rw-r--r--sound/soc/codecs/ak4535.c31
-rw-r--r--sound/soc/codecs/ak4641.c33
-rw-r--r--sound/soc/codecs/ak4642.c16
-rw-r--r--sound/soc/codecs/ak4671.c13
-rw-r--r--sound/soc/codecs/alc5623.c22
-rw-r--r--sound/soc/codecs/alc5632.c22
-rw-r--r--sound/soc/codecs/arizona.c34
-rw-r--r--sound/soc/codecs/cq93vc.c33
-rw-r--r--sound/soc/codecs/cs4265.c2
-rw-r--r--sound/soc/codecs/cs4271-i2c.c62
-rw-r--r--sound/soc/codecs/cs4271-spi.c55
-rw-r--r--sound/soc/codecs/cs4271.c155
-rw-r--r--sound/soc/codecs/cs4271.h11
-rw-r--r--sound/soc/codecs/cs42l51.c6
-rw-r--r--sound/soc/codecs/cs42l73.c6
-rw-r--r--sound/soc/codecs/hdmi.c2
-rw-r--r--sound/soc/codecs/lm49453.c8
-rw-r--r--sound/soc/codecs/max98088.c31
-rw-r--r--sound/soc/codecs/max98090.c201
-rw-r--r--sound/soc/codecs/max98090.h8
-rw-r--r--sound/soc/codecs/max98095.c23
-rw-r--r--sound/soc/codecs/max9850.c22
-rw-r--r--sound/soc/codecs/rt286.c231
-rw-r--r--sound/soc/codecs/rt5631.c38
-rw-r--r--sound/soc/codecs/rt5645.c200
-rw-r--r--sound/soc/codecs/rt5645.h19
-rw-r--r--sound/soc/codecs/rt5670.c136
-rw-r--r--sound/soc/codecs/rt5670.h6
-rw-r--r--sound/soc/codecs/rt5677-spi.c130
-rw-r--r--sound/soc/codecs/rt5677-spi.h21
-rw-r--r--sound/soc/codecs/rt5677.c1198
-rw-r--r--sound/soc/codecs/rt5677.h162
-rw-r--r--sound/soc/codecs/sgtl5000.c111
-rw-r--r--sound/soc/codecs/sigmadsp-i2c.c81
-rw-r--r--sound/soc/codecs/sigmadsp-regmap.c46
-rw-r--r--sound/soc/codecs/sigmadsp.c711
-rw-r--r--sound/soc/codecs/sigmadsp.h59
-rw-r--r--sound/soc/codecs/sirf-audio-codec.c6
-rw-r--r--sound/soc/codecs/sn95031.c15
-rw-r--r--sound/soc/codecs/ssm4567.c128
-rw-r--r--sound/soc/codecs/sta32x.c21
-rw-r--r--sound/soc/codecs/sta350.c21
-rw-r--r--sound/soc/codecs/sta529.c35
-rw-r--r--sound/soc/codecs/stac9766.c60
-rw-r--r--sound/soc/codecs/tas2552.c10
-rw-r--r--sound/soc/codecs/tfa9879.c328
-rw-r--r--sound/soc/codecs/tfa9879.h202
-rw-r--r--sound/soc/codecs/tlv320aic23.c21
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c31
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c24
-rw-r--r--sound/soc/codecs/tlv320aic3x.c228
-rw-r--r--sound/soc/codecs/tlv320aic3x.h1
-rw-r--r--sound/soc/codecs/tlv320dac33.c2
-rw-r--r--sound/soc/codecs/ts3a227e.c314
-rw-r--r--sound/soc/codecs/ts3a227e.h17
-rw-r--r--sound/soc/codecs/twl4030.c2
-rw-r--r--sound/soc/codecs/twl6040.c23
-rw-r--r--sound/soc/codecs/uda134x.c32
-rw-r--r--sound/soc/codecs/uda1380.c20
-rw-r--r--sound/soc/codecs/wl1273.c10
-rw-r--r--sound/soc/codecs/wm5102.c18
-rw-r--r--sound/soc/codecs/wm8350.c21
-rw-r--r--sound/soc/codecs/wm8400.c34
-rw-r--r--sound/soc/codecs/wm8510.c26
-rw-r--r--sound/soc/codecs/wm8523.c29
-rw-r--r--sound/soc/codecs/wm8580.c4
-rw-r--r--sound/soc/codecs/wm8711.c27
-rw-r--r--sound/soc/codecs/wm8728.c34
-rw-r--r--sound/soc/codecs/wm8731.c37
-rw-r--r--sound/soc/codecs/wm8737.c49
-rw-r--r--sound/soc/codecs/wm8750.c25
-rw-r--r--sound/soc/codecs/wm8776.c31
-rw-r--r--sound/soc/codecs/wm8804.c3
-rw-r--r--sound/soc/codecs/wm8900.c8
-rw-r--r--sound/soc/codecs/wm8903.c43
-rw-r--r--sound/soc/codecs/wm8940.c22
-rw-r--r--sound/soc/codecs/wm8955.c33
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c12
-rw-r--r--sound/soc/codecs/wm8960.c113
-rw-r--r--sound/soc/codecs/wm8961.c34
-rw-r--r--sound/soc/codecs/wm8962.c11
-rw-r--r--sound/soc/codecs/wm8974.c25
-rw-r--r--sound/soc/codecs/wm8978.c10
-rw-r--r--sound/soc/codecs/wm8983.c27
-rw-r--r--sound/soc/codecs/wm8985.c28
-rw-r--r--sound/soc/codecs/wm8988.c27
-rw-r--r--sound/soc/codecs/wm8990.c24
-rw-r--r--sound/soc/codecs/wm8991.c32
-rw-r--r--sound/soc/codecs/wm8993.c12
-rw-r--r--sound/soc/codecs/wm8994.c4
-rw-r--r--sound/soc/codecs/wm8994.h2
-rw-r--r--sound/soc/codecs/wm8995.c17
-rw-r--r--sound/soc/codecs/wm9081.c7
-rw-r--r--sound/soc/codecs/wm9090.c32
-rw-r--r--sound/soc/codecs/wm9705.c46
-rw-r--r--sound/soc/codecs/wm9712.c219
-rw-r--r--sound/soc/codecs/wm9713.c230
-rw-r--r--sound/soc/codecs/wm_adsp.c97
-rw-r--r--sound/soc/davinci/davinci-mcasp.c339
-rw-r--r--sound/soc/davinci/davinci-mcasp.h17
-rw-r--r--sound/soc/dwc/designware_i2s.c46
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c5
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c19
-rw-r--r--sound/soc/fsl/fsl_dma.c9
-rw-r--r--sound/soc/fsl/fsl_esai.c12
-rw-r--r--sound/soc/fsl/fsl_ssi.c17
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c6
-rw-r--r--sound/soc/fsl/imx-spdif.c3
-rw-r--r--sound/soc/fsl/imx-ssi.c2
-rw-r--r--sound/soc/fsl/imx-wm8962.c6
-rw-r--r--sound/soc/fsl/mpc5200_dma.c3
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c6
-rw-r--r--sound/soc/generic/simple-card.c160
-rw-r--r--sound/soc/intel/Kconfig44
-rw-r--r--sound/soc/intel/Makefile7
-rw-r--r--sound/soc/intel/broadwell.c64
-rw-r--r--sound/soc/intel/bytcr_dpcm_rt5640.c230
-rw-r--r--sound/soc/intel/cht_bsw_rt5672.c285
-rw-r--r--sound/soc/intel/haswell.c14
-rw-r--r--sound/soc/intel/sst-atom-controls.c1208
-rw-r--r--sound/soc/intel/sst-atom-controls.h428
-rw-r--r--sound/soc/intel/sst-baytrail-dsp.c24
-rw-r--r--sound/soc/intel/sst-dsp-priv.h136
-rw-r--r--sound/soc/intel/sst-dsp.c31
-rw-r--r--sound/soc/intel/sst-dsp.h28
-rw-r--r--sound/soc/intel/sst-firmware.c937
-rw-r--r--sound/soc/intel/sst-haswell-dsp.c295
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c413
-rw-r--r--sound/soc/intel/sst-haswell-ipc.h25
-rw-r--r--sound/soc/intel/sst-haswell-pcm.c419
-rw-r--r--sound/soc/intel/sst-mfld-platform-compress.c8
-rw-r--r--sound/soc/intel/sst-mfld-platform-pcm.c154
-rw-r--r--sound/soc/intel/sst-mfld-platform.h5
-rw-r--r--sound/soc/intel/sst/Makefile7
-rw-r--r--sound/soc/intel/sst/sst.c437
-rw-r--r--sound/soc/intel/sst/sst.h546
-rw-r--r--sound/soc/intel/sst/sst_acpi.c383
-rw-r--r--sound/soc/intel/sst/sst_drv_interface.c686
-rw-r--r--sound/soc/intel/sst/sst_ipc.c373
-rw-r--r--sound/soc/intel/sst/sst_loader.c456
-rw-r--r--sound/soc/intel/sst/sst_pci.c209
-rw-r--r--sound/soc/intel/sst/sst_pvt.c449
-rw-r--r--sound/soc/intel/sst/sst_stream.c437
-rw-r--r--sound/soc/jz4740/qi_lb60.c11
-rw-r--r--sound/soc/mxs/mxs-saif.c2
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c7
-rw-r--r--sound/soc/nuc900/nuc900-ac97.c2
-rw-r--r--sound/soc/omap/Kconfig8
-rw-r--r--sound/soc/omap/mcbsp.c3
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c7
-rw-r--r--sound/soc/pxa/pxa-ssp.c16
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c6
-rw-r--r--sound/soc/pxa/spitz.c52
-rw-r--r--sound/soc/rockchip/Kconfig9
-rw-r--r--sound/soc/samsung/Kconfig8
-rw-r--r--sound/soc/samsung/Makefile2
-rw-r--r--sound/soc/samsung/ac97.c4
-rw-r--r--sound/soc/samsung/arndale_rt5631.c150
-rw-r--r--sound/soc/samsung/i2s-regs.h10
-rw-r--r--sound/soc/samsung/i2s.c244
-rw-r--r--sound/soc/samsung/odroidx2_max98090.c4
-rw-r--r--sound/soc/sh/fsi.c9
-rw-r--r--sound/soc/sh/hac.c2
-rw-r--r--sound/soc/sh/rcar/adg.c2
-rw-r--r--sound/soc/sh/rcar/core.c236
-rw-r--r--sound/soc/sh/rcar/dvc.c215
-rw-r--r--sound/soc/sh/rcar/gen.c30
-rw-r--r--sound/soc/sh/rcar/rsnd.h92
-rw-r--r--sound/soc/sh/rcar/src.c101
-rw-r--r--sound/soc/sh/rcar/ssi.c233
-rw-r--r--sound/soc/soc-ac97.c256
-rw-r--r--sound/soc/soc-cache.c149
-rw-r--r--sound/soc/soc-compress.c11
-rw-r--r--sound/soc/soc-core.c1642
-rw-r--r--sound/soc/soc-dapm.c755
-rw-r--r--sound/soc/soc-jack.c11
-rw-r--r--sound/soc/soc-ops.c952
-rw-r--r--sound/soc/soc-pcm.c23
-rw-r--r--sound/soc/tegra/tegra20_ac97.c2
-rw-r--r--sound/soc/tegra/tegra_rt5640.c6
-rw-r--r--sound/soc/txx9/txx9aclc-ac97.c2
-rw-r--r--sound/soc/txx9/txx9aclc.c2
-rw-r--r--sound/soc/ux500/mop500.c8
209 files changed, 16830 insertions, 5574 deletions
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index d88edfced8c4..865e090c8061 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,10 +1,14 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
-snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o
+snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o
ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
snd-soc-core-objs += soc-generic-dmaengine-pcm.o
endif
+ifneq ($(CONFIG_SND_SOC_AC97_BUS),)
+snd-soc-core-objs += soc-ac97.o
+endif
+
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
obj-$(CONFIG_SND_SOC) += generic/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 27e3fc4a536b..fb3878312bf8 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -52,12 +52,3 @@ config SND_AT91_SOC_SAM9X5_WM8731
help
Say Y if you want to add support for audio SoC on an
at91sam9x5 based board that is using WM8731 codec.
-
-config SND_AT91_SOC_AFEB9260
- tristate "SoC Audio support for AFEB9260 board"
- depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
- select SND_ATMEL_SOC_PDC
- select SND_ATMEL_SOC_SSC
- select SND_SOC_TLV320AIC23_I2C
- help
- Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index 5baabc8bde3a..466a821da98c 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
-obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index b79a2a864513..33fb3bb133df 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -80,9 +80,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
/* stop RX and capture: will be enabled again at restart */
ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable);
- snd_pcm_stream_lock(substream);
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- snd_pcm_stream_unlock(substream);
+ snd_pcm_stop_xrun(substream);
/* now drain RHR and read status to remove xrun condition */
ssc_readx(prtd->ssc->regs, SSC_RHR);
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index f403f399808a..b1cc2a4a7fc0 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
* transmit and receive, so if a value has already
* been set, it must match this value.
*/
- if (ssc_p->cmr_div == 0)
+ if (ssc_p->dir_mask !=
+ (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE))
+ ssc_p->cmr_div = div;
+ else if (ssc_p->cmr_div == 0)
ssc_p->cmr_div = div;
else
if (div != ssc_p->cmr_div)
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
deleted file mode 100644
index 9579799ace54..000000000000
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- * afeb9260.c -- SoC audio for AFEB9260
- *
- * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-
-#include <linux/atmel-ssc.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <linux/gpio.h>
-
-#include "../codecs/tlv320aic23.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-#define CODEC_CLOCK 12000000
-
-static int afeb9260_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int err;
-
- /* Set the codec system clock for DAC and ADC */
- err =
- snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
-
- if (err < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return err;
- }
-
- return err;
-}
-
-static struct snd_soc_ops afeb9260_ops = {
- .hw_params = afeb9260_hw_params,
-};
-
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_LINE("Line In", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
- {"Headphone Jack", NULL, "LHPOUT"},
- {"Headphone Jack", NULL, "RHPOUT"},
-
- {"LLINEIN", NULL, "Line In"},
- {"RLINEIN", NULL, "Line In"},
-
- {"MICIN", NULL, "Mic Jack"},
-};
-
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link afeb9260_dai = {
- .name = "TLV320AIC23",
- .stream_name = "AIC23",
- .cpu_dai_name = "atmel-ssc-dai.0",
- .codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "atmel_pcm-audio",
- .codec_name = "tlv320aic23-codec.0-001a",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &afeb9260_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_machine_afeb9260 = {
- .name = "AFEB9260",
- .owner = THIS_MODULE,
- .dai_link = &afeb9260_dai,
- .num_links = 1,
-
- .dapm_widgets = tlv320aic23_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
- .dapm_routes = afeb9260_audio_map,
- .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
-};
-
-static struct platform_device *afeb9260_snd_device;
-
-static int __init afeb9260_soc_init(void)
-{
- int err;
- struct device *dev;
-
- if (!(machine_is_afeb9260()))
- return -ENODEV;
-
-
- afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
- if (!afeb9260_snd_device) {
- printk(KERN_ERR "ASoC: Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260);
- err = platform_device_add(afeb9260_snd_device);
- if (err)
- goto err1;
-
- dev = &afeb9260_snd_device->dev;
-
- return 0;
-err1:
- platform_device_put(afeb9260_snd_device);
- return err;
-}
-
-static void __exit afeb9260_soc_exit(void)
-{
- platform_device_unregister(afeb9260_snd_device);
-}
-
-module_init(afeb9260_soc_init);
-module_exit(afeb9260_soc_exit);
-
-MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
-MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
-MODULE_LICENSE("GPL");
-
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
index c8a2de103c5f..5159a50a45a6 100644
--- a/sound/soc/au1x/ac97c.c
+++ b/sound/soc/au1x/ac97c.c
@@ -205,7 +205,7 @@ static int au1xac97c_dai_probe(struct snd_soc_dai *dai)
static struct snd_soc_dai_driver au1xac97c_dai_driver = {
.name = "alchemy-ac97c",
- .ac97_control = 1,
+ .bus_control = true,
.probe = au1xac97c_dai_probe,
.playback = {
.rates = AC97_RATES,
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 84f31e1f9d24..c6daec98ff89 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -343,7 +343,7 @@ static const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
};
static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = {
- .ac97_control = 1,
+ .bus_control = true,
.probe = au1xpsc_ac97_probe,
.playback = {
.rates = AC97_RATES,
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index e82eb373a731..6bf21a6c02e4 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -260,7 +260,7 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
#endif
static struct snd_soc_dai_driver bfin_ac97_dai = {
- .ac97_control = 1,
+ .bus_control = true,
.suspend = bf5xx_ac97_suspend,
.resume = bf5xx_ac97_resume,
.playback = {
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
index 3450e8f9080d..0fa81a523b8a 100644
--- a/sound/soc/blackfin/bf5xx-ad1980.c
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -46,8 +46,6 @@
#include <linux/gpio.h>
#include <asm/portmux.h>
-#include "../codecs/ad1980.h"
-
#include "bf5xx-ac97.h"
static struct snd_soc_card bf5xx_board;
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 5477c5475923..7b7fbcd49e5e 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -36,7 +36,8 @@ config SND_EP93XX_SOC_EDB93XX
tristate "SoC Audio support for Cirrus Logic EDB93xx boards"
depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A)
select SND_EP93XX_SOC_I2S
- select SND_SOC_CS4271
+ select SND_SOC_CS4271_I2C if I2C
+ select SND_SOC_CS4271_SPI if SPI_MASTER
help
Say Y or M here if you want to add support for I2S audio on the
Cirrus Logic EDB93xx boards.
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index f30dadf85b99..6b8a366b0211 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -338,7 +338,7 @@ static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = {
static struct snd_soc_dai_driver ep93xx_ac97_dai = {
.name = "ep93xx-ac97",
.id = 0,
- .ac97_control = 1,
+ .bus_control = true,
.probe = ep93xx_ac97_dai_probe,
.playback = {
.stream_name = "AC97 Playback",
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a68d1731a8fd..883c5778b309 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -50,7 +50,8 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS42L73 if I2C
select SND_SOC_CS4265 if I2C
select SND_SOC_CS4270 if I2C
- select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_CS4271_I2C if I2C
+ select SND_SOC_CS4271_SPI if SPI_MASTER
select SND_SOC_CS42XX8_I2C if I2C
select SND_SOC_CX20442 if TTY
select SND_SOC_DA7210 if I2C
@@ -85,7 +86,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_RT5645 if I2C
select SND_SOC_RT5651 if I2C
select SND_SOC_RT5670 if I2C
- select SND_SOC_RT5677 if I2C
+ select SND_SOC_RT5677 if I2C && SPI_MASTER
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
select SND_SOC_SIRF_AUDIO_CODEC
@@ -101,6 +102,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TAS2552 if I2C
select SND_SOC_TAS5086 if I2C
+ select SND_SOC_TFA9879 if I2C
select SND_SOC_TLV320AIC23_I2C if I2C
select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -109,6 +111,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_TPA6130A2 if I2C
select SND_SOC_TLV320DAC33 if I2C
+ select SND_SOC_TS3A227E if I2C
select SND_SOC_TWL4030 if TWL4030_CORE
select SND_SOC_TWL6040 if TWL6040_CORE
select SND_SOC_UDA134X
@@ -223,6 +226,7 @@ config SND_SOC_AD193X_I2C
select SND_SOC_AD193X
config SND_SOC_AD1980
+ select REGMAP_AC97
tristate
config SND_SOC_AD73311
@@ -336,7 +340,8 @@ config SND_SOC_CS42L51
tristate
config SND_SOC_CS42L51_I2C
- tristate
+ tristate "Cirrus Logic CS42L51 CODEC (I2C)"
+ depends on I2C
select SND_SOC_CS42L51
config SND_SOC_CS42L52
@@ -370,8 +375,19 @@ config SND_SOC_CS4270_VD33_ERRATA
depends on SND_SOC_CS4270
config SND_SOC_CS4271
- tristate "Cirrus Logic CS4271 CODEC"
- depends on SND_SOC_I2C_AND_SPI
+ tristate
+
+config SND_SOC_CS4271_I2C
+ tristate "Cirrus Logic CS4271 CODEC (I2C)"
+ depends on I2C
+ select SND_SOC_CS4271
+ select REGMAP_I2C
+
+config SND_SOC_CS4271_SPI
+ tristate "Cirrus Logic CS4271 CODEC (SPI)"
+ depends on SPI_MASTER
+ select SND_SOC_CS4271
+ select REGMAP_SPI
config SND_SOC_CS42XX8
tristate
@@ -487,7 +503,8 @@ config SND_SOC_RT286
depends on I2C
config SND_SOC_RT5631
- tristate
+ tristate "Realtek ALC5631/RT5631 CODEC"
+ depends on I2C
config SND_SOC_RT5640
tristate
@@ -504,6 +521,10 @@ config SND_SOC_RT5670
config SND_SOC_RT5677
tristate
+config SND_SOC_RT5677_SPI
+ tristate
+ default SND_SOC_RT5677
+
#Freescale sgtl5000 codec
config SND_SOC_SGTL5000
tristate "Freescale SGTL5000 CODEC"
@@ -577,15 +598,21 @@ config SND_SOC_TAS5086
tristate "Texas Instruments TAS5086 speaker amplifier"
depends on I2C
+config SND_SOC_TFA9879
+ tristate "NXP Semiconductors TFA9879 amplifier"
+ depends on I2C
+
config SND_SOC_TLV320AIC23
tristate
config SND_SOC_TLV320AIC23_I2C
- tristate
+ tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C"
+ depends on I2C
select SND_SOC_TLV320AIC23
config SND_SOC_TLV320AIC23_SPI
- tristate
+ tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI"
+ depends on SPI_MASTER
select SND_SOC_TLV320AIC23
config SND_SOC_TLV320AIC26
@@ -607,6 +634,10 @@ config SND_SOC_TLV320AIC3X
config SND_SOC_TLV320DAC33
tristate
+config SND_SOC_TS3A227E
+ tristate "TI Headset/Mic detect and keypress chip"
+ depends on I2C
+
config SND_SOC_TWL4030
select MFD_TWL4030_AUDIO
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5dce451661e4..bbdfd1e1c182 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -41,6 +41,8 @@ snd-soc-cs42l73-objs := cs42l73.o
snd-soc-cs4265-objs := cs4265.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
+snd-soc-cs4271-i2c-objs := cs4271-i2c.o
+snd-soc-cs4271-spi-objs := cs4271-spi.o
snd-soc-cs42xx8-objs := cs42xx8.o
snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o
snd-soc-cx20442-objs := cx20442.o
@@ -80,6 +82,7 @@ snd-soc-rt5645-objs := rt5645.o
snd-soc-rt5651-objs := rt5651.o
snd-soc-rt5670-objs := rt5670.o
snd-soc-rt5677-objs := rt5677.o
+snd-soc-rt5677-spi-objs := rt5677-spi.o
snd-soc-sgtl5000-objs := sgtl5000.o
snd-soc-alc5623-objs := alc5623.o
snd-soc-alc5632-objs := alc5632.o
@@ -101,6 +104,7 @@ snd-soc-sta350-objs := sta350.o
snd-soc-sta529-objs := sta529.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tas5086-objs := tas5086.o
+snd-soc-tfa9879-objs := tfa9879.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
@@ -109,6 +113,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
+snd-soc-ts3a227e-objs := ts3a227e.o
snd-soc-twl4030-objs := twl4030.o
snd-soc-twl6040-objs := twl6040.o
snd-soc-uda134x-objs := uda134x.o
@@ -217,6 +222,8 @@ obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o
obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
+obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o
+obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o
obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o
obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
@@ -256,6 +263,7 @@ obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o
obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o
obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o
obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o
+obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o
obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o
@@ -274,6 +282,7 @@ obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o
obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
+obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o
obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o
@@ -282,6 +291,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o
obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
+obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o
obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index fd43827bb856..7dfbc9921e91 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -126,13 +126,13 @@ struct ab8500_codec_drvdata_dbg {
/* Private data for AB8500 device-driver */
struct ab8500_codec_drvdata {
struct regmap *regmap;
+ struct mutex ctrl_lock;
/* Sidetone */
long *sid_fir_values;
enum sid_state sid_status;
/* ANC */
- struct mutex anc_lock;
long *anc_fir_values;
long *anc_iir_values;
enum anc_state anc_status;
@@ -1129,9 +1129,9 @@ static int sid_status_control_get(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
- mutex_lock(&codec->mutex);
+ mutex_lock(&drvdata->ctrl_lock);
ucontrol->value.integer.value[0] = drvdata->sid_status;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&drvdata->ctrl_lock);
return 0;
}
@@ -1154,7 +1154,7 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol,
return -EIO;
}
- mutex_lock(&codec->mutex);
+ mutex_lock(&drvdata->ctrl_lock);
sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF);
if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) {
@@ -1185,7 +1185,7 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol,
drvdata->sid_status = SID_FIR_CONFIGURED;
out:
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&drvdata->ctrl_lock);
dev_dbg(codec->dev, "%s: Exit\n", __func__);
@@ -1198,9 +1198,9 @@ static int anc_status_control_get(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
- mutex_lock(&codec->mutex);
+ mutex_lock(&drvdata->ctrl_lock);
ucontrol->value.integer.value[0] = drvdata->anc_status;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&drvdata->ctrl_lock);
return 0;
}
@@ -1217,7 +1217,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
dev_dbg(dev, "%s: Enter.\n", __func__);
- mutex_lock(&drvdata->anc_lock);
+ mutex_lock(&drvdata->ctrl_lock);
req = ucontrol->value.integer.value[0];
if (req >= ARRAY_SIZE(enum_anc_state)) {
@@ -1244,9 +1244,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
}
snd_soc_dapm_sync(&codec->dapm);
- mutex_lock(&codec->mutex);
anc_configure(codec, apply_fir, apply_iir);
- mutex_unlock(&codec->mutex);
if (apply_fir) {
if (drvdata->anc_status == ANC_IIR_CONFIGURED)
@@ -1265,7 +1263,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
snd_soc_dapm_sync(&codec->dapm);
cleanup:
- mutex_unlock(&drvdata->anc_lock);
+ mutex_unlock(&drvdata->ctrl_lock);
if (status < 0)
dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n",
@@ -1294,14 +1292,15 @@ static int filter_control_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct ab8500_codec_drvdata *drvdata = snd_soc_codec_get_drvdata(codec);
struct filter_control *fc =
(struct filter_control *)kcontrol->private_value;
unsigned int i;
- mutex_lock(&codec->mutex);
+ mutex_lock(&drvdata->ctrl_lock);
for (i = 0; i < fc->count; i++)
ucontrol->value.integer.value[i] = fc->value[i];
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&drvdata->ctrl_lock);
return 0;
}
@@ -1310,14 +1309,15 @@ static int filter_control_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct ab8500_codec_drvdata *drvdata = snd_soc_codec_get_drvdata(codec);
struct filter_control *fc =
(struct filter_control *)kcontrol->private_value;
unsigned int i;
- mutex_lock(&codec->mutex);
+ mutex_lock(&drvdata->ctrl_lock);
for (i = 0; i < fc->count; i++)
fc->value[i] = ucontrol->value.integer.value[i];
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&drvdata->ctrl_lock);
return 0;
}
@@ -2545,7 +2545,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
(void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
- mutex_init(&drvdata->anc_lock);
+ mutex_init(&drvdata->ctrl_lock);
return status;
}
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index bd9b1839c8b0..c6e5a313ebf4 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -37,10 +37,11 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
- return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate);
+ return snd_ac97_set_rate(ac97, reg, substream->runtime->rate);
}
#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
@@ -53,7 +54,6 @@ static const struct snd_soc_dai_ops ac97_dai_ops = {
static struct snd_soc_dai_driver ac97_dai = {
.name = "ac97-hifi",
- .ac97_control = 1,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 1,
@@ -71,6 +71,7 @@ static struct snd_soc_dai_driver ac97_dai = {
static int ac97_soc_probe(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97;
struct snd_ac97_bus *ac97_bus;
struct snd_ac97_template ac97_template;
int ret;
@@ -82,24 +83,31 @@ static int ac97_soc_probe(struct snd_soc_codec *codec)
return ret;
memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
- ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97);
+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97);
if (ret < 0)
return ret;
+ snd_soc_codec_set_drvdata(codec, ac97);
+
return 0;
}
#ifdef CONFIG_PM
static int ac97_soc_suspend(struct snd_soc_codec *codec)
{
- snd_ac97_suspend(codec->ac97);
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ snd_ac97_suspend(ac97);
return 0;
}
static int ac97_soc_resume(struct snd_soc_codec *codec)
{
- snd_ac97_resume(codec->ac97);
+
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ snd_ac97_resume(ac97);
return 0;
}
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 6844d0b2af68..387530b0b0fd 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -72,11 +72,13 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = {
};
static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = {
- SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1),
+ SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0),
+ SND_SOC_DAPM_VMID("VMID"),
SND_SOC_DAPM_OUTPUT("DAC1OUT"),
SND_SOC_DAPM_OUTPUT("DAC2OUT"),
SND_SOC_DAPM_OUTPUT("DAC3OUT"),
@@ -87,13 +89,15 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = {
static const struct snd_soc_dapm_route audio_paths[] = {
{ "DAC", NULL, "SYSCLK" },
+ { "DAC Output", NULL, "DAC" },
+ { "DAC Output", NULL, "VMID" },
{ "ADC", NULL, "SYSCLK" },
{ "DAC", NULL, "ADC_PWR" },
{ "ADC", NULL, "ADC_PWR" },
- { "DAC1OUT", NULL, "DAC" },
- { "DAC2OUT", NULL, "DAC" },
- { "DAC3OUT", NULL, "DAC" },
- { "DAC4OUT", NULL, "DAC" },
+ { "DAC1OUT", NULL, "DAC Output" },
+ { "DAC2OUT", NULL, "DAC Output" },
+ { "DAC3OUT", NULL, "DAC Output" },
+ { "DAC4OUT", NULL, "DAC Output" },
{ "ADC", NULL, "ADC1IN" },
{ "ADC", NULL, "ADC2IN" },
{ "SYSCLK", NULL, "PLL_PWR" },
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 304d3003339a..2860eef8610c 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -24,34 +24,86 @@
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
+#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
-#include "ad1980.h"
+static const struct reg_default ad1980_reg_defaults[] = {
+ { 0x02, 0x8000 },
+ { 0x04, 0x8000 },
+ { 0x06, 0x8000 },
+ { 0x0c, 0x8008 },
+ { 0x0e, 0x8008 },
+ { 0x10, 0x8808 },
+ { 0x12, 0x8808 },
+ { 0x16, 0x8808 },
+ { 0x18, 0x8808 },
+ { 0x1a, 0x0000 },
+ { 0x1c, 0x8000 },
+ { 0x20, 0x0000 },
+ { 0x28, 0x03c7 },
+ { 0x2c, 0xbb80 },
+ { 0x2e, 0xbb80 },
+ { 0x30, 0xbb80 },
+ { 0x32, 0xbb80 },
+ { 0x36, 0x8080 },
+ { 0x38, 0x8080 },
+ { 0x3a, 0x2000 },
+ { 0x60, 0x0000 },
+ { 0x62, 0x0000 },
+ { 0x72, 0x0000 },
+ { 0x74, 0x1001 },
+ { 0x76, 0x0000 },
+};
-/*
- * AD1980 register cache
- */
-static const u16 ad1980_reg[] = {
- 0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6 */
- 0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e */
- 0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */
- 0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */
- 0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */
- 0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */
- 0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */
- 0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */
- 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
- 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
- 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
- 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
- 0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */
- 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
- 0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */
- 0x0000, 0x0000, 0x4144, 0x5370 /* 78 - 7e */
+static bool ad1980_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case AC97_RESET ... AC97_MASTER_MONO:
+ case AC97_PHONE ... AC97_CD:
+ case AC97_AUX ... AC97_GENERAL_PURPOSE:
+ case AC97_POWERDOWN ... AC97_PCM_LR_ADC_RATE:
+ case AC97_SPDIF:
+ case AC97_CODEC_CLASS_REV:
+ case AC97_PCI_SVID:
+ case AC97_AD_CODEC_CFG:
+ case AC97_AD_JACK_SPDIF:
+ case AC97_AD_SERIAL_CFG:
+ case AC97_VENDOR_ID1:
+ case AC97_VENDOR_ID2:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool ad1980_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case AC97_VENDOR_ID1:
+ case AC97_VENDOR_ID2:
+ return false;
+ default:
+ return ad1980_readable_reg(dev, reg);
+ }
+}
+
+static const struct regmap_config ad1980_regmap_config = {
+ .reg_bits = 16,
+ .reg_stride = 2,
+ .val_bits = 16,
+ .max_register = 0x7e,
+ .cache_type = REGCACHE_RBTREE,
+
+ .volatile_reg = regmap_ac97_default_volatile,
+ .readable_reg = ad1980_readable_reg,
+ .writeable_reg = ad1980_writeable_reg,
+
+ .reg_defaults = ad1980_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(ad1980_reg_defaults),
};
static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
@@ -134,45 +186,8 @@ static const struct snd_soc_dapm_route ad1980_dapm_routes[] = {
{ "HP_OUT_R", NULL, "Playback" },
};
-static unsigned int ac97_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
-
- switch (reg) {
- case AC97_RESET:
- case AC97_INT_PAGING:
- case AC97_POWERDOWN:
- case AC97_EXTENDED_STATUS:
- case AC97_VENDOR_ID1:
- case AC97_VENDOR_ID2:
- return soc_ac97_ops->read(codec->ac97, reg);
- default:
- reg = reg >> 1;
-
- if (reg >= ARRAY_SIZE(ad1980_reg))
- return -EINVAL;
-
- return cache[reg];
- }
-}
-
-static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int val)
-{
- u16 *cache = codec->reg_cache;
-
- soc_ac97_ops->write(codec->ac97, reg, val);
- reg = reg >> 1;
- if (reg < ARRAY_SIZE(ad1980_reg))
- cache[reg] = val;
-
- return 0;
-}
-
static struct snd_soc_dai_driver ad1980_dai = {
.name = "ad1980-hifi",
- .ac97_control = 1,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
@@ -189,108 +204,115 @@ static struct snd_soc_dai_driver ad1980_dai = {
static int ad1980_reset(struct snd_soc_codec *codec, int try_warm)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
unsigned int retry_cnt = 0;
do {
if (try_warm && soc_ac97_ops->warm_reset) {
- soc_ac97_ops->warm_reset(codec->ac97);
- if (ac97_read(codec, AC97_RESET) == 0x0090)
+ soc_ac97_ops->warm_reset(ac97);
+ if (snd_soc_read(codec, AC97_RESET) == 0x0090)
return 1;
}
- soc_ac97_ops->reset(codec->ac97);
+ soc_ac97_ops->reset(ac97);
/*
* Set bit 16slot in register 74h, then every slot will has only
* 16 bits. This command is sent out in 20bit mode, in which
* case the first nibble of data is eaten by the addr. (Tag is
* always 16 bit)
*/
- ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
+ snd_soc_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
- if (ac97_read(codec, AC97_RESET) == 0x0090)
+ if (snd_soc_read(codec, AC97_RESET) == 0x0090)
return 0;
} while (retry_cnt++ < 10);
- printk(KERN_ERR "AD1980 AC97 reset failed\n");
+ dev_err(codec->dev, "Failed to reset: AC97 link error\n");
+
return -EIO;
}
static int ad1980_soc_probe(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97;
+ struct regmap *regmap;
int ret;
u16 vendor_id2;
u16 ext_status;
- printk(KERN_INFO "AD1980 SoC Audio Codec\n");
-
- ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0);
- if (ret < 0) {
- printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
+ ac97 = snd_soc_new_ac97_codec(codec);
+ if (IS_ERR(ac97)) {
+ ret = PTR_ERR(ac97);
+ dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret);
return ret;
}
+ regmap = regmap_init_ac97(ac97, &ad1980_regmap_config);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ goto err_free_ac97;
+ }
+
+ snd_soc_codec_init_regmap(codec, regmap);
+ snd_soc_codec_set_drvdata(codec, ac97);
+
ret = ad1980_reset(codec, 0);
- if (ret < 0) {
- printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
+ if (ret < 0)
goto reset_err;
- }
/* Read out vendor ID to make sure it is ad1980 */
- if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) {
+ if (snd_soc_read(codec, AC97_VENDOR_ID1) != 0x4144) {
ret = -ENODEV;
goto reset_err;
}
- vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2);
+ vendor_id2 = snd_soc_read(codec, AC97_VENDOR_ID2);
if (vendor_id2 != 0x5370) {
if (vendor_id2 != 0x5374) {
ret = -ENODEV;
goto reset_err;
} else {
- printk(KERN_WARNING "ad1980: "
- "Found AD1981 - only 2/2 IN/OUT Channels "
- "supported\n");
+ dev_warn(codec->dev,
+ "Found AD1981 - only 2/2 IN/OUT Channels supported\n");
}
}
/* unmute captures and playbacks volume */
- ac97_write(codec, AC97_MASTER, 0x0000);
- ac97_write(codec, AC97_PCM, 0x0000);
- ac97_write(codec, AC97_REC_GAIN, 0x0000);
- ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
- ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
+ snd_soc_write(codec, AC97_MASTER, 0x0000);
+ snd_soc_write(codec, AC97_PCM, 0x0000);
+ snd_soc_write(codec, AC97_REC_GAIN, 0x0000);
+ snd_soc_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
+ snd_soc_write(codec, AC97_SURROUND_MASTER, 0x0000);
/*power on LFE/CENTER/Surround DACs*/
- ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
- ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
-
- snd_soc_add_codec_controls(codec, ad1980_snd_ac97_controls,
- ARRAY_SIZE(ad1980_snd_ac97_controls));
+ ext_status = snd_soc_read(codec, AC97_EXTENDED_STATUS);
+ snd_soc_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
return 0;
reset_err:
- snd_soc_free_ac97_codec(codec);
+ snd_soc_codec_exit_regmap(codec);
+err_free_ac97:
+ snd_soc_free_ac97_codec(ac97);
return ret;
}
static int ad1980_soc_remove(struct snd_soc_codec *codec)
{
- snd_soc_free_ac97_codec(codec);
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_codec_exit_regmap(codec);
+ snd_soc_free_ac97_codec(ac97);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
.probe = ad1980_soc_probe,
.remove = ad1980_soc_remove,
- .reg_cache_size = ARRAY_SIZE(ad1980_reg),
- .reg_word_size = sizeof(u16),
- .reg_cache_default = ad1980_reg,
- .reg_cache_step = 2,
- .write = ac97_write,
- .read = ac97_read,
+ .controls = ad1980_snd_ac97_controls,
+ .num_controls = ARRAY_SIZE(ad1980_snd_ac97_controls),
.dapm_widgets = ad1980_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets),
.dapm_routes = ad1980_dapm_routes,
diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h
deleted file mode 100644
index eb0af44ad3df..000000000000
--- a/sound/soc/codecs/ad1980.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * ad1980.h -- ad1980 Soc Audio driver
- *
- * WARNING:
- *
- * Because Analog Devices Inc. discontinued the ad1980 sound chip since
- * Sep. 2009, this ad1980 driver is not maintained, tested and supported
- * by ADI now.
- */
-
-#ifndef _AD1980_H
-#define _AD1980_H
-/* Bit definition of Power-Down Control/Status Register */
-#define ADC 0x0001
-#define DAC 0x0002
-#define ANL 0x0004
-#define REF 0x0008
-#define PR0 0x0100
-#define PR1 0x0200
-#define PR2 0x0400
-#define PR3 0x0800
-#define PR4 0x1000
-#define PR5 0x2000
-#define PR6 0x4000
-
-#endif
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 7c784ad3e8b2..783dcb57043a 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -551,7 +551,7 @@ static const struct snd_kcontrol_new adau1373_drc_controls[] = {
static int adau1373_pll_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
unsigned int pll_id = w->name[3] - '1';
unsigned int val;
@@ -823,7 +823,7 @@ static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = {
static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct snd_soc_codec *codec = source->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
unsigned int dai;
const char *clk;
@@ -844,7 +844,7 @@ static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source,
static int adau1373_check_src(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct snd_soc_codec *codec = source->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
unsigned int dai;
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 370b742117ef..d4e219b6b98f 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -22,9 +22,14 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <asm/unaligned.h>
+
#include "sigmadsp.h"
#include "adau1701.h"
+#define ADAU1701_SAFELOAD_DATA(i) (0x0810 + (i))
+#define ADAU1701_SAFELOAD_ADDR(i) (0x0815 + (i))
+
#define ADAU1701_DSPCTRL 0x081c
#define ADAU1701_SEROCTL 0x081e
#define ADAU1701_SERICTL 0x081f
@@ -42,6 +47,7 @@
#define ADAU1701_DSPCTRL_CR (1 << 2)
#define ADAU1701_DSPCTRL_DAM (1 << 3)
#define ADAU1701_DSPCTRL_ADM (1 << 4)
+#define ADAU1701_DSPCTRL_IST (1 << 5)
#define ADAU1701_DSPCTRL_SR_48 0x00
#define ADAU1701_DSPCTRL_SR_96 0x01
#define ADAU1701_DSPCTRL_SR_192 0x02
@@ -102,7 +108,10 @@ struct adau1701 {
unsigned int pll_clkdiv;
unsigned int sysclk;
struct regmap *regmap;
+ struct i2c_client *client;
u8 pin_config[12];
+
+ struct sigmadsp *sigmadsp;
};
static const struct snd_kcontrol_new adau1701_controls[] = {
@@ -159,6 +168,7 @@ static bool adau1701_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case ADAU1701_DACSET:
+ case ADAU1701_DSPCTRL:
return true;
default:
return false;
@@ -238,12 +248,58 @@ static int adau1701_reg_read(void *context, unsigned int reg,
return 0;
}
-static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
+static int adau1701_safeload(struct sigmadsp *sigmadsp, unsigned int addr,
+ const uint8_t bytes[], size_t len)
+{
+ struct i2c_client *client = to_i2c_client(sigmadsp->dev);
+ struct adau1701 *adau1701 = i2c_get_clientdata(client);
+ unsigned int val;
+ unsigned int i;
+ uint8_t buf[10];
+ int ret;
+
+ ret = regmap_read(adau1701->regmap, ADAU1701_DSPCTRL, &val);
+ if (ret)
+ return ret;
+
+ if (val & ADAU1701_DSPCTRL_IST)
+ msleep(50);
+
+ for (i = 0; i < len / 4; i++) {
+ put_unaligned_le16(ADAU1701_SAFELOAD_DATA(i), buf);
+ buf[2] = 0x00;
+ memcpy(buf + 3, bytes + i * 4, 4);
+ ret = i2c_master_send(client, buf, 7);
+ if (ret < 0)
+ return ret;
+ else if (ret != 7)
+ return -EIO;
+
+ put_unaligned_le16(ADAU1701_SAFELOAD_ADDR(i), buf);
+ put_unaligned_le16(addr + i, buf + 2);
+ ret = i2c_master_send(client, buf, 4);
+ if (ret < 0)
+ return ret;
+ else if (ret != 4)
+ return -EIO;
+ }
+
+ return regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL,
+ ADAU1701_DSPCTRL_IST, ADAU1701_DSPCTRL_IST);
+}
+
+static const struct sigmadsp_ops adau1701_sigmadsp_ops = {
+ .safeload = adau1701_safeload,
+};
+
+static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv,
+ unsigned int rate)
{
struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *client = to_i2c_client(codec->dev);
int ret;
+ sigmadsp_reset(adau1701->sigmadsp);
+
if (clkdiv != ADAU1707_CLKDIV_UNSET &&
gpio_is_valid(adau1701->gpio_pll_mode[0]) &&
gpio_is_valid(adau1701->gpio_pll_mode[1])) {
@@ -284,7 +340,7 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
* know the correct PLL setup
*/
if (clkdiv != ADAU1707_CLKDIV_UNSET) {
- ret = process_sigma_firmware(client, ADAU1701_FIRMWARE);
+ ret = sigmadsp_setup(adau1701->sigmadsp, rate);
if (ret) {
dev_warn(codec->dev, "Failed to load firmware\n");
return ret;
@@ -385,7 +441,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream,
* firmware upload.
*/
if (clkdiv != adau1701->pll_clkdiv) {
- ret = adau1701_reset(codec, clkdiv);
+ ret = adau1701_reset(codec, clkdiv, params_rate(params));
if (ret < 0)
return ret;
}
@@ -554,6 +610,14 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id,
return 0;
}
+static int adau1701_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(dai->codec);
+
+ return sigmadsp_restrict_params(adau1701->sigmadsp, substream);
+}
+
#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
SNDRV_PCM_RATE_192000)
@@ -564,6 +628,7 @@ static const struct snd_soc_dai_ops adau1701_dai_ops = {
.set_fmt = adau1701_set_dai_fmt,
.hw_params = adau1701_hw_params,
.digital_mute = adau1701_digital_mute,
+ .startup = adau1701_startup,
};
static struct snd_soc_dai_driver adau1701_dai = {
@@ -600,6 +665,10 @@ static int adau1701_probe(struct snd_soc_codec *codec)
unsigned int val;
struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ ret = sigmadsp_attach(adau1701->sigmadsp, &codec->component);
+ if (ret)
+ return ret;
+
/*
* Let the pll_clkdiv variable default to something that won't happen
* at runtime. That way, we can postpone the firmware download from
@@ -609,7 +678,7 @@ static int adau1701_probe(struct snd_soc_codec *codec)
adau1701->pll_clkdiv = ADAU1707_CLKDIV_UNSET;
/* initalize with pre-configured pll mode settings */
- ret = adau1701_reset(codec, adau1701->pll_clkdiv);
+ ret = adau1701_reset(codec, adau1701->pll_clkdiv, 0);
if (ret < 0)
return ret;
@@ -667,6 +736,7 @@ static int adau1701_i2c_probe(struct i2c_client *client,
if (!adau1701)
return -ENOMEM;
+ adau1701->client = client;
adau1701->regmap = devm_regmap_init(dev, NULL, client,
&adau1701_regmap);
if (IS_ERR(adau1701->regmap))
@@ -722,6 +792,12 @@ static int adau1701_i2c_probe(struct i2c_client *client,
adau1701->gpio_pll_mode[1] = gpio_pll_mode[1];
i2c_set_clientdata(client, adau1701);
+
+ adau1701->sigmadsp = devm_sigmadsp_init_i2c(client,
+ &adau1701_sigmadsp_ops, ADAU1701_FIRMWARE);
+ if (IS_ERR(adau1701->sigmadsp))
+ return PTR_ERR(adau1701->sigmadsp);
+
ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv,
&adau1701_dai, 1);
return ret;
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index 91f60282fd2f..a1baeee160f4 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -255,7 +255,8 @@ static const struct snd_kcontrol_new adau1761_input_mux_control =
static int adau1761_dejitter_fixup(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct adau *adau = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct adau *adau = snd_soc_codec_get_drvdata(codec);
/* After any power changes have been made the dejitter circuit
* has to be reinitialized. */
@@ -702,11 +703,6 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(adau1761_dapm_routes));
if (ret)
return ret;
-
- ret = adau17x1_load_firmware(adau, codec->dev,
- ADAU1761_FIRMWARE);
- if (ret)
- dev_warn(codec->dev, "Failed to firmware\n");
}
ret = adau17x1_add_routes(codec);
@@ -775,16 +771,20 @@ int adau1761_probe(struct device *dev, struct regmap *regmap,
enum adau17x1_type type, void (*switch_mode)(struct device *dev))
{
struct snd_soc_dai_driver *dai_drv;
+ const char *firmware_name;
int ret;
- ret = adau17x1_probe(dev, regmap, type, switch_mode);
- if (ret)
- return ret;
-
- if (type == ADAU1361)
+ if (type == ADAU1361) {
dai_drv = &adau1361_dai_driver;
- else
+ firmware_name = NULL;
+ } else {
dai_drv = &adau1761_dai_driver;
+ firmware_name = ADAU1761_FIRMWARE;
+ }
+
+ ret = adau17x1_probe(dev, regmap, type, switch_mode, firmware_name);
+ if (ret)
+ return ret;
return snd_soc_register_codec(dev, &adau1761_codec_driver, dai_drv, 1);
}
@@ -798,6 +798,7 @@ const struct regmap_config adau1761_regmap_config = {
.num_reg_defaults = ARRAY_SIZE(adau1761_reg_defaults),
.readable_reg = adau1761_readable_register,
.volatile_reg = adau17x1_volatile_register,
+ .precious_reg = adau17x1_precious_register,
.cache_type = REGCACHE_RBTREE,
};
EXPORT_SYMBOL_GPL(adau1761_regmap_config);
diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c
index e9fc00fb13dd..35581f43fa6d 100644
--- a/sound/soc/codecs/adau1781.c
+++ b/sound/soc/codecs/adau1781.c
@@ -174,7 +174,7 @@ static const struct snd_kcontrol_new adau1781_mono_mixer_controls[] = {
static int adau1781_dejitter_fixup(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct adau *adau = snd_soc_codec_get_drvdata(codec);
/* After any power changes have been made the dejitter circuit
@@ -385,7 +385,6 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec)
{
struct adau1781_platform_data *pdata = dev_get_platdata(codec->dev);
struct adau *adau = snd_soc_codec_get_drvdata(codec);
- const char *firmware;
int ret;
ret = adau17x1_add_widgets(codec);
@@ -422,25 +421,10 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec)
return ret;
}
- switch (adau->type) {
- case ADAU1381:
- firmware = ADAU1381_FIRMWARE;
- break;
- case ADAU1781:
- firmware = ADAU1781_FIRMWARE;
- break;
- default:
- return -EINVAL;
- }
-
ret = adau17x1_add_routes(codec);
if (ret < 0)
return ret;
- ret = adau17x1_load_firmware(adau, codec->dev, firmware);
- if (ret)
- dev_warn(codec->dev, "Failed to load firmware\n");
-
return 0;
}
@@ -488,6 +472,7 @@ const struct regmap_config adau1781_regmap_config = {
.num_reg_defaults = ARRAY_SIZE(adau1781_reg_defaults),
.readable_reg = adau1781_readable_register,
.volatile_reg = adau17x1_volatile_register,
+ .precious_reg = adau17x1_precious_register,
.cache_type = REGCACHE_RBTREE,
};
EXPORT_SYMBOL_GPL(adau1781_regmap_config);
@@ -495,9 +480,21 @@ EXPORT_SYMBOL_GPL(adau1781_regmap_config);
int adau1781_probe(struct device *dev, struct regmap *regmap,
enum adau17x1_type type, void (*switch_mode)(struct device *dev))
{
+ const char *firmware_name;
int ret;
- ret = adau17x1_probe(dev, regmap, type, switch_mode);
+ switch (type) {
+ case ADAU1381:
+ firmware_name = ADAU1381_FIRMWARE;
+ break;
+ case ADAU1781:
+ firmware_name = ADAU1781_FIRMWARE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = adau17x1_probe(dev, regmap, type, switch_mode, firmware_name);
if (ret)
return ret;
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 3e16c1c64115..fa2e690e51c8 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -61,7 +61,8 @@ static const struct snd_kcontrol_new adau17x1_controls[] = {
static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct adau *adau = snd_soc_codec_get_drvdata(w->codec);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct adau *adau = snd_soc_codec_get_drvdata(codec);
int ret;
if (SND_SOC_DAPM_EVENT_ON(event)) {
@@ -307,6 +308,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
struct adau *adau = snd_soc_codec_get_drvdata(codec);
unsigned int val, div, dsp_div;
unsigned int freq;
+ int ret;
if (adau->clk_src == ADAU17X1_CLK_SRC_PLL)
freq = adau->pll_freq;
@@ -356,6 +358,12 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div);
}
+ if (adau->sigmadsp) {
+ ret = adau17x1_setup_firmware(adau, params_rate(params));
+ if (ret < 0)
+ return ret;
+ }
+
if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
return 0;
@@ -661,12 +669,24 @@ static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai,
return 0;
}
+static int adau17x1_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
+
+ if (adau->sigmadsp)
+ return sigmadsp_restrict_params(adau->sigmadsp, substream);
+
+ return 0;
+}
+
const struct snd_soc_dai_ops adau17x1_dai_ops = {
.hw_params = adau17x1_hw_params,
.set_sysclk = adau17x1_set_dai_sysclk,
.set_fmt = adau17x1_set_dai_fmt,
.set_pll = adau17x1_set_dai_pll,
.set_tdm_slot = adau17x1_set_dai_tdm_slot,
+ .startup = adau17x1_startup,
};
EXPORT_SYMBOL_GPL(adau17x1_dai_ops);
@@ -687,8 +707,22 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec,
}
EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage);
+bool adau17x1_precious_register(struct device *dev, unsigned int reg)
+{
+ /* SigmaDSP parameter memory */
+ if (reg < 0x400)
+ return true;
+
+ return false;
+}
+EXPORT_SYMBOL_GPL(adau17x1_precious_register);
+
bool adau17x1_readable_register(struct device *dev, unsigned int reg)
{
+ /* SigmaDSP parameter memory */
+ if (reg < 0x400)
+ return true;
+
switch (reg) {
case ADAU17X1_CLOCK_CONTROL:
case ADAU17X1_PLL_CONTROL:
@@ -745,8 +779,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
}
EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
-int adau17x1_load_firmware(struct adau *adau, struct device *dev,
- const char *firmware)
+int adau17x1_setup_firmware(struct adau *adau, unsigned int rate)
{
int ret;
int dspsr;
@@ -758,7 +791,7 @@ int adau17x1_load_firmware(struct adau *adau, struct device *dev,
regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1);
regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf);
- ret = process_sigma_firmware_regmap(dev, adau->regmap, firmware);
+ ret = sigmadsp_setup(adau->sigmadsp, rate);
if (ret) {
regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0);
return ret;
@@ -767,7 +800,7 @@ int adau17x1_load_firmware(struct adau *adau, struct device *dev,
return 0;
}
-EXPORT_SYMBOL_GPL(adau17x1_load_firmware);
+EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
int adau17x1_add_widgets(struct snd_soc_codec *codec)
{
@@ -787,8 +820,21 @@ int adau17x1_add_widgets(struct snd_soc_codec *codec)
ret = snd_soc_dapm_new_controls(&codec->dapm,
adau17x1_dsp_dapm_widgets,
ARRAY_SIZE(adau17x1_dsp_dapm_widgets));
+ if (ret)
+ return ret;
+
+ if (!adau->sigmadsp)
+ return 0;
+
+ ret = sigmadsp_attach(adau->sigmadsp, &codec->component);
+ if (ret) {
+ dev_err(codec->dev, "Failed to attach firmware: %d\n",
+ ret);
+ return ret;
+ }
}
- return ret;
+
+ return 0;
}
EXPORT_SYMBOL_GPL(adau17x1_add_widgets);
@@ -829,7 +875,8 @@ int adau17x1_resume(struct snd_soc_codec *codec)
EXPORT_SYMBOL_GPL(adau17x1_resume);
int adau17x1_probe(struct device *dev, struct regmap *regmap,
- enum adau17x1_type type, void (*switch_mode)(struct device *dev))
+ enum adau17x1_type type, void (*switch_mode)(struct device *dev),
+ const char *firmware_name)
{
struct adau *adau;
@@ -846,6 +893,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
dev_set_drvdata(dev, adau);
+ if (firmware_name) {
+ adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL,
+ firmware_name);
+ if (IS_ERR(adau->sigmadsp)) {
+ dev_warn(dev, "Could not find firmware file: %ld\n",
+ PTR_ERR(adau->sigmadsp));
+ adau->sigmadsp = NULL;
+ }
+ }
+
if (switch_mode)
switch_mode(dev);
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index e4a557fd7155..e13583e6ff56 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -4,6 +4,8 @@
#include <linux/regmap.h>
#include <linux/platform_data/adau17x1.h>
+#include "sigmadsp.h"
+
enum adau17x1_type {
ADAU1361,
ADAU1761,
@@ -42,22 +44,24 @@ struct adau {
bool dsp_bypass[2];
struct regmap *regmap;
+ struct sigmadsp *sigmadsp;
};
int adau17x1_add_widgets(struct snd_soc_codec *codec);
int adau17x1_add_routes(struct snd_soc_codec *codec);
int adau17x1_probe(struct device *dev, struct regmap *regmap,
- enum adau17x1_type type, void (*switch_mode)(struct device *dev));
+ enum adau17x1_type type, void (*switch_mode)(struct device *dev),
+ const char *firmware_name);
int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec,
enum adau17x1_micbias_voltage micbias);
bool adau17x1_readable_register(struct device *dev, unsigned int reg);
bool adau17x1_volatile_register(struct device *dev, unsigned int reg);
+bool adau17x1_precious_register(struct device *dev, unsigned int reg);
int adau17x1_resume(struct snd_soc_codec *codec);
extern const struct snd_soc_dai_ops adau17x1_dai_ops;
-int adau17x1_load_firmware(struct adau *adau, struct device *dev,
- const char *firmware);
+int adau17x1_setup_firmware(struct adau *adau, unsigned int rate);
bool adau17x1_has_dsp(struct adau *adau);
#define ADAU17X1_CLOCK_CONTROL 0x4000
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index ce3cdca9fc62..b67480f1b1aa 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -212,7 +212,7 @@ static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = {
static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct snd_soc_codec *codec = source->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
const char *clk;
@@ -236,7 +236,7 @@ static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source,
static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
- struct snd_soc_codec *codec = source->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL;
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 30e297890fec..9130d916f2f4 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -373,33 +373,9 @@ static struct snd_soc_dai_driver ak4535_dai = {
.ops = &ak4535_dai_ops,
};
-static int ak4535_suspend(struct snd_soc_codec *codec)
-{
- ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static int ak4535_resume(struct snd_soc_codec *codec)
{
snd_soc_cache_sync(codec);
- ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
-static int ak4535_probe(struct snd_soc_codec *codec)
-{
- /* power on device */
- ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- snd_soc_add_codec_controls(codec, ak4535_snd_controls,
- ARRAY_SIZE(ak4535_snd_controls));
- return 0;
-}
-
-/* power down chip */
-static int ak4535_remove(struct snd_soc_codec *codec)
-{
- ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -416,11 +392,12 @@ static const struct regmap_config ak4535_regmap = {
};
static struct snd_soc_codec_driver soc_codec_dev_ak4535 = {
- .probe = ak4535_probe,
- .remove = ak4535_remove,
- .suspend = ak4535_suspend,
.resume = ak4535_resume,
.set_bias_level = ak4535_set_bias_level,
+ .suspend_bias_off = true,
+
+ .controls = ak4535_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4535_snd_controls),
.dapm_widgets = ak4535_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets),
.dapm_routes = ak4535_audio_map,
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 7afe8f482088..70861c7b1631 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -505,39 +505,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = {
},
};
-static int ak4641_suspend(struct snd_soc_codec *codec)
-{
- ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int ak4641_resume(struct snd_soc_codec *codec)
-{
- ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
-static int ak4641_probe(struct snd_soc_codec *codec)
-{
- /* power on device */
- ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int ak4641_remove(struct snd_soc_codec *codec)
-{
- ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-
static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
- .probe = ak4641_probe,
- .remove = ak4641_remove,
- .suspend = ak4641_suspend,
- .resume = ak4641_resume,
.controls = ak4641_snd_controls,
.num_controls = ARRAY_SIZE(ak4641_snd_controls),
.dapm_widgets = ak4641_dapm_widgets,
@@ -545,6 +513,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
.dapm_routes = ak4641_audio_map,
.num_dapm_routes = ARRAY_SIZE(ak4641_audio_map),
.set_bias_level = ak4641_set_bias_level,
+ .suspend_bias_off = true,
};
static const struct regmap_config ak4641_regmap = {
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 041712592e29..dde8b49c19ad 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -491,23 +491,7 @@ static int ak4642_resume(struct snd_soc_codec *codec)
return 0;
}
-
-static int ak4642_probe(struct snd_soc_codec *codec)
-{
- ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int ak4642_remove(struct snd_soc_codec *codec)
-{
- ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
- .probe = ak4642_probe,
- .remove = ak4642_remove,
.resume = ak4642_resume,
.set_bias_level = ak4642_set_bias_level,
.controls = ak4642_snd_controls,
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 998fa0c5a0b9..686cacb0e835 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -611,20 +611,7 @@ static struct snd_soc_dai_driver ak4671_dai = {
.ops = &ak4671_dai_ops,
};
-static int ak4671_probe(struct snd_soc_codec *codec)
-{
- return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-}
-
-static int ak4671_remove(struct snd_soc_codec *codec)
-{
- ak4671_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_ak4671 = {
- .probe = ak4671_probe,
- .remove = ak4671_remove,
.set_bias_level = ak4671_set_bias_level,
.controls = ak4671_snd_controls,
.num_controls = ARRAY_SIZE(ak4671_snd_controls),
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 9d0755aa1d16..bdf8c5ac8ca4 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec)
{
struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
- alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
regcache_cache_only(alc5623->regmap, true);
return 0;
@@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec)
return ret;
}
- alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- /* charge alc5623 caps */
- if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
- alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- codec->dapm.bias_level = SND_SOC_BIAS_ON;
- alc5623_set_bias_level(codec, codec->dapm.bias_level);
- }
-
return 0;
}
@@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec)
alc5623_reset(codec);
- /* power on device */
- alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
if (alc5623->add_ctrl) {
snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
alc5623->add_ctrl);
@@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec)
return 0;
}
-/* power down chip */
-static int alc5623_remove(struct snd_soc_codec *codec)
-{
- alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
.probe = alc5623_probe,
- .remove = alc5623_remove,
.suspend = alc5623_suspend,
.resume = alc5623_resume,
.set_bias_level = alc5623_set_bias_level,
+ .suspend_bias_off = true,
};
static const struct regmap_config alc5623_regmap = {
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 85942ca36cbf..d1fdbc266631 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -1038,23 +1038,15 @@ static struct snd_soc_dai_driver alc5632_dai = {
};
#ifdef CONFIG_PM
-static int alc5632_suspend(struct snd_soc_codec *codec)
-{
- alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static int alc5632_resume(struct snd_soc_codec *codec)
{
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
regcache_sync(alc5632->regmap);
- alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
#else
-#define alc5632_suspend NULL
#define alc5632_resume NULL
#endif
@@ -1062,9 +1054,6 @@ static int alc5632_probe(struct snd_soc_codec *codec)
{
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
- /* power on device */
- alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
switch (alc5632->id) {
case 0x5c:
snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls,
@@ -1077,19 +1066,12 @@ static int alc5632_probe(struct snd_soc_codec *codec)
return 0;
}
-/* power down chip */
-static int alc5632_remove(struct snd_soc_codec *codec)
-{
- alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_device_alc5632 = {
.probe = alc5632_probe,
- .remove = alc5632_remove,
- .suspend = alc5632_suspend,
.resume = alc5632_resume,
.set_bias_level = alc5632_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = alc5632_snd_controls,
.num_controls = ARRAY_SIZE(alc5632_snd_controls),
.dapm_widgets = alc5632_dapm_widgets,
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 0c05e7a7945f..9550d7433ad0 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -61,6 +61,11 @@
#define ARIZONA_FLL_MIN_OUTDIV 2
#define ARIZONA_FLL_MAX_OUTDIV 7
+#define ARIZONA_FMT_DSP_MODE_A 0
+#define ARIZONA_FMT_DSP_MODE_B 1
+#define ARIZONA_FMT_I2S_MODE 2
+#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3
+
#define arizona_fll_err(_fll, fmt, ...) \
dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_fll_warn(_fll, fmt, ...) \
@@ -648,7 +653,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum,
EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum);
static const char * const arizona_in_dmic_osr_text[] = {
- "1.536MHz", "3.072MHz", "6.144MHz",
+ "1.536MHz", "3.072MHz", "6.144MHz", "768kHz",
};
const struct soc_enum arizona_in_dmic_osr[] = {
@@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
- mode = 0;
+ mode = ARIZONA_FMT_DSP_MODE_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK)
+ != SND_SOC_DAIFMT_CBM_CFM) {
+ arizona_aif_err(dai, "DSP_B not valid in slave mode\n");
+ return -EINVAL;
+ }
+ mode = ARIZONA_FMT_DSP_MODE_B;
break;
case SND_SOC_DAIFMT_I2S:
- mode = 2;
+ mode = ARIZONA_FMT_I2S_MODE;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK)
+ != SND_SOC_DAIFMT_CBM_CFM) {
+ arizona_aif_err(dai, "LEFT_J not valid in slave mode\n");
+ return -EINVAL;
+ }
+ mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE;
break;
default:
arizona_aif_err(dai, "Unsupported DAI format %d\n",
@@ -1164,13 +1185,13 @@ static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec,
{ 0x80, 0x0 },
};
- mutex_lock(&codec->mutex);
+ mutex_lock(&arizona->dac_comp_lock);
dac_comp[1].def = arizona->dac_comp_coeff;
if (rate >= 176400)
dac_comp[2].def = arizona->dac_comp_enabled;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&arizona->dac_comp_lock);
regmap_multi_reg_write(arizona->regmap,
dac_comp,
@@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
/* Force multiple of 2 channels for I2S mode */
val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
- if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) {
+ val &= ARIZONA_AIF1_FMT_MASK;
+ if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) {
arizona_aif_dbg(dai, "Forcing stereo mode\n");
bclk_target /= channels;
bclk_target *= channels + 1;
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 537327c7f7f1..8d638e8aa8eb 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -62,14 +62,10 @@ static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
- struct snd_soc_codec *codec = codec_dai->codec;
- struct davinci_vc *davinci_vc = codec->dev->platform_data;
-
switch (freq) {
case 22579200:
case 27000000:
case 33868800:
- davinci_vc->cq93vc.sysclk = freq;
return 0;
}
@@ -126,32 +122,6 @@ static struct snd_soc_dai_driver cq93vc_dai = {
.ops = &cq93vc_dai_ops,
};
-static int cq93vc_resume(struct snd_soc_codec *codec)
-{
- cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int cq93vc_probe(struct snd_soc_codec *codec)
-{
- struct davinci_vc *davinci_vc = codec->dev->platform_data;
-
- davinci_vc->cq93vc.codec = codec;
-
- /* Off, with power on */
- cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int cq93vc_remove(struct snd_soc_codec *codec)
-{
- cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
static struct regmap *cq93vc_get_regmap(struct device *dev)
{
struct davinci_vc *davinci_vc = dev->platform_data;
@@ -161,9 +131,6 @@ static struct regmap *cq93vc_get_regmap(struct device *dev)
static struct snd_soc_codec_driver soc_codec_dev_cq93vc = {
.set_bias_level = cq93vc_set_bias_level,
- .probe = cq93vc_probe,
- .remove = cq93vc_remove,
- .resume = cq93vc_resume,
.get_regmap = cq93vc_get_regmap,
.controls = cq93vc_snd_controls,
.num_controls = ARRAY_SIZE(cq93vc_snd_controls),
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 4fdd47d700e3..ce6086835ebd 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -32,7 +32,6 @@
#include "cs4265.h"
struct cs4265_private {
- struct device *dev;
struct regmap *regmap;
struct gpio_desc *reset_gpio;
u8 format;
@@ -598,7 +597,6 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client,
GFP_KERNEL);
if (cs4265 == NULL)
return -ENOMEM;
- cs4265->dev = &i2c_client->dev;
cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap);
if (IS_ERR(cs4265->regmap)) {
diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c
new file mode 100644
index 000000000000..b264da030340
--- /dev/null
+++ b/sound/soc/codecs/cs4271-i2c.c
@@ -0,0 +1,62 @@
+/*
+ * CS4271 I2C audio driver
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include "cs4271.h"
+
+static int cs4271_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap_config config;
+
+ config = cs4271_regmap_config;
+ config.reg_bits = 8;
+ config.val_bits = 8;
+
+ return cs4271_probe(&client->dev,
+ devm_regmap_init_i2c(client, &config));
+}
+
+static int cs4271_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id cs4271_i2c_id[] = {
+ { "cs4271", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
+
+static struct i2c_driver cs4271_i2c_driver = {
+ .driver = {
+ .name = "cs4271",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(cs4271_dt_ids),
+ },
+ .probe = cs4271_i2c_probe,
+ .remove = cs4271_i2c_remove,
+ .id_table = cs4271_i2c_id,
+};
+module_i2c_driver(cs4271_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS4271 I2C Driver");
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4271-spi.c b/sound/soc/codecs/cs4271-spi.c
new file mode 100644
index 000000000000..acd49d86e706
--- /dev/null
+++ b/sound/soc/codecs/cs4271-spi.c
@@ -0,0 +1,55 @@
+/*
+ * CS4271 SPI audio driver
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include "cs4271.h"
+
+static int cs4271_spi_probe(struct spi_device *spi)
+{
+ struct regmap_config config;
+
+ config = cs4271_regmap_config;
+ config.reg_bits = 16;
+ config.val_bits = 8;
+ config.read_flag_mask = 0x21;
+ config.write_flag_mask = 0x20;
+
+ return cs4271_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+}
+
+static int cs4271_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver cs4271_spi_driver = {
+ .driver = {
+ .name = "cs4271",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(cs4271_dt_ids),
+ },
+ .probe = cs4271_spi_probe,
+ .remove = cs4271_spi_remove,
+};
+module_spi_driver(cs4271_spi_driver);
+
+MODULE_DESCRIPTION("ASoC CS4271 SPI Driver");
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 93cec52f4733..79a4efcb894c 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -23,8 +23,6 @@
#include <linux/slab.h>
#include <linux/delay.h>
#include <linux/gpio.h>
-#include <linux/i2c.h>
-#include <linux/spi/spi.h>
#include <linux/of.h>
#include <linux/of_device.h>
#include <linux/of_gpio.h>
@@ -32,6 +30,7 @@
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/cs4271.h>
+#include "cs4271.h"
#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
@@ -527,14 +526,15 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec)
#endif /* CONFIG_PM */
#ifdef CONFIG_OF
-static const struct of_device_id cs4271_dt_ids[] = {
+const struct of_device_id cs4271_dt_ids[] = {
{ .compatible = "cirrus,cs4271", },
{ }
};
MODULE_DEVICE_TABLE(of, cs4271_dt_ids);
+EXPORT_SYMBOL_GPL(cs4271_dt_ids);
#endif
-static int cs4271_probe(struct snd_soc_codec *codec)
+static int cs4271_codec_probe(struct snd_soc_codec *codec)
{
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
@@ -587,7 +587,7 @@ static int cs4271_probe(struct snd_soc_codec *codec)
return 0;
}
-static int cs4271_remove(struct snd_soc_codec *codec)
+static int cs4271_codec_remove(struct snd_soc_codec *codec)
{
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
@@ -599,8 +599,8 @@ static int cs4271_remove(struct snd_soc_codec *codec)
};
static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
- .probe = cs4271_probe,
- .remove = cs4271_remove,
+ .probe = cs4271_codec_probe,
+ .remove = cs4271_codec_remove,
.suspend = cs4271_soc_suspend,
.resume = cs4271_soc_resume,
@@ -642,14 +642,8 @@ static int cs4271_common_probe(struct device *dev,
return 0;
}
-#if defined(CONFIG_SPI_MASTER)
-
-static const struct regmap_config cs4271_spi_regmap = {
- .reg_bits = 16,
- .val_bits = 8,
+const struct regmap_config cs4271_regmap_config = {
.max_register = CS4271_LASTREG,
- .read_flag_mask = 0x21,
- .write_flag_mask = 0x20,
.reg_defaults = cs4271_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults),
@@ -657,140 +651,27 @@ static const struct regmap_config cs4271_spi_regmap = {
.volatile_reg = cs4271_volatile_reg,
};
+EXPORT_SYMBOL_GPL(cs4271_regmap_config);
-static int cs4271_spi_probe(struct spi_device *spi)
+int cs4271_probe(struct device *dev, struct regmap *regmap)
{
struct cs4271_private *cs4271;
int ret;
- ret = cs4271_common_probe(&spi->dev, &cs4271);
- if (ret < 0)
- return ret;
-
- spi_set_drvdata(spi, cs4271);
- cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap);
- if (IS_ERR(cs4271->regmap))
- return PTR_ERR(cs4271->regmap);
-
- return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271,
- &cs4271_dai, 1);
-}
-
-static int cs4271_spi_remove(struct spi_device *spi)
-{
- snd_soc_unregister_codec(&spi->dev);
- return 0;
-}
-
-static struct spi_driver cs4271_spi_driver = {
- .driver = {
- .name = "cs4271",
- .owner = THIS_MODULE,
- .of_match_table = of_match_ptr(cs4271_dt_ids),
- },
- .probe = cs4271_spi_probe,
- .remove = cs4271_spi_remove,
-};
-#endif /* defined(CONFIG_SPI_MASTER) */
-
-#if IS_ENABLED(CONFIG_I2C)
-static const struct i2c_device_id cs4271_i2c_id[] = {
- {"cs4271", 0},
- {}
-};
-MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
-static const struct regmap_config cs4271_i2c_regmap = {
- .reg_bits = 8,
- .val_bits = 8,
- .max_register = CS4271_LASTREG,
-
- .reg_defaults = cs4271_reg_defaults,
- .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults),
- .cache_type = REGCACHE_RBTREE,
-
- .volatile_reg = cs4271_volatile_reg,
-};
-
-static int cs4271_i2c_probe(struct i2c_client *client,
- const struct i2c_device_id *id)
-{
- struct cs4271_private *cs4271;
- int ret;
-
- ret = cs4271_common_probe(&client->dev, &cs4271);
+ ret = cs4271_common_probe(dev, &cs4271);
if (ret < 0)
return ret;
- i2c_set_clientdata(client, cs4271);
- cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap);
- if (IS_ERR(cs4271->regmap))
- return PTR_ERR(cs4271->regmap);
+ dev_set_drvdata(dev, cs4271);
+ cs4271->regmap = regmap;
- return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271,
- &cs4271_dai, 1);
-}
-
-static int cs4271_i2c_remove(struct i2c_client *client)
-{
- snd_soc_unregister_codec(&client->dev);
- return 0;
-}
-
-static struct i2c_driver cs4271_i2c_driver = {
- .driver = {
- .name = "cs4271",
- .owner = THIS_MODULE,
- .of_match_table = of_match_ptr(cs4271_dt_ids),
- },
- .id_table = cs4271_i2c_id,
- .probe = cs4271_i2c_probe,
- .remove = cs4271_i2c_remove,
-};
-#endif /* IS_ENABLED(CONFIG_I2C) */
-
-/*
- * We only register our serial bus driver here without
- * assignment to particular chip. So if any of the below
- * fails, there is some problem with I2C or SPI subsystem.
- * In most cases this module will be compiled with support
- * of only one serial bus.
- */
-static int __init cs4271_modinit(void)
-{
- int ret;
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&cs4271_i2c_driver);
- if (ret) {
- pr_err("Failed to register CS4271 I2C driver: %d\n", ret);
- return ret;
- }
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&cs4271_spi_driver);
- if (ret) {
- pr_err("Failed to register CS4271 SPI driver: %d\n", ret);
- return ret;
- }
-#endif
-
- return 0;
-}
-module_init(cs4271_modinit);
-
-static void __exit cs4271_modexit(void)
-{
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&cs4271_spi_driver);
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&cs4271_i2c_driver);
-#endif
+ return snd_soc_register_codec(dev, &soc_codec_dev_cs4271, &cs4271_dai,
+ 1);
}
-module_exit(cs4271_modexit);
+EXPORT_SYMBOL_GPL(cs4271_probe);
MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver");
diff --git a/sound/soc/codecs/cs4271.h b/sound/soc/codecs/cs4271.h
new file mode 100644
index 000000000000..9adad8eefdc9
--- /dev/null
+++ b/sound/soc/codecs/cs4271.h
@@ -0,0 +1,11 @@
+#ifndef _CS4271_PRIV_H
+#define _CS4271_PRIV_H
+
+#include <linux/regmap.h>
+
+extern const struct of_device_id cs4271_dt_ids[];
+extern const struct regmap_config cs4271_regmap_config;
+
+int cs4271_probe(struct device *dev, struct regmap *regmap);
+
+#endif
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 669c38fc3034..b3951524339f 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -153,15 +153,17 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
switch (event) {
case SND_SOC_DAPM_PRE_PMD:
- snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1,
+ snd_soc_update_bits(codec, CS42L51_POWER_CTL1,
CS42L51_POWER_CTL1_PDN,
CS42L51_POWER_CTL1_PDN);
break;
default:
case SND_SOC_DAPM_POST_PMD:
- snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1,
+ snd_soc_update_bits(codec, CS42L51_POWER_CTL1,
CS42L51_POWER_CTL1_PDN, 0);
break;
}
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 2f8b94683e83..7c55537c69cf 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -584,7 +584,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMD:
@@ -600,7 +600,7 @@ static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w,
static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMD:
@@ -618,7 +618,7 @@ static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w,
static int cs42l73_hp_amp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMD:
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
index 1087fd5f9917..1391ad50f95d 100644
--- a/sound/soc/codecs/hdmi.c
+++ b/sound/soc/codecs/hdmi.c
@@ -47,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = {
SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
},
.capture = {
.stream_name = "Capture",
@@ -75,6 +76,7 @@ static struct snd_soc_codec_driver hdmi_codec = {
.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
.dapm_routes = hdmi_routes,
.num_dapm_routes = ARRAY_SIZE(hdmi_routes),
+ .ignore_pmdown_time = true,
};
static int hdmi_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index c1ae5764983f..c4dfde9bdf1c 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1395,15 +1395,7 @@ static struct snd_soc_dai_driver lm49453_dai[] = {
},
};
-/* power down chip */
-static int lm49453_remove(struct snd_soc_codec *codec)
-{
- lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
- .remove = lm49453_remove,
.set_bias_level = lm49453_set_bias_level,
.controls = lm49453_snd_controls,
.num_controls = ARRAY_SIZE(lm49453_snd_controls),
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 2cd3e5427441..805b3f8cd39d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -875,7 +875,7 @@ static const struct snd_kcontrol_new max98088_right_ADC_mixer_controls[] = {
static int max98088_mic_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -905,7 +905,7 @@ static int max98088_mic_event(struct snd_soc_dapm_widget *w,
static int max98088_line_pga(struct snd_soc_dapm_widget *w,
int event, int line, u8 channel)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
u8 *state;
@@ -1887,25 +1887,6 @@ static void max98088_handle_pdata(struct snd_soc_codec *codec)
max98088_handle_eq_pdata(codec);
}
-#ifdef CONFIG_PM
-static int max98088_suspend(struct snd_soc_codec *codec)
-{
- max98088_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int max98088_resume(struct snd_soc_codec *codec)
-{
- max98088_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define max98088_suspend NULL
-#define max98088_resume NULL
-#endif
-
static int max98088_probe(struct snd_soc_codec *codec)
{
struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
@@ -1946,9 +1927,6 @@ static int max98088_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV);
- /* initialize registers cache to hardware default */
- max98088_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
snd_soc_write(codec, M98088_REG_0F_IRQ_ENABLE, 0x00);
snd_soc_write(codec, M98088_REG_22_MIX_DAC,
@@ -1974,7 +1952,6 @@ static int max98088_remove(struct snd_soc_codec *codec)
{
struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
- max98088_set_bias_level(codec, SND_SOC_BIAS_OFF);
kfree(max98088->eq_texts);
return 0;
@@ -1983,9 +1960,9 @@ static int max98088_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_max98088 = {
.probe = max98088_probe,
.remove = max98088_remove,
- .suspend = max98088_suspend,
- .resume = max98088_resume,
.set_bias_level = max98088_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = max98088_snd_controls,
.num_controls = ARRAY_SIZE(max98088_snd_controls),
.dapm_widgets = max98088_dapm_widgets,
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 1229554f1464..151f718241ea 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -806,7 +806,7 @@ static const struct snd_kcontrol_new max98091_snd_controls[] = {
static int max98090_micinput_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
unsigned int val = snd_soc_read(codec, w->reg);
@@ -1311,6 +1311,10 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"MIC1 Input", NULL, "MIC1"},
{"MIC2 Input", NULL, "MIC2"},
+ {"DMICL", NULL, "DMICL_ENA"},
+ {"DMICL", NULL, "DMICR_ENA"},
+ {"DMICR", NULL, "DMICL_ENA"},
+ {"DMICR", NULL, "DMICR_ENA"},
{"DMICL", NULL, "AHPF"},
{"DMICR", NULL, "AHPF"},
@@ -1368,8 +1372,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"DMIC Mux", "ADC", "ADCR"},
{"DMIC Mux", "DMIC", "DMICL"},
{"DMIC Mux", "DMIC", "DMICR"},
- {"DMIC Mux", "DMIC", "DMICL_ENA"},
- {"DMIC Mux", "DMIC", "DMICR_ENA"},
{"LBENL Mux", "Normal", "DMIC Mux"},
{"LBENL Mux", "Loopback", "LTENL Mux"},
@@ -1395,8 +1397,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"STENL Mux", "Sidetone Left", "DMICL"},
{"STENR Mux", "Sidetone Right", "ADCR"},
{"STENR Mux", "Sidetone Right", "DMICR"},
- {"DACL", "NULL", "STENL Mux"},
- {"DACR", "NULL", "STENL Mux"},
+ {"DACL", NULL, "STENL Mux"},
+ {"DACR", NULL, "STENR Mux"},
{"AIFINL", NULL, "SHDN"},
{"AIFINR", NULL, "SHDN"},
@@ -1826,27 +1828,155 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-static const int comp_pclk_rates[] = {
- 11289600, 12288000, 12000000, 13000000, 19200000
-};
-
-static const int dmic_micclk[] = {
- 2, 2, 2, 2, 4, 2
-};
+static const int dmic_divisors[] = { 2, 3, 4, 5, 6, 8 };
static const int comp_lrclk_rates[] = {
8000, 16000, 32000, 44100, 48000, 96000
};
-static const int dmic_comp[6][6] = {
- {7, 8, 3, 3, 3, 3},
- {7, 8, 3, 3, 3, 3},
- {7, 8, 3, 3, 3, 3},
- {7, 8, 3, 1, 1, 1},
- {7, 8, 3, 1, 2, 2},
- {7, 8, 3, 3, 3, 3}
+struct dmic_table {
+ int pclk;
+ struct {
+ int freq;
+ int comp[6]; /* One each for 8, 16, 32, 44.1, 48, and 96 kHz */
+ } settings[6]; /* One for each dmic divisor. */
};
+static const struct dmic_table dmic_table[] = { /* One for each pclk freq. */
+ {
+ .pclk = 11289600,
+ .settings = {
+ { .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 6, 6, 6, 6 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ },
+ },
+ {
+ .pclk = 12000000,
+ .settings = {
+ { .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ }
+ },
+ {
+ .pclk = 12288000,
+ .settings = {
+ { .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 6, 6, 6, 6 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+ }
+ },
+ {
+ .pclk = 13000000,
+ .settings = {
+ { .freq = 2, .comp = { 7, 8, 1, 1, 1, 1 } },
+ { .freq = 1, .comp = { 7, 8, 0, 0, 0, 0 } },
+ { .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } },
+ { .freq = 0, .comp = { 7, 8, 4, 4, 5, 5 } },
+ { .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } },
+ { .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } },
+ }
+ },
+ {
+ .pclk = 19200000,
+ .settings = {
+ { .freq = 2, .comp = { 0, 0, 0, 0, 0, 0 } },
+ { .freq = 1, .comp = { 7, 8, 1, 1, 1, 1 } },
+ { .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } },
+ { .freq = 0, .comp = { 7, 8, 2, 2, 3, 3 } },
+ { .freq = 0, .comp = { 7, 8, 1, 1, 2, 2 } },
+ { .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } },
+ }
+ },
+};
+
+static int max98090_find_divisor(int target_freq, int pclk)
+{
+ int current_diff = INT_MAX;
+ int test_diff = INT_MAX;
+ int divisor_index = 0;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(dmic_divisors); i++) {
+ test_diff = abs(target_freq - (pclk / dmic_divisors[i]));
+ if (test_diff < current_diff) {
+ current_diff = test_diff;
+ divisor_index = i;
+ }
+ }
+
+ return divisor_index;
+}
+
+static int max98090_find_closest_pclk(int pclk)
+{
+ int m1;
+ int m2;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(dmic_table); i++) {
+ if (pclk == dmic_table[i].pclk)
+ return i;
+ if (pclk < dmic_table[i].pclk) {
+ if (i == 0)
+ return i;
+ m1 = pclk - dmic_table[i-1].pclk;
+ m2 = dmic_table[i].pclk - pclk;
+ if (m1 < m2)
+ return i - 1;
+ else
+ return i;
+ }
+ }
+
+ return -EINVAL;
+}
+
+static int max98090_configure_dmic(struct max98090_priv *max98090,
+ int target_dmic_clk, int pclk, int fs)
+{
+ int micclk_index;
+ int pclk_index;
+ int dmic_freq;
+ int dmic_comp;
+ int i;
+
+ pclk_index = max98090_find_closest_pclk(pclk);
+ if (pclk_index < 0)
+ return pclk_index;
+
+ micclk_index = max98090_find_divisor(target_dmic_clk, pclk);
+
+ for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) {
+ if (fs <= (comp_lrclk_rates[i] + comp_lrclk_rates[i+1]) / 2)
+ break;
+ }
+
+ dmic_freq = dmic_table[pclk_index].settings[micclk_index].freq;
+ dmic_comp = dmic_table[pclk_index].settings[micclk_index].comp[i];
+
+ regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_ENABLE,
+ M98090_MICCLK_MASK,
+ micclk_index << M98090_MICCLK_SHIFT);
+
+ regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_CONFIG,
+ M98090_DMIC_COMP_MASK | M98090_DMIC_FREQ_MASK,
+ dmic_comp << M98090_DMIC_COMP_SHIFT |
+ dmic_freq << M98090_DMIC_FREQ_SHIFT);
+
+ return 0;
+}
+
static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1854,7 +1984,6 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
struct max98090_cdata *cdata;
- int i, j;
cdata = &max98090->dai[0];
max98090->bclk = snd_soc_params_to_bclk(params);
@@ -1893,27 +2022,8 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG,
M98090_DHF_MASK, M98090_DHF_MASK);
- /* Check for supported PCLK to LRCLK ratios */
- for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) {
- if (comp_pclk_rates[j] == max98090->sysclk) {
- break;
- }
- }
-
- for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) {
- if (max98090->lrclk <= (comp_lrclk_rates[i] +
- comp_lrclk_rates[i + 1]) / 2) {
- break;
- }
- }
-
- snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE,
- M98090_MICCLK_MASK,
- dmic_micclk[j] << M98090_MICCLK_SHIFT);
-
- snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG,
- M98090_DMIC_COMP_MASK,
- dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT);
+ max98090_configure_dmic(max98090, max98090->dmic_freq, max98090->pclk,
+ max98090->lrclk);
return 0;
}
@@ -1944,12 +2054,15 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai,
if ((freq >= 10000000) && (freq <= 20000000)) {
snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK,
M98090_PSCLK_DIV1);
+ max98090->pclk = freq;
} else if ((freq > 20000000) && (freq <= 40000000)) {
snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK,
M98090_PSCLK_DIV2);
+ max98090->pclk = freq >> 1;
} else if ((freq > 40000000) && (freq <= 60000000)) {
snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK,
M98090_PSCLK_DIV4);
+ max98090->pclk = freq >> 2;
} else {
dev_err(codec->dev, "Invalid master clock frequency\n");
return -EINVAL;
@@ -2324,6 +2437,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
/* Initialize private data */
max98090->sysclk = (unsigned)-1;
+ max98090->pclk = (unsigned)-1;
max98090->master = false;
cdata = &max98090->dai[0];
@@ -2463,6 +2577,11 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, max98090);
max98090->pdata = i2c->dev.platform_data;
+ ret = of_property_read_u32(i2c->dev.of_node, "maxim,dmic-freq",
+ &max98090->dmic_freq);
+ if (ret < 0)
+ max98090->dmic_freq = MAX98090_DEFAULT_DMIC_FREQ;
+
max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap);
if (IS_ERR(max98090->regmap)) {
ret = PTR_ERR(max98090->regmap);
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index a5f6bada06da..21ff743f5af2 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -12,6 +12,12 @@
#define _MAX98090_H
/*
+ * The default operating frequency for a DMIC attached to the codec.
+ * This can be overridden by a device tree property.
+ */
+#define MAX98090_DEFAULT_DMIC_FREQ 2500000
+
+/*
* MAX98090 Register Definitions
*/
@@ -1518,8 +1524,10 @@ struct max98090_priv {
struct max98090_pdata *pdata;
struct clk *mclk;
unsigned int sysclk;
+ unsigned int pclk;
unsigned int bclk;
unsigned int lrclk;
+ u32 dmic_freq;
struct max98090_cdata dai[1];
int jack_state;
struct delayed_work jack_work;
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 0ee6797d5083..8fba0c3db798 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -16,6 +16,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/clk.h>
+#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -57,6 +58,7 @@ struct max98095_priv {
unsigned int mic2pre;
struct snd_soc_jack *headphone_jack;
struct snd_soc_jack *mic_jack;
+ struct mutex lock;
};
static const struct reg_default max98095_reg_def[] = {
@@ -864,7 +866,7 @@ static const struct snd_kcontrol_new max98095_right_ADC_mixer_controls[] = {
static int max98095_mic_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
switch (event) {
@@ -894,7 +896,7 @@ static int max98095_mic_event(struct snd_soc_dapm_widget *w,
static int max98095_line_pga(struct snd_soc_dapm_widget *w,
int event, u8 channel)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
u8 *state;
@@ -942,7 +944,7 @@ static int max98095_pga_in2_event(struct snd_soc_dapm_widget *w,
static int max98095_lineout_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- struct snd_soc_codec *codec = w->codec;
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -1803,7 +1805,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
regsave = snd_soc_read(codec, M98095_088_CFG_LEVEL);
snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, 0);
- mutex_lock(&codec->mutex);
+ mutex_lock(&max98095->lock);
snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, M98095_SEG);
m98095_eq_band(codec, channel, 0, coef_set->band1);
m98095_eq_band(codec, channel, 1, coef_set->band2);
@@ -1811,7 +1813,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
m98095_eq_band(codec, channel, 3, coef_set->band4);
m98095_eq_band(codec, channel, 4, coef_set->band5);
snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, 0);
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&max98095->lock);
/* Restore the original on/off state */
snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, regsave);
@@ -1957,12 +1959,12 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
regsave = snd_soc_read(codec, M98095_088_CFG_LEVEL);
snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, 0);
- mutex_lock(&codec->mutex);
+ mutex_lock(&max98095->lock);
snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, M98095_SEG);
m98095_biquad_band(codec, channel, 0, coef_set->band1);
m98095_biquad_band(codec, channel, 1, coef_set->band2);
snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, 0);
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&max98095->lock);
/* Restore the original on/off state */
snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, regsave);
@@ -2317,9 +2319,6 @@ static int max98095_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV);
- /* initialize registers cache to hardware default */
- max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
snd_soc_write(codec, M98095_048_MIX_DAC_LR,
M98095_DAI1L_TO_DACL|M98095_DAI1R_TO_DACR);
@@ -2359,8 +2358,6 @@ static int max98095_remove(struct snd_soc_codec *codec)
struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
struct i2c_client *client = to_i2c_client(codec->dev);
- max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
if (max98095->headphone_jack || max98095->mic_jack)
max98095_jack_detect_disable(codec);
@@ -2395,6 +2392,8 @@ static int max98095_i2c_probe(struct i2c_client *i2c,
if (max98095 == NULL)
return -ENOMEM;
+ mutex_init(&max98095->lock);
+
max98095->regmap = devm_regmap_init_i2c(i2c, &max98095_regmap);
if (IS_ERR(max98095->regmap)) {
ret = PTR_ERR(max98095->regmap);
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 4fdf5aaa236f..10f8e47ce2c2 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -291,25 +291,6 @@ static struct snd_soc_dai_driver max9850_dai = {
.ops = &max9850_dai_ops,
};
-#ifdef CONFIG_PM
-static int max9850_suspend(struct snd_soc_codec *codec)
-{
- max9850_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int max9850_resume(struct snd_soc_codec *codec)
-{
- max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define max9850_suspend NULL
-#define max9850_resume NULL
-#endif
-
static int max9850_probe(struct snd_soc_codec *codec)
{
/* enable zero-detect */
@@ -324,9 +305,8 @@ static int max9850_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
.probe = max9850_probe,
- .suspend = max9850_suspend,
- .resume = max9850_resume,
.set_bias_level = max9850_set_bias_level,
+ .suspend_bias_off = true,
.controls = max9850_controls,
.num_controls = ARRAY_SIZE(max9850_controls),
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 4aa555cbcca8..2cd4fe463102 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -17,6 +17,7 @@
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
+#include <linux/dmi.h>
#include <linux/acpi.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -36,11 +37,13 @@
struct rt286_priv {
struct regmap *regmap;
+ struct snd_soc_codec *codec;
struct rt286_platform_data pdata;
struct i2c_client *i2c;
struct snd_soc_jack *jack;
struct delayed_work jack_detect_work;
int sys_clk;
+ int clk_id;
struct reg_default *index_cache;
};
@@ -188,7 +191,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
u8 data[4];
int ret, i;
- /*handle index registers*/
+ /* handle index registers */
if (reg <= 0xff) {
rt286_hw_write(client, RT286_COEF_INDEX, reg);
for (i = 0; i < INDEX_CACHE_SIZE; i++) {
@@ -231,7 +234,7 @@ static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value)
__be32 be_reg;
unsigned int index, vid, buf = 0x0;
- /*handle index registers*/
+ /* handle index registers */
if (reg <= 0xff) {
rt286_hw_write(client, RT286_COEF_INDEX, reg);
reg = RT286_PROC_COEF;
@@ -298,7 +301,6 @@ static int rt286_support_power_controls[] = {
static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
{
unsigned int val, buf;
- int i;
*hp = false;
*mic = false;
@@ -309,67 +311,44 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
if (*hp) {
/* power on HV,VERF */
regmap_update_bits(rt286->regmap,
- RT286_POWER_CTRL1, 0x1001, 0x0);
+ RT286_DC_GAIN, 0x200, 0x200);
+
+ snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
+ "HV");
+ snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
+ "VREF");
/* power LDO1 */
- regmap_update_bits(rt286->regmap,
- RT286_POWER_CTRL2, 0x4, 0x4);
- regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24);
- regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val);
+ snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
+ "LDO1");
+ snd_soc_dapm_sync(&rt286->codec->dapm);
- msleep(200);
- i = 40;
- while (((val & 0x0800) == 0) && (i > 0)) {
- regmap_read(rt286->regmap,
- RT286_CBJ_CTRL2, &val);
- i--;
- msleep(20);
- }
+ regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24);
+ msleep(50);
- if (0x0400 == (val & 0x0700)) {
- *mic = false;
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0xfcc0, 0xd400);
+ msleep(300);
+ regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val);
- regmap_write(rt286->regmap,
- RT286_SET_MIC1, 0x20);
- /* power off HV,VERF */
- regmap_update_bits(rt286->regmap,
- RT286_POWER_CTRL1, 0x1001, 0x1001);
- regmap_update_bits(rt286->regmap,
- RT286_A_BIAS_CTRL3, 0xc000, 0x0000);
- regmap_update_bits(rt286->regmap,
- RT286_CBJ_CTRL1, 0x0030, 0x0000);
- regmap_update_bits(rt286->regmap,
- RT286_A_BIAS_CTRL2, 0xc000, 0x0000);
- } else if ((0x0200 == (val & 0x0700)) ||
- (0x0100 == (val & 0x0700))) {
+ if (0x0070 == (val & 0x0070)) {
*mic = true;
- regmap_update_bits(rt286->regmap,
- RT286_A_BIAS_CTRL3, 0xc000, 0x8000);
- regmap_update_bits(rt286->regmap,
- RT286_CBJ_CTRL1, 0x0030, 0x0020);
- regmap_update_bits(rt286->regmap,
- RT286_A_BIAS_CTRL2, 0xc000, 0x8000);
} else {
- *mic = false;
+ regmap_update_bits(rt286->regmap,
+ RT286_CBJ_CTRL1, 0xfcc0, 0xe400);
+ msleep(300);
+ regmap_read(rt286->regmap,
+ RT286_CBJ_CTRL2, &val);
+ if (0x0070 == (val & 0x0070))
+ *mic = true;
+ else
+ *mic = false;
}
-
- regmap_update_bits(rt286->regmap,
- RT286_MISC_CTRL1,
- 0x0060, 0x0000);
- } else {
- regmap_update_bits(rt286->regmap,
- RT286_MISC_CTRL1,
- 0x0060, 0x0020);
regmap_update_bits(rt286->regmap,
- RT286_A_BIAS_CTRL3,
- 0xc000, 0x8000);
- regmap_update_bits(rt286->regmap,
- RT286_CBJ_CTRL1,
- 0x0030, 0x0020);
- regmap_update_bits(rt286->regmap,
- RT286_A_BIAS_CTRL2,
- 0xc000, 0x8000);
+ RT286_DC_GAIN, 0x200, 0x0);
+ } else {
*mic = false;
+ regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20);
}
} else {
regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
@@ -378,6 +357,12 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
*mic = buf & 0x80000000;
}
+ snd_soc_dapm_disable_pin(&rt286->codec->dapm, "HV");
+ snd_soc_dapm_disable_pin(&rt286->codec->dapm, "VREF");
+ if (!*hp)
+ snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1");
+ snd_soc_dapm_sync(&rt286->codec->dapm);
+
return 0;
}
@@ -415,6 +400,17 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
}
EXPORT_SYMBOL_GPL(rt286_mic_detect);
+static int is_mclk_mode(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(source->codec);
+
+ if (rt286->clk_id == RT286_SCLK_S_MCLK)
+ return 1;
+ else
+ return 0;
+}
+
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0);
static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0);
@@ -568,7 +564,84 @@ static int rt286_adc_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int rt286_vref_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec,
+ RT286_CBJ_CTRL1, 0x0400, 0x0000);
+ mdelay(50);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt286_ldo2_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x08);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x30);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt286_mic1_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL3, 0xc000, 0x8000);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL2, 0xc000, 0x8000);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL3, 0xc000, 0x0000);
+ snd_soc_update_bits(codec,
+ RT286_A_BIAS_CTRL2, 0xc000, 0x0000);
+ break;
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1,
+ 12, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1,
+ 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2,
+ 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1,
+ 13, 1, rt286_ldo2_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SUPPLY("MCLK MODE", RT286_PLL_CTRL1,
+ 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM,
+ 0, 0, rt286_mic1_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
/* Input Lines */
SND_SOC_DAPM_INPUT("DMIC1 Pin"),
SND_SOC_DAPM_INPUT("DMIC2 Pin"),
@@ -642,6 +715,25 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route rt286_dapm_routes[] = {
+ {"ADC 0", NULL, "MCLK MODE", is_mclk_mode},
+ {"ADC 1", NULL, "MCLK MODE", is_mclk_mode},
+ {"Front", NULL, "MCLK MODE", is_mclk_mode},
+ {"Surround", NULL, "MCLK MODE", is_mclk_mode},
+
+ {"HP Power", NULL, "LDO1"},
+ {"HP Power", NULL, "LDO2"},
+
+ {"MIC1", NULL, "LDO1"},
+ {"MIC1", NULL, "LDO2"},
+ {"MIC1", NULL, "HV"},
+ {"MIC1", NULL, "VREF"},
+ {"MIC1", NULL, "MIC1 Input Buffer"},
+
+ {"SPO", NULL, "LDO1"},
+ {"SPO", NULL, "LDO2"},
+ {"SPO", NULL, "HV"},
+ {"SPO", NULL, "VREF"},
+
{"DMIC1", NULL, "DMIC1 Pin"},
{"DMIC2", NULL, "DMIC2 Pin"},
{"DMIC1", NULL, "DMIC Receiver"},
@@ -880,6 +972,7 @@ static int rt286_set_dai_sysclk(struct snd_soc_dai *dai,
}
rt286->sys_clk = freq;
+ rt286->clk_id = clk_id;
return 0;
}
@@ -915,13 +1008,18 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_ON:
mdelay(10);
+ snd_soc_update_bits(codec,
+ RT286_CBJ_CTRL1, 0x0400, 0x0400);
+ snd_soc_update_bits(codec,
+ RT286_DC_GAIN, 0x200, 0x0);
+
break;
case SND_SOC_BIAS_STANDBY:
snd_soc_write(codec,
RT286_SET_AUDIO_POWER, AC_PWRST_D3);
snd_soc_update_bits(codec,
- RT286_DC_GAIN, 0x200, 0x0);
+ RT286_CBJ_CTRL1, 0x0400, 0x0000);
break;
default:
@@ -962,6 +1060,7 @@ static int rt286_probe(struct snd_soc_codec *codec)
{
struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
+ rt286->codec = codec;
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
if (rt286->i2c->irq) {
@@ -1107,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = {
};
MODULE_DEVICE_TABLE(acpi, rt286_acpi_match);
+static struct dmi_system_id force_combo_jack_table[] = {
+ {
+ .ident = "Intel Wilson Beach",
+ .matches = {
+ DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS")
+ }
+ },
+ { }
+};
+
static int rt286_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1142,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt286->pdata = *pdata;
+ if (dmi_check_system(force_combo_jack_table))
+ rt286->pdata.cbj_en = true;
+
regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
for (i = 0; i < RT286_POWER_REG_LEN; i++)
@@ -1152,7 +1264,6 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
if (!rt286->pdata.cbj_en) {
regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000);
regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816);
- regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000);
regmap_update_bits(rt286->regmap,
RT286_CBJ_CTRL1, 0xf000, 0xb000);
} else {
@@ -1169,10 +1280,12 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
mdelay(10);
- /*Power down LDO2*/
- regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0);
+ regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000);
+ /* Power down LDO, VREF */
+ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0);
+ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001);
- /*Set depop parameter*/
+ /* Set depop parameter */
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a);
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 1ba27db660a6..6d7b7ca7d530 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1612,29 +1612,6 @@ static int rt5631_probe(struct snd_soc_codec *codec)
return 0;
}
-static int rt5631_remove(struct snd_soc_codec *codec)
-{
- rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-#ifdef CONFIG_PM
-static int rt5631_suspend(struct snd_soc_codec *codec)
-{
- rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int rt5631_resume(struct snd_soc_codec *codec)
-{
- rt5631_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define rt5631_suspend NULL
-#define rt5631_resume NULL
-#endif
-
#define RT5631_STEREO_RATES SNDRV_PCM_RATE_8000_96000
#define RT5631_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S20_3LE | \
@@ -1672,10 +1649,8 @@ static struct snd_soc_dai_driver rt5631_dai[] = {
static struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
.probe = rt5631_probe,
- .remove = rt5631_remove,
- .suspend = rt5631_suspend,
- .resume = rt5631_resume,
.set_bias_level = rt5631_set_bias_level,
+ .suspend_bias_off = true,
.controls = rt5631_snd_controls,
.num_controls = ARRAY_SIZE(rt5631_snd_controls),
.dapm_widgets = rt5631_dapm_widgets,
@@ -1686,10 +1661,20 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
static const struct i2c_device_id rt5631_i2c_id[] = {
{ "rt5631", 0 },
+ { "alc5631", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id);
+#ifdef CONFIG_OF
+static struct of_device_id rt5631_i2c_dt_ids[] = {
+ { .compatible = "realtek,rt5631"},
+ { .compatible = "realtek,alc5631"},
+ { }
+};
+MODULE_DEVICE_TABLE(of, rt5631_i2c_dt_ids);
+#endif
+
static const struct regmap_config rt5631_regmap_config = {
.reg_bits = 8,
.val_bits = 16,
@@ -1734,6 +1719,7 @@ static struct i2c_driver rt5631_i2c_driver = {
.driver = {
.name = "rt5631",
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(rt5631_i2c_dt_ids),
},
.probe = rt5631_i2c_probe,
.remove = rt5631_i2c_remove,
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index d16331e0b64d..a7789a8726e3 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -554,6 +554,53 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
return 0;
}
+static int is_using_asrc(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ unsigned int reg, shift, val;
+
+ switch (source->shift) {
+ case 0:
+ reg = RT5645_ASRC_3;
+ shift = 0;
+ break;
+ case 1:
+ reg = RT5645_ASRC_3;
+ shift = 4;
+ break;
+ case 3:
+ reg = RT5645_ASRC_2;
+ shift = 0;
+ break;
+ case 8:
+ reg = RT5645_ASRC_2;
+ shift = 4;
+ break;
+ case 9:
+ reg = RT5645_ASRC_2;
+ shift = 8;
+ break;
+ case 10:
+ reg = RT5645_ASRC_2;
+ shift = 12;
+ break;
+ default:
+ return 0;
+ }
+
+ val = (snd_soc_read(source->codec, reg) >> shift) & 0xf;
+ switch (val) {
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ return 1;
+ default:
+ return 0;
+ }
+
+}
+
/* Digital Mixer */
static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = {
SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER,
@@ -1246,6 +1293,30 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL,
RT5645_PWR_MIC_DET_BIT, 0, NULL, 0),
+ /* ASRC */
+ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5645_ASRC_1,
+ 11, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5645_ASRC_1,
+ 12, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5645_ASRC_1,
+ 10, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5645_ASRC_1,
+ 9, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5645_ASRC_1,
+ 8, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5645_ASRC_1,
+ 7, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5645_ASRC_1,
+ 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5645_ASRC_1,
+ 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5645_ASRC_1,
+ 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5645_ASRC_1,
+ 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5645_ASRC_1,
+ 0, 0, NULL, 0),
+
/* Input Side */
/* micbias */
SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2,
@@ -1504,6 +1575,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route rt5645_dapm_routes[] = {
+ { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc },
+ { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc },
+ { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc },
+ { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc },
+ { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc },
+ { "dac mono right filter", NULL, "DAC MONO R ASRC", is_using_asrc },
+ { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc },
+
+ { "I2S1", NULL, "I2S1 ASRC" },
+ { "I2S2", NULL, "I2S2 ASRC" },
+
{ "IN1P", NULL, "LDO2" },
{ "IN2P", NULL, "LDO2" },
@@ -1550,12 +1632,15 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = {
{ "Stereo1 DMIC Mux", "DMIC1", "DMIC1" },
{ "Stereo1 DMIC Mux", "DMIC2", "DMIC2" },
+ { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC" },
{ "Mono DMIC L Mux", "DMIC1", "DMIC L1" },
{ "Mono DMIC L Mux", "DMIC2", "DMIC L2" },
+ { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC" },
{ "Mono DMIC R Mux", "DMIC1", "DMIC R1" },
{ "Mono DMIC R Mux", "DMIC2", "DMIC R2" },
+ { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC" },
{ "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" },
{ "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" },
@@ -2029,8 +2114,11 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
struct snd_soc_codec *codec = dai->codec;
unsigned int val = 0;
- if (rx_mask || tx_mask)
+ if (rx_mask || tx_mask) {
val |= (1 << 14);
+ snd_soc_update_bits(codec, RT5645_BASS_BACK,
+ RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB);
+ }
switch (slots) {
case 4:
@@ -2071,8 +2159,8 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
- case SND_SOC_BIAS_STANDBY:
- if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
snd_soc_update_bits(codec, RT5645_PWR_ANLG1,
RT5645_PWR_VREF1 | RT5645_PWR_MB |
RT5645_PWR_BG | RT5645_PWR_VREF2,
@@ -2087,15 +2175,24 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
}
break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, RT5645_PWR_ANLG1,
+ RT5645_PWR_VREF1 | RT5645_PWR_MB |
+ RT5645_PWR_BG | RT5645_PWR_VREF2,
+ RT5645_PWR_VREF1 | RT5645_PWR_MB |
+ RT5645_PWR_BG | RT5645_PWR_VREF2);
+ snd_soc_update_bits(codec, RT5645_PWR_ANLG1,
+ RT5645_PWR_FV1 | RT5645_PWR_FV2,
+ RT5645_PWR_FV1 | RT5645_PWR_FV2);
+ break;
+
case SND_SOC_BIAS_OFF:
snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100);
snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128);
- snd_soc_write(codec, RT5645_PWR_DIG1, 0x0000);
- snd_soc_write(codec, RT5645_PWR_DIG2, 0x0000);
- snd_soc_write(codec, RT5645_PWR_VOL, 0x0000);
- snd_soc_write(codec, RT5645_PWR_MIXER, 0x0000);
- snd_soc_write(codec, RT5645_PWR_ANLG1, 0x0000);
- snd_soc_write(codec, RT5645_PWR_ANLG2, 0x0000);
+ snd_soc_update_bits(codec, RT5645_PWR_ANLG1,
+ RT5645_PWR_VREF1 | RT5645_PWR_MB |
+ RT5645_PWR_BG | RT5645_PWR_VREF2 |
+ RT5645_PWR_FV1 | RT5645_PWR_FV2, 0x0);
break;
default:
@@ -2106,8 +2203,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-static int rt5645_jack_detect(struct snd_soc_codec *codec,
- struct snd_soc_jack *jack)
+static int rt5645_jack_detect(struct snd_soc_codec *codec)
{
struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
int gpio_state, jack_type = 0;
@@ -2145,34 +2241,44 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec,
snd_soc_dapm_disable_pin(&codec->dapm, "micbias1");
snd_soc_dapm_disable_pin(&codec->dapm, "micbias2");
- snd_soc_dapm_disable_pin(&codec->dapm, "LDO2");
+ if (rt5645->pdata.jd_mode == 0)
+ snd_soc_dapm_disable_pin(&codec->dapm, "LDO2");
snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power");
snd_soc_dapm_sync(&codec->dapm);
}
- snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET);
-
+ snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE);
+ snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE);
return 0;
}
int rt5645_set_jack_detect(struct snd_soc_codec *codec,
- struct snd_soc_jack *jack)
+ struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack)
{
struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
- rt5645->jack = jack;
-
- rt5645_jack_detect(codec, rt5645->jack);
+ rt5645->hp_jack = hp_jack;
+ rt5645->mic_jack = mic_jack;
+ rt5645_jack_detect(codec);
return 0;
}
EXPORT_SYMBOL_GPL(rt5645_set_jack_detect);
+static void rt5645_jack_detect_work(struct work_struct *work)
+{
+ struct rt5645_priv *rt5645 =
+ container_of(work, struct rt5645_priv, jack_detect_work.work);
+
+ rt5645_jack_detect(rt5645->codec);
+}
+
static irqreturn_t rt5645_irq(int irq, void *data)
{
struct rt5645_priv *rt5645 = data;
- rt5645_jack_detect(rt5645->codec, rt5645->jack);
+ queue_delayed_work(system_power_efficient_wq,
+ &rt5645->jack_detect_work, msecs_to_jiffies(250));
return IRQ_HANDLED;
}
@@ -2187,6 +2293,13 @@ static int rt5645_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
+ /* for JD function */
+ if (rt5645->pdata.en_jd_func) {
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power");
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2");
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+
return 0;
}
@@ -2420,6 +2533,51 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
}
+ if (rt5645->pdata.en_jd_func) {
+ regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3,
+ RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU,
+ RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU);
+ regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1,
+ RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN);
+ regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3,
+ RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL,
+ RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL);
+ regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+ RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT);
+ }
+
+ if (rt5645->pdata.jd_mode) {
+ regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+ RT5645_IRQ_JD_1_1_EN, RT5645_IRQ_JD_1_1_EN);
+ regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3,
+ RT5645_JD_PSV_MODE, RT5645_JD_PSV_MODE);
+ regmap_update_bits(rt5645->regmap, RT5645_HPO_MIXER,
+ RT5645_IRQ_PSV_MODE, RT5645_IRQ_PSV_MODE);
+ regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+ RT5645_MIC2_OVCD_EN, RT5645_MIC2_OVCD_EN);
+ regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
+ RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
+ switch (rt5645->pdata.jd_mode) {
+ case 1:
+ regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1,
+ RT5645_JD1_MODE_MASK,
+ RT5645_JD1_MODE_0);
+ break;
+ case 2:
+ regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1,
+ RT5645_JD1_MODE_MASK,
+ RT5645_JD1_MODE_1);
+ break;
+ case 3:
+ regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1,
+ RT5645_JD1_MODE_MASK,
+ RT5645_JD1_MODE_2);
+ break;
+ default:
+ break;
+ }
+ }
+
if (rt5645->i2c->irq) {
ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq,
IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
@@ -2438,6 +2596,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n");
}
+ INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
+
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645,
rt5645_dai, ARRAY_SIZE(rt5645_dai));
}
@@ -2449,6 +2609,8 @@ static int rt5645_i2c_remove(struct i2c_client *i2c)
if (i2c->irq)
free_irq(i2c->irq, rt5645);
+ cancel_delayed_work_sync(&rt5645->jack_detect_work);
+
if (gpio_is_valid(rt5645->pdata.hp_det_gpio))
gpio_free(rt5645->pdata.hp_det_gpio);
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 50c62c5668ea..a815e36a2bdb 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -594,6 +594,7 @@
#define RT5645_M_DAC1_HM_SFT 14
#define RT5645_M_HPVOL_HM (0x1 << 13)
#define RT5645_M_HPVOL_HM_SFT 13
+#define RT5645_IRQ_PSV_MODE (0x1 << 12)
/* SPK Left Mixer Control (0x46) */
#define RT5645_G_RM_L_SM_L_MASK (0x3 << 14)
@@ -1348,6 +1349,12 @@
#define RT5645_PWR_CLK25M_SFT 4
#define RT5645_PWR_CLK25M_PD (0x0 << 4)
#define RT5645_PWR_CLK25M_PU (0x1 << 4)
+#define RT5645_IRQ_CLK_MCLK (0x0 << 3)
+#define RT5645_IRQ_CLK_INT (0x1 << 3)
+#define RT5645_JD1_MODE_MASK (0x3 << 0)
+#define RT5645_JD1_MODE_0 (0x0 << 0)
+#define RT5645_JD1_MODE_1 (0x1 << 0)
+#define RT5645_JD1_MODE_2 (0x2 << 0)
/* VAD Control 4 (0x9d) */
#define RT5645_VAD_SEL_MASK (0x3 << 8)
@@ -1636,6 +1643,7 @@
#define RT5645_OT_P_SFT 10
#define RT5645_OT_P_NOR (0x0 << 10)
#define RT5645_OT_P_INV (0x1 << 10)
+#define RT5645_IRQ_JD_1_1_EN (0x1 << 9)
/* IRQ Control 2 (0xbe) */
#define RT5645_IRQ_MB1_OC_MASK (0x1 << 15)
@@ -1853,6 +1861,7 @@
#define RT5645_M_BB_HPF_R_SFT 6
#define RT5645_G_BB_BST_MASK (0x3f)
#define RT5645_G_BB_BST_SFT 0
+#define RT5645_G_BB_BST_25DB 0x14
/* MP3 Plus Control 1 (0xd0) */
#define RT5645_M_MP3_L_MASK (0x1 << 15)
@@ -2116,6 +2125,10 @@ enum {
#define RT5645_RXDP2_SEL_ADC (0x1 << 3)
#define RT5645_RXDP2_SEL_SFT (3)
+/* General Control3 (0xfc) */
+#define RT5645_JD_PSV_MODE (0x1 << 12)
+#define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11)
+#define RT5645_MICINDET_MANU (0x1 << 7)
/* Vendor ID (0xfd) */
#define RT5645_VER_C 0x2
@@ -2167,7 +2180,9 @@ struct rt5645_priv {
struct rt5645_platform_data pdata;
struct regmap *regmap;
struct i2c_client *i2c;
- struct snd_soc_jack *jack;
+ struct snd_soc_jack *hp_jack;
+ struct snd_soc_jack *mic_jack;
+ struct delayed_work jack_detect_work;
int sysclk;
int sysclk_src;
@@ -2181,6 +2196,6 @@ struct rt5645_priv {
};
int rt5645_set_jack_detect(struct snd_soc_codec *codec,
- struct snd_soc_jack *jack);
+ struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack);
#endif /* __RT5645_H__ */
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 9bd8b4f63303..8a0833de1665 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -16,6 +16,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
+#include <linux/acpi.h>
#include <linux/spi/spi.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -575,6 +576,18 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source,
}
+static int can_use_asrc(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm);
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
+ if (rt5670->sysclk > rt5670->lrck[RT5670_AIF1] * 384)
+ return 1;
+
+ return 0;
+}
+
/* Digital Mixer */
static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = {
SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER,
@@ -1281,6 +1294,14 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
9, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1,
8, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5670_ASRC_1,
+ 7, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC STO2 ASRC", 1, RT5670_ASRC_1,
+ 6, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5670_ASRC_1,
+ 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5670_ASRC_1,
+ 4, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1,
3, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1,
@@ -1595,29 +1616,40 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
/* PDM */
SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2,
RT5670_PWR_PDM1_BIT, 0, NULL, 0),
- SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2,
- RT5670_PWR_PDM2_BIT, 0, NULL, 0),
SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL,
RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux),
SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL,
RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux),
- SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL,
- RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux),
- SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL,
- RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux),
/* Output Lines */
SND_SOC_DAPM_OUTPUT("HPOL"),
SND_SOC_DAPM_OUTPUT("HPOR"),
SND_SOC_DAPM_OUTPUT("LOUTL"),
SND_SOC_DAPM_OUTPUT("LOUTR"),
+};
+
+static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2,
+ RT5670_PWR_PDM2_BIT, 0, NULL, 0),
+ SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL,
+ RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux),
+ SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL,
+ RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux),
SND_SOC_DAPM_OUTPUT("PDM1L"),
SND_SOC_DAPM_OUTPUT("PDM1R"),
SND_SOC_DAPM_OUTPUT("PDM2L"),
SND_SOC_DAPM_OUTPUT("PDM2R"),
};
+static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = {
+ SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("SPOLP"),
+ SND_SOC_DAPM_OUTPUT("SPOLN"),
+ SND_SOC_DAPM_OUTPUT("SPORP"),
+ SND_SOC_DAPM_OUTPUT("SPORN"),
+};
+
static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
{ "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc },
{ "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc },
@@ -1626,9 +1658,13 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
{ "DAC Mono Left Filter", NULL, "DAC MONO L ASRC", is_using_asrc },
{ "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc },
{ "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc },
+ { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", can_use_asrc },
+ { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", can_use_asrc },
+ { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", can_use_asrc },
+ { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", can_use_asrc },
- { "I2S1", NULL, "I2S1 ASRC" },
- { "I2S2", NULL, "I2S2 ASRC" },
+ { "I2S1", NULL, "I2S1 ASRC", can_use_asrc},
+ { "I2S2", NULL, "I2S2 ASRC", can_use_asrc},
{ "DMIC1", NULL, "DMIC L1" },
{ "DMIC1", NULL, "DMIC R1" },
@@ -1970,12 +2006,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
{ "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" },
{ "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" },
{ "PDM1 R Mux", NULL, "PDM1 Power" },
- { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" },
- { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" },
- { "PDM2 L Mux", NULL, "PDM2 Power" },
- { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" },
- { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" },
- { "PDM2 R Mux", NULL, "PDM2 Power" },
{ "HP Amp", NULL, "HPO MIX" },
{ "HP Amp", NULL, "Mic Det Power" },
@@ -1993,13 +2023,30 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = {
{ "LOUTR", NULL, "LOUT R Playback" },
{ "LOUTL", NULL, "Improve HP Amp Drv" },
{ "LOUTR", NULL, "Improve HP Amp Drv" },
+};
+static const struct snd_soc_dapm_route rt5670_specific_dapm_routes[] = {
+ { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" },
+ { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" },
+ { "PDM2 L Mux", NULL, "PDM2 Power" },
+ { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" },
+ { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" },
+ { "PDM2 R Mux", NULL, "PDM2 Power" },
{ "PDM1L", NULL, "PDM1 L Mux" },
{ "PDM1R", NULL, "PDM1 R Mux" },
{ "PDM2L", NULL, "PDM2 L Mux" },
{ "PDM2R", NULL, "PDM2 R Mux" },
};
+static const struct snd_soc_dapm_route rt5672_specific_dapm_routes[] = {
+ { "SPO Amp", NULL, "PDM1 L Mux" },
+ { "SPO Amp", NULL, "PDM1 R Mux" },
+ { "SPOLP", NULL, "SPO Amp" },
+ { "SPOLN", NULL, "SPO Amp" },
+ { "SPORP", NULL, "SPO Amp" },
+ { "SPORN", NULL, "SPO Amp" },
+};
+
static int rt5670_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
@@ -2287,6 +2334,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
static int rt5670_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
@@ -2308,16 +2357,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
}
break;
case SND_SOC_BIAS_STANDBY:
- snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000);
- snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001);
- snd_soc_write(codec, RT5670_PWR_VOL, 0x0000);
- snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001);
- snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800);
- snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004);
- snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0);
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_PWR_VREF1 | RT5670_PWR_VREF2 |
+ RT5670_PWR_FV1 | RT5670_PWR_FV2, 0);
snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
RT5670_LDO_SEL_MASK, 0x1);
break;
+ case SND_SOC_BIAS_OFF:
+ if (rt5670->pdata.jd_mode)
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_PWR_VREF1 | RT5670_PWR_MB |
+ RT5670_PWR_BG | RT5670_PWR_VREF2 |
+ RT5670_PWR_FV1 | RT5670_PWR_FV2,
+ RT5670_PWR_MB | RT5670_PWR_BG);
+ else
+ snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
+ RT5670_PWR_VREF1 | RT5670_PWR_MB |
+ RT5670_PWR_BG | RT5670_PWR_VREF2 |
+ RT5670_PWR_FV1 | RT5670_PWR_FV2, 0);
+
+ snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0);
+ break;
default:
break;
@@ -2331,6 +2391,29 @@ static int rt5670_probe(struct snd_soc_codec *codec)
{
struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
+ switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) {
+ case RT5670_ID_5670:
+ case RT5670_ID_5671:
+ snd_soc_dapm_new_controls(&codec->dapm,
+ rt5670_specific_dapm_widgets,
+ ARRAY_SIZE(rt5670_specific_dapm_widgets));
+ snd_soc_dapm_add_routes(&codec->dapm,
+ rt5670_specific_dapm_routes,
+ ARRAY_SIZE(rt5670_specific_dapm_routes));
+ break;
+ case RT5670_ID_5672:
+ snd_soc_dapm_new_controls(&codec->dapm,
+ rt5672_specific_dapm_widgets,
+ ARRAY_SIZE(rt5672_specific_dapm_widgets));
+ snd_soc_dapm_add_routes(&codec->dapm,
+ rt5672_specific_dapm_routes,
+ ARRAY_SIZE(rt5672_specific_dapm_routes));
+ break;
+ default:
+ dev_err(codec->dev,
+ "The driver is for RT5670 RT5671 or RT5672 only\n");
+ return -ENODEV;
+ }
rt5670->codec = codec;
return 0;
@@ -2452,10 +2535,20 @@ static const struct regmap_config rt5670_regmap = {
static const struct i2c_device_id rt5670_i2c_id[] = {
{ "rt5670", 0 },
+ { "rt5671", 0 },
+ { "rt5672", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id);
+#ifdef CONFIG_ACPI
+static struct acpi_device_id rt5670_acpi_match[] = {
+ { "10EC5670", 0},
+ { },
+};
+MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match);
+#endif
+
static int rt5670_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -2644,6 +2737,7 @@ static struct i2c_driver rt5670_i2c_driver = {
.driver = {
.name = "rt5670",
.owner = THIS_MODULE,
+ .acpi_match_table = ACPI_PTR(rt5670_acpi_match),
},
.probe = rt5670_i2c_probe,
.remove = rt5670_i2c_remove,
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index a0b5c855b492..d11b9c207e26 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -228,6 +228,12 @@
#define RT5670_R_VOL_MASK (0x3f)
#define RT5670_R_VOL_SFT 0
+/* SW Reset & Device ID (0x00) */
+#define RT5670_ID_MASK (0x3 << 1)
+#define RT5670_ID_5670 (0x0 << 1)
+#define RT5670_ID_5672 (0x1 << 1)
+#define RT5670_ID_5671 (0x2 << 1)
+
/* Combo Jack Control 1 (0x0a) */
#define RT5670_CBJ_BST1_MASK (0xf << 12)
#define RT5670_CBJ_BST1_SFT (12)
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
new file mode 100644
index 000000000000..ef6348cb9157
--- /dev/null
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -0,0 +1,130 @@
+/*
+ * rt5677-spi.c -- RT5677 ALSA SoC audio codec driver
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/input.h>
+#include <linux/spi/spi.h>
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/irq.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/sched.h>
+#include <linux/kthread.h>
+#include <linux/uaccess.h>
+#include <linux/miscdevice.h>
+#include <linux/regulator/consumer.h>
+#include <linux/pm_qos.h>
+#include <linux/sysfs.h>
+#include <linux/clk.h>
+#include <linux/firmware.h>
+
+#include "rt5677-spi.h"
+
+static struct spi_device *g_spi;
+
+/**
+ * rt5677_spi_write - Write data to SPI.
+ * @txbuf: Data Buffer for writing.
+ * @len: Data length.
+ *
+ *
+ * Returns true for success.
+ */
+int rt5677_spi_write(u8 *txbuf, size_t len)
+{
+ int status;
+
+ status = spi_write(g_spi, txbuf, len);
+
+ if (status)
+ dev_err(&g_spi->dev, "rt5677_spi_write error %d\n", status);
+
+ return status;
+}
+EXPORT_SYMBOL_GPL(rt5677_spi_write);
+
+/**
+ * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address.
+ * @addr: Start address.
+ * @txbuf: Data Buffer for writng.
+ * @len: Data length, it must be a multiple of 8.
+ *
+ *
+ * Returns true for success.
+ */
+int rt5677_spi_burst_write(u32 addr, const struct firmware *fw)
+{
+ u8 spi_cmd = RT5677_SPI_CMD_BURST_WRITE;
+ u8 *write_buf;
+ unsigned int i, end, offset = 0;
+
+ write_buf = kmalloc(RT5677_SPI_BUF_LEN + 6, GFP_KERNEL);
+
+ if (write_buf == NULL)
+ return -ENOMEM;
+
+ while (offset < fw->size) {
+ if (offset + RT5677_SPI_BUF_LEN <= fw->size)
+ end = RT5677_SPI_BUF_LEN;
+ else
+ end = fw->size % RT5677_SPI_BUF_LEN;
+
+ write_buf[0] = spi_cmd;
+ write_buf[1] = ((addr + offset) & 0xff000000) >> 24;
+ write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16;
+ write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8;
+ write_buf[4] = ((addr + offset) & 0x000000ff) >> 0;
+
+ for (i = 0; i < end; i += 8) {
+ write_buf[i + 12] = fw->data[offset + i + 0];
+ write_buf[i + 11] = fw->data[offset + i + 1];
+ write_buf[i + 10] = fw->data[offset + i + 2];
+ write_buf[i + 9] = fw->data[offset + i + 3];
+ write_buf[i + 8] = fw->data[offset + i + 4];
+ write_buf[i + 7] = fw->data[offset + i + 5];
+ write_buf[i + 6] = fw->data[offset + i + 6];
+ write_buf[i + 5] = fw->data[offset + i + 7];
+ }
+
+ write_buf[end + 5] = spi_cmd;
+
+ rt5677_spi_write(write_buf, end + 6);
+
+ offset += RT5677_SPI_BUF_LEN;
+ }
+
+ kfree(write_buf);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(rt5677_spi_burst_write);
+
+static int rt5677_spi_probe(struct spi_device *spi)
+{
+ g_spi = spi;
+ return 0;
+}
+
+static struct spi_driver rt5677_spi_driver = {
+ .driver = {
+ .name = "rt5677",
+ .owner = THIS_MODULE,
+ },
+ .probe = rt5677_spi_probe,
+};
+module_spi_driver(rt5677_spi_driver);
+
+MODULE_DESCRIPTION("ASoC RT5677 SPI driver");
+MODULE_AUTHOR("Oder Chiou <oder_chiou@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h
new file mode 100644
index 000000000000..ec41b2b3b2ca
--- /dev/null
+++ b/sound/soc/codecs/rt5677-spi.h
@@ -0,0 +1,21 @@
+/*
+ * rt5677-spi.h -- RT5677 ALSA SoC audio codec driver
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __RT5677_SPI_H__
+#define __RT5677_SPI_H__
+
+#define RT5677_SPI_BUF_LEN 240
+#define RT5677_SPI_CMD_BURST_WRITE 0x05
+
+int rt5677_spi_write(u8 *txbuf, size_t len);
+int rt5677_spi_burst_write(u32 addr, const struct firmware *fw);
+
+#endif /* __RT5677_SPI_H__ */
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 16aa4d99a713..81fe1464d268 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -20,6 +20,7 @@
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
+#include <linux/firmware.h>
#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -31,6 +32,7 @@
#include "rl6231.h"
#include "rt5677.h"
+#include "rt5677-spi.h"
#define RT5677_DEVICE_ID 0x6327
@@ -53,12 +55,13 @@ static const struct regmap_range_cfg rt5677_ranges[] = {
};
static const struct reg_default init_list[] = {
+ {RT5677_ASRC_12, 0x0018},
{RT5677_PR_BASE + 0x3d, 0x364d},
- {RT5677_PR_BASE + 0x17, 0x4fc0},
- {RT5677_PR_BASE + 0x13, 0x0312},
- {RT5677_PR_BASE + 0x1e, 0x0000},
- {RT5677_PR_BASE + 0x12, 0x0eaa},
- {RT5677_PR_BASE + 0x14, 0x018a},
+ {RT5677_PR_BASE + 0x17, 0x4fc0},
+ {RT5677_PR_BASE + 0x13, 0x0312},
+ {RT5677_PR_BASE + 0x1e, 0x0000},
+ {RT5677_PR_BASE + 0x12, 0x0eaa},
+ {RT5677_PR_BASE + 0x14, 0x018a},
};
#define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list)
@@ -171,7 +174,7 @@ static const struct reg_default rt5677_reg[] = {
{RT5677_ASRC_9 , 0x0000},
{RT5677_ASRC_10 , 0x0000},
{RT5677_ASRC_11 , 0x0000},
- {RT5677_ASRC_12 , 0x0008},
+ {RT5677_ASRC_12 , 0x0018},
{RT5677_ASRC_13 , 0x0000},
{RT5677_ASRC_14 , 0x0000},
{RT5677_ASRC_15 , 0x0000},
@@ -537,10 +540,232 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg)
}
}
+/**
+ * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode.
+ * @rt5677: Private Data.
+ * @addr: Address index.
+ * @value: Address data.
+ *
+ *
+ * Returns 0 for success or negative error code.
+ */
+static int rt5677_dsp_mode_i2c_write_addr(struct rt5677_priv *rt5677,
+ unsigned int addr, unsigned int value, unsigned int opcode)
+{
+ struct snd_soc_codec *codec = rt5677->codec;
+ int ret;
+
+ mutex_lock(&rt5677->dsp_cmd_lock);
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_MSB,
+ addr >> 16);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_LSB,
+ addr & 0xffff);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_MSB,
+ value >> 16);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set data msb value: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_LSB,
+ value & 0xffff);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_OP_CODE,
+ opcode);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set op code value: %d\n", ret);
+ goto err;
+ }
+
+err:
+ mutex_unlock(&rt5677->dsp_cmd_lock);
+
+ return ret;
+}
+
+/**
+ * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode.
+ * rt5677: Private Data.
+ * @addr: Address index.
+ * @value: Address data.
+ *
+ *
+ * Returns 0 for success or negative error code.
+ */
+static int rt5677_dsp_mode_i2c_read_addr(
+ struct rt5677_priv *rt5677, unsigned int addr, unsigned int *value)
+{
+ struct snd_soc_codec *codec = rt5677->codec;
+ int ret;
+ unsigned int msb, lsb;
+
+ mutex_lock(&rt5677->dsp_cmd_lock);
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_MSB,
+ addr >> 16);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_LSB,
+ addr & 0xffff);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_OP_CODE,
+ 0x0002);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set op code value: %d\n", ret);
+ goto err;
+ }
+
+ regmap_read(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_MSB, &msb);
+ regmap_read(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_LSB, &lsb);
+ *value = (msb << 16) | lsb;
+
+err:
+ mutex_unlock(&rt5677->dsp_cmd_lock);
+
+ return ret;
+}
+
+/**
+ * rt5677_dsp_mode_i2c_write - Write register on DSP mode.
+ * rt5677: Private Data.
+ * @reg: Register index.
+ * @value: Register data.
+ *
+ *
+ * Returns 0 for success or negative error code.
+ */
+static int rt5677_dsp_mode_i2c_write(struct rt5677_priv *rt5677,
+ unsigned int reg, unsigned int value)
+{
+ return rt5677_dsp_mode_i2c_write_addr(rt5677, 0x18020000 + reg * 2,
+ value, 0x0001);
+}
+
+/**
+ * rt5677_dsp_mode_i2c_read - Read register on DSP mode.
+ * @codec: SoC audio codec device.
+ * @reg: Register index.
+ * @value: Register data.
+ *
+ *
+ * Returns 0 for success or negative error code.
+ */
+static int rt5677_dsp_mode_i2c_read(
+ struct rt5677_priv *rt5677, unsigned int reg, unsigned int *value)
+{
+ int ret = rt5677_dsp_mode_i2c_read_addr(rt5677, 0x18020000 + reg * 2,
+ value);
+
+ *value &= 0xffff;
+
+ return ret;
+}
+
+static void rt5677_set_dsp_mode(struct snd_soc_codec *codec, bool on)
+{
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+
+ if (on) {
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x2);
+ rt5677->is_dsp_mode = true;
+ } else {
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x0);
+ rt5677->is_dsp_mode = false;
+ }
+}
+
+static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on)
+{
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ static bool activity;
+ int ret;
+
+ if (on && !activity) {
+ activity = true;
+
+ regcache_cache_only(rt5677->regmap, false);
+ regcache_cache_bypass(rt5677->regmap, true);
+
+ regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x1);
+ regmap_update_bits(rt5677->regmap,
+ RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0f00);
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
+ RT5677_LDO1_SEL_MASK, 0x0);
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2,
+ RT5677_PWR_LDO1, RT5677_PWR_LDO1);
+ regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1,
+ RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC);
+ regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2,
+ RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK,
+ RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS);
+ regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff);
+ regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd);
+ rt5677_set_dsp_mode(codec, true);
+
+ ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1,
+ codec->dev);
+ if (ret == 0) {
+ rt5677_spi_burst_write(0x50000000, rt5677->fw1);
+ release_firmware(rt5677->fw1);
+ }
+
+ ret = request_firmware(&rt5677->fw2, RT5677_FIRMWARE2,
+ codec->dev);
+ if (ret == 0) {
+ rt5677_spi_burst_write(0x60000000, rt5677->fw2);
+ release_firmware(rt5677->fw2);
+ }
+
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x0);
+
+ regcache_cache_bypass(rt5677->regmap, false);
+ regcache_cache_only(rt5677->regmap, true);
+ } else if (!on && activity) {
+ activity = false;
+
+ regcache_cache_only(rt5677->regmap, false);
+ regcache_cache_bypass(rt5677->regmap, true);
+
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x1);
+ rt5677_set_dsp_mode(codec, false);
+ regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x0001);
+
+ regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec);
+
+ regcache_cache_bypass(rt5677->regmap, false);
+ regcache_mark_dirty(rt5677->regmap);
+ regcache_sync(rt5677->regmap);
+ }
+
+ return 0;
+}
+
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0);
@@ -556,6 +781,31 @@ static unsigned int bst_tlv[] = {
8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0),
};
+static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = rt5677->dsp_vad_en;
+
+ return 0;
+}
+
+static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+
+ rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0];
+
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ rt5677_set_dsp_vad(codec, rt5677->dsp_vad_en);
+
+ return 0;
+}
+
static const struct snd_kcontrol_new rt5677_snd_controls[] = {
/* OUTPUT Control */
SOC_SINGLE("OUT1 Playback Switch", RT5677_LOUT1,
@@ -567,13 +817,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
/* DAC Digital Volume */
SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL,
- RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv),
+ RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv),
SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL,
- RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv),
+ RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv),
SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL,
- RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv),
+ RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv),
SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL,
- RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv),
+ RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv),
/* IN1/IN2 Control */
SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv),
@@ -592,19 +842,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1),
SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL,
- RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0,
+ RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0,
adc_vol_tlv),
SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL,
- RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0,
+ RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0,
adc_vol_tlv),
SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL,
- RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0,
+ RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0,
adc_vol_tlv),
SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL,
- RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0,
+ RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0,
adc_vol_tlv),
SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL,
- RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0,
+ RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 63, 0,
adc_vol_tlv),
/* Sidetone Control */
@@ -627,6 +877,9 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2,
RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
+
+ SOC_SINGLE_EXT("DSP VAD Switch", SND_SOC_NOPM, 0, 1, 0,
+ rt5677_dsp_vad_get, rt5677_dsp_vad_put),
};
/**
@@ -1086,7 +1339,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux =
SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum);
-/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */
+/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */
static const char * const rt5677_stereo_adc2_src[] = {
"DD MIX1", "DMIC", "Stereo DAC MIX"
};
@@ -1171,7 +1424,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux =
SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum);
-/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */
+/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */
static const char * const rt5677_stereo_adc1_src[] = {
"DD MIX1", "ADC1/2", "Stereo DAC MIX"
};
@@ -1443,7 +1696,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_pdm2_r_mux =
SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum);
-/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/
+/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0] */
static const char * const rt5677_if12_adc1_src[] = {
"STO1 ADC MIX", "OB01", "VAD ADC"
};
@@ -1521,7 +1774,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_slb_adc3_mux =
SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum);
-/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */
+/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */
static const char * const rt5677_if12_adc4_src[] = {
"STO4 ADC MIX", "OB67", "OB01"
};
@@ -1547,7 +1800,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_slb_adc4_mux =
SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum);
-/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/
+/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4] */
static const char * const rt5677_if34_adc_src[] = {
"STO1 ADC MIX", "STO2 ADC MIX", "STO3 ADC MIX", "STO4 ADC MIX",
"MONO ADC MIX", "OB01", "OB23", "VAD ADC"
@@ -1567,6 +1820,213 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_if4_adc_mux =
SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum);
+/* TDM IF1/2 ADC Data Selection */ /* MX-3B MX-40 [7:6][5:4][3:2][1:0] */
+static const char * const rt5677_if12_adc_swap_src[] = {
+ "L/R", "R/L", "L/L", "R/R"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_adc1_swap_enum, RT5677_TDM1_CTRL1,
+ RT5677_IF1_ADC1_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if1_adc1_swap_mux =
+ SOC_DAPM_ENUM("IF1 ADC1 Swap Source", rt5677_if1_adc1_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_adc2_swap_enum, RT5677_TDM1_CTRL1,
+ RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if1_adc2_swap_mux =
+ SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if1_adc2_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_adc3_swap_enum, RT5677_TDM1_CTRL1,
+ RT5677_IF1_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if1_adc3_swap_mux =
+ SOC_DAPM_ENUM("IF1 ADC3 Swap Source", rt5677_if1_adc3_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_adc4_swap_enum, RT5677_TDM1_CTRL1,
+ RT5677_IF1_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if1_adc4_swap_mux =
+ SOC_DAPM_ENUM("IF1 ADC4 Swap Source", rt5677_if1_adc4_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_adc1_swap_enum, RT5677_TDM2_CTRL1,
+ RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if2_adc1_swap_mux =
+ SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if2_adc1_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_adc2_swap_enum, RT5677_TDM2_CTRL1,
+ RT5677_IF2_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if2_adc2_swap_mux =
+ SOC_DAPM_ENUM("IF2 ADC2 Swap Source", rt5677_if2_adc2_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_adc3_swap_enum, RT5677_TDM2_CTRL1,
+ RT5677_IF2_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if2_adc3_swap_mux =
+ SOC_DAPM_ENUM("IF2 ADC3 Swap Source", rt5677_if2_adc3_swap_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_adc4_swap_enum, RT5677_TDM2_CTRL1,
+ RT5677_IF2_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux =
+ SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum);
+
+/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */
+static const char * const rt5677_if1_adc_tdm_swap_src[] = {
+ "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3",
+ "3/1/2/4", "3/4/1/2"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_adc_tdm_swap_enum, RT5677_TDM1_CTRL2,
+ RT5677_IF1_ADC_CTRL_SFT, rt5677_if1_adc_tdm_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if1_adc_tdm_swap_mux =
+ SOC_DAPM_ENUM("IF1 ADC TDM Swap Source", rt5677_if1_adc_tdm_swap_enum);
+
+/* TDM IF2 ADC Data Selection */ /* MX-41[2:0] */
+static const char * const rt5677_if2_adc_tdm_swap_src[] = {
+ "1/2/3/4", "2/1/3/4", "3/1/2/4", "4/1/2/3", "1/3/2/4", "1/4/2/3",
+ "2/3/1/4", "3/4/1/2"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_adc_tdm_swap_enum, RT5677_TDM2_CTRL2,
+ RT5677_IF2_ADC_CTRL_SFT, rt5677_if2_adc_tdm_swap_src);
+
+static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux =
+ SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum);
+
+/* TDM IF1/2 DAC Data Selection */ /* MX-3E[14:12][10:8][6:4][2:0]
+ MX-3F[14:12][10:8][6:4][2:0]
+ MX-43[14:12][10:8][6:4][2:0]
+ MX-44[14:12][10:8][6:4][2:0] */
+static const char * const rt5677_if12_dac_tdm_sel_src[] = {
+ "Slot0", "Slot1", "Slot2", "Slot3", "Slot4", "Slot5", "Slot6", "Slot7"
+};
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac0_tdm_sel_enum, RT5677_TDM1_CTRL4,
+ RT5677_IF1_DAC0_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac0_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC0 TDM Source", rt5677_if1_dac0_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac1_tdm_sel_enum, RT5677_TDM1_CTRL4,
+ RT5677_IF1_DAC1_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac1_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC1 TDM Source", rt5677_if1_dac1_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac2_tdm_sel_enum, RT5677_TDM1_CTRL4,
+ RT5677_IF1_DAC2_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac2_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC2 TDM Source", rt5677_if1_dac2_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac3_tdm_sel_enum, RT5677_TDM1_CTRL4,
+ RT5677_IF1_DAC3_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac3_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC3 TDM Source", rt5677_if1_dac3_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac4_tdm_sel_enum, RT5677_TDM1_CTRL5,
+ RT5677_IF1_DAC4_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac4_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC4 TDM Source", rt5677_if1_dac4_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac5_tdm_sel_enum, RT5677_TDM1_CTRL5,
+ RT5677_IF1_DAC5_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac5_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC5 TDM Source", rt5677_if1_dac5_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac6_tdm_sel_enum, RT5677_TDM1_CTRL5,
+ RT5677_IF1_DAC6_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac6_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC6 TDM Source", rt5677_if1_dac6_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if1_dac7_tdm_sel_enum, RT5677_TDM1_CTRL5,
+ RT5677_IF1_DAC7_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if1_dac7_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF1 DAC7 TDM Source", rt5677_if1_dac7_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac0_tdm_sel_enum, RT5677_TDM2_CTRL4,
+ RT5677_IF2_DAC0_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac0_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC0 TDM Source", rt5677_if2_dac0_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac1_tdm_sel_enum, RT5677_TDM2_CTRL4,
+ RT5677_IF2_DAC1_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac1_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC1 TDM Source", rt5677_if2_dac1_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac2_tdm_sel_enum, RT5677_TDM2_CTRL4,
+ RT5677_IF2_DAC2_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac2_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC2 TDM Source", rt5677_if2_dac2_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac3_tdm_sel_enum, RT5677_TDM2_CTRL4,
+ RT5677_IF2_DAC3_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac3_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC3 TDM Source", rt5677_if2_dac3_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac4_tdm_sel_enum, RT5677_TDM2_CTRL5,
+ RT5677_IF2_DAC4_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac4_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC4 TDM Source", rt5677_if2_dac4_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac5_tdm_sel_enum, RT5677_TDM2_CTRL5,
+ RT5677_IF2_DAC5_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac5_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC5 TDM Source", rt5677_if2_dac5_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac6_tdm_sel_enum, RT5677_TDM2_CTRL5,
+ RT5677_IF2_DAC6_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac6_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC6 TDM Source", rt5677_if2_dac6_tdm_sel_enum);
+
+static SOC_ENUM_SINGLE_DECL(
+ rt5677_if2_dac7_tdm_sel_enum, RT5677_TDM2_CTRL5,
+ RT5677_IF2_DAC7_SFT, rt5677_if12_dac_tdm_sel_src);
+
+static const struct snd_kcontrol_new rt5677_if2_dac7_tdm_sel_mux =
+ SOC_DAPM_ENUM("IF2 DAC7 TDM Source", rt5677_if2_dac7_tdm_sel_enum);
+
static int rt5677_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -1678,6 +2138,77 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ unsigned int value;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ regmap_read(rt5677->regmap, RT5677_TDM1_CTRL2, &value);
+ if (value & RT5677_IF1_ADC_CTRL_MASK)
+ regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1,
+ RT5677_IF1_ADC_MODE_MASK,
+ RT5677_IF1_ADC_MODE_TDM);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ unsigned int value;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ regmap_read(rt5677->regmap, RT5677_TDM2_CTRL2, &value);
+ if (value & RT5677_IF2_ADC_CTRL_MASK)
+ regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1,
+ RT5677_IF2_ADC_MODE_MASK,
+ RT5677_IF2_ADC_MODE_TDM);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (codec->dapm.bias_level != SND_SOC_BIAS_ON &&
+ !rt5677->is_vref_slow) {
+ mdelay(20);
+ regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
+ RT5677_PWR_FV1 | RT5677_PWR_FV2,
+ RT5677_PWR_FV1 | RT5677_PWR_FV2);
+ rt5677->is_vref_slow = true;
+ }
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU),
@@ -1837,10 +2368,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_PGA("Stereo4 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0),
/* DSP */
SND_SOC_DAPM_MUX("IB9 Mux", SND_SOC_NOPM, 0, 0,
@@ -1963,6 +2492,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
&rt5677_if1_adc3_mux),
SND_SOC_DAPM_MUX("IF1 ADC4 Mux", SND_SOC_NOPM, 0, 0,
&rt5677_if1_adc4_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_adc1_swap_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_adc2_swap_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_adc3_swap_mux),
+ SND_SOC_DAPM_MUX("IF1 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_adc4_swap_mux),
+ SND_SOC_DAPM_MUX_E("IF1 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_adc_tdm_swap_mux, rt5677_if1_adc_tdm_event,
+ SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_MUX("IF2 ADC1 Mux", SND_SOC_NOPM, 0, 0,
&rt5677_if2_adc1_mux),
SND_SOC_DAPM_MUX("IF2 ADC2 Mux", SND_SOC_NOPM, 0, 0,
@@ -1971,6 +2511,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
&rt5677_if2_adc3_mux),
SND_SOC_DAPM_MUX("IF2 ADC4 Mux", SND_SOC_NOPM, 0, 0,
&rt5677_if2_adc4_mux),
+ SND_SOC_DAPM_MUX("IF2 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_adc1_swap_mux),
+ SND_SOC_DAPM_MUX("IF2 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_adc2_swap_mux),
+ SND_SOC_DAPM_MUX("IF2 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_adc3_swap_mux),
+ SND_SOC_DAPM_MUX("IF2 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_adc4_swap_mux),
+ SND_SOC_DAPM_MUX_E("IF2 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_adc_tdm_swap_mux, rt5677_if2_adc_tdm_event,
+ SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_MUX("IF3 ADC Mux", SND_SOC_NOPM, 0, 0,
&rt5677_if3_adc_mux),
SND_SOC_DAPM_MUX("IF4 ADC Mux", SND_SOC_NOPM, 0, 0,
@@ -1984,6 +2535,40 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_MUX("SLB ADC4 Mux", SND_SOC_NOPM, 0, 0,
&rt5677_slb_adc4_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC0 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac0_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac1_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac2_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC3 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac3_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC4 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac4_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC5 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac5_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC6 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac6_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF1 DAC7 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if1_dac7_tdm_sel_mux),
+
+ SND_SOC_DAPM_MUX("IF2 DAC0 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac0_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC1 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac1_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC2 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac2_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC3 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac3_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC4 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac4_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC5 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac5_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC6 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac6_tdm_sel_mux),
+ SND_SOC_DAPM_MUX("IF2 DAC7 Mux", SND_SOC_NOPM, 0, 0,
+ &rt5677_if2_dac7_tdm_sel_mux),
+
/* Audio Interface */
SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
@@ -2022,7 +2607,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
rt5677_ob_7_mix, ARRAY_SIZE(rt5677_ob_7_mix)),
/* Output Side */
- /* DAC mixer before sound effect */
+ /* DAC mixer before sound effect */
SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0,
rt5677_dac_l_mix, ARRAY_SIZE(rt5677_dac_l_mix)),
SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0,
@@ -2109,13 +2694,20 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_MUX("PDM2 R Mux", RT5677_PDM_OUT_CTRL, RT5677_M_PDM2_R_SFT,
1, &rt5677_pdm2_r_mux),
- SND_SOC_DAPM_PGA_S("LOUT1 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO1_BIT,
+ SND_SOC_DAPM_PGA_S("LOUT1 amp", 0, RT5677_PWR_ANLG1, RT5677_PWR_LO1_BIT,
0, NULL, 0),
- SND_SOC_DAPM_PGA_S("LOUT2 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO2_BIT,
+ SND_SOC_DAPM_PGA_S("LOUT2 amp", 0, RT5677_PWR_ANLG1, RT5677_PWR_LO2_BIT,
0, NULL, 0),
- SND_SOC_DAPM_PGA_S("LOUT3 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO3_BIT,
+ SND_SOC_DAPM_PGA_S("LOUT3 amp", 0, RT5677_PWR_ANLG1, RT5677_PWR_LO3_BIT,
0, NULL, 0),
+ SND_SOC_DAPM_PGA_S("LOUT1 vref", 1, SND_SOC_NOPM, 0, 0,
+ rt5677_vref_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_S("LOUT2 vref", 1, SND_SOC_NOPM, 0, 0,
+ rt5677_vref_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_S("LOUT3 vref", 1, SND_SOC_NOPM, 0, 0,
+ rt5677_vref_event, SND_SOC_DAPM_POST_PMU),
+
/* Output Lines */
SND_SOC_DAPM_OUTPUT("LOUT1"),
SND_SOC_DAPM_OUTPUT("LOUT2"),
@@ -2124,6 +2716,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("PDM1R"),
SND_SOC_DAPM_OUTPUT("PDM2L"),
SND_SOC_DAPM_OUTPUT("PDM2R"),
+
+ SND_SOC_DAPM_POST("vref", rt5677_vref_event),
};
static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
@@ -2354,11 +2948,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IF1 ADC4 Mux", "OB67", "OB67" },
{ "IF1 ADC4 Mux", "OB01", "OB01 Bypass Mux" },
+ { "IF1 ADC1 Swap Mux", "L/R", "IF1 ADC1 Mux" },
+ { "IF1 ADC1 Swap Mux", "R/L", "IF1 ADC1 Mux" },
+ { "IF1 ADC1 Swap Mux", "L/L", "IF1 ADC1 Mux" },
+ { "IF1 ADC1 Swap Mux", "R/R", "IF1 ADC1 Mux" },
+
+ { "IF1 ADC2 Swap Mux", "L/R", "IF1 ADC2 Mux" },
+ { "IF1 ADC2 Swap Mux", "R/L", "IF1 ADC2 Mux" },
+ { "IF1 ADC2 Swap Mux", "L/L", "IF1 ADC2 Mux" },
+ { "IF1 ADC2 Swap Mux", "R/R", "IF1 ADC2 Mux" },
+
+ { "IF1 ADC3 Swap Mux", "L/R", "IF1 ADC3 Mux" },
+ { "IF1 ADC3 Swap Mux", "R/L", "IF1 ADC3 Mux" },
+ { "IF1 ADC3 Swap Mux", "L/L", "IF1 ADC3 Mux" },
+ { "IF1 ADC3 Swap Mux", "R/R", "IF1 ADC3 Mux" },
+
+ { "IF1 ADC4 Swap Mux", "L/R", "IF1 ADC4 Mux" },
+ { "IF1 ADC4 Swap Mux", "R/L", "IF1 ADC4 Mux" },
+ { "IF1 ADC4 Swap Mux", "L/L", "IF1 ADC4 Mux" },
+ { "IF1 ADC4 Swap Mux", "R/R", "IF1 ADC4 Mux" },
+
+ { "IF1 ADC", NULL, "IF1 ADC1 Swap Mux" },
+ { "IF1 ADC", NULL, "IF1 ADC2 Swap Mux" },
+ { "IF1 ADC", NULL, "IF1 ADC3 Swap Mux" },
+ { "IF1 ADC", NULL, "IF1 ADC4 Swap Mux" },
+
+ { "IF1 ADC TDM Swap Mux", "1/2/3/4", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "2/1/3/4", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "2/3/1/4", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "4/1/2/3", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "1/3/2/4", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "1/4/2/3", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "3/1/2/4", "IF1 ADC" },
+ { "IF1 ADC TDM Swap Mux", "3/4/1/2", "IF1 ADC" },
+
{ "AIF1TX", NULL, "I2S1" },
- { "AIF1TX", NULL, "IF1 ADC1 Mux" },
- { "AIF1TX", NULL, "IF1 ADC2 Mux" },
- { "AIF1TX", NULL, "IF1 ADC3 Mux" },
- { "AIF1TX", NULL, "IF1 ADC4 Mux" },
+ { "AIF1TX", NULL, "IF1 ADC TDM Swap Mux" },
{ "IF2 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" },
{ "IF2 ADC1 Mux", "OB01", "OB01 Bypass Mux" },
@@ -2375,11 +3000,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IF2 ADC4 Mux", "OB67", "OB67" },
{ "IF2 ADC4 Mux", "OB01", "OB01 Bypass Mux" },
+ { "IF2 ADC1 Swap Mux", "L/R", "IF2 ADC1 Mux" },
+ { "IF2 ADC1 Swap Mux", "R/L", "IF2 ADC1 Mux" },
+ { "IF2 ADC1 Swap Mux", "L/L", "IF2 ADC1 Mux" },
+ { "IF2 ADC1 Swap Mux", "R/R", "IF2 ADC1 Mux" },
+
+ { "IF2 ADC2 Swap Mux", "L/R", "IF2 ADC2 Mux" },
+ { "IF2 ADC2 Swap Mux", "R/L", "IF2 ADC2 Mux" },
+ { "IF2 ADC2 Swap Mux", "L/L", "IF2 ADC2 Mux" },
+ { "IF2 ADC2 Swap Mux", "R/R", "IF2 ADC2 Mux" },
+
+ { "IF2 ADC3 Swap Mux", "L/R", "IF2 ADC3 Mux" },
+ { "IF2 ADC3 Swap Mux", "R/L", "IF2 ADC3 Mux" },
+ { "IF2 ADC3 Swap Mux", "L/L", "IF2 ADC3 Mux" },
+ { "IF2 ADC3 Swap Mux", "R/R", "IF2 ADC3 Mux" },
+
+ { "IF2 ADC4 Swap Mux", "L/R", "IF2 ADC4 Mux" },
+ { "IF2 ADC4 Swap Mux", "R/L", "IF2 ADC4 Mux" },
+ { "IF2 ADC4 Swap Mux", "L/L", "IF2 ADC4 Mux" },
+ { "IF2 ADC4 Swap Mux", "R/R", "IF2 ADC4 Mux" },
+
+ { "IF2 ADC", NULL, "IF2 ADC1 Swap Mux" },
+ { "IF2 ADC", NULL, "IF2 ADC2 Swap Mux" },
+ { "IF2 ADC", NULL, "IF2 ADC3 Swap Mux" },
+ { "IF2 ADC", NULL, "IF2 ADC4 Swap Mux" },
+
+ { "IF2 ADC TDM Swap Mux", "1/2/3/4", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "2/1/3/4", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "3/1/2/4", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "4/1/2/3", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "1/3/2/4", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "1/4/2/3", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "2/3/1/4", "IF2 ADC" },
+ { "IF2 ADC TDM Swap Mux", "3/4/1/2", "IF2 ADC" },
+
{ "AIF2TX", NULL, "I2S2" },
- { "AIF2TX", NULL, "IF2 ADC1 Mux" },
- { "AIF2TX", NULL, "IF2 ADC2 Mux" },
- { "AIF2TX", NULL, "IF2 ADC3 Mux" },
- { "AIF2TX", NULL, "IF2 ADC4 Mux" },
+ { "AIF2TX", NULL, "IF2 ADC TDM Swap Mux" },
{ "IF3 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" },
{ "IF3 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" },
@@ -2569,14 +3225,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IF1 DAC6", NULL, "I2S1" },
{ "IF1 DAC7", NULL, "I2S1" },
- { "IF1 DAC01", NULL, "IF1 DAC0" },
- { "IF1 DAC01", NULL, "IF1 DAC1" },
- { "IF1 DAC23", NULL, "IF1 DAC2" },
- { "IF1 DAC23", NULL, "IF1 DAC3" },
- { "IF1 DAC45", NULL, "IF1 DAC4" },
- { "IF1 DAC45", NULL, "IF1 DAC5" },
- { "IF1 DAC67", NULL, "IF1 DAC6" },
- { "IF1 DAC67", NULL, "IF1 DAC7" },
+ { "IF1 DAC0 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC0 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC0 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC0 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC0 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC0 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC0 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC0 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC1 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC1 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC1 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC1 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC1 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC1 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC1 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC1 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC2 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC2 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC2 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC2 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC2 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC2 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC2 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC2 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC3 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC3 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC3 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC3 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC3 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC3 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC3 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC3 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC4 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC4 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC4 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC4 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC4 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC4 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC4 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC4 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC5 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC5 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC5 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC5 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC5 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC5 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC5 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC5 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC6 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC6 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC6 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC6 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC6 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC6 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC6 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC6 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC7 Mux", "Slot0", "IF1 DAC0" },
+ { "IF1 DAC7 Mux", "Slot1", "IF1 DAC1" },
+ { "IF1 DAC7 Mux", "Slot2", "IF1 DAC2" },
+ { "IF1 DAC7 Mux", "Slot3", "IF1 DAC3" },
+ { "IF1 DAC7 Mux", "Slot4", "IF1 DAC4" },
+ { "IF1 DAC7 Mux", "Slot5", "IF1 DAC5" },
+ { "IF1 DAC7 Mux", "Slot6", "IF1 DAC6" },
+ { "IF1 DAC7 Mux", "Slot7", "IF1 DAC7" },
+
+ { "IF1 DAC01", NULL, "IF1 DAC0 Mux" },
+ { "IF1 DAC01", NULL, "IF1 DAC1 Mux" },
+ { "IF1 DAC23", NULL, "IF1 DAC2 Mux" },
+ { "IF1 DAC23", NULL, "IF1 DAC3 Mux" },
+ { "IF1 DAC45", NULL, "IF1 DAC4 Mux" },
+ { "IF1 DAC45", NULL, "IF1 DAC5 Mux" },
+ { "IF1 DAC67", NULL, "IF1 DAC6 Mux" },
+ { "IF1 DAC67", NULL, "IF1 DAC7 Mux" },
{ "IF2 DAC0", NULL, "AIF2RX" },
{ "IF2 DAC1", NULL, "AIF2RX" },
@@ -2595,14 +3323,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "IF2 DAC6", NULL, "I2S2" },
{ "IF2 DAC7", NULL, "I2S2" },
- { "IF2 DAC01", NULL, "IF2 DAC0" },
- { "IF2 DAC01", NULL, "IF2 DAC1" },
- { "IF2 DAC23", NULL, "IF2 DAC2" },
- { "IF2 DAC23", NULL, "IF2 DAC3" },
- { "IF2 DAC45", NULL, "IF2 DAC4" },
- { "IF2 DAC45", NULL, "IF2 DAC5" },
- { "IF2 DAC67", NULL, "IF2 DAC6" },
- { "IF2 DAC67", NULL, "IF2 DAC7" },
+ { "IF2 DAC0 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC0 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC0 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC0 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC0 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC0 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC0 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC0 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC1 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC1 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC1 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC1 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC1 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC1 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC1 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC1 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC2 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC2 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC2 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC2 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC2 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC2 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC2 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC2 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC3 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC3 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC3 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC3 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC3 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC3 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC3 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC3 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC4 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC4 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC4 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC4 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC4 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC4 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC4 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC4 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC5 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC5 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC5 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC5 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC5 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC5 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC5 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC5 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC6 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC6 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC6 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC6 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC6 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC6 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC6 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC6 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC7 Mux", "Slot0", "IF2 DAC0" },
+ { "IF2 DAC7 Mux", "Slot1", "IF2 DAC1" },
+ { "IF2 DAC7 Mux", "Slot2", "IF2 DAC2" },
+ { "IF2 DAC7 Mux", "Slot3", "IF2 DAC3" },
+ { "IF2 DAC7 Mux", "Slot4", "IF2 DAC4" },
+ { "IF2 DAC7 Mux", "Slot5", "IF2 DAC5" },
+ { "IF2 DAC7 Mux", "Slot6", "IF2 DAC6" },
+ { "IF2 DAC7 Mux", "Slot7", "IF2 DAC7" },
+
+ { "IF2 DAC01", NULL, "IF2 DAC0 Mux" },
+ { "IF2 DAC01", NULL, "IF2 DAC1 Mux" },
+ { "IF2 DAC23", NULL, "IF2 DAC2 Mux" },
+ { "IF2 DAC23", NULL, "IF2 DAC3 Mux" },
+ { "IF2 DAC45", NULL, "IF2 DAC4 Mux" },
+ { "IF2 DAC45", NULL, "IF2 DAC5 Mux" },
+ { "IF2 DAC67", NULL, "IF2 DAC6 Mux" },
+ { "IF2 DAC67", NULL, "IF2 DAC7 Mux" },
{ "IF3 DAC", NULL, "AIF3RX" },
{ "IF3 DAC", NULL, "I2S3" },
@@ -2806,9 +3606,13 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "LOUT2 amp", NULL, "DAC 2" },
{ "LOUT3 amp", NULL, "DAC 3" },
- { "LOUT1", NULL, "LOUT1 amp" },
- { "LOUT2", NULL, "LOUT2 amp" },
- { "LOUT3", NULL, "LOUT3 amp" },
+ { "LOUT1 vref", NULL, "LOUT1 amp" },
+ { "LOUT2 vref", NULL, "LOUT2 amp" },
+ { "LOUT3 vref", NULL, "LOUT3 amp" },
+
+ { "LOUT1", NULL, "LOUT1 vref" },
+ { "LOUT2", NULL, "LOUT2 vref" },
+ { "LOUT3", NULL, "LOUT3 vref" },
{ "PDM1L", NULL, "PDM1 L Mux" },
{ "PDM1R", NULL, "PDM1 R Mux" },
@@ -2837,7 +3641,8 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream,
rt5677->lrck[dai->id] = params_rate(params);
pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]);
if (pre_div < 0) {
- dev_err(codec->dev, "Unsupported clock setting\n");
+ dev_err(codec->dev, "Unsupported clock setting: sysclk=%dHz lrck=%dHz\n",
+ rt5677->sysclk, rt5677->lrck[dai->id]);
return -EINVAL;
}
frame_size = snd_soc_params_to_frame_size(params);
@@ -3181,6 +3986,8 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+ rt5677_set_dsp_vad(codec, false);
+
regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK,
0x0055);
@@ -3188,14 +3995,12 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
RT5677_PR_BASE + RT5677_BIAS_CUR4,
0x0f00, 0x0f00);
regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
+ RT5677_PWR_FV1 | RT5677_PWR_FV2 |
RT5677_PWR_VREF1 | RT5677_PWR_MB |
RT5677_PWR_BG | RT5677_PWR_VREF2,
RT5677_PWR_VREF1 | RT5677_PWR_MB |
RT5677_PWR_BG | RT5677_PWR_VREF2);
- mdelay(20);
- regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
- RT5677_PWR_FV1 | RT5677_PWR_FV2,
- RT5677_PWR_FV1 | RT5677_PWR_FV2);
+ rt5677->is_vref_slow = false;
regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2,
RT5677_PWR_CORE, RT5677_PWR_CORE);
regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC,
@@ -3214,6 +4019,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000);
regmap_update_bits(rt5677->regmap,
RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000);
+
+ if (rt5677->dsp_vad_en)
+ rt5677_set_dsp_vad(codec, true);
break;
default:
@@ -3309,6 +4117,78 @@ static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
return 0;
}
+/** Configures the gpio as
+ * 0 - floating
+ * 1 - pull down
+ * 2 - pull up
+ */
+static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset,
+ int value)
+{
+ int shift;
+
+ switch (offset) {
+ case RT5677_GPIO1 ... RT5677_GPIO2:
+ shift = 2 * (1 - offset);
+ regmap_update_bits(rt5677->regmap,
+ RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL2,
+ 0x3 << shift,
+ (value & 0x3) << shift);
+ break;
+
+ case RT5677_GPIO3 ... RT5677_GPIO6:
+ shift = 2 * (9 - offset);
+ regmap_update_bits(rt5677->regmap,
+ RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL3,
+ 0x3 << shift,
+ (value & 0x3) << shift);
+ break;
+
+ default:
+ break;
+ }
+}
+
+static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset)
+{
+ struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+ struct regmap_irq_chip_data *data = rt5677->irq_data;
+ int irq;
+
+ if (offset >= RT5677_GPIO1 && offset <= RT5677_GPIO3) {
+ if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) ||
+ (rt5677->pdata.jd1_gpio == 2 &&
+ offset == RT5677_GPIO2) ||
+ (rt5677->pdata.jd1_gpio == 3 &&
+ offset == RT5677_GPIO3)) {
+ irq = RT5677_IRQ_JD1;
+ } else {
+ return -ENXIO;
+ }
+ }
+
+ if (offset >= RT5677_GPIO4 && offset <= RT5677_GPIO6) {
+ if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) ||
+ (rt5677->pdata.jd2_gpio == 2 &&
+ offset == RT5677_GPIO5) ||
+ (rt5677->pdata.jd2_gpio == 3 &&
+ offset == RT5677_GPIO6)) {
+ irq = RT5677_IRQ_JD2;
+ } else if ((rt5677->pdata.jd3_gpio == 1 &&
+ offset == RT5677_GPIO4) ||
+ (rt5677->pdata.jd3_gpio == 2 &&
+ offset == RT5677_GPIO5) ||
+ (rt5677->pdata.jd3_gpio == 3 &&
+ offset == RT5677_GPIO6)) {
+ irq = RT5677_IRQ_JD3;
+ } else {
+ return -ENXIO;
+ }
+ }
+
+ return regmap_irq_get_virq(data, irq);
+}
+
static struct gpio_chip rt5677_template_chip = {
.label = "rt5677",
.owner = THIS_MODULE,
@@ -3316,6 +4196,7 @@ static struct gpio_chip rt5677_template_chip = {
.set = rt5677_gpio_set,
.direction_input = rt5677_gpio_direction_in,
.get = rt5677_gpio_get,
+ .to_irq = rt5677_to_irq,
.can_sleep = 1,
};
@@ -3341,6 +4222,11 @@ static void rt5677_free_gpio(struct i2c_client *i2c)
gpiochip_remove(&rt5677->gpio_chip);
}
#else
+static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset,
+ int value)
+{
+}
+
static void rt5677_init_gpio(struct i2c_client *i2c)
{
}
@@ -3353,6 +4239,7 @@ static void rt5677_free_gpio(struct i2c_client *i2c)
static int rt5677_probe(struct snd_soc_codec *codec)
{
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+ int i;
rt5677->codec = codec;
@@ -3371,6 +4258,37 @@ static int rt5677_probe(struct snd_soc_codec *codec)
regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020);
regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00);
+ for (i = 0; i < RT5677_GPIO_NUM; i++)
+ rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]);
+
+ if (rt5677->irq_data) {
+ regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, 0x8000,
+ 0x8000);
+ regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x0018,
+ 0x0008);
+
+ if (rt5677->pdata.jd1_gpio)
+ regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1,
+ RT5677_SEL_GPIO_JD1_MASK,
+ rt5677->pdata.jd1_gpio <<
+ RT5677_SEL_GPIO_JD1_SFT);
+
+ if (rt5677->pdata.jd2_gpio)
+ regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1,
+ RT5677_SEL_GPIO_JD2_MASK,
+ rt5677->pdata.jd2_gpio <<
+ RT5677_SEL_GPIO_JD2_SFT);
+
+ if (rt5677->pdata.jd3_gpio)
+ regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1,
+ RT5677_SEL_GPIO_JD3_MASK,
+ rt5677->pdata.jd3_gpio <<
+ RT5677_SEL_GPIO_JD3_SFT);
+ }
+
+ mutex_init(&rt5677->dsp_cmd_lock);
+ mutex_init(&rt5677->dsp_pri_lock);
+
return 0;
}
@@ -3390,8 +4308,11 @@ static int rt5677_suspend(struct snd_soc_codec *codec)
{
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
- regcache_cache_only(rt5677->regmap, true);
- regcache_mark_dirty(rt5677->regmap);
+ if (!rt5677->dsp_vad_en) {
+ regcache_cache_only(rt5677->regmap, true);
+ regcache_mark_dirty(rt5677->regmap);
+ }
+
if (gpio_is_valid(rt5677->pow_ldo2))
gpio_set_value_cansleep(rt5677->pow_ldo2, 0);
@@ -3406,8 +4327,11 @@ static int rt5677_resume(struct snd_soc_codec *codec)
gpio_set_value_cansleep(rt5677->pow_ldo2, 1);
msleep(10);
}
- regcache_cache_only(rt5677->regmap, false);
- regcache_sync(rt5677->regmap);
+
+ if (!rt5677->dsp_vad_en) {
+ regcache_cache_only(rt5677->regmap, false);
+ regcache_sync(rt5677->regmap);
+ }
return 0;
}
@@ -3416,6 +4340,51 @@ static int rt5677_resume(struct snd_soc_codec *codec)
#define rt5677_resume NULL
#endif
+static int rt5677_read(void *context, unsigned int reg, unsigned int *val)
+{
+ struct i2c_client *client = context;
+ struct rt5677_priv *rt5677 = i2c_get_clientdata(client);
+
+ if (rt5677->is_dsp_mode) {
+ if (reg > 0xff) {
+ mutex_lock(&rt5677->dsp_pri_lock);
+ rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX,
+ reg & 0xff);
+ rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val);
+ mutex_unlock(&rt5677->dsp_pri_lock);
+ } else {
+ rt5677_dsp_mode_i2c_read(rt5677, reg, val);
+ }
+ } else {
+ regmap_read(rt5677->regmap_physical, reg, val);
+ }
+
+ return 0;
+}
+
+static int rt5677_write(void *context, unsigned int reg, unsigned int val)
+{
+ struct i2c_client *client = context;
+ struct rt5677_priv *rt5677 = i2c_get_clientdata(client);
+
+ if (rt5677->is_dsp_mode) {
+ if (reg > 0xff) {
+ mutex_lock(&rt5677->dsp_pri_lock);
+ rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX,
+ reg & 0xff);
+ rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA,
+ val);
+ mutex_unlock(&rt5677->dsp_pri_lock);
+ } else {
+ rt5677_dsp_mode_i2c_write(rt5677, reg, val);
+ }
+ } else {
+ regmap_write(rt5677->regmap_physical, reg, val);
+ }
+
+ return 0;
+}
+
#define RT5677_STEREO_RATES SNDRV_PCM_RATE_8000_96000
#define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
@@ -3541,6 +4510,20 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5677 = {
.num_dapm_routes = ARRAY_SIZE(rt5677_dapm_routes),
};
+static const struct regmap_config rt5677_regmap_physical = {
+ .name = "physical",
+ .reg_bits = 8,
+ .val_bits = 16,
+
+ .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) *
+ RT5677_PR_SPACING),
+ .readable_reg = rt5677_readable_register,
+
+ .cache_type = REGCACHE_NONE,
+ .ranges = rt5677_ranges,
+ .num_ranges = ARRAY_SIZE(rt5677_ranges),
+};
+
static const struct regmap_config rt5677_regmap = {
.reg_bits = 8,
.val_bits = 16,
@@ -3550,6 +4533,8 @@ static const struct regmap_config rt5677_regmap = {
.volatile_reg = rt5677_volatile_register,
.readable_reg = rt5677_readable_register,
+ .reg_read = rt5677_read,
+ .reg_write = rt5677_write,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = rt5677_reg,
@@ -3590,9 +4575,77 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np)
(rt5677->pow_ldo2 != -ENOENT))
return rt5677->pow_ldo2;
+ of_property_read_u8_array(np, "realtek,gpio-config",
+ rt5677->pdata.gpio_config, RT5677_GPIO_NUM);
+
+ of_property_read_u32(np, "realtek,jd1-gpio", &rt5677->pdata.jd1_gpio);
+ of_property_read_u32(np, "realtek,jd2-gpio", &rt5677->pdata.jd2_gpio);
+ of_property_read_u32(np, "realtek,jd3-gpio", &rt5677->pdata.jd3_gpio);
+
+ return 0;
+}
+
+static struct regmap_irq rt5677_irqs[] = {
+ [RT5677_IRQ_JD1] = {
+ .reg_offset = 0,
+ .mask = RT5677_EN_IRQ_GPIO_JD1,
+ },
+ [RT5677_IRQ_JD2] = {
+ .reg_offset = 0,
+ .mask = RT5677_EN_IRQ_GPIO_JD2,
+ },
+ [RT5677_IRQ_JD3] = {
+ .reg_offset = 0,
+ .mask = RT5677_EN_IRQ_GPIO_JD3,
+ },
+};
+
+static struct regmap_irq_chip rt5677_irq_chip = {
+ .name = "rt5677",
+ .irqs = rt5677_irqs,
+ .num_irqs = ARRAY_SIZE(rt5677_irqs),
+
+ .num_regs = 1,
+ .status_base = RT5677_IRQ_CTRL1,
+ .mask_base = RT5677_IRQ_CTRL1,
+ .mask_invert = 1,
+};
+
+static int rt5677_init_irq(struct i2c_client *i2c)
+{
+ int ret;
+ struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+
+ if (!rt5677->pdata.jd1_gpio &&
+ !rt5677->pdata.jd2_gpio &&
+ !rt5677->pdata.jd3_gpio)
+ return 0;
+
+ if (!i2c->irq) {
+ dev_err(&i2c->dev, "No interrupt specified\n");
+ return -EINVAL;
+ }
+
+ ret = regmap_add_irq_chip(rt5677->regmap, i2c->irq,
+ IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, 0,
+ &rt5677_irq_chip, &rt5677->irq_data);
+
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register IRQ chip: %d\n", ret);
+ return ret;
+ }
+
return 0;
}
+static void rt5677_free_irq(struct i2c_client *i2c)
+{
+ struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+
+ if (rt5677->irq_data)
+ regmap_del_irq_chip(i2c->irq, rt5677->irq_data);
+}
+
static int rt5677_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -3638,7 +4691,16 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
msleep(10);
}
- rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap);
+ rt5677->regmap_physical = devm_regmap_init_i2c(i2c,
+ &rt5677_regmap_physical);
+ if (IS_ERR(rt5677->regmap_physical)) {
+ ret = PTR_ERR(rt5677->regmap_physical);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ rt5677->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5677_regmap);
if (IS_ERR(rt5677->regmap)) {
ret = PTR_ERR(rt5677->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -3690,6 +4752,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
}
rt5677_init_gpio(i2c);
+ rt5677_init_irq(i2c);
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
rt5677_dai, ARRAY_SIZE(rt5677_dai));
@@ -3698,6 +4761,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
static int rt5677_i2c_remove(struct i2c_client *i2c)
{
snd_soc_unregister_codec(&i2c->dev);
+ rt5677_free_irq(i2c);
rt5677_free_gpio(i2c);
return 0;
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index d4eb6d5e6746..c0a625f290cc 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -13,6 +13,7 @@
#define __RT5677_H__
#include <sound/rt5677.h>
+#include <linux/gpio/driver.h>
/* Info */
#define RT5677_RESET 0x00
@@ -305,10 +306,10 @@
#define RT5677_R_MUTE_SFT 7
#define RT5677_VOL_R_MUTE (0x1 << 6)
#define RT5677_VOL_R_SFT 6
-#define RT5677_L_VOL_MASK (0x3f << 8)
-#define RT5677_L_VOL_SFT 8
-#define RT5677_R_VOL_MASK (0x3f)
-#define RT5677_R_VOL_SFT 0
+#define RT5677_L_VOL_MASK (0x7f << 9)
+#define RT5677_L_VOL_SFT 9
+#define RT5677_R_VOL_MASK (0x7f << 1)
+#define RT5677_R_VOL_SFT 1
/* LOUT1 Control (0x01) */
#define RT5677_LOUT1_L_MUTE (0x1 << 15)
@@ -446,16 +447,16 @@
#define RT5677_SEL_DAC2_R_SRC_SFT 0
/* Stereo1 ADC Digital Volume Control (0x1c) */
-#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8)
-#define RT5677_STO1_ADC_L_VOL_SFT 8
-#define RT5677_STO1_ADC_R_VOL_MASK (0x7f)
-#define RT5677_STO1_ADC_R_VOL_SFT 0
+#define RT5677_STO1_ADC_L_VOL_MASK (0x3f << 9)
+#define RT5677_STO1_ADC_L_VOL_SFT 9
+#define RT5677_STO1_ADC_R_VOL_MASK (0x3f << 1)
+#define RT5677_STO1_ADC_R_VOL_SFT 1
/* Mono ADC Digital Volume Control (0x1d) */
-#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8)
-#define RT5677_MONO_ADC_L_VOL_SFT 8
-#define RT5677_MONO_ADC_R_VOL_MASK (0x7f)
-#define RT5677_MONO_ADC_R_VOL_SFT 0
+#define RT5677_MONO_ADC_L_VOL_MASK (0x3f << 9)
+#define RT5677_MONO_ADC_L_VOL_SFT 9
+#define RT5677_MONO_ADC_R_VOL_MASK (0x3f << 1)
+#define RT5677_MONO_ADC_R_VOL_SFT 1
/* Stereo 1/2 ADC Boost Gain Control (0x1e) */
#define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14)
@@ -798,7 +799,21 @@
#define RT5677_PDM2_I2C_EXE (0x1 << 1)
#define RT5677_PDM2_I2C_BUSY (0x1 << 0)
-/* MX3C TDM1 control 1 (0x3c) */
+/* TDM1 control 1 (0x3b) */
+#define RT5677_IF1_ADC_MODE_MASK (0x1 << 12)
+#define RT5677_IF1_ADC_MODE_SFT 12
+#define RT5677_IF1_ADC_MODE_I2S (0x0 << 12)
+#define RT5677_IF1_ADC_MODE_TDM (0x1 << 12)
+#define RT5677_IF1_ADC1_SWAP_MASK (0x3 << 6)
+#define RT5677_IF1_ADC1_SWAP_SFT 6
+#define RT5677_IF1_ADC2_SWAP_MASK (0x3 << 4)
+#define RT5677_IF1_ADC2_SWAP_SFT 4
+#define RT5677_IF1_ADC3_SWAP_MASK (0x3 << 2)
+#define RT5677_IF1_ADC3_SWAP_SFT 2
+#define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0)
+#define RT5677_IF1_ADC4_SWAP_SFT 0
+
+/* TDM1 control 2 (0x3c) */
#define RT5677_IF1_ADC4_MASK (0x3 << 10)
#define RT5677_IF1_ADC4_SFT 10
#define RT5677_IF1_ADC3_MASK (0x3 << 8)
@@ -807,8 +822,44 @@
#define RT5677_IF1_ADC2_SFT 6
#define RT5677_IF1_ADC1_MASK (0x3 << 4)
#define RT5677_IF1_ADC1_SFT 4
-
-/* MX41 TDM2 control 1 (0x41) */
+#define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0)
+#define RT5677_IF1_ADC_CTRL_SFT 0
+
+/* TDM1 control 4 (0x3e) */
+#define RT5677_IF1_DAC0_MASK (0x7 << 12)
+#define RT5677_IF1_DAC0_SFT 12
+#define RT5677_IF1_DAC1_MASK (0x7 << 8)
+#define RT5677_IF1_DAC1_SFT 8
+#define RT5677_IF1_DAC2_MASK (0x7 << 4)
+#define RT5677_IF1_DAC2_SFT 4
+#define RT5677_IF1_DAC3_MASK (0x7 << 0)
+#define RT5677_IF1_DAC3_SFT 0
+
+/* TDM1 control 5 (0x3f) */
+#define RT5677_IF1_DAC4_MASK (0x7 << 12)
+#define RT5677_IF1_DAC4_SFT 12
+#define RT5677_IF1_DAC5_MASK (0x7 << 8)
+#define RT5677_IF1_DAC5_SFT 8
+#define RT5677_IF1_DAC6_MASK (0x7 << 4)
+#define RT5677_IF1_DAC6_SFT 4
+#define RT5677_IF1_DAC7_MASK (0x7 << 0)
+#define RT5677_IF1_DAC7_SFT 0
+
+/* TDM2 control 1 (0x40) */
+#define RT5677_IF2_ADC_MODE_MASK (0x1 << 12)
+#define RT5677_IF2_ADC_MODE_SFT 12
+#define RT5677_IF2_ADC_MODE_I2S (0x0 << 12)
+#define RT5677_IF2_ADC_MODE_TDM (0x1 << 12)
+#define RT5677_IF2_ADC1_SWAP_MASK (0x3 << 6)
+#define RT5677_IF2_ADC1_SWAP_SFT 6
+#define RT5677_IF2_ADC2_SWAP_MASK (0x3 << 4)
+#define RT5677_IF2_ADC2_SWAP_SFT 4
+#define RT5677_IF2_ADC3_SWAP_MASK (0x3 << 2)
+#define RT5677_IF2_ADC3_SWAP_SFT 2
+#define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0)
+#define RT5677_IF2_ADC4_SWAP_SFT 0
+
+/* TDM2 control 2 (0x41) */
#define RT5677_IF2_ADC4_MASK (0x3 << 10)
#define RT5677_IF2_ADC4_SFT 10
#define RT5677_IF2_ADC3_MASK (0x3 << 8)
@@ -817,6 +868,28 @@
#define RT5677_IF2_ADC2_SFT 6
#define RT5677_IF2_ADC1_MASK (0x3 << 4)
#define RT5677_IF2_ADC1_SFT 4
+#define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0)
+#define RT5677_IF2_ADC_CTRL_SFT 0
+
+/* TDM2 control 4 (0x43) */
+#define RT5677_IF2_DAC0_MASK (0x7 << 12)
+#define RT5677_IF2_DAC0_SFT 12
+#define RT5677_IF2_DAC1_MASK (0x7 << 8)
+#define RT5677_IF2_DAC1_SFT 8
+#define RT5677_IF2_DAC2_MASK (0x7 << 4)
+#define RT5677_IF2_DAC2_SFT 4
+#define RT5677_IF2_DAC3_MASK (0x7 << 0)
+#define RT5677_IF2_DAC3_SFT 0
+
+/* TDM2 control 5 (0x44) */
+#define RT5677_IF2_DAC4_MASK (0x7 << 12)
+#define RT5677_IF2_DAC4_SFT 12
+#define RT5677_IF2_DAC5_MASK (0x7 << 8)
+#define RT5677_IF2_DAC5_SFT 8
+#define RT5677_IF2_DAC6_MASK (0x7 << 4)
+#define RT5677_IF2_DAC6_SFT 4
+#define RT5677_IF2_DAC7_MASK (0x7 << 0)
+#define RT5677_IF2_DAC7_SFT 0
/* Digital Microphone Control 1 (0x50) */
#define RT5677_DMIC_1_EN_MASK (0x1 << 15)
@@ -1367,6 +1440,48 @@
#define RT5677_SEL_SRC_IB01 (0x1 << 0)
#define RT5677_SEL_SRC_IB01_SFT 0
+/* Jack Detect Control 1 (0xb5) */
+#define RT5677_SEL_GPIO_JD1_MASK (0x3 << 14)
+#define RT5677_SEL_GPIO_JD1_SFT 14
+#define RT5677_SEL_GPIO_JD2_MASK (0x3 << 12)
+#define RT5677_SEL_GPIO_JD2_SFT 12
+#define RT5677_SEL_GPIO_JD3_MASK (0x3 << 10)
+#define RT5677_SEL_GPIO_JD3_SFT 10
+
+/* IRQ Control 1 (0xbd) */
+#define RT5677_STA_GPIO_JD1 (0x1 << 15)
+#define RT5677_STA_GPIO_JD1_SFT 15
+#define RT5677_EN_IRQ_GPIO_JD1 (0x1 << 14)
+#define RT5677_EN_IRQ_GPIO_JD1_SFT 14
+#define RT5677_EN_GPIO_JD1_STICKY (0x1 << 13)
+#define RT5677_EN_GPIO_JD1_STICKY_SFT 13
+#define RT5677_INV_GPIO_JD1 (0x1 << 12)
+#define RT5677_INV_GPIO_JD1_SFT 12
+#define RT5677_STA_GPIO_JD2 (0x1 << 11)
+#define RT5677_STA_GPIO_JD2_SFT 11
+#define RT5677_EN_IRQ_GPIO_JD2 (0x1 << 10)
+#define RT5677_EN_IRQ_GPIO_JD2_SFT 10
+#define RT5677_EN_GPIO_JD2_STICKY (0x1 << 9)
+#define RT5677_EN_GPIO_JD2_STICKY_SFT 9
+#define RT5677_INV_GPIO_JD2 (0x1 << 8)
+#define RT5677_INV_GPIO_JD2_SFT 8
+#define RT5677_STA_MICBIAS1_OVCD (0x1 << 7)
+#define RT5677_STA_MICBIAS1_OVCD_SFT 7
+#define RT5677_EN_IRQ_MICBIAS1_OVCD (0x1 << 6)
+#define RT5677_EN_IRQ_MICBIAS1_OVCD_SFT 6
+#define RT5677_EN_MICBIAS1_OVCD_STICKY (0x1 << 5)
+#define RT5677_EN_MICBIAS1_OVCD_STICKY_SFT 5
+#define RT5677_INV_MICBIAS1_OVCD (0x1 << 4)
+#define RT5677_INV_MICBIAS1_OVCD_SFT 4
+#define RT5677_STA_GPIO_JD3 (0x1 << 3)
+#define RT5677_STA_GPIO_JD3_SFT 3
+#define RT5677_EN_IRQ_GPIO_JD3 (0x1 << 2)
+#define RT5677_EN_IRQ_GPIO_JD3_SFT 2
+#define RT5677_EN_GPIO_JD3_STICKY (0x1 << 1)
+#define RT5677_EN_GPIO_JD3_STICKY_SFT 1
+#define RT5677_INV_GPIO_JD3 (0x1 << 0)
+#define RT5677_INV_GPIO_JD3_SFT 0
+
/* GPIO status (0xbf) */
#define RT5677_GPIO6_STATUS_MASK (0x1 << 5)
#define RT5677_GPIO6_STATUS_SFT 5
@@ -1506,6 +1621,9 @@
#define RT5677_GPIO5_FUNC_GPIO (0x0 << 9)
#define RT5677_GPIO5_FUNC_DMIC (0x1 << 9)
+#define RT5677_FIRMWARE1 "rt5677_dsp_fw1.bin"
+#define RT5677_FIRMWARE2 "rt5677_dsp_fw2.bin"
+
/* System Clock Source */
enum {
RT5677_SCLK_S_MCLK,
@@ -1541,10 +1659,18 @@ enum {
RT5677_GPIO_NUM,
};
+enum {
+ RT5677_IRQ_JD1,
+ RT5677_IRQ_JD2,
+ RT5677_IRQ_JD3,
+};
+
struct rt5677_priv {
struct snd_soc_codec *codec;
struct rt5677_platform_data pdata;
- struct regmap *regmap;
+ struct regmap *regmap, *regmap_physical;
+ const struct firmware *fw1, *fw2;
+ struct mutex dsp_cmd_lock, dsp_pri_lock;
int sysclk;
int sysclk_src;
@@ -1558,6 +1684,10 @@ struct rt5677_priv {
#ifdef CONFIG_GPIOLIB
struct gpio_chip gpio_chip;
#endif
+ bool dsp_vad_en;
+ struct regmap_irq_chip_data *irq_data;
+ bool is_dsp_mode;
+ bool is_vref_slow;
};
#endif /* __RT5677_H__ */
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index dab9b15304af..29cf7ce610f4 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -16,6 +16,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/clk.h>
+#include <linux/log2.h>
#include <linux/regmap.h>
#include <linux/regulator/driver.h>
#include <linux/regulator/machine.h>
@@ -121,6 +122,13 @@ struct ldo_regulator {
bool enabled;
};
+enum sgtl5000_micbias_resistor {
+ SGTL5000_MICBIAS_OFF = 0,
+ SGTL5000_MICBIAS_2K = 2,
+ SGTL5000_MICBIAS_4K = 4,
+ SGTL5000_MICBIAS_8K = 8,
+};
+
/* sgtl5000 private structure in codec */
struct sgtl5000_priv {
int sysclk; /* sysclk rate */
@@ -131,6 +139,8 @@ struct sgtl5000_priv {
struct regmap *regmap;
struct clk *mclk;
int revision;
+ u8 micbias_resistor;
+ u8 micbias_voltage;
};
/*
@@ -145,12 +155,14 @@ struct sgtl5000_priv {
static int mic_bias_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec);
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- /* change mic bias resistor to 4Kohm */
+ /* change mic bias resistor */
snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
- SGTL5000_BIAS_R_MASK,
- SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT);
+ SGTL5000_BIAS_R_MASK,
+ sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
break;
case SND_SOC_DAPM_PRE_PMD:
@@ -530,16 +542,16 @@ static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai,
/*
* set clock according to i2s frame clock,
- * sgtl5000 provide 2 clock sources.
- * 1. sys_mclk. sample freq can only configure to
+ * sgtl5000 provides 2 clock sources:
+ * 1. sys_mclk: sample freq can only be configured to
* 1/256, 1/384, 1/512 of sys_mclk.
- * 2. pll. can derive any audio clocks.
+ * 2. pll: can derive any audio clocks.
*
* clock setting rules:
- * 1. in slave mode, only sys_mclk can use.
- * 2. as constraint by sys_mclk, sample freq should
- * set to 32k, 44.1k and above.
- * 3. using sys_mclk prefer to pll to save power.
+ * 1. in slave mode, only sys_mclk can be used
+ * 2. as constraint by sys_mclk, sample freq should be set to 32 kHz, 44.1 kHz
+ * and above.
+ * 3. usage of sys_mclk is preferred over pll to save power.
*/
static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
{
@@ -549,8 +561,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
/*
* sample freq should be divided by frame clock,
- * if frame clock lower than 44.1khz, sample feq should set to
- * 32khz or 44.1khz.
+ * if frame clock is lower than 44.1 kHz, sample freq should be set to
+ * 32 kHz or 44.1 kHz.
*/
switch (frame_rate) {
case 8000:
@@ -603,9 +615,10 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
/*
* calculate the divider of mclk/sample_freq,
- * factor of freq =96k can only be 256, since mclk in range (12m,27m)
+ * factor of freq = 96 kHz can only be 256, since mclk is in the range
+ * of 8 MHz - 27 MHz
*/
- switch (sgtl5000->sysclk / sys_fs) {
+ switch (sgtl5000->sysclk / frame_rate) {
case 256:
clk_ctl |= SGTL5000_MCLK_FREQ_256FS <<
SGTL5000_MCLK_FREQ_SHIFT;
@@ -619,7 +632,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
SGTL5000_MCLK_FREQ_SHIFT;
break;
default:
- /* if mclk not satisify the divider, use pll */
+ /* if mclk does not satisfy the divider, use pll */
if (sgtl5000->master) {
clk_ctl |= SGTL5000_MCLK_FREQ_PLL <<
SGTL5000_MCLK_FREQ_SHIFT;
@@ -628,7 +641,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
"PLL not supported in slave mode\n");
dev_err(codec->dev, "%d ratio is not supported. "
"SYS_MCLK needs to be 256, 384 or 512 * fs\n",
- sgtl5000->sysclk / sys_fs);
+ sgtl5000->sysclk / frame_rate);
return -EINVAL;
}
}
@@ -795,7 +808,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev)
SGTL5000_LINEREG_D_POWERUP,
SGTL5000_LINEREG_D_POWERUP);
- /* when internal ldo enabled, simple digital power can be disabled */
+ /* when internal ldo is enabled, simple digital power can be disabled */
snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
SGTL5000_LINREG_SIMPLE_POWERUP,
0);
@@ -1079,7 +1092,7 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg)
/*
* sgtl5000 has 3 internal power supplies:
* 1. VAG, normally set to vdda/2
- * 2. chargepump, set to different value
+ * 2. charge pump, set to different value
* according to voltage of vdda and vddio
* 3. line out VAG, normally set to vddio/2
*
@@ -1325,8 +1338,13 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
SGTL5000_HP_ZCD_EN |
SGTL5000_ADC_ZCD_EN);
- snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
+ snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
+ SGTL5000_BIAS_R_MASK,
+ sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
+ snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
+ SGTL5000_BIAS_R_MASK,
+ sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
/*
* disable DAP
* TODO:
@@ -1416,10 +1434,10 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
{
struct sgtl5000_priv *sgtl5000;
int ret, reg, rev;
- unsigned int mclk;
+ struct device_node *np = client->dev.of_node;
+ u32 value;
- sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv),
- GFP_KERNEL);
+ sgtl5000 = devm_kzalloc(&client->dev, sizeof(*sgtl5000), GFP_KERNEL);
if (!sgtl5000)
return -ENOMEM;
@@ -1440,14 +1458,6 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
return ret;
}
- /* SGTL5000 SYS_MCLK should be between 8 and 27 MHz */
- mclk = clk_get_rate(sgtl5000->mclk);
- if (mclk < 8000000 || mclk > 27000000) {
- dev_err(&client->dev, "Invalid SYS_CLK frequency: %u.%03uMHz\n",
- mclk / 1000000, mclk / 1000 % 1000);
- return -EINVAL;
- }
-
ret = clk_prepare_enable(sgtl5000->mclk);
if (ret)
return ret;
@@ -1469,6 +1479,47 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev);
sgtl5000->revision = rev;
+ if (np) {
+ if (!of_property_read_u32(np,
+ "micbias-resistor-k-ohms", &value)) {
+ switch (value) {
+ case SGTL5000_MICBIAS_OFF:
+ sgtl5000->micbias_resistor = 0;
+ break;
+ case SGTL5000_MICBIAS_2K:
+ sgtl5000->micbias_resistor = 1;
+ break;
+ case SGTL5000_MICBIAS_4K:
+ sgtl5000->micbias_resistor = 2;
+ break;
+ case SGTL5000_MICBIAS_8K:
+ sgtl5000->micbias_resistor = 3;
+ break;
+ default:
+ sgtl5000->micbias_resistor = 2;
+ dev_err(&client->dev,
+ "Unsuitable MicBias resistor\n");
+ }
+ } else {
+ /* default is 4Kohms */
+ sgtl5000->micbias_resistor = 2;
+ }
+ if (!of_property_read_u32(np,
+ "micbias-voltage-m-volts", &value)) {
+ /* 1250mV => 0 */
+ /* steps of 250mV */
+ if ((value >= 1250) && (value <= 3000))
+ sgtl5000->micbias_voltage = (value / 250) - 5;
+ else {
+ sgtl5000->micbias_voltage = 0;
+ dev_err(&client->dev,
+ "Unsuitable MicBias resistor\n");
+ }
+ } else {
+ sgtl5000->micbias_voltage = 0;
+ }
+ }
+
i2c_set_clientdata(client, sgtl5000);
/* Ensure sgtl5000 will start with sane register values */
diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c
index 246081aae8ca..21ca3a5e9f66 100644
--- a/sound/soc/codecs/sigmadsp-i2c.c
+++ b/sound/soc/codecs/sigmadsp-i2c.c
@@ -6,29 +6,88 @@
* Licensed under the GPL-2 or later.
*/
-#include <linux/i2c.h>
#include <linux/export.h>
+#include <linux/i2c.h>
#include <linux/module.h>
+#include <linux/slab.h>
+#include <asm/unaligned.h>
#include "sigmadsp.h"
-static int sigma_action_write_i2c(void *control_data,
- const struct sigma_action *sa, size_t len)
+static int sigmadsp_write_i2c(void *control_data,
+ unsigned int addr, const uint8_t data[], size_t len)
+{
+ uint8_t *buf;
+ int ret;
+
+ buf = kzalloc(2 + len, GFP_KERNEL | GFP_DMA);
+ if (!buf)
+ return -ENOMEM;
+
+ put_unaligned_be16(addr, buf);
+ memcpy(buf + 2, data, len);
+
+ ret = i2c_master_send(control_data, buf, len + 2);
+
+ kfree(buf);
+
+ return ret;
+}
+
+static int sigmadsp_read_i2c(void *control_data,
+ unsigned int addr, uint8_t data[], size_t len)
{
- return i2c_master_send(control_data, (const unsigned char *)&sa->addr,
- len);
+ struct i2c_client *client = control_data;
+ struct i2c_msg msgs[2];
+ uint8_t buf[2];
+ int ret;
+
+ put_unaligned_be16(addr, buf);
+
+ msgs[0].addr = client->addr;
+ msgs[0].len = sizeof(buf);
+ msgs[0].buf = buf;
+ msgs[0].flags = 0;
+
+ msgs[1].addr = client->addr;
+ msgs[1].len = len;
+ msgs[1].buf = data;
+ msgs[1].flags = I2C_M_RD;
+
+ ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs));
+ if (ret < 0)
+ return ret;
+ else if (ret != ARRAY_SIZE(msgs))
+ return -EIO;
+ return 0;
}
-int process_sigma_firmware(struct i2c_client *client, const char *name)
+/**
+ * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance
+ * @client: The parent I2C device
+ * @ops: The sigmadsp_ops to use for this instance
+ * @firmware_name: Name of the firmware file to load
+ *
+ * Allocates a SigmaDSP instance and loads the specified firmware file.
+ *
+ * Returns a pointer to a struct sigmadsp on success, or a PTR_ERR() on error.
+ */
+struct sigmadsp *devm_sigmadsp_init_i2c(struct i2c_client *client,
+ const struct sigmadsp_ops *ops, const char *firmware_name)
{
- struct sigma_firmware ssfw;
+ struct sigmadsp *sigmadsp;
+
+ sigmadsp = devm_sigmadsp_init(&client->dev, ops, firmware_name);
+ if (IS_ERR(sigmadsp))
+ return sigmadsp;
- ssfw.control_data = client;
- ssfw.write = sigma_action_write_i2c;
+ sigmadsp->control_data = client;
+ sigmadsp->write = sigmadsp_write_i2c;
+ sigmadsp->read = sigmadsp_read_i2c;
- return _process_sigma_firmware(&client->dev, &ssfw, name);
+ return sigmadsp;
}
-EXPORT_SYMBOL(process_sigma_firmware);
+EXPORT_SYMBOL_GPL(devm_sigmadsp_init_i2c);
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_DESCRIPTION("SigmaDSP I2C firmware loader");
diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c
index f78ed8d2cfb2..912861be5b87 100644
--- a/sound/soc/codecs/sigmadsp-regmap.c
+++ b/sound/soc/codecs/sigmadsp-regmap.c
@@ -12,24 +12,48 @@
#include "sigmadsp.h"
-static int sigma_action_write_regmap(void *control_data,
- const struct sigma_action *sa, size_t len)
+static int sigmadsp_write_regmap(void *control_data,
+ unsigned int addr, const uint8_t data[], size_t len)
{
- return regmap_raw_write(control_data, be16_to_cpu(sa->addr),
- sa->payload, len - 2);
+ return regmap_raw_write(control_data, addr,
+ data, len);
}
-int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap,
- const char *name)
+static int sigmadsp_read_regmap(void *control_data,
+ unsigned int addr, uint8_t data[], size_t len)
{
- struct sigma_firmware ssfw;
+ return regmap_raw_read(control_data, addr,
+ data, len);
+}
+
+/**
+ * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance
+ * @dev: The parent device
+ * @regmap: Regmap instance to use
+ * @ops: The sigmadsp_ops to use for this instance
+ * @firmware_name: Name of the firmware file to load
+ *
+ * Allocates a SigmaDSP instance and loads the specified firmware file.
+ *
+ * Returns a pointer to a struct sigmadsp on success, or a PTR_ERR() on error.
+ */
+struct sigmadsp *devm_sigmadsp_init_regmap(struct device *dev,
+ struct regmap *regmap, const struct sigmadsp_ops *ops,
+ const char *firmware_name)
+{
+ struct sigmadsp *sigmadsp;
+
+ sigmadsp = devm_sigmadsp_init(dev, ops, firmware_name);
+ if (IS_ERR(sigmadsp))
+ return sigmadsp;
- ssfw.control_data = regmap;
- ssfw.write = sigma_action_write_regmap;
+ sigmadsp->control_data = regmap;
+ sigmadsp->write = sigmadsp_write_regmap;
+ sigmadsp->read = sigmadsp_read_regmap;
- return _process_sigma_firmware(dev, &ssfw, name);
+ return sigmadsp;
}
-EXPORT_SYMBOL(process_sigma_firmware_regmap);
+EXPORT_SYMBOL_GPL(devm_sigmadsp_init_regmap);
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_DESCRIPTION("SigmaDSP regmap firmware loader");
diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c
index f2de7e049bc6..d53680ac78e4 100644
--- a/sound/soc/codecs/sigmadsp.c
+++ b/sound/soc/codecs/sigmadsp.c
@@ -1,23 +1,74 @@
/*
* Load Analog Devices SigmaStudio firmware files
*
- * Copyright 2009-2011 Analog Devices Inc.
+ * Copyright 2009-2014 Analog Devices Inc.
*
* Licensed under the GPL-2 or later.
*/
#include <linux/crc32.h>
-#include <linux/delay.h>
#include <linux/firmware.h>
#include <linux/kernel.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/module.h>
+#include <linux/slab.h>
+
+#include <sound/control.h>
+#include <sound/soc.h>
#include "sigmadsp.h"
#define SIGMA_MAGIC "ADISIGM"
+#define SIGMA_FW_CHUNK_TYPE_DATA 0
+#define SIGMA_FW_CHUNK_TYPE_CONTROL 1
+#define SIGMA_FW_CHUNK_TYPE_SAMPLERATES 2
+
+struct sigmadsp_control {
+ struct list_head head;
+ uint32_t samplerates;
+ unsigned int addr;
+ unsigned int num_bytes;
+ const char *name;
+ struct snd_kcontrol *kcontrol;
+ bool cached;
+ uint8_t cache[];
+};
+
+struct sigmadsp_data {
+ struct list_head head;
+ uint32_t samplerates;
+ unsigned int addr;
+ unsigned int length;
+ uint8_t data[];
+};
+
+struct sigma_fw_chunk {
+ __le32 length;
+ __le32 tag;
+ __le32 samplerates;
+} __packed;
+
+struct sigma_fw_chunk_data {
+ struct sigma_fw_chunk chunk;
+ __le16 addr;
+ uint8_t data[];
+} __packed;
+
+struct sigma_fw_chunk_control {
+ struct sigma_fw_chunk chunk;
+ __le16 type;
+ __le16 addr;
+ __le16 num_bytes;
+ const char name[];
+} __packed;
+
+struct sigma_fw_chunk_samplerate {
+ struct sigma_fw_chunk chunk;
+ __le32 samplerates[];
+} __packed;
+
struct sigma_firmware_header {
unsigned char magic[7];
u8 version;
@@ -28,12 +79,286 @@ enum {
SIGMA_ACTION_WRITEXBYTES = 0,
SIGMA_ACTION_WRITESINGLE,
SIGMA_ACTION_WRITESAFELOAD,
- SIGMA_ACTION_DELAY,
- SIGMA_ACTION_PLLWAIT,
- SIGMA_ACTION_NOOP,
SIGMA_ACTION_END,
};
+struct sigma_action {
+ u8 instr;
+ u8 len_hi;
+ __le16 len;
+ __be16 addr;
+ unsigned char payload[];
+} __packed;
+
+static int sigmadsp_write(struct sigmadsp *sigmadsp, unsigned int addr,
+ const uint8_t data[], size_t len)
+{
+ return sigmadsp->write(sigmadsp->control_data, addr, data, len);
+}
+
+static int sigmadsp_read(struct sigmadsp *sigmadsp, unsigned int addr,
+ uint8_t data[], size_t len)
+{
+ return sigmadsp->read(sigmadsp->control_data, addr, data, len);
+}
+
+static int sigmadsp_ctrl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *info)
+{
+ struct sigmadsp_control *ctrl = (void *)kcontrol->private_value;
+
+ info->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ info->count = ctrl->num_bytes;
+
+ return 0;
+}
+
+static int sigmadsp_ctrl_write(struct sigmadsp *sigmadsp,
+ struct sigmadsp_control *ctrl, void *data)
+{
+ /* safeload loads up to 20 bytes in a atomic operation */
+ if (ctrl->num_bytes > 4 && ctrl->num_bytes <= 20 && sigmadsp->ops &&
+ sigmadsp->ops->safeload)
+ return sigmadsp->ops->safeload(sigmadsp, ctrl->addr, data,
+ ctrl->num_bytes);
+ else
+ return sigmadsp_write(sigmadsp, ctrl->addr, data,
+ ctrl->num_bytes);
+}
+
+static int sigmadsp_ctrl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct sigmadsp_control *ctrl = (void *)kcontrol->private_value;
+ struct sigmadsp *sigmadsp = snd_kcontrol_chip(kcontrol);
+ uint8_t *data;
+ int ret = 0;
+
+ mutex_lock(&sigmadsp->lock);
+
+ data = ucontrol->value.bytes.data;
+
+ if (!(kcontrol->vd[0].access & SNDRV_CTL_ELEM_ACCESS_INACTIVE))
+ ret = sigmadsp_ctrl_write(sigmadsp, ctrl, data);
+
+ if (ret == 0) {
+ memcpy(ctrl->cache, data, ctrl->num_bytes);
+ ctrl->cached = true;
+ }
+
+ mutex_unlock(&sigmadsp->lock);
+
+ return ret;
+}
+
+static int sigmadsp_ctrl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct sigmadsp_control *ctrl = (void *)kcontrol->private_value;
+ struct sigmadsp *sigmadsp = snd_kcontrol_chip(kcontrol);
+ int ret = 0;
+
+ mutex_lock(&sigmadsp->lock);
+
+ if (!ctrl->cached) {
+ ret = sigmadsp_read(sigmadsp, ctrl->addr, ctrl->cache,
+ ctrl->num_bytes);
+ }
+
+ if (ret == 0) {
+ ctrl->cached = true;
+ memcpy(ucontrol->value.bytes.data, ctrl->cache,
+ ctrl->num_bytes);
+ }
+
+ mutex_unlock(&sigmadsp->lock);
+
+ return ret;
+}
+
+static void sigmadsp_control_free(struct snd_kcontrol *kcontrol)
+{
+ struct sigmadsp_control *ctrl = (void *)kcontrol->private_value;
+
+ ctrl->kcontrol = NULL;
+}
+
+static bool sigma_fw_validate_control_name(const char *name, unsigned int len)
+{
+ unsigned int i;
+
+ for (i = 0; i < len; i++) {
+ /* Normal ASCII characters are valid */
+ if (name[i] < ' ' || name[i] > '~')
+ return false;
+ }
+
+ return true;
+}
+
+static int sigma_fw_load_control(struct sigmadsp *sigmadsp,
+ const struct sigma_fw_chunk *chunk, unsigned int length)
+{
+ const struct sigma_fw_chunk_control *ctrl_chunk;
+ struct sigmadsp_control *ctrl;
+ unsigned int num_bytes;
+ size_t name_len;
+ char *name;
+ int ret;
+
+ if (length <= sizeof(*ctrl_chunk))
+ return -EINVAL;
+
+ ctrl_chunk = (const struct sigma_fw_chunk_control *)chunk;
+
+ name_len = length - sizeof(*ctrl_chunk);
+ if (name_len >= SNDRV_CTL_ELEM_ID_NAME_MAXLEN)
+ name_len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - 1;
+
+ /* Make sure there are no non-displayable characaters in the string */
+ if (!sigma_fw_validate_control_name(ctrl_chunk->name, name_len))
+ return -EINVAL;
+
+ num_bytes = le16_to_cpu(ctrl_chunk->num_bytes);
+ ctrl = kzalloc(sizeof(*ctrl) + num_bytes, GFP_KERNEL);
+ if (!ctrl)
+ return -ENOMEM;
+
+ name = kzalloc(name_len + 1, GFP_KERNEL);
+ if (!name) {
+ ret = -ENOMEM;
+ goto err_free_ctrl;
+ }
+ memcpy(name, ctrl_chunk->name, name_len);
+ name[name_len] = '\0';
+ ctrl->name = name;
+
+ ctrl->addr = le16_to_cpu(ctrl_chunk->addr);
+ ctrl->num_bytes = num_bytes;
+ ctrl->samplerates = le32_to_cpu(chunk->samplerates);
+
+ list_add_tail(&ctrl->head, &sigmadsp->ctrl_list);
+
+ return 0;
+
+err_free_ctrl:
+ kfree(ctrl);
+
+ return ret;
+}
+
+static int sigma_fw_load_data(struct sigmadsp *sigmadsp,
+ const struct sigma_fw_chunk *chunk, unsigned int length)
+{
+ const struct sigma_fw_chunk_data *data_chunk;
+ struct sigmadsp_data *data;
+
+ if (length <= sizeof(*data_chunk))
+ return -EINVAL;
+
+ data_chunk = (struct sigma_fw_chunk_data *)chunk;
+
+ length -= sizeof(*data_chunk);
+
+ data = kzalloc(sizeof(*data) + length, GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ data->addr = le16_to_cpu(data_chunk->addr);
+ data->length = length;
+ data->samplerates = le32_to_cpu(chunk->samplerates);
+ memcpy(data->data, data_chunk->data, length);
+ list_add_tail(&data->head, &sigmadsp->data_list);
+
+ return 0;
+}
+
+static int sigma_fw_load_samplerates(struct sigmadsp *sigmadsp,
+ const struct sigma_fw_chunk *chunk, unsigned int length)
+{
+ const struct sigma_fw_chunk_samplerate *rate_chunk;
+ unsigned int num_rates;
+ unsigned int *rates;
+ unsigned int i;
+
+ rate_chunk = (const struct sigma_fw_chunk_samplerate *)chunk;
+
+ num_rates = (length - sizeof(*rate_chunk)) / sizeof(__le32);
+
+ if (num_rates > 32 || num_rates == 0)
+ return -EINVAL;
+
+ /* We only allow one samplerates block per file */
+ if (sigmadsp->rate_constraints.count)
+ return -EINVAL;
+
+ rates = kcalloc(num_rates, sizeof(*rates), GFP_KERNEL);
+ if (!rates)
+ return -ENOMEM;
+
+ for (i = 0; i < num_rates; i++)
+ rates[i] = le32_to_cpu(rate_chunk->samplerates[i]);
+
+ sigmadsp->rate_constraints.count = num_rates;
+ sigmadsp->rate_constraints.list = rates;
+
+ return 0;
+}
+
+static int sigmadsp_fw_load_v2(struct sigmadsp *sigmadsp,
+ const struct firmware *fw)
+{
+ struct sigma_fw_chunk *chunk;
+ unsigned int length, pos;
+ int ret;
+
+ /*
+ * Make sure that there is at least one chunk to avoid integer
+ * underflows later on. Empty firmware is still valid though.
+ */
+ if (fw->size < sizeof(*chunk) + sizeof(struct sigma_firmware_header))
+ return 0;
+
+ pos = sizeof(struct sigma_firmware_header);
+
+ while (pos < fw->size - sizeof(*chunk)) {
+ chunk = (struct sigma_fw_chunk *)(fw->data + pos);
+
+ length = le32_to_cpu(chunk->length);
+
+ if (length > fw->size - pos || length < sizeof(*chunk))
+ return -EINVAL;
+
+ switch (le32_to_cpu(chunk->tag)) {
+ case SIGMA_FW_CHUNK_TYPE_DATA:
+ ret = sigma_fw_load_data(sigmadsp, chunk, length);
+ break;
+ case SIGMA_FW_CHUNK_TYPE_CONTROL:
+ ret = sigma_fw_load_control(sigmadsp, chunk, length);
+ break;
+ case SIGMA_FW_CHUNK_TYPE_SAMPLERATES:
+ ret = sigma_fw_load_samplerates(sigmadsp, chunk, length);
+ break;
+ default:
+ dev_warn(sigmadsp->dev, "Unknown chunk type: %d\n",
+ chunk->tag);
+ ret = 0;
+ break;
+ }
+
+ if (ret)
+ return ret;
+
+ /*
+ * This can not overflow since if length is larger than the
+ * maximum firmware size (0x4000000) we'll error out earilier.
+ */
+ pos += ALIGN(length, sizeof(__le32));
+ }
+
+ return 0;
+}
+
static inline u32 sigma_action_len(struct sigma_action *sa)
{
return (sa->len_hi << 16) | le16_to_cpu(sa->len);
@@ -62,11 +387,11 @@ static size_t sigma_action_size(struct sigma_action *sa)
* Returns a negative error value in case of an error, 0 if processing of
* the firmware should be stopped after this action, 1 otherwise.
*/
-static int
-process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa)
+static int process_sigma_action(struct sigmadsp *sigmadsp,
+ struct sigma_action *sa)
{
size_t len = sigma_action_len(sa);
- int ret;
+ struct sigmadsp_data *data;
pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__,
sa->instr, sa->addr, len);
@@ -75,13 +400,17 @@ process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa)
case SIGMA_ACTION_WRITEXBYTES:
case SIGMA_ACTION_WRITESINGLE:
case SIGMA_ACTION_WRITESAFELOAD:
- ret = ssfw->write(ssfw->control_data, sa, len);
- if (ret < 0)
+ if (len < 3)
return -EINVAL;
- break;
- case SIGMA_ACTION_DELAY:
- udelay(len);
- len = 0;
+
+ data = kzalloc(sizeof(*data) + len - 2, GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ data->addr = be16_to_cpu(sa->addr);
+ data->length = len - 2;
+ memcpy(data->data, sa->payload, data->length);
+ list_add_tail(&data->head, &sigmadsp->data_list);
break;
case SIGMA_ACTION_END:
return 0;
@@ -92,22 +421,24 @@ process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa)
return 1;
}
-static int
-process_sigma_actions(struct sigma_firmware *ssfw)
+static int sigmadsp_fw_load_v1(struct sigmadsp *sigmadsp,
+ const struct firmware *fw)
{
struct sigma_action *sa;
- size_t size;
+ size_t size, pos;
int ret;
- while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) {
- sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos);
+ pos = sizeof(struct sigma_firmware_header);
+
+ while (pos + sizeof(*sa) <= fw->size) {
+ sa = (struct sigma_action *)(fw->data + pos);
size = sigma_action_size(sa);
- ssfw->pos += size;
- if (ssfw->pos > ssfw->fw->size || size == 0)
+ pos += size;
+ if (pos > fw->size || size == 0)
break;
- ret = process_sigma_action(ssfw, sa);
+ ret = process_sigma_action(sigmadsp, sa);
pr_debug("%s: action returned %i\n", __func__, ret);
@@ -115,29 +446,47 @@ process_sigma_actions(struct sigma_firmware *ssfw)
return ret;
}
- if (ssfw->pos != ssfw->fw->size)
+ if (pos != fw->size)
return -EINVAL;
return 0;
}
-int _process_sigma_firmware(struct device *dev,
- struct sigma_firmware *ssfw, const char *name)
+static void sigmadsp_firmware_release(struct sigmadsp *sigmadsp)
{
- int ret;
- struct sigma_firmware_header *ssfw_head;
+ struct sigmadsp_control *ctrl, *_ctrl;
+ struct sigmadsp_data *data, *_data;
+
+ list_for_each_entry_safe(ctrl, _ctrl, &sigmadsp->ctrl_list, head) {
+ kfree(ctrl->name);
+ kfree(ctrl);
+ }
+
+ list_for_each_entry_safe(data, _data, &sigmadsp->data_list, head)
+ kfree(data);
+
+ INIT_LIST_HEAD(&sigmadsp->ctrl_list);
+ INIT_LIST_HEAD(&sigmadsp->data_list);
+}
+
+static void devm_sigmadsp_release(struct device *dev, void *res)
+{
+ sigmadsp_firmware_release((struct sigmadsp *)res);
+}
+
+static int sigmadsp_firmware_load(struct sigmadsp *sigmadsp, const char *name)
+{
+ const struct sigma_firmware_header *ssfw_head;
const struct firmware *fw;
+ int ret;
u32 crc;
- pr_debug("%s: loading firmware %s\n", __func__, name);
-
/* first load the blob */
- ret = request_firmware(&fw, name, dev);
+ ret = request_firmware(&fw, name, sigmadsp->dev);
if (ret) {
pr_debug("%s: request_firmware() failed with %i\n", __func__, ret);
- return ret;
+ goto done;
}
- ssfw->fw = fw;
/* then verify the header */
ret = -EINVAL;
@@ -149,13 +498,13 @@ int _process_sigma_firmware(struct device *dev,
* overflows later in the loading process.
*/
if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) {
- dev_err(dev, "Failed to load firmware: Invalid size\n");
+ dev_err(sigmadsp->dev, "Failed to load firmware: Invalid size\n");
goto done;
}
ssfw_head = (void *)fw->data;
if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) {
- dev_err(dev, "Failed to load firmware: Invalid magic\n");
+ dev_err(sigmadsp->dev, "Failed to load firmware: Invalid magic\n");
goto done;
}
@@ -163,23 +512,303 @@ int _process_sigma_firmware(struct device *dev,
fw->size - sizeof(*ssfw_head));
pr_debug("%s: crc=%x\n", __func__, crc);
if (crc != le32_to_cpu(ssfw_head->crc)) {
- dev_err(dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n",
+ dev_err(sigmadsp->dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n",
le32_to_cpu(ssfw_head->crc), crc);
goto done;
}
- ssfw->pos = sizeof(*ssfw_head);
+ switch (ssfw_head->version) {
+ case 1:
+ ret = sigmadsp_fw_load_v1(sigmadsp, fw);
+ break;
+ case 2:
+ ret = sigmadsp_fw_load_v2(sigmadsp, fw);
+ break;
+ default:
+ dev_err(sigmadsp->dev,
+ "Failed to load firmware: Invalid version %d. Supported firmware versions: 1, 2\n",
+ ssfw_head->version);
+ ret = -EINVAL;
+ break;
+ }
- /* finally process all of the actions */
- ret = process_sigma_actions(ssfw);
+ if (ret)
+ sigmadsp_firmware_release(sigmadsp);
- done:
+done:
release_firmware(fw);
- pr_debug("%s: loaded %s\n", __func__, name);
+ return ret;
+}
+
+static int sigmadsp_init(struct sigmadsp *sigmadsp, struct device *dev,
+ const struct sigmadsp_ops *ops, const char *firmware_name)
+{
+ sigmadsp->ops = ops;
+ sigmadsp->dev = dev;
+
+ INIT_LIST_HEAD(&sigmadsp->ctrl_list);
+ INIT_LIST_HEAD(&sigmadsp->data_list);
+ mutex_init(&sigmadsp->lock);
+
+ return sigmadsp_firmware_load(sigmadsp, firmware_name);
+}
+
+/**
+ * devm_sigmadsp_init() - Initialize SigmaDSP instance
+ * @dev: The parent device
+ * @ops: The sigmadsp_ops to use for this instance
+ * @firmware_name: Name of the firmware file to load
+ *
+ * Allocates a SigmaDSP instance and loads the specified firmware file.
+ *
+ * Returns a pointer to a struct sigmadsp on success, or a PTR_ERR() on error.
+ */
+struct sigmadsp *devm_sigmadsp_init(struct device *dev,
+ const struct sigmadsp_ops *ops, const char *firmware_name)
+{
+ struct sigmadsp *sigmadsp;
+ int ret;
+
+ sigmadsp = devres_alloc(devm_sigmadsp_release, sizeof(*sigmadsp),
+ GFP_KERNEL);
+ if (!sigmadsp)
+ return ERR_PTR(-ENOMEM);
+
+ ret = sigmadsp_init(sigmadsp, dev, ops, firmware_name);
+ if (ret) {
+ devres_free(sigmadsp);
+ return ERR_PTR(ret);
+ }
+
+ devres_add(dev, sigmadsp);
+
+ return sigmadsp;
+}
+EXPORT_SYMBOL_GPL(devm_sigmadsp_init);
+
+static int sigmadsp_rate_to_index(struct sigmadsp *sigmadsp, unsigned int rate)
+{
+ unsigned int i;
+
+ for (i = 0; i < sigmadsp->rate_constraints.count; i++) {
+ if (sigmadsp->rate_constraints.list[i] == rate)
+ return i;
+ }
+
+ return -EINVAL;
+}
+
+static unsigned int sigmadsp_get_samplerate_mask(struct sigmadsp *sigmadsp,
+ unsigned int samplerate)
+{
+ int samplerate_index;
+
+ if (samplerate == 0)
+ return 0;
+
+ if (sigmadsp->rate_constraints.count) {
+ samplerate_index = sigmadsp_rate_to_index(sigmadsp, samplerate);
+ if (samplerate_index < 0)
+ return 0;
+
+ return BIT(samplerate_index);
+ } else {
+ return ~0;
+ }
+}
+
+static bool sigmadsp_samplerate_valid(unsigned int supported,
+ unsigned int requested)
+{
+ /* All samplerates are supported */
+ if (!supported)
+ return true;
+
+ return supported & requested;
+}
+
+static int sigmadsp_alloc_control(struct sigmadsp *sigmadsp,
+ struct sigmadsp_control *ctrl, unsigned int samplerate_mask)
+{
+ struct snd_kcontrol_new template;
+ struct snd_kcontrol *kcontrol;
+
+ memset(&template, 0, sizeof(template));
+ template.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ template.name = ctrl->name;
+ template.info = sigmadsp_ctrl_info;
+ template.get = sigmadsp_ctrl_get;
+ template.put = sigmadsp_ctrl_put;
+ template.private_value = (unsigned long)ctrl;
+ template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ if (!sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask))
+ template.access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+
+ kcontrol = snd_ctl_new1(&template, sigmadsp);
+ if (!kcontrol)
+ return -ENOMEM;
+
+ kcontrol->private_free = sigmadsp_control_free;
+ ctrl->kcontrol = kcontrol;
+
+ return snd_ctl_add(sigmadsp->component->card->snd_card, kcontrol);
+}
+
+static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp,
+ struct sigmadsp_control *ctrl, unsigned int samplerate_mask)
+{
+ struct snd_card *card = sigmadsp->component->card->snd_card;
+ struct snd_kcontrol_volatile *vd;
+ struct snd_ctl_elem_id id;
+ bool active;
+ bool changed = false;
+
+ active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask);
+
+ down_write(&card->controls_rwsem);
+ if (!ctrl->kcontrol) {
+ up_write(&card->controls_rwsem);
+ return;
+ }
+
+ id = ctrl->kcontrol->id;
+ vd = &ctrl->kcontrol->vd[0];
+ if (active == (bool)(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)) {
+ vd->access ^= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ changed = true;
+ }
+ up_write(&card->controls_rwsem);
+
+ if (active && changed) {
+ mutex_lock(&sigmadsp->lock);
+ if (ctrl->cached)
+ sigmadsp_ctrl_write(sigmadsp, ctrl, ctrl->cache);
+ mutex_unlock(&sigmadsp->lock);
+ }
+
+ if (changed)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &id);
+}
+
+/**
+ * sigmadsp_attach() - Attach a sigmadsp instance to a ASoC component
+ * @sigmadsp: The sigmadsp instance to attach
+ * @component: The component to attach to
+ *
+ * Typically called in the components probe callback.
+ *
+ * Note, once this function has been called the firmware must not be released
+ * until after the ALSA snd_card that the component belongs to has been
+ * disconnected, even if sigmadsp_attach() returns an error.
+ */
+int sigmadsp_attach(struct sigmadsp *sigmadsp,
+ struct snd_soc_component *component)
+{
+ struct sigmadsp_control *ctrl;
+ unsigned int samplerate_mask;
+ int ret;
+
+ sigmadsp->component = component;
+
+ samplerate_mask = sigmadsp_get_samplerate_mask(sigmadsp,
+ sigmadsp->current_samplerate);
+
+ list_for_each_entry(ctrl, &sigmadsp->ctrl_list, head) {
+ ret = sigmadsp_alloc_control(sigmadsp, ctrl, samplerate_mask);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sigmadsp_attach);
+
+/**
+ * sigmadsp_setup() - Setup the DSP for the specified samplerate
+ * @sigmadsp: The sigmadsp instance to configure
+ * @samplerate: The samplerate the DSP should be configured for
+ *
+ * Loads the appropriate firmware program and parameter memory (if not already
+ * loaded) and enables the controls for the specified samplerate. Any control
+ * parameter changes that have been made previously will be restored.
+ *
+ * Returns 0 on success, a negative error code otherwise.
+ */
+int sigmadsp_setup(struct sigmadsp *sigmadsp, unsigned int samplerate)
+{
+ struct sigmadsp_control *ctrl;
+ unsigned int samplerate_mask;
+ struct sigmadsp_data *data;
+ int ret;
+
+ if (sigmadsp->current_samplerate == samplerate)
+ return 0;
+
+ samplerate_mask = sigmadsp_get_samplerate_mask(sigmadsp, samplerate);
+ if (samplerate_mask == 0)
+ return -EINVAL;
+
+ list_for_each_entry(data, &sigmadsp->data_list, head) {
+ if (!sigmadsp_samplerate_valid(data->samplerates,
+ samplerate_mask))
+ continue;
+ ret = sigmadsp_write(sigmadsp, data->addr, data->data,
+ data->length);
+ if (ret)
+ goto err;
+ }
+
+ list_for_each_entry(ctrl, &sigmadsp->ctrl_list, head)
+ sigmadsp_activate_ctrl(sigmadsp, ctrl, samplerate_mask);
+
+ sigmadsp->current_samplerate = samplerate;
+
+ return 0;
+err:
+ sigmadsp_reset(sigmadsp);
return ret;
}
-EXPORT_SYMBOL_GPL(_process_sigma_firmware);
+EXPORT_SYMBOL_GPL(sigmadsp_setup);
+
+/**
+ * sigmadsp_reset() - Notify the sigmadsp instance that the DSP has been reset
+ * @sigmadsp: The sigmadsp instance to reset
+ *
+ * Should be called whenever the DSP has been reset and parameter and program
+ * memory need to be re-loaded.
+ */
+void sigmadsp_reset(struct sigmadsp *sigmadsp)
+{
+ struct sigmadsp_control *ctrl;
+
+ list_for_each_entry(ctrl, &sigmadsp->ctrl_list, head)
+ sigmadsp_activate_ctrl(sigmadsp, ctrl, false);
+
+ sigmadsp->current_samplerate = 0;
+}
+EXPORT_SYMBOL_GPL(sigmadsp_reset);
+
+/**
+ * sigmadsp_restrict_params() - Applies DSP firmware specific constraints
+ * @sigmadsp: The sigmadsp instance
+ * @substream: The substream to restrict
+ *
+ * Applies samplerate constraints that may be required by the firmware Should
+ * typically be called from the CODEC/component drivers startup callback.
+ *
+ * Returns 0 on success, a negative error code otherwise.
+ */
+int sigmadsp_restrict_params(struct sigmadsp *sigmadsp,
+ struct snd_pcm_substream *substream)
+{
+ if (sigmadsp->rate_constraints.count == 0)
+ return 0;
+
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &sigmadsp->rate_constraints);
+}
+EXPORT_SYMBOL_GPL(sigmadsp_restrict_params);
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h
index c47cd23e9827..614475cbb823 100644
--- a/sound/soc/codecs/sigmadsp.h
+++ b/sound/soc/codecs/sigmadsp.h
@@ -11,31 +11,56 @@
#include <linux/device.h>
#include <linux/regmap.h>
+#include <linux/list.h>
-struct sigma_action {
- u8 instr;
- u8 len_hi;
- __le16 len;
- __be16 addr;
- unsigned char payload[];
-} __packed;
+#include <sound/pcm.h>
-struct sigma_firmware {
- const struct firmware *fw;
- size_t pos;
+struct sigmadsp;
+struct snd_soc_component;
+struct snd_pcm_substream;
+
+struct sigmadsp_ops {
+ int (*safeload)(struct sigmadsp *sigmadsp, unsigned int addr,
+ const uint8_t *data, size_t len);
+};
+
+struct sigmadsp {
+ const struct sigmadsp_ops *ops;
+
+ struct list_head ctrl_list;
+ struct list_head data_list;
+
+ struct snd_pcm_hw_constraint_list rate_constraints;
+
+ unsigned int current_samplerate;
+ struct snd_soc_component *component;
+ struct device *dev;
+
+ struct mutex lock;
void *control_data;
- int (*write)(void *control_data, const struct sigma_action *sa,
- size_t len);
+ int (*write)(void *, unsigned int, const uint8_t *, size_t);
+ int (*read)(void *, unsigned int, uint8_t *, size_t);
};
-int _process_sigma_firmware(struct device *dev,
- struct sigma_firmware *ssfw, const char *name);
+struct sigmadsp *devm_sigmadsp_init(struct device *dev,
+ const struct sigmadsp_ops *ops, const char *firmware_name);
+void sigmadsp_reset(struct sigmadsp *sigmadsp);
+
+int sigmadsp_restrict_params(struct sigmadsp *sigmadsp,
+ struct snd_pcm_substream *substream);
struct i2c_client;
-extern int process_sigma_firmware(struct i2c_client *client, const char *name);
-extern int process_sigma_firmware_regmap(struct device *dev,
- struct regmap *regmap, const char *name);
+struct sigmadsp *devm_sigmadsp_init_regmap(struct device *dev,
+ struct regmap *regmap, const struct sigmadsp_ops *ops,
+ const char *firmware_name);
+struct sigmadsp *devm_sigmadsp_init_i2c(struct i2c_client *client,
+ const struct sigmadsp_ops *ops, const char *firmware_name);
+
+int sigmadsp_attach(struct sigmadsp *sigmadsp,
+ struct snd_soc_component *component);
+int sigmadsp_setup(struct sigmadsp *sigmadsp, unsigned int rate);
+void sigmadsp_reset(struct sigmadsp *sigmadsp);
#endif
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index 06ba4923fd5a..07eea20e6645 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -120,7 +120,8 @@ static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
{
#define ATLAS6_CODEC_ENABLE_BITS (1 << 29)
#define ATLAS6_CODEC_RESET_BITS (1 << 28)
- struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
enable_and_reset_codec(sirf_audio_codec->regmap,
@@ -142,7 +143,8 @@ static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
{
#define PRIMA2_CODEC_ENABLE_BITS (1 << 27)
#define PRIMA2_CODEC_RESET_BITS (1 << 26)
- struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
enable_and_reset_codec(sirf_audio_codec->regmap,
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index cf8fa40662f0..31d97cd5e59b 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -867,25 +867,16 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, SN95031_SSR2, 0x10);
snd_soc_write(codec, SN95031_SSR3, 0x40);
- snd_soc_add_codec_controls(codec, sn95031_snd_controls,
- ARRAY_SIZE(sn95031_snd_controls));
-
- return 0;
-}
-
-static int sn95031_codec_remove(struct snd_soc_codec *codec)
-{
- pr_debug("codec_remove called\n");
- sn95031_set_vaud_bias(codec, SND_SOC_BIAS_OFF);
-
return 0;
}
static struct snd_soc_codec_driver sn95031_codec = {
.probe = sn95031_codec_probe,
- .remove = sn95031_codec_remove,
.set_bias_level = sn95031_set_vaud_bias,
.idle_bias_off = true,
+
+ .controls = sn95031_snd_controls,
+ .num_controls = ARRAY_SIZE(sn95031_snd_controls),
.dapm_widgets = sn95031_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets),
.dapm_routes = sn95031_audio_map,
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index 4b5c17f8507e..a984485108cd 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -69,6 +69,22 @@
#define SSM4567_DAC_FS_64000_96000 0x3
#define SSM4567_DAC_FS_128000_192000 0x4
+/* SAI_CTRL_1 */
+#define SSM4567_SAI_CTRL_1_BCLK BIT(6)
+#define SSM4567_SAI_CTRL_1_TDM_BLCKS_MASK (0x3 << 4)
+#define SSM4567_SAI_CTRL_1_TDM_BLCKS_32 (0x0 << 4)
+#define SSM4567_SAI_CTRL_1_TDM_BLCKS_48 (0x1 << 4)
+#define SSM4567_SAI_CTRL_1_TDM_BLCKS_64 (0x2 << 4)
+#define SSM4567_SAI_CTRL_1_FSYNC BIT(3)
+#define SSM4567_SAI_CTRL_1_LJ BIT(2)
+#define SSM4567_SAI_CTRL_1_TDM BIT(1)
+#define SSM4567_SAI_CTRL_1_PDM BIT(0)
+
+/* SAI_CTRL_2 */
+#define SSM4567_SAI_CTRL_2_AUTO_SLOT BIT(3)
+#define SSM4567_SAI_CTRL_2_TDM_SLOT_MASK 0x7
+#define SSM4567_SAI_CTRL_2_TDM_SLOT(x) (x)
+
struct ssm4567 {
struct regmap *regmap;
};
@@ -145,15 +161,24 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = {
SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0,
0xff, 1, ssm4567_vol_tlv),
SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0),
+ SOC_SINGLE("DAC High Pass Filter Switch", SSM4567_REG_DAC_CTRL,
+ 5, 1, 0),
};
+static const struct snd_kcontrol_new ssm4567_amplifier_boost_control =
+ SOC_DAPM_SINGLE("Switch", SSM4567_REG_POWER_CTRL, 1, 1, 1);
+
static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1),
+ SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1,
+ &ssm4567_amplifier_boost_control),
SND_SOC_DAPM_OUTPUT("OUT"),
};
static const struct snd_soc_dapm_route ssm4567_routes[] = {
+ { "OUT", NULL, "Amplifier Boost" },
+ { "Amplifier Boost", "Switch", "DAC" },
{ "OUT", NULL, "DAC" },
};
@@ -192,6 +217,107 @@ static int ssm4567_mute(struct snd_soc_dai *dai, int mute)
SSM4567_DAC_MUTE, val);
}
+static int ssm4567_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int width)
+{
+ struct ssm4567 *ssm4567 = snd_soc_dai_get_drvdata(dai);
+ unsigned int blcks;
+ int slot;
+ int ret;
+
+ if (tx_mask == 0)
+ return -EINVAL;
+
+ if (rx_mask && rx_mask != tx_mask)
+ return -EINVAL;
+
+ slot = __ffs(tx_mask);
+ if (tx_mask != BIT(slot))
+ return -EINVAL;
+
+ switch (width) {
+ case 32:
+ blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_32;
+ break;
+ case 48:
+ blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_48;
+ break;
+ case 64:
+ blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_64;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_2,
+ SSM4567_SAI_CTRL_2_AUTO_SLOT | SSM4567_SAI_CTRL_2_TDM_SLOT_MASK,
+ SSM4567_SAI_CTRL_2_TDM_SLOT(slot));
+ if (ret)
+ return ret;
+
+ return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1,
+ SSM4567_SAI_CTRL_1_TDM_BLCKS_MASK, blcks);
+}
+
+static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct ssm4567 *ssm4567 = snd_soc_dai_get_drvdata(dai);
+ unsigned int ctrl1 = 0;
+ bool invert_fclk;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ invert_fclk = false;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ctrl1 |= SSM4567_SAI_CTRL_1_BCLK;
+ invert_fclk = false;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC;
+ invert_fclk = true;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ ctrl1 |= SSM4567_SAI_CTRL_1_BCLK;
+ invert_fclk = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl1 |= SSM4567_SAI_CTRL_1_LJ;
+ invert_fclk = !invert_fclk;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ ctrl1 |= SSM4567_SAI_CTRL_1_TDM;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ ctrl1 |= SSM4567_SAI_CTRL_1_TDM | SSM4567_SAI_CTRL_1_LJ;
+ break;
+ case SND_SOC_DAIFMT_PDM:
+ ctrl1 |= SSM4567_SAI_CTRL_1_PDM;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (invert_fclk)
+ ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC;
+
+ return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1);
+}
+
static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable)
{
int ret = 0;
@@ -246,6 +372,8 @@ static int ssm4567_set_bias_level(struct snd_soc_codec *codec,
static const struct snd_soc_dai_ops ssm4567_dai_ops = {
.hw_params = ssm4567_hw_params,
.digital_mute = ssm4567_mute,
+ .set_fmt = ssm4567_set_dai_fmt,
+ .set_tdm_slot = ssm4567_set_tdm_slot,
};
static struct snd_soc_dai_driver ssm4567_dai = {
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 48740855566d..7e18200dd6a9 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -833,23 +833,6 @@ static struct snd_soc_dai_driver sta32x_dai = {
.ops = &sta32x_dai_ops,
};
-#ifdef CONFIG_PM
-static int sta32x_suspend(struct snd_soc_codec *codec)
-{
- sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int sta32x_resume(struct snd_soc_codec *codec)
-{
- sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define sta32x_suspend NULL
-#define sta32x_resume NULL
-#endif
-
static int sta32x_probe(struct snd_soc_codec *codec)
{
struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
@@ -936,7 +919,6 @@ static int sta32x_remove(struct snd_soc_codec *codec)
struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
sta32x_watchdog_stop(sta32x);
- sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
return 0;
@@ -955,9 +937,8 @@ static bool sta32x_reg_is_volatile(struct device *dev, unsigned int reg)
static const struct snd_soc_codec_driver sta32x_codec = {
.probe = sta32x_probe,
.remove = sta32x_remove,
- .suspend = sta32x_suspend,
- .resume = sta32x_resume,
.set_bias_level = sta32x_set_bias_level,
+ .suspend_bias_off = true,
.controls = sta32x_snd_controls,
.num_controls = ARRAY_SIZE(sta32x_snd_controls),
.dapm_widgets = sta32x_dapm_widgets,
diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c
index cc97dd52aa9c..bda2ee18769e 100644
--- a/sound/soc/codecs/sta350.c
+++ b/sound/soc/codecs/sta350.c
@@ -912,23 +912,6 @@ static struct snd_soc_dai_driver sta350_dai = {
.ops = &sta350_dai_ops,
};
-#ifdef CONFIG_PM
-static int sta350_suspend(struct snd_soc_codec *codec)
-{
- sta350_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int sta350_resume(struct snd_soc_codec *codec)
-{
- sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define sta350_suspend NULL
-#define sta350_resume NULL
-#endif
-
static int sta350_probe(struct snd_soc_codec *codec)
{
struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec);
@@ -1065,7 +1048,6 @@ static int sta350_remove(struct snd_soc_codec *codec)
{
struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec);
- sta350_set_bias_level(codec, SND_SOC_BIAS_OFF);
regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies);
return 0;
@@ -1074,9 +1056,8 @@ static int sta350_remove(struct snd_soc_codec *codec)
static const struct snd_soc_codec_driver sta350_codec = {
.probe = sta350_probe,
.remove = sta350_remove,
- .suspend = sta350_suspend,
- .resume = sta350_resume,
.set_bias_level = sta350_set_bias_level,
+ .suspend_bias_off = true,
.controls = sta350_snd_controls,
.num_controls = ARRAY_SIZE(sta350_snd_controls),
.dapm_widgets = sta350_dapm_widgets,
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 89c748dd3d6e..b0f436d10125 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -319,41 +319,10 @@ static struct snd_soc_dai_driver sta529_dai = {
.ops = &sta529_dai_ops,
};
-static int sta529_probe(struct snd_soc_codec *codec)
-{
- sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-/* power down chip */
-static int sta529_remove(struct snd_soc_codec *codec)
-{
- sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int sta529_suspend(struct snd_soc_codec *codec)
-{
- sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int sta529_resume(struct snd_soc_codec *codec)
-{
- sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
static const struct snd_soc_codec_driver sta529_codec_driver = {
- .probe = sta529_probe,
- .remove = sta529_remove,
.set_bias_level = sta529_set_bias_level,
- .suspend = sta529_suspend,
- .resume = sta529_resume,
+ .suspend_bias_off = true,
+
.controls = sta529_snd_controls,
.num_controls = ARRAY_SIZE(sta529_snd_controls),
};
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 53b810d23fea..dbff0c89be48 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -139,18 +139,19 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
- soc_ac97_ops->write(codec->ac97, reg, val);
+ soc_ac97_ops->write(ac97, reg, val);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return 0;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
- soc_ac97_ops->write(codec->ac97, reg, val);
+ soc_ac97_ops->write(ac97, reg, val);
cache[reg / 2] = val;
return 0;
}
@@ -158,11 +159,12 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 val = 0, *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
- val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0);
+ val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return val;
}
@@ -173,7 +175,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
reg == AC97_VENDOR_ID2) {
- val = soc_ac97_ops->read(codec->ac97, reg);
+ val = soc_ac97_ops->read(ac97, reg);
return val;
}
return cache[reg / 2];
@@ -240,45 +242,41 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec,
static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
if (try_warm && soc_ac97_ops->warm_reset) {
- soc_ac97_ops->warm_reset(codec->ac97);
+ soc_ac97_ops->warm_reset(ac97);
if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
return 1;
}
- soc_ac97_ops->reset(codec->ac97);
+ soc_ac97_ops->reset(ac97);
if (soc_ac97_ops->warm_reset)
- soc_ac97_ops->warm_reset(codec->ac97);
+ soc_ac97_ops->warm_reset(ac97);
if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
return -EIO;
return 0;
}
-static int stac9766_codec_suspend(struct snd_soc_codec *codec)
-{
- stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static int stac9766_codec_resume(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 id, reset;
reset = 0;
/* give the codec an AC97 warm reset to start the link */
reset:
if (reset > 5) {
- printk(KERN_ERR "stac9766 failed to resume");
+ dev_err(codec->dev, "Failed to resume\n");
return -EIO;
}
- codec->ac97->bus->ops->warm_reset(codec->ac97);
- id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2);
+ ac97->bus->ops->warm_reset(ac97);
+ id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2);
if (id != 0x4c13) {
stac9766_reset(codec, 0);
reset++;
goto reset;
}
- stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
@@ -294,7 +292,6 @@ static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
static struct snd_soc_dai_driver stac9766_dai[] = {
{
.name = "stac9766-hifi-analog",
- .ac97_control = 1,
/* stream cababilities */
.playback = {
@@ -316,7 +313,6 @@ static struct snd_soc_dai_driver stac9766_dai[] = {
},
{
.name = "stac9766-hifi-IEC958",
- .ac97_control = 1,
/* stream cababilities */
.playback = {
@@ -334,46 +330,48 @@ static struct snd_soc_dai_driver stac9766_dai[] = {
static int stac9766_codec_probe(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97;
int ret = 0;
- ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0);
- if (ret < 0)
- goto codec_err;
+ ac97 = snd_soc_new_ac97_codec(codec);
+ if (IS_ERR(ac97))
+ return PTR_ERR(ac97);
+
+ snd_soc_codec_set_drvdata(codec, ac97);
/* do a cold reset for the controller and then try
* a warm reset followed by an optional cold reset for codec */
stac9766_reset(codec, 0);
ret = stac9766_reset(codec, 1);
if (ret < 0) {
- printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
+ dev_err(codec->dev, "Failed to reset: AC97 link error\n");
goto codec_err;
}
- stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
- ARRAY_SIZE(stac9766_snd_ac97_controls));
-
return 0;
codec_err:
- snd_soc_free_ac97_codec(codec);
+ snd_soc_free_ac97_codec(ac97);
return ret;
}
static int stac9766_codec_remove(struct snd_soc_codec *codec)
{
- snd_soc_free_ac97_codec(codec);
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_free_ac97_codec(ac97);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
+ .controls = stac9766_snd_ac97_controls,
+ .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls),
.write = stac9766_ac97_write,
.read = stac9766_ac97_read,
.set_bias_level = stac9766_set_bias_level,
+ .suspend_bias_off = true,
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
- .suspend = stac9766_codec_suspend,
.resume = stac9766_codec_resume,
.reg_cache_size = ARRAY_SIZE(stac9766_reg),
.reg_word_size = sizeof(u16),
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index f039dc825971..b505212019e2 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -345,7 +345,6 @@ static const struct reg_default tas2552_init_regs[] = {
static int tas2552_codec_probe(struct snd_soc_codec *codec)
{
struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
- struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
tas2552->codec = codec;
@@ -390,11 +389,6 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, TAS2552_CFG_2, TAS2552_BOOST_EN |
TAS2552_APT_EN | TAS2552_LIM_EN);
- snd_soc_dapm_new_controls(dapm, tas2552_dapm_widgets,
- ARRAY_SIZE(tas2552_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, tas2552_audio_map,
- ARRAY_SIZE(tas2552_audio_map));
-
return 0;
patch_fail:
@@ -462,6 +456,10 @@ static struct snd_soc_codec_driver soc_codec_dev_tas2552 = {
.resume = tas2552_resume,
.controls = tas2552_snd_controls,
.num_controls = ARRAY_SIZE(tas2552_snd_controls),
+ .dapm_widgets = tas2552_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tas2552_dapm_widgets),
+ .dapm_routes = tas2552_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(tas2552_audio_map),
};
static const struct regmap_config tas2552_regmap_config = {
diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c
new file mode 100644
index 000000000000..16f1b71edb55
--- /dev/null
+++ b/sound/soc/codecs/tfa9879.c
@@ -0,0 +1,328 @@
+/*
+ * tfa9879.c -- driver for NXP Semiconductors TFA9879
+ *
+ * Copyright (C) 2014 Axentia Technologies AB
+ * Author: Peter Rosin <peda@axentia.se>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/pcm_params.h>
+
+#include "tfa9879.h"
+
+struct tfa9879_priv {
+ struct regmap *regmap;
+ int lsb_justified;
+};
+
+static int tfa9879_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct tfa9879_priv *tfa9879 = snd_soc_codec_get_drvdata(codec);
+ int fs;
+ int i2s_set = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs = TFA9879_I2S_FS_8000;
+ break;
+ case 11025:
+ fs = TFA9879_I2S_FS_11025;
+ break;
+ case 12000:
+ fs = TFA9879_I2S_FS_12000;
+ break;
+ case 16000:
+ fs = TFA9879_I2S_FS_16000;
+ break;
+ case 22050:
+ fs = TFA9879_I2S_FS_22050;
+ break;
+ case 24000:
+ fs = TFA9879_I2S_FS_24000;
+ break;
+ case 32000:
+ fs = TFA9879_I2S_FS_32000;
+ break;
+ case 44100:
+ fs = TFA9879_I2S_FS_44100;
+ break;
+ case 48000:
+ fs = TFA9879_I2S_FS_48000;
+ break;
+ case 64000:
+ fs = TFA9879_I2S_FS_64000;
+ break;
+ case 88200:
+ fs = TFA9879_I2S_FS_88200;
+ break;
+ case 96000:
+ fs = TFA9879_I2S_FS_96000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (params_width(params)) {
+ case 16:
+ i2s_set = TFA9879_I2S_SET_LSB_J_16;
+ break;
+ case 24:
+ i2s_set = TFA9879_I2S_SET_LSB_J_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (tfa9879->lsb_justified)
+ snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1,
+ TFA9879_I2S_SET_MASK,
+ i2s_set << TFA9879_I2S_SET_SHIFT);
+
+ snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1,
+ TFA9879_I2S_FS_MASK,
+ fs << TFA9879_I2S_FS_SHIFT);
+ return 0;
+}
+
+static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ snd_soc_update_bits(codec, TFA9879_MISC_CONTROL,
+ TFA9879_S_MUTE_MASK,
+ !!mute << TFA9879_S_MUTE_SHIFT);
+
+ return 0;
+}
+
+static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct tfa9879_priv *tfa9879 = snd_soc_codec_get_drvdata(codec);
+ int i2s_set;
+ int sck_pol;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sck_pol = TFA9879_SCK_POL_NORMAL;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sck_pol = TFA9879_SCK_POL_INVERSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ tfa9879->lsb_justified = 0;
+ i2s_set = TFA9879_I2S_SET_I2S_24;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ tfa9879->lsb_justified = 0;
+ i2s_set = TFA9879_I2S_SET_MSB_J_24;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ tfa9879->lsb_justified = 1;
+ i2s_set = TFA9879_I2S_SET_LSB_J_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1,
+ TFA9879_SCK_POL_MASK,
+ sck_pol << TFA9879_SCK_POL_SHIFT);
+ snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1,
+ TFA9879_I2S_SET_MASK,
+ i2s_set << TFA9879_I2S_SET_SHIFT);
+ return 0;
+}
+
+static struct reg_default tfa9879_regs[] = {
+ { TFA9879_DEVICE_CONTROL, 0x0000 }, /* 0x00 */
+ { TFA9879_SERIAL_INTERFACE_1, 0x0a18 }, /* 0x01 */
+ { TFA9879_PCM_IOM2_FORMAT_1, 0x0007 }, /* 0x02 */
+ { TFA9879_SERIAL_INTERFACE_2, 0x0a18 }, /* 0x03 */
+ { TFA9879_PCM_IOM2_FORMAT_2, 0x0007 }, /* 0x04 */
+ { TFA9879_EQUALIZER_A1, 0x59dd }, /* 0x05 */
+ { TFA9879_EQUALIZER_A2, 0xc63e }, /* 0x06 */
+ { TFA9879_EQUALIZER_B1, 0x651a }, /* 0x07 */
+ { TFA9879_EQUALIZER_B2, 0xe53e }, /* 0x08 */
+ { TFA9879_EQUALIZER_C1, 0x4616 }, /* 0x09 */
+ { TFA9879_EQUALIZER_C2, 0xd33e }, /* 0x0a */
+ { TFA9879_EQUALIZER_D1, 0x4df3 }, /* 0x0b */
+ { TFA9879_EQUALIZER_D2, 0xea3e }, /* 0x0c */
+ { TFA9879_EQUALIZER_E1, 0x5ee0 }, /* 0x0d */
+ { TFA9879_EQUALIZER_E2, 0xf93e }, /* 0x0e */
+ { TFA9879_BYPASS_CONTROL, 0x0093 }, /* 0x0f */
+ { TFA9879_DYNAMIC_RANGE_COMPR, 0x92ba }, /* 0x10 */
+ { TFA9879_BASS_TREBLE, 0x12a5 }, /* 0x11 */
+ { TFA9879_HIGH_PASS_FILTER, 0x0004 }, /* 0x12 */
+ { TFA9879_VOLUME_CONTROL, 0x10bd }, /* 0x13 */
+ { TFA9879_MISC_CONTROL, 0x0000 }, /* 0x14 */
+};
+
+static bool tfa9879_volatile_reg(struct device *dev, unsigned int reg)
+{
+ return reg == TFA9879_MISC_STATUS;
+}
+
+static const DECLARE_TLV_DB_SCALE(volume_tlv, -7050, 50, 1);
+static const DECLARE_TLV_DB_SCALE(tb_gain_tlv, -1800, 200, 0);
+static const char * const tb_freq_text[] = {
+ "Low", "Mid", "High"
+};
+static const struct soc_enum treble_freq_enum =
+ SOC_ENUM_SINGLE(TFA9879_BASS_TREBLE, TFA9879_F_TRBLE_SHIFT,
+ ARRAY_SIZE(tb_freq_text), tb_freq_text);
+static const struct soc_enum bass_freq_enum =
+ SOC_ENUM_SINGLE(TFA9879_BASS_TREBLE, TFA9879_F_BASS_SHIFT,
+ ARRAY_SIZE(tb_freq_text), tb_freq_text);
+
+static const struct snd_kcontrol_new tfa9879_controls[] = {
+ SOC_SINGLE_TLV("PCM Playback Volume", TFA9879_VOLUME_CONTROL,
+ TFA9879_VOL_SHIFT, 0xbd, 1, volume_tlv),
+ SOC_SINGLE_TLV("Treble Volume", TFA9879_BASS_TREBLE,
+ TFA9879_G_TRBLE_SHIFT, 18, 0, tb_gain_tlv),
+ SOC_SINGLE_TLV("Bass Volume", TFA9879_BASS_TREBLE,
+ TFA9879_G_BASS_SHIFT, 18, 0, tb_gain_tlv),
+ SOC_ENUM("Treble Corner Freq", treble_freq_enum),
+ SOC_ENUM("Bass Corner Freq", bass_freq_enum),
+};
+
+static const struct snd_soc_dapm_widget tfa9879_dapm_widgets[] = {
+SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_DAC("DAC", NULL, TFA9879_DEVICE_CONTROL, TFA9879_OPMODE_SHIFT, 0),
+SND_SOC_DAPM_OUTPUT("LINEOUT"),
+SND_SOC_DAPM_SUPPLY("POWER", TFA9879_DEVICE_CONTROL, TFA9879_POWERUP_SHIFT, 0,
+ NULL, 0),
+};
+
+static const struct snd_soc_dapm_route tfa9879_dapm_routes[] = {
+ { "DAC", NULL, "AIFINL" },
+ { "DAC", NULL, "AIFINR" },
+
+ { "LINEOUT", NULL, "DAC" },
+
+ { "DAC", NULL, "POWER" },
+};
+
+static const struct snd_soc_codec_driver tfa9879_codec = {
+ .controls = tfa9879_controls,
+ .num_controls = ARRAY_SIZE(tfa9879_controls),
+
+ .dapm_widgets = tfa9879_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tfa9879_dapm_widgets),
+ .dapm_routes = tfa9879_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(tfa9879_dapm_routes),
+};
+
+static const struct regmap_config tfa9879_regmap = {
+ .reg_bits = 8,
+ .val_bits = 16,
+
+ .volatile_reg = tfa9879_volatile_reg,
+ .max_register = TFA9879_MISC_STATUS,
+ .reg_defaults = tfa9879_regs,
+ .num_reg_defaults = ARRAY_SIZE(tfa9879_regs),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static const struct snd_soc_dai_ops tfa9879_dai_ops = {
+ .hw_params = tfa9879_hw_params,
+ .digital_mute = tfa9879_digital_mute,
+ .set_fmt = tfa9879_set_fmt,
+};
+
+#define TFA9879_RATES SNDRV_PCM_RATE_8000_96000
+
+#define TFA9879_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver tfa9879_dai = {
+ .name = "tfa9879-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = TFA9879_RATES,
+ .formats = TFA9879_FORMATS, },
+ .ops = &tfa9879_dai_ops,
+};
+
+static int tfa9879_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct tfa9879_priv *tfa9879;
+ int i;
+
+ tfa9879 = devm_kzalloc(&i2c->dev, sizeof(*tfa9879), GFP_KERNEL);
+ if (IS_ERR(tfa9879))
+ return PTR_ERR(tfa9879);
+
+ i2c_set_clientdata(i2c, tfa9879);
+
+ tfa9879->regmap = devm_regmap_init_i2c(i2c, &tfa9879_regmap);
+ if (IS_ERR(tfa9879->regmap))
+ return PTR_ERR(tfa9879->regmap);
+
+ /* Ensure the device is in reset state */
+ for (i = 0; i < ARRAY_SIZE(tfa9879_regs); i++)
+ regmap_write(tfa9879->regmap,
+ tfa9879_regs[i].reg, tfa9879_regs[i].def);
+
+ return snd_soc_register_codec(&i2c->dev, &tfa9879_codec,
+ &tfa9879_dai, 1);
+}
+
+static int tfa9879_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id tfa9879_i2c_id[] = {
+ { "tfa9879", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id);
+
+static struct i2c_driver tfa9879_i2c_driver = {
+ .driver = {
+ .name = "tfa9879",
+ .owner = THIS_MODULE,
+ },
+ .probe = tfa9879_i2c_probe,
+ .remove = tfa9879_i2c_remove,
+ .id_table = tfa9879_i2c_id,
+};
+
+module_i2c_driver(tfa9879_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC NXP Semiconductors TFA9879 driver");
+MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tfa9879.h b/sound/soc/codecs/tfa9879.h
new file mode 100644
index 000000000000..3408c90c4628
--- /dev/null
+++ b/sound/soc/codecs/tfa9879.h
@@ -0,0 +1,202 @@
+/*
+ * tfa9879.h -- driver for NXP Semiconductors TFA9879
+ *
+ * Copyright (C) 2014 Axentia Technologies AB
+ * Author: Peter Rosin <peda@axentia.se>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _TFA9879_H
+#define _TFA9879_H
+
+#define TFA9879_DEVICE_CONTROL 0x00
+#define TFA9879_SERIAL_INTERFACE_1 0x01
+#define TFA9879_PCM_IOM2_FORMAT_1 0x02
+#define TFA9879_SERIAL_INTERFACE_2 0x03
+#define TFA9879_PCM_IOM2_FORMAT_2 0x04
+#define TFA9879_EQUALIZER_A1 0x05
+#define TFA9879_EQUALIZER_A2 0x06
+#define TFA9879_EQUALIZER_B1 0x07
+#define TFA9879_EQUALIZER_B2 0x08
+#define TFA9879_EQUALIZER_C1 0x09
+#define TFA9879_EQUALIZER_C2 0x0a
+#define TFA9879_EQUALIZER_D1 0x0b
+#define TFA9879_EQUALIZER_D2 0x0c
+#define TFA9879_EQUALIZER_E1 0x0d
+#define TFA9879_EQUALIZER_E2 0x0e
+#define TFA9879_BYPASS_CONTROL 0x0f
+#define TFA9879_DYNAMIC_RANGE_COMPR 0x10
+#define TFA9879_BASS_TREBLE 0x11
+#define TFA9879_HIGH_PASS_FILTER 0x12
+#define TFA9879_VOLUME_CONTROL 0x13
+#define TFA9879_MISC_CONTROL 0x14
+#define TFA9879_MISC_STATUS 0x15
+
+/* TFA9879_DEVICE_CONTROL */
+#define TFA9879_INPUT_SEL_MASK 0x0010
+#define TFA9879_INPUT_SEL_SHIFT 4
+#define TFA9879_OPMODE_MASK 0x0008
+#define TFA9879_OPMODE_SHIFT 3
+#define TFA9879_RESET_MASK 0x0002
+#define TFA9879_RESET_SHIFT 1
+#define TFA9879_POWERUP_MASK 0x0001
+#define TFA9879_POWERUP_SHIFT 0
+
+/* TFA9879_SERIAL_INTERFACE */
+#define TFA9879_MONO_SEL_MASK 0x0c00
+#define TFA9879_MONO_SEL_SHIFT 10
+#define TFA9879_MONO_SEL_LEFT 0
+#define TFA9879_MONO_SEL_RIGHT 1
+#define TFA9879_MONO_SEL_BOTH 2
+#define TFA9879_I2S_FS_MASK 0x03c0
+#define TFA9879_I2S_FS_SHIFT 6
+#define TFA9879_I2S_FS_8000 0
+#define TFA9879_I2S_FS_11025 1
+#define TFA9879_I2S_FS_12000 2
+#define TFA9879_I2S_FS_16000 3
+#define TFA9879_I2S_FS_22050 4
+#define TFA9879_I2S_FS_24000 5
+#define TFA9879_I2S_FS_32000 6
+#define TFA9879_I2S_FS_44100 7
+#define TFA9879_I2S_FS_48000 8
+#define TFA9879_I2S_FS_64000 9
+#define TFA9879_I2S_FS_88200 10
+#define TFA9879_I2S_FS_96000 11
+#define TFA9879_I2S_SET_MASK 0x0038
+#define TFA9879_I2S_SET_SHIFT 3
+#define TFA9879_I2S_SET_MSB_J_24 2
+#define TFA9879_I2S_SET_I2S_24 3
+#define TFA9879_I2S_SET_LSB_J_16 4
+#define TFA9879_I2S_SET_LSB_J_18 5
+#define TFA9879_I2S_SET_LSB_J_20 6
+#define TFA9879_I2S_SET_LSB_J_24 7
+#define TFA9879_SCK_POL_MASK 0x0004
+#define TFA9879_SCK_POL_SHIFT 2
+#define TFA9879_SCK_POL_NORMAL 0
+#define TFA9879_SCK_POL_INVERSE 1
+#define TFA9879_I_MODE_MASK 0x0003
+#define TFA9879_I_MODE_SHIFT 0
+#define TFA9879_I_MODE_I2S 0
+#define TFA9879_I_MODE_PCM_IOM2_SHORT 1
+#define TFA9879_I_MODE_PCM_IOM2_LONG 2
+
+/* TFA9879_PCM_IOM2_FORMAT */
+#define TFA9879_PCM_FS_MASK 0x0800
+#define TFA9879_PCM_FS_SHIFT 11
+#define TFA9879_A_LAW_MASK 0x0400
+#define TFA9879_A_LAW_SHIFT 10
+#define TFA9879_PCM_COMP_MASK 0x0200
+#define TFA9879_PCM_COMP_SHIFT 9
+#define TFA9879_PCM_DL_MASK 0x0100
+#define TFA9879_PCM_DL_SHIFT 8
+#define TFA9879_D1_SLOT_MASK 0x00f0
+#define TFA9879_D1_SLOT_SHIFT 4
+#define TFA9879_D2_SLOT_MASK 0x000f
+#define TFA9879_D2_SLOT_SHIFT 0
+
+/* TFA9879_EQUALIZER_X1 */
+#define TFA9879_T1_MASK 0x8000
+#define TFA9879_T1_SHIFT 15
+#define TFA9879_K1M_MASK 0x7ff0
+#define TFA9879_K1M_SHIFT 4
+#define TFA9879_K1E_MASK 0x000f
+#define TFA9879_K1E_SHIFT 0
+
+/* TFA9879_EQUALIZER_X2 */
+#define TFA9879_T2_MASK 0x8000
+#define TFA9879_T2_SHIFT 15
+#define TFA9879_K2M_MASK 0x7800
+#define TFA9879_K2M_SHIFT 11
+#define TFA9879_K2E_MASK 0x0700
+#define TFA9879_K2E_SHIFT 8
+#define TFA9879_K0_MASK 0x00fe
+#define TFA9879_K0_SHIFT 1
+#define TFA9879_S_MASK 0x0001
+#define TFA9879_S_SHIFT 0
+
+/* TFA9879_BYPASS_CONTROL */
+#define TFA9879_L_OCP_MASK 0x00c0
+#define TFA9879_L_OCP_SHIFT 6
+#define TFA9879_L_OTP_MASK 0x0030
+#define TFA9879_L_OTP_SHIFT 4
+#define TFA9879_CLIPCTRL_MASK 0x0008
+#define TFA9879_CLIPCTRL_SHIFT 3
+#define TFA9879_HPF_BP_MASK 0x0004
+#define TFA9879_HPF_BP_SHIFT 2
+#define TFA9879_DRC_BP_MASK 0x0002
+#define TFA9879_DRC_BP_SHIFT 1
+#define TFA9879_EQ_BP_MASK 0x0001
+#define TFA9879_EQ_BP_SHIFT 0
+
+/* TFA9879_DYNAMIC_RANGE_COMPR */
+#define TFA9879_AT_LVL_MASK 0xf000
+#define TFA9879_AT_LVL_SHIFT 12
+#define TFA9879_AT_RATE_MASK 0x0f00
+#define TFA9879_AT_RATE_SHIFT 8
+#define TFA9879_RL_LVL_MASK 0x00f0
+#define TFA9879_RL_LVL_SHIFT 4
+#define TFA9879_RL_RATE_MASK 0x000f
+#define TFA9879_RL_RATE_SHIFT 0
+
+/* TFA9879_BASS_TREBLE */
+#define TFA9879_G_TRBLE_MASK 0x3e00
+#define TFA9879_G_TRBLE_SHIFT 9
+#define TFA9879_F_TRBLE_MASK 0x0180
+#define TFA9879_F_TRBLE_SHIFT 7
+#define TFA9879_G_BASS_MASK 0x007c
+#define TFA9879_G_BASS_SHIFT 2
+#define TFA9879_F_BASS_MASK 0x0003
+#define TFA9879_F_BASS_SHIFT 0
+
+/* TFA9879_HIGH_PASS_FILTER */
+#define TFA9879_HP_CTRL_MASK 0x00ff
+#define TFA9879_HP_CTRL_SHIFT 0
+
+/* TFA9879_VOLUME_CONTROL */
+#define TFA9879_ZR_CRSS_MASK 0x1000
+#define TFA9879_ZR_CRSS_SHIFT 12
+#define TFA9879_VOL_MASK 0x00ff
+#define TFA9879_VOL_SHIFT 0
+
+/* TFA9879_MISC_CONTROL */
+#define TFA9879_DE_PHAS_MASK 0x0c00
+#define TFA9879_DE_PHAS_SHIFT 10
+#define TFA9879_H_MUTE_MASK 0x0200
+#define TFA9879_H_MUTE_SHIFT 9
+#define TFA9879_S_MUTE_MASK 0x0100
+#define TFA9879_S_MUTE_SHIFT 8
+#define TFA9879_P_LIM_MASK 0x00ff
+#define TFA9879_P_LIM_SHIFT 0
+
+/* TFA9879_MISC_STATUS */
+#define TFA9879_PS_MASK 0x4000
+#define TFA9879_PS_SHIFT 14
+#define TFA9879_PORA_MASK 0x2000
+#define TFA9879_PORA_SHIFT 13
+#define TFA9879_AMP_MASK 0x0600
+#define TFA9879_AMP_SHIFT 9
+#define TFA9879_IBP_2_MASK 0x0100
+#define TFA9879_IBP_2_SHIFT 8
+#define TFA9879_OFP_2_MASK 0x0080
+#define TFA9879_OFP_2_SHIFT 7
+#define TFA9879_UFP_2_MASK 0x0040
+#define TFA9879_UFP_2_SHIFT 6
+#define TFA9879_IBP_1_MASK 0x0020
+#define TFA9879_IBP_1_SHIFT 5
+#define TFA9879_OFP_1_MASK 0x0010
+#define TFA9879_OFP_1_SHIFT 4
+#define TFA9879_UFP_1_MASK 0x0008
+#define TFA9879_UFP_1_SHIFT 3
+#define TFA9879_OCPOKA_MASK 0x0004
+#define TFA9879_OCPOKA_SHIFT 2
+#define TFA9879_OCPOKB_MASK 0x0002
+#define TFA9879_OCPOKB_SHIFT 1
+#define TFA9879_OTPOK_MASK 0x0001
+#define TFA9879_OTPOK_SHIFT 0
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index d67167920c2f..cc17e7e5126e 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -540,19 +540,11 @@ static struct snd_soc_dai_driver tlv320aic23_dai = {
.ops = &tlv320aic23_dai_ops,
};
-static int tlv320aic23_suspend(struct snd_soc_codec *codec)
-{
- tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
static int tlv320aic23_resume(struct snd_soc_codec *codec)
{
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
regcache_mark_dirty(aic23->regmap);
regcache_sync(aic23->regmap);
- tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
@@ -562,9 +554,6 @@ static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
/* Reset codec */
snd_soc_write(codec, TLV320AIC23_RESET, 0);
- /* power on device */
- tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
/* Unmute input */
@@ -589,18 +578,12 @@ static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
return 0;
}
-static int tlv320aic23_remove(struct snd_soc_codec *codec)
-{
- tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
.probe = tlv320aic23_codec_probe,
- .remove = tlv320aic23_remove,
- .suspend = tlv320aic23_suspend,
.resume = tlv320aic23_resume,
.set_bias_level = tlv320aic23_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = tlv320aic23_snd_controls,
.num_controls = ARRAY_SIZE(tlv320aic23_snd_controls),
.dapm_widgets = tlv320aic23_dapm_widgets,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 145fe5b253d4..dc3223d6eca1 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -911,12 +911,13 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
}
aic31xx->p_div = i;
- for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) {
- if (i == ARRAY_SIZE(aic31xx_divs)) {
- dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
- __func__, freq);
- return -EINVAL;
- }
+ for (i = 0; i < ARRAY_SIZE(aic31xx_divs) &&
+ aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++)
+ ;
+ if (i == ARRAY_SIZE(aic31xx_divs)) {
+ dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
+ __func__, freq);
+ return -EINVAL;
}
/* set clock on MCLK, BCLK, or GPIO1 as PLL input */
@@ -1056,18 +1057,6 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-static int aic31xx_suspend(struct snd_soc_codec *codec)
-{
- aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int aic31xx_resume(struct snd_soc_codec *codec)
-{
- aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int aic31xx_codec_probe(struct snd_soc_codec *codec)
{
int ret = 0;
@@ -1110,8 +1099,6 @@ static int aic31xx_codec_remove(struct snd_soc_codec *codec)
{
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
int i;
- /* power down chip */
- aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
regulator_unregister_notifier(aic31xx->supplies[i].consumer,
@@ -1123,9 +1110,9 @@ static int aic31xx_codec_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
.probe = aic31xx_codec_probe,
.remove = aic31xx_codec_remove,
- .suspend = aic31xx_suspend,
- .resume = aic31xx_resume,
.set_bias_level = aic31xx_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = aic31xx_snd_controls,
.num_controls = ARRAY_SIZE(aic31xx_snd_controls),
.dapm_widgets = aic31xx_dapm_widgets,
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 6ea662db2410..015467ed606b 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -597,18 +597,6 @@ static struct snd_soc_dai_driver aic32x4_dai = {
.symmetric_rates = 1,
};
-static int aic32x4_suspend(struct snd_soc_codec *codec)
-{
- aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int aic32x4_resume(struct snd_soc_codec *codec)
-{
- aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int aic32x4_probe(struct snd_soc_codec *codec)
{
struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
@@ -654,8 +642,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, AIC32X4_RMICPGANIN,
AIC32X4_RMICPGANIN_CM1R_10K);
- aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
/*
* Workaround: for an unknown reason, the ADC needs to be powered up
* and down for the first capture to work properly. It seems related to
@@ -669,18 +655,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
return 0;
}
-static int aic32x4_remove(struct snd_soc_codec *codec)
-{
- aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
.probe = aic32x4_probe,
- .remove = aic32x4_remove,
- .suspend = aic32x4_suspend,
- .resume = aic32x4_resume,
.set_bias_level = aic32x4_set_bias_level,
+ .suspend_bias_off = true,
.controls = aic32x4_snd_controls,
.num_controls = ARRAY_SIZE(aic32x4_snd_controls),
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index f7c2a575a892..b7ebce054b4e 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -78,6 +78,8 @@ struct aic3x_priv {
struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES];
struct aic3x_setup_data *setup;
unsigned int sysclk;
+ unsigned int dai_fmt;
+ unsigned int tdm_delay;
struct list_head list;
int master;
int gpio_reset;
@@ -214,61 +216,78 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" };
-static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" };
-static const char *aic3x_left_hpcom_mux[] =
- { "differential of HPLOUT", "constant VCM", "single-ended" };
-static const char *aic3x_right_hpcom_mux[] =
- { "differential of HPROUT", "constant VCM", "single-ended",
- "differential of HPLCOM", "external feedback" };
-static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
-static const char *aic3x_adc_hpf[] =
- { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
-
-#define LDAC_ENUM 0
-#define RDAC_ENUM 1
-#define LHPCOM_ENUM 2
-#define RHPCOM_ENUM 3
-#define LINE1L_2_L_ENUM 4
-#define LINE1L_2_R_ENUM 5
-#define LINE1R_2_L_ENUM 6
-#define LINE1R_2_R_ENUM 7
-#define LINE2L_ENUM 8
-#define LINE2R_ENUM 9
-#define ADC_HPF_ENUM 10
-
-static const struct soc_enum aic3x_enum[] = {
- SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
- SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux),
- SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux),
- SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux),
- SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
- SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
- SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
- SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
- SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
- SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
- SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
-};
-
-static const char *aic3x_agc_level[] =
- { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" };
-static const struct soc_enum aic3x_agc_level_enum[] = {
- SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level),
- SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level),
-};
-
-static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" };
-static const struct soc_enum aic3x_agc_attack_enum[] = {
- SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack),
- SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack),
-};
-
-static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" };
-static const struct soc_enum aic3x_agc_decay_enum[] = {
- SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay),
- SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
-};
+static const char * const aic3x_left_dac_mux[] = {
+ "DAC_L1", "DAC_L3", "DAC_L2" };
+static SOC_ENUM_SINGLE_DECL(aic3x_left_dac_enum, DAC_LINE_MUX, 6,
+ aic3x_left_dac_mux);
+
+static const char * const aic3x_right_dac_mux[] = {
+ "DAC_R1", "DAC_R3", "DAC_R2" };
+static SOC_ENUM_SINGLE_DECL(aic3x_right_dac_enum, DAC_LINE_MUX, 4,
+ aic3x_right_dac_mux);
+
+static const char * const aic3x_left_hpcom_mux[] = {
+ "differential of HPLOUT", "constant VCM", "single-ended" };
+static SOC_ENUM_SINGLE_DECL(aic3x_left_hpcom_enum, HPLCOM_CFG, 4,
+ aic3x_left_hpcom_mux);
+
+static const char * const aic3x_right_hpcom_mux[] = {
+ "differential of HPROUT", "constant VCM", "single-ended",
+ "differential of HPLCOM", "external feedback" };
+static SOC_ENUM_SINGLE_DECL(aic3x_right_hpcom_enum, HPRCOM_CFG, 3,
+ aic3x_right_hpcom_mux);
+
+static const char * const aic3x_linein_mode_mux[] = {
+ "single-ended", "differential" };
+static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_l_enum, LINE1L_2_LADC_CTRL, 7,
+ aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_r_enum, LINE1L_2_RADC_CTRL, 7,
+ aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_l_enum, LINE1R_2_LADC_CTRL, 7,
+ aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_r_enum, LINE1R_2_RADC_CTRL, 7,
+ aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line2l_2_ldac_enum, LINE2L_2_LADC_CTRL, 7,
+ aic3x_linein_mode_mux);
+static SOC_ENUM_SINGLE_DECL(aic3x_line2r_2_rdac_enum, LINE2R_2_RADC_CTRL, 7,
+ aic3x_linein_mode_mux);
+
+static const char * const aic3x_adc_hpf[] = {
+ "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
+static SOC_ENUM_DOUBLE_DECL(aic3x_adc_hpf_enum, AIC3X_CODEC_DFILT_CTRL, 6, 4,
+ aic3x_adc_hpf);
+
+static const char * const aic3x_agc_level[] = {
+ "-5.5dB", "-8dB", "-10dB", "-12dB",
+ "-14dB", "-17dB", "-20dB", "-24dB" };
+static SOC_ENUM_SINGLE_DECL(aic3x_lagc_level_enum, LAGC_CTRL_A, 4,
+ aic3x_agc_level);
+static SOC_ENUM_SINGLE_DECL(aic3x_ragc_level_enum, RAGC_CTRL_A, 4,
+ aic3x_agc_level);
+
+static const char * const aic3x_agc_attack[] = {
+ "8ms", "11ms", "16ms", "20ms" };
+static SOC_ENUM_SINGLE_DECL(aic3x_lagc_attack_enum, LAGC_CTRL_A, 2,
+ aic3x_agc_attack);
+static SOC_ENUM_SINGLE_DECL(aic3x_ragc_attack_enum, RAGC_CTRL_A, 2,
+ aic3x_agc_attack);
+
+static const char * const aic3x_agc_decay[] = {
+ "100ms", "200ms", "400ms", "500ms" };
+static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0,
+ aic3x_agc_decay);
+static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0,
+ aic3x_agc_decay);
+
+static const char * const aic3x_poweron_time[] = {
+ "0us", "10us", "100us", "1ms", "10ms", "50ms",
+ "100ms", "200ms", "400ms", "800ms", "2s", "4s" };
+static SOC_ENUM_SINGLE_DECL(aic3x_poweron_time_enum, HPOUT_POP_REDUCTION, 4,
+ aic3x_poweron_time);
+
+static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" };
+static SOC_ENUM_SINGLE_DECL(aic3x_rampup_step_enum, HPOUT_POP_REDUCTION, 2,
+ aic3x_rampup_step);
/*
* DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
@@ -383,12 +402,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
* adjust PGA to max value when ADC is on and will never go back.
*/
SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
- SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]),
- SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]),
- SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]),
- SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]),
- SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]),
- SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]),
+ SOC_ENUM("Left AGC Target level", aic3x_lagc_level_enum),
+ SOC_ENUM("Right AGC Target level", aic3x_ragc_level_enum),
+ SOC_ENUM("Left AGC Attack time", aic3x_lagc_attack_enum),
+ SOC_ENUM("Right AGC Attack time", aic3x_ragc_attack_enum),
+ SOC_ENUM("Left AGC Decay time", aic3x_lagc_decay_enum),
+ SOC_ENUM("Right AGC Decay time", aic3x_ragc_decay_enum),
/* De-emphasis */
SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
@@ -398,7 +417,11 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
0, 119, 0, adc_tlv),
SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
- SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
+ SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum),
+
+ /* Pop reduction */
+ SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum),
+ SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum),
};
static const struct snd_kcontrol_new aic3x_mono_controls[] = {
@@ -425,19 +448,19 @@ static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
/* Left DAC Mux */
static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_left_dac_enum);
/* Right DAC Mux */
static const struct snd_kcontrol_new aic3x_right_dac_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_right_dac_enum);
/* Left HPCOM Mux */
static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_left_hpcom_enum);
/* Right HPCOM Mux */
static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum);
/* Left Line Mixer */
static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = {
@@ -529,23 +552,23 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
/* Left Line1 Mux */
static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum);
static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1l_2_r_enum);
/* Right Line1 Mux */
static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1r_2_r_enum);
static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line1r_2_l_enum);
/* Left Line2 Mux */
static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line2l_2_ldac_enum);
/* Right Line2 Mux */
static const struct snd_kcontrol_new aic3x_right_line2_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
+SOC_DAPM_ENUM("Route", aic3x_line2r_2_rdac_enum);
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
/* Left DAC to Left Outputs */
@@ -1009,6 +1032,25 @@ found:
return 0;
}
+static int aic3x_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+ int delay = 0;
+
+ /* TDM slot selection only valid in DSP_A/_B mode */
+ if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
+ delay += (aic3x->tdm_delay + 1);
+ else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
+ delay += aic3x->tdm_delay;
+
+ /* Configure data delay */
+ snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay);
+
+ return 0;
+}
+
static int aic3x_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
@@ -1048,7 +1090,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
u8 iface_areg, iface_breg;
- int delay = 0;
iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
@@ -1076,7 +1117,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
- delay = 1;
case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
@@ -1090,10 +1130,45 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
+ aic3x->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
/* set iface */
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
- snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
+
+ return 0;
+}
+
+static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+ unsigned int lsb;
+
+ if (tx_mask != rx_mask) {
+ dev_err(codec->dev, "tx and rx masks must be symmetric\n");
+ return -EINVAL;
+ }
+
+ if (unlikely(!tx_mask)) {
+ dev_err(codec->dev, "tx and rx masks need to be non 0\n");
+ return -EINVAL;
+ }
+
+ /* TDM based on DSP mode requires slots to be adjacent */
+ lsb = __ffs(tx_mask);
+ if ((lsb + 1) != __fls(tx_mask)) {
+ dev_err(codec->dev, "Invalid mask, slots must be adjacent\n");
+ return -EINVAL;
+ }
+
+ aic3x->tdm_delay = lsb * slot_width;
+
+ /* DOUT in high-impedance on inactive bit clocks */
+ snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
+ DOUT_TRISTATE, DOUT_TRISTATE);
return 0;
}
@@ -1212,9 +1287,11 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
static const struct snd_soc_dai_ops aic3x_dai_ops = {
.hw_params = aic3x_hw_params,
+ .prepare = aic3x_prepare,
.digital_mute = aic3x_mute,
.set_sysclk = aic3x_set_dai_sysclk,
.set_fmt = aic3x_set_dai_fmt,
+ .set_tdm_slot = aic3x_set_dai_tdm_slot,
};
static struct snd_soc_dai_driver aic3x_dai = {
@@ -1414,7 +1491,6 @@ static int aic3x_remove(struct snd_soc_codec *codec)
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int i;
- aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
list_del(&aic3x->list);
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
regulator_unregister_notifier(aic3x->supplies[i].consumer,
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index e521ac3ddde8..89fa692df206 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -169,6 +169,7 @@
/* Audio serial data interface control register A bits */
#define BIT_CLK_MASTER 0x80
#define WORD_CLK_MASTER 0x40
+#define DOUT_TRISTATE 0x20
/* Codec Datapath setup register 7 */
#define FSREF_44100 (1 << 7)
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index e21ed934bdbf..0fe2ced5b09f 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1436,8 +1436,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec)
{
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
- dac33_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
if (dac33->irq >= 0) {
free_irq(dac33->irq, dac33->codec);
destroy_workqueue(dac33->dac33_wq);
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
new file mode 100644
index 000000000000..1d1205702d23
--- /dev/null
+++ b/sound/soc/codecs/ts3a227e.c
@@ -0,0 +1,314 @@
+/*
+ * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch
+ *
+ * Copyright (C) 2014 Google, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/init.h>
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/of_gpio.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+
+struct ts3a227e {
+ struct regmap *regmap;
+ struct snd_soc_jack *jack;
+ bool plugged;
+ bool mic_present;
+ unsigned int buttons_held;
+};
+
+/* Button values to be reported on the jack */
+static const int ts3a227e_buttons[] = {
+ SND_JACK_BTN_0,
+ SND_JACK_BTN_1,
+ SND_JACK_BTN_2,
+ SND_JACK_BTN_3,
+};
+
+#define TS3A227E_NUM_BUTTONS 4
+#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \
+ SND_JACK_MICROPHONE | \
+ SND_JACK_BTN_0 | \
+ SND_JACK_BTN_1 | \
+ SND_JACK_BTN_2 | \
+ SND_JACK_BTN_3)
+
+/* TS3A227E registers */
+#define TS3A227E_REG_DEVICE_ID 0x00
+#define TS3A227E_REG_INTERRUPT 0x01
+#define TS3A227E_REG_KP_INTERRUPT 0x02
+#define TS3A227E_REG_INTERRUPT_DISABLE 0x03
+#define TS3A227E_REG_SETTING_1 0x04
+#define TS3A227E_REG_SETTING_2 0x05
+#define TS3A227E_REG_SETTING_3 0x06
+#define TS3A227E_REG_SWITCH_CONTROL_1 0x07
+#define TS3A227E_REG_SWITCH_CONTROL_2 0x08
+#define TS3A227E_REG_SWITCH_STATUS_1 0x09
+#define TS3A227E_REG_SWITCH_STATUS_2 0x0a
+#define TS3A227E_REG_ACCESSORY_STATUS 0x0b
+#define TS3A227E_REG_ADC_OUTPUT 0x0c
+#define TS3A227E_REG_KP_THRESHOLD_1 0x0d
+#define TS3A227E_REG_KP_THRESHOLD_2 0x0e
+#define TS3A227E_REG_KP_THRESHOLD_3 0x0f
+
+/* TS3A227E_REG_INTERRUPT 0x01 */
+#define INS_REM_EVENT 0x01
+#define DETECTION_COMPLETE_EVENT 0x02
+
+/* TS3A227E_REG_KP_INTERRUPT 0x02 */
+#define PRESS_MASK(idx) (0x01 << (2 * (idx)))
+#define RELEASE_MASK(idx) (0x02 << (2 * (idx)))
+
+/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */
+#define INS_REM_INT_DISABLE 0x01
+#define DETECTION_COMPLETE_INT_DISABLE 0x02
+#define ADC_COMPLETE_INT_DISABLE 0x04
+#define INTB_DISABLE 0x08
+
+/* TS3A227E_REG_SETTING_2 0x05 */
+#define KP_ENABLE 0x04
+
+/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */
+#define TYPE_3_POLE 0x01
+#define TYPE_4_POLE_OMTP 0x02
+#define TYPE_4_POLE_STANDARD 0x04
+#define JACK_INSERTED 0x08
+#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD)
+
+static const struct reg_default ts3a227e_reg_defaults[] = {
+ { TS3A227E_REG_DEVICE_ID, 0x10 },
+ { TS3A227E_REG_INTERRUPT, 0x00 },
+ { TS3A227E_REG_KP_INTERRUPT, 0x00 },
+ { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 },
+ { TS3A227E_REG_SETTING_1, 0x23 },
+ { TS3A227E_REG_SETTING_2, 0x00 },
+ { TS3A227E_REG_SETTING_3, 0x0e },
+ { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 },
+ { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 },
+ { TS3A227E_REG_SWITCH_STATUS_1, 0x0c },
+ { TS3A227E_REG_SWITCH_STATUS_2, 0x00 },
+ { TS3A227E_REG_ACCESSORY_STATUS, 0x00 },
+ { TS3A227E_REG_ADC_OUTPUT, 0x00 },
+ { TS3A227E_REG_KP_THRESHOLD_1, 0x20 },
+ { TS3A227E_REG_KP_THRESHOLD_2, 0x40 },
+ { TS3A227E_REG_KP_THRESHOLD_3, 0x68 },
+};
+
+static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2:
+ case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE:
+ case TS3A227E_REG_SETTING_2:
+ case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static void ts3a227e_jack_report(struct ts3a227e *ts3a227e)
+{
+ unsigned int i;
+ int report = 0;
+
+ if (!ts3a227e->jack)
+ return;
+
+ if (ts3a227e->plugged)
+ report = SND_JACK_HEADPHONE;
+ if (ts3a227e->mic_present)
+ report |= SND_JACK_MICROPHONE;
+ for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) {
+ if (ts3a227e->buttons_held & (1 << i))
+ report |= ts3a227e_buttons[i];
+ }
+ snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK);
+}
+
+static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg)
+{
+ bool plugged, mic_present;
+
+ plugged = !!(acc_reg & JACK_INSERTED);
+ mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK);
+
+ ts3a227e->plugged = plugged;
+
+ if (mic_present != ts3a227e->mic_present) {
+ ts3a227e->mic_present = mic_present;
+ ts3a227e->buttons_held = 0;
+ if (mic_present) {
+ /* Enable key press detection. */
+ regmap_update_bits(ts3a227e->regmap,
+ TS3A227E_REG_SETTING_2,
+ KP_ENABLE, KP_ENABLE);
+ }
+ }
+}
+
+static irqreturn_t ts3a227e_interrupt(int irq, void *data)
+{
+ struct ts3a227e *ts3a227e = (struct ts3a227e *)data;
+ struct regmap *regmap = ts3a227e->regmap;
+ unsigned int int_reg, kp_int_reg, acc_reg, i;
+
+ /* Check for plug/unplug. */
+ regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg);
+ if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) {
+ regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+ ts3a227e_new_jack_state(ts3a227e, acc_reg);
+ }
+
+ /* Report any key events. */
+ regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg);
+ for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) {
+ if (kp_int_reg & PRESS_MASK(i))
+ ts3a227e->buttons_held |= (1 << i);
+ if (kp_int_reg & RELEASE_MASK(i))
+ ts3a227e->buttons_held &= ~(1 << i);
+ }
+
+ ts3a227e_jack_report(ts3a227e);
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * ts3a227e_enable_jack_detect - Specify a jack for event reporting
+ *
+ * @component: component to register the jack with
+ * @jack: jack to use to report headset and button events on
+ *
+ * After this function has been called the headset insert/remove and button
+ * events 0-3 will be routed to the given jack. Jack can be null to stop
+ * reporting.
+ */
+int ts3a227e_enable_jack_detect(struct snd_soc_component *component,
+ struct snd_soc_jack *jack)
+{
+ struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component);
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+ ts3a227e->jack = jack;
+ ts3a227e_jack_report(ts3a227e);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect);
+
+static struct snd_soc_component_driver ts3a227e_soc_driver;
+
+static const struct regmap_config ts3a227e_regmap_config = {
+ .val_bits = 8,
+ .reg_bits = 8,
+
+ .max_register = TS3A227E_REG_KP_THRESHOLD_3,
+ .readable_reg = ts3a227e_readable_reg,
+ .writeable_reg = ts3a227e_writeable_reg,
+ .volatile_reg = ts3a227e_volatile_reg,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = ts3a227e_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults),
+};
+
+static int ts3a227e_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ts3a227e *ts3a227e;
+ struct device *dev = &i2c->dev;
+ int ret;
+
+ ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
+ if (ts3a227e == NULL)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, ts3a227e);
+
+ ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config);
+ if (IS_ERR(ts3a227e->regmap))
+ return PTR_ERR(ts3a227e->regmap);
+
+ ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt,
+ IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+ "TS3A227E", ts3a227e);
+ if (ret) {
+ dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret);
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver,
+ NULL, 0);
+ if (ret)
+ return ret;
+
+ /* Enable interrupts except for ADC complete. */
+ regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE,
+ INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
+ ADC_COMPLETE_INT_DISABLE);
+
+ return 0;
+}
+
+static const struct i2c_device_id ts3a227e_i2c_ids[] = {
+ { "ts3a227e", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids);
+
+static const struct of_device_id ts3a227e_of_match[] = {
+ { .compatible = "ti,ts3a227e", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, ts3a227e_of_match);
+
+static struct i2c_driver ts3a227e_driver = {
+ .driver = {
+ .name = "ts3a227e",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(ts3a227e_of_match),
+ },
+ .probe = ts3a227e_i2c_probe,
+ .id_table = ts3a227e_i2c_ids,
+};
+module_i2c_driver(ts3a227e_driver);
+
+MODULE_DESCRIPTION("ASoC ts3a227e driver");
+MODULE_AUTHOR("Dylan Reid <dgreid@chromium.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h
new file mode 100644
index 000000000000..e2acf9c5bebe
--- /dev/null
+++ b/sound/soc/codecs/ts3a227e.h
@@ -0,0 +1,17 @@
+/*
+ * TS3A227E Autonous Audio Accessory Detection and Configureation Switch
+ *
+ * Copyright (C) 2014 Google, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _TS3A227E_H
+#define _TS3A227E_H
+
+int ts3a227e_enable_jack_detect(struct snd_soc_component *component,
+ struct snd_soc_jack *jack);
+
+#endif
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index b6b0cb399599..27f3b21effb2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2177,8 +2177,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec)
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
struct twl4030_codec_data *pdata = twl4030->pdata;
- twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio))
gpio_free(pdata->hs_extmute_gpio);
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 0f6067f04e29..5ff2b1e4638e 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1095,25 +1095,6 @@ static struct snd_soc_dai_driver twl6040_dai[] = {
},
};
-#ifdef CONFIG_PM
-static int twl6040_suspend(struct snd_soc_codec *codec)
-{
- twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int twl6040_resume(struct snd_soc_codec *codec)
-{
- twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define twl6040_suspend NULL
-#define twl6040_resume NULL
-#endif
-
static int twl6040_probe(struct snd_soc_codec *codec)
{
struct twl6040_data *priv;
@@ -1160,7 +1141,6 @@ static int twl6040_remove(struct snd_soc_codec *codec)
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
free_irq(priv->plug_irq, codec);
- twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -1168,11 +1148,10 @@ static int twl6040_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_twl6040 = {
.probe = twl6040_probe,
.remove = twl6040_remove,
- .suspend = twl6040_suspend,
- .resume = twl6040_resume,
.read = twl6040_read,
.write = twl6040_write,
.set_bias_level = twl6040_set_bias_level,
+ .suspend_bias_off = true,
.ignore_pmdown_time = true,
.controls = twl6040_snd_controls,
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 32b2f78aa62c..4056260a502e 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -518,11 +518,6 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
uda134x_reset(codec);
- if (pd->is_powered_on_standby)
- uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
- else
- uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
if (pd->model == UDA134X_UDA1341) {
widgets = uda1341_dapm_widgets;
num_widgets = ARRAY_SIZE(uda1341_dapm_widgets);
@@ -574,44 +569,21 @@ static int uda134x_soc_remove(struct snd_soc_codec *codec)
{
struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
- uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
kfree(uda134x);
return 0;
}
-#if defined(CONFIG_PM)
-static int uda134x_soc_suspend(struct snd_soc_codec *codec)
-{
- uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int uda134x_soc_resume(struct snd_soc_codec *codec)
-{
- uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
- uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
- return 0;
-}
-#else
-#define uda134x_soc_suspend NULL
-#define uda134x_soc_resume NULL
-#endif /* CONFIG_PM */
-
static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
.probe = uda134x_soc_probe,
.remove = uda134x_soc_remove,
- .suspend = uda134x_soc_suspend,
- .resume = uda134x_soc_resume,
.reg_cache_size = sizeof(uda134x_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = uda134x_reg,
.reg_cache_step = 1,
.read = uda134x_read_reg_cache,
- .write = uda134x_write,
.set_bias_level = uda134x_set_bias_level,
+ .suspend_bias_off = true,
+
.dapm_widgets = uda134x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets),
.dapm_routes = uda134x_dapm_routes,
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index e62e70781ec2..dc7778b6dd7f 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -693,18 +693,6 @@ static struct snd_soc_dai_driver uda1380_dai[] = {
},
};
-static int uda1380_suspend(struct snd_soc_codec *codec)
-{
- uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int uda1380_resume(struct snd_soc_codec *codec)
-{
- uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int uda1380_probe(struct snd_soc_codec *codec)
{
struct uda1380_platform_data *pdata =codec->dev->platform_data;
@@ -739,8 +727,6 @@ static int uda1380_probe(struct snd_soc_codec *codec)
INIT_WORK(&uda1380->work, uda1380_flush_work);
- /* power on device */
- uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* set clock input */
switch (pdata->dac_clk) {
case UDA1380_DAC_CLK_SYSCLK:
@@ -766,8 +752,6 @@ static int uda1380_remove(struct snd_soc_codec *codec)
{
struct uda1380_platform_data *pdata =codec->dev->platform_data;
- uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
gpio_free(pdata->gpio_reset);
gpio_free(pdata->gpio_power);
@@ -777,11 +761,11 @@ static int uda1380_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_uda1380 = {
.probe = uda1380_probe,
.remove = uda1380_remove,
- .suspend = uda1380_suspend,
- .resume = uda1380_resume,
.read = uda1380_read_reg_cache,
.write = uda1380_write,
.set_bias_level = uda1380_set_bias_level,
+ .suspend_bias_off = true,
+
.reg_cache_size = ARRAY_SIZE(uda1380_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = uda1380_reg,
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index f3d4e88d0b7b..00aea4100bb3 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -452,7 +452,6 @@ static int wl1273_probe(struct snd_soc_codec *codec)
{
struct wl1273_core **core = codec->dev->platform_data;
struct wl1273_priv *wl1273;
- int r;
dev_dbg(codec->dev, "%s.\n", __func__);
@@ -470,12 +469,7 @@ static int wl1273_probe(struct snd_soc_codec *codec)
snd_soc_codec_set_drvdata(codec, wl1273);
- r = snd_soc_add_codec_controls(codec, wl1273_controls,
- ARRAY_SIZE(wl1273_controls));
- if (r)
- kfree(wl1273);
-
- return r;
+ return 0;
}
static int wl1273_remove(struct snd_soc_codec *codec)
@@ -492,6 +486,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
.probe = wl1273_probe,
.remove = wl1273_remove,
+ .controls = wl1273_controls,
+ .num_controls = ARRAY_SIZE(wl1273_controls),
.dapm_widgets = wl1273_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets),
.dapm_routes = wl1273_dapm_routes,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index f60234962527..d78fb8dffc8c 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -619,10 +619,10 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol,
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
uint16_t data;
- mutex_lock(&codec->mutex);
+ mutex_lock(&arizona->dac_comp_lock);
data = cpu_to_be16(arizona->dac_comp_coeff);
memcpy(ucontrol->value.bytes.data, &data, sizeof(data));
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&arizona->dac_comp_lock);
return 0;
}
@@ -633,11 +633,11 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- mutex_lock(&codec->mutex);
+ mutex_lock(&arizona->dac_comp_lock);
memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data,
sizeof(arizona->dac_comp_coeff));
arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff);
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&arizona->dac_comp_lock);
return 0;
}
@@ -648,9 +648,9 @@ static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- mutex_lock(&codec->mutex);
+ mutex_lock(&arizona->dac_comp_lock);
ucontrol->value.integer.value[0] = arizona->dac_comp_enabled;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&arizona->dac_comp_lock);
return 0;
}
@@ -661,9 +661,9 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- mutex_lock(&codec->mutex);
+ mutex_lock(&arizona->dac_comp_lock);
arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&arizona->dac_comp_lock);
return 0;
}
@@ -1900,6 +1900,8 @@ static int wm5102_probe(struct platform_device *pdev)
return -ENOMEM;
platform_set_drvdata(pdev, wm5102);
+ mutex_init(&arizona->dac_comp_lock);
+
wm5102->core.arizona = arizona;
wm5102->core.num_inputs = 6;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 628ec774cf22..87f664b9cc7d 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1242,19 +1242,6 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-static int wm8350_suspend(struct snd_soc_codec *codec)
-{
- wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8350_resume(struct snd_soc_codec *codec)
-{
- wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
static void wm8350_hp_work(struct wm8350_data *priv,
struct wm8350_jack_data *jack,
u16 mask)
@@ -1565,9 +1552,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD,
wm8350_mic_handler, 0, "Microphone detect", priv);
-
- wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
return 0;
}
@@ -1596,8 +1580,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec)
* wait for its completion */
flush_delayed_work(&codec->dapm.delayed_work);
- wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
return 0;
@@ -1613,10 +1595,9 @@ static struct regmap *wm8350_get_regmap(struct device *dev)
static struct snd_soc_codec_driver soc_codec_dev_wm8350 = {
.probe = wm8350_codec_probe,
.remove = wm8350_codec_remove,
- .suspend = wm8350_suspend,
- .resume = wm8350_resume,
.get_regmap = wm8350_get_regmap,
.set_bias_level = wm8350_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8350_snd_controls,
.num_controls = ARRAY_SIZE(wm8350_snd_controls),
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 72471bef2e9a..385894f6e264 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -58,12 +58,10 @@ static struct regulator_bulk_data power[] = {
/* codec private data */
struct wm8400_priv {
- struct snd_soc_codec *codec;
struct wm8400 *wm8400;
u16 fake_register;
unsigned int sysclk;
unsigned int pcmclk;
- struct work_struct work;
int fll_in, fll_out;
};
@@ -1278,30 +1276,6 @@ static struct snd_soc_dai_driver wm8400_dai = {
.ops = &wm8400_dai_ops,
};
-static int wm8400_suspend(struct snd_soc_codec *codec)
-{
- wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int wm8400_resume(struct snd_soc_codec *codec)
-{
- wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static void wm8400_probe_deferred(struct work_struct *work)
-{
- struct wm8400_priv *priv = container_of(work, struct wm8400_priv,
- work);
- struct snd_soc_codec *codec = priv->codec;
-
- /* charge output caps */
- wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-}
-
static int wm8400_codec_probe(struct snd_soc_codec *codec)
{
struct wm8400 *wm8400 = dev_get_platdata(codec->dev);
@@ -1316,7 +1290,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
snd_soc_codec_set_drvdata(codec, priv);
priv->wm8400 = wm8400;
- priv->codec = codec;
ret = devm_regulator_bulk_get(wm8400->dev,
ARRAY_SIZE(power), &power[0]);
@@ -1325,8 +1298,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
return ret;
}
- INIT_WORK(&priv->work, wm8400_probe_deferred);
-
wm8400_codec_reset(codec);
reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1);
@@ -1343,8 +1314,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
- if (!schedule_work(&priv->work))
- return -EINVAL;
return 0;
}
@@ -1369,10 +1338,9 @@ static struct regmap *wm8400_get_regmap(struct device *dev)
static struct snd_soc_codec_driver soc_codec_dev_wm8400 = {
.probe = wm8400_codec_probe,
.remove = wm8400_codec_remove,
- .suspend = wm8400_suspend,
- .resume = wm8400_resume,
.get_regmap = wm8400_get_regmap,
.set_bias_level = wm8400_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8400_snd_controls,
.num_controls = ARRAY_SIZE(wm8400_snd_controls),
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index e11127f9069e..8736ad094b24 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -575,41 +575,17 @@ static struct snd_soc_dai_driver wm8510_dai = {
.symmetric_rates = 1,
};
-static int wm8510_suspend(struct snd_soc_codec *codec)
-{
- wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8510_resume(struct snd_soc_codec *codec)
-{
- wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int wm8510_probe(struct snd_soc_codec *codec)
{
wm8510_reset(codec);
- /* power on device */
- wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-/* power down chip */
-static int wm8510_remove(struct snd_soc_codec *codec)
-{
- wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm8510 = {
.probe = wm8510_probe,
- .remove = wm8510_remove,
- .suspend = wm8510_suspend,
- .resume = wm8510_resume,
.set_bias_level = wm8510_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8510_snd_controls,
.num_controls = ARRAY_SIZE(wm8510_snd_controls),
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index ec1f5740dbd0..b1cc94f5fc4b 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -372,23 +372,6 @@ static struct snd_soc_dai_driver wm8523_dai = {
.ops = &wm8523_dai_ops,
};
-#ifdef CONFIG_PM
-static int wm8523_suspend(struct snd_soc_codec *codec)
-{
- wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8523_resume(struct snd_soc_codec *codec)
-{
- wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define wm8523_suspend NULL
-#define wm8523_resume NULL
-#endif
-
static int wm8523_probe(struct snd_soc_codec *codec)
{
struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
@@ -402,23 +385,13 @@ static int wm8523_probe(struct snd_soc_codec *codec)
WM8523_DACR_VU, WM8523_DACR_VU);
snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC);
- wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int wm8523_remove(struct snd_soc_codec *codec)
-{
- wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm8523 = {
.probe = wm8523_probe,
- .remove = wm8523_remove,
- .suspend = wm8523_suspend,
- .resume = wm8523_resume,
.set_bias_level = wm8523_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8523_controls,
.num_controls = ARRAY_SIZE(wm8523_controls),
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 911605ee25b0..0a887c5ec83a 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -882,8 +882,6 @@ static int wm8580_probe(struct snd_soc_codec *codec)
goto err_regulator_enable;
}
- wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
return 0;
err_regulator_enable:
@@ -897,8 +895,6 @@ static int wm8580_remove(struct snd_soc_codec *codec)
{
struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec);
- wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
return 0;
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 32187e739b4f..121e46d53779 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -350,19 +350,6 @@ static struct snd_soc_dai_driver wm8711_dai = {
.ops = &wm8711_ops,
};
-static int wm8711_suspend(struct snd_soc_codec *codec)
-{
- snd_soc_write(codec, WM8711_ACTIVE, 0x0);
- wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8711_resume(struct snd_soc_codec *codec)
-{
- wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int wm8711_probe(struct snd_soc_codec *codec)
{
int ret;
@@ -373,8 +360,6 @@ static int wm8711_probe(struct snd_soc_codec *codec)
return ret;
}
- wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
/* Latch the update bits */
snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100);
snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100);
@@ -383,19 +368,11 @@ static int wm8711_probe(struct snd_soc_codec *codec)
}
-/* power down chip */
-static int wm8711_remove(struct snd_soc_codec *codec)
-{
- wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8711 = {
.probe = wm8711_probe,
- .remove = wm8711_remove,
- .suspend = wm8711_suspend,
- .resume = wm8711_resume,
.set_bias_level = wm8711_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = wm8711_snd_controls,
.num_controls = ARRAY_SIZE(wm8711_snd_controls),
.dapm_widgets = wm8711_dapm_widgets,
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 38ff826f589a..55c7fb4fc786 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -212,40 +212,10 @@ static struct snd_soc_dai_driver wm8728_dai = {
.ops = &wm8728_dai_ops,
};
-static int wm8728_suspend(struct snd_soc_codec *codec)
-{
- wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int wm8728_resume(struct snd_soc_codec *codec)
-{
- wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int wm8728_probe(struct snd_soc_codec *codec)
-{
- /* power on device */
- wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int wm8728_remove(struct snd_soc_codec *codec)
-{
- wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8728 = {
- .probe = wm8728_probe,
- .remove = wm8728_remove,
- .suspend = wm8728_suspend,
- .resume = wm8728_resume,
.set_bias_level = wm8728_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = wm8728_snd_controls,
.num_controls = ARRAY_SIZE(wm8728_snd_controls),
.dapm_widgets = wm8728_dapm_widgets,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index eebb3280bfad..b9211b42f6e9 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -24,6 +24,7 @@
#include <linux/regulator/consumer.h>
#include <linux/spi/spi.h>
#include <linux/of_device.h>
+#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -50,6 +51,8 @@ struct wm8731_priv {
int sysclk_type;
int playback_fs;
bool deemph;
+
+ struct mutex lock;
};
@@ -138,7 +141,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol,
if (deemph > 1)
return -EINVAL;
- mutex_lock(&codec->mutex);
+ mutex_lock(&wm8731->lock);
if (wm8731->deemph != deemph) {
wm8731->deemph = deemph;
@@ -146,7 +149,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol,
ret = 1;
}
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&wm8731->lock);
return ret;
}
@@ -559,25 +562,6 @@ static struct snd_soc_dai_driver wm8731_dai = {
.symmetric_rates = 1,
};
-#ifdef CONFIG_PM
-static int wm8731_suspend(struct snd_soc_codec *codec)
-{
- wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int wm8731_resume(struct snd_soc_codec *codec)
-{
- wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define wm8731_suspend NULL
-#define wm8731_resume NULL
-#endif
-
static int wm8731_probe(struct snd_soc_codec *codec)
{
struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
@@ -633,8 +617,6 @@ static int wm8731_remove(struct snd_soc_codec *codec)
{
struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
- wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
return 0;
@@ -643,9 +625,9 @@ static int wm8731_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_wm8731 = {
.probe = wm8731_probe,
.remove = wm8731_remove,
- .suspend = wm8731_suspend,
- .resume = wm8731_resume,
.set_bias_level = wm8731_set_bias_level,
+ .suspend_bias_off = true,
+
.dapm_widgets = wm8731_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
.dapm_routes = wm8731_intercon,
@@ -680,11 +662,12 @@ static int wm8731_spi_probe(struct spi_device *spi)
struct wm8731_priv *wm8731;
int ret;
- wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv),
- GFP_KERNEL);
+ wm8731 = devm_kzalloc(&spi->dev, sizeof(*wm8731), GFP_KERNEL);
if (wm8731 == NULL)
return -ENOMEM;
+ mutex_init(&wm8731->lock);
+
wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap);
if (IS_ERR(wm8731->regmap)) {
ret = PTR_ERR(wm8731->regmap);
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 744a422ecb05..ada9ac1ba2c6 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -277,17 +277,6 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "AIF", NULL, "ADCR" },
};
-static int wm8737_add_widgets(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, wm8737_dapm_widgets,
- ARRAY_SIZE(wm8737_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
- return 0;
-}
-
/* codec mclk clock divider coefficients */
static const struct {
u32 mclk;
@@ -548,23 +537,6 @@ static struct snd_soc_dai_driver wm8737_dai = {
.ops = &wm8737_dai_ops,
};
-#ifdef CONFIG_PM
-static int wm8737_suspend(struct snd_soc_codec *codec)
-{
- wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8737_resume(struct snd_soc_codec *codec)
-{
- wm8737_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define wm8737_suspend NULL
-#define wm8737_resume NULL
-#endif
-
static int wm8737_probe(struct snd_soc_codec *codec)
{
struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec);
@@ -593,10 +565,6 @@ static int wm8737_probe(struct snd_soc_codec *codec)
/* Bias level configuration will have done an extra enable */
regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies);
- snd_soc_add_codec_controls(codec, wm8737_snd_controls,
- ARRAY_SIZE(wm8737_snd_controls));
- wm8737_add_widgets(codec);
-
return 0;
err_enable:
@@ -605,18 +573,17 @@ err_get:
return ret;
}
-static int wm8737_remove(struct snd_soc_codec *codec)
-{
- wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8737 = {
.probe = wm8737_probe,
- .remove = wm8737_remove,
- .suspend = wm8737_suspend,
- .resume = wm8737_resume,
.set_bias_level = wm8737_set_bias_level,
+ .suspend_bias_off = true,
+
+ .controls = wm8737_snd_controls,
+ .num_controls = ARRAY_SIZE(wm8737_snd_controls),
+ .dapm_widgets = wm8737_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8737_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
};
static const struct of_device_id wm8737_of_match[] = {
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 67653a2db223..f6847fdd6ddd 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -686,18 +686,6 @@ static struct snd_soc_dai_driver wm8750_dai = {
.ops = &wm8750_dai_ops,
};
-static int wm8750_suspend(struct snd_soc_codec *codec)
-{
- wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8750_resume(struct snd_soc_codec *codec)
-{
- wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int wm8750_probe(struct snd_soc_codec *codec)
{
int ret;
@@ -708,9 +696,6 @@ static int wm8750_probe(struct snd_soc_codec *codec)
return ret;
}
- /* charge output caps */
- wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
/* set the update bits */
snd_soc_update_bits(codec, WM8750_LDAC, 0x0100, 0x0100);
snd_soc_update_bits(codec, WM8750_RDAC, 0x0100, 0x0100);
@@ -724,18 +709,10 @@ static int wm8750_probe(struct snd_soc_codec *codec)
return ret;
}
-static int wm8750_remove(struct snd_soc_codec *codec)
-{
- wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8750 = {
.probe = wm8750_probe,
- .remove = wm8750_remove,
- .suspend = wm8750_suspend,
- .resume = wm8750_resume,
.set_bias_level = wm8750_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8750_snd_controls,
.num_controls = ARRAY_SIZE(wm8750_snd_controls),
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 70952ceb278b..c13050b77931 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -408,24 +408,6 @@ static struct snd_soc_dai_driver wm8776_dai[] = {
},
};
-#ifdef CONFIG_PM
-static int wm8776_suspend(struct snd_soc_codec *codec)
-{
- wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int wm8776_resume(struct snd_soc_codec *codec)
-{
- wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define wm8776_suspend NULL
-#define wm8776_resume NULL
-#endif
-
static int wm8776_probe(struct snd_soc_codec *codec)
{
int ret = 0;
@@ -436,8 +418,6 @@ static int wm8776_probe(struct snd_soc_codec *codec)
return ret;
}
- wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
/* Latch the update bits; right channel only since we always
* update both. */
snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100);
@@ -446,19 +426,10 @@ static int wm8776_probe(struct snd_soc_codec *codec)
return ret;
}
-/* power down chip */
-static int wm8776_remove(struct snd_soc_codec *codec)
-{
- wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8776 = {
.probe = wm8776_probe,
- .remove = wm8776_remove,
- .suspend = wm8776_suspend,
- .resume = wm8776_resume,
.set_bias_level = wm8776_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8776_snd_controls,
.num_controls = ARRAY_SIZE(wm8776_snd_controls),
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 3addc5fe5cb2..1315f7642503 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -524,7 +524,6 @@ static int wm8804_remove(struct snd_soc_codec *codec)
int i;
wm8804 = snd_soc_codec_get_drvdata(codec);
- wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF);
for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i)
regulator_unregister_notifier(wm8804->supplies[i].consumer,
@@ -606,8 +605,6 @@ static int wm8804_probe(struct snd_soc_codec *codec)
goto err_reg_enable;
}
- wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
return 0;
err_reg_enable:
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 44a5f1511f0f..3a0d4b7d692f 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1209,16 +1209,8 @@ static int wm8900_probe(struct snd_soc_codec *codec)
return 0;
}
-/* power down chip */
-static int wm8900_remove(struct snd_soc_codec *codec)
-{
- wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8900 = {
.probe = wm8900_probe,
- .remove = wm8900_remove,
.suspend = wm8900_suspend,
.resume = wm8900_resume,
.set_bias_level = wm8900_set_bias_level,
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index c038b3e04398..cc6b0ef98a34 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -26,6 +26,7 @@
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/irq.h>
+#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
@@ -117,12 +118,12 @@ static const struct reg_default wm8903_reg_defaults[] = {
struct wm8903_priv {
struct wm8903_platform_data *pdata;
struct device *dev;
- struct snd_soc_codec *codec;
struct regmap *regmap;
int sysclk;
int irq;
+ struct mutex lock;
int fs;
int deemph;
@@ -457,7 +458,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
if (deemph > 1)
return -EINVAL;
- mutex_lock(&codec->mutex);
+ mutex_lock(&wm8903->lock);
if (wm8903->deemph != deemph) {
wm8903->deemph = deemph;
@@ -465,7 +466,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
ret = 1;
}
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&wm8903->lock);
return ret;
}
@@ -1757,21 +1758,12 @@ static struct snd_soc_dai_driver wm8903_dai = {
.symmetric_rates = 1,
};
-static int wm8903_suspend(struct snd_soc_codec *codec)
-{
- wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
static int wm8903_resume(struct snd_soc_codec *codec)
{
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
regcache_sync(wm8903->regmap);
- wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
return 0;
}
@@ -1889,33 +1881,12 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903)
}
#endif
-static int wm8903_probe(struct snd_soc_codec *codec)
-{
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
- wm8903->codec = codec;
-
- /* power on device */
- wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-/* power down chip */
-static int wm8903_remove(struct snd_soc_codec *codec)
-{
- wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8903 = {
- .probe = wm8903_probe,
- .remove = wm8903_remove,
- .suspend = wm8903_suspend,
.resume = wm8903_resume,
.set_bias_level = wm8903_set_bias_level,
.seq_notifier = wm8903_seq_notifier,
+ .suspend_bias_off = true,
+
.controls = wm8903_snd_controls,
.num_controls = ARRAY_SIZE(wm8903_snd_controls),
.dapm_widgets = wm8903_dapm_widgets,
@@ -2023,6 +1994,8 @@ static int wm8903_i2c_probe(struct i2c_client *i2c,
GFP_KERNEL);
if (wm8903 == NULL)
return -ENOMEM;
+
+ mutex_init(&wm8903->lock);
wm8903->dev = &i2c->dev;
wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 52011043e54c..e4142b4309eb 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -695,17 +695,6 @@ static struct snd_soc_dai_driver wm8940_dai = {
.symmetric_rates = 1,
};
-static int wm8940_suspend(struct snd_soc_codec *codec)
-{
- return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
-}
-
-static int wm8940_resume(struct snd_soc_codec *codec)
-{
- wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int wm8940_probe(struct snd_soc_codec *codec)
{
struct wm8940_setup_data *pdata = codec->dev->platform_data;
@@ -736,18 +725,11 @@ static int wm8940_probe(struct snd_soc_codec *codec)
return ret;
}
-static int wm8940_remove(struct snd_soc_codec *codec)
-{
- wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8940 = {
.probe = wm8940_probe,
- .remove = wm8940_remove,
- .suspend = wm8940_suspend,
- .resume = wm8940_resume,
.set_bias_level = wm8940_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = wm8940_snd_controls,
.num_controls = ARRAY_SIZE(wm8940_snd_controls),
.dapm_widgets = wm8940_dapm_widgets,
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 09d91d9dc4ee..1173f7fef5a7 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -866,29 +866,6 @@ static struct snd_soc_dai_driver wm8955_dai = {
.ops = &wm8955_dai_ops,
};
-#ifdef CONFIG_PM
-static int wm8955_suspend(struct snd_soc_codec *codec)
-{
- struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
-
- wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- regcache_mark_dirty(wm8955->regmap);
-
- return 0;
-}
-
-static int wm8955_resume(struct snd_soc_codec *codec)
-{
- wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define wm8955_suspend NULL
-#define wm8955_resume NULL
-#endif
-
static int wm8955_probe(struct snd_soc_codec *codec)
{
struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
@@ -964,18 +941,10 @@ err_enable:
return ret;
}
-static int wm8955_remove(struct snd_soc_codec *codec)
-{
- wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8955 = {
.probe = wm8955_probe,
- .remove = wm8955_remove,
- .suspend = wm8955_suspend,
- .resume = wm8955_resume,
.set_bias_level = wm8955_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8955_snd_controls,
.num_controls = ARRAY_SIZE(wm8955_snd_controls),
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 0dada7f0105e..3cbc82b33292 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -867,9 +867,9 @@ static void wm8958_enh_eq_loaded(const struct firmware *fw, void *context)
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (fw && (wm8958_dsp2_fw(codec, "ENH_EQ", fw, true) == 0)) {
- mutex_lock(&codec->mutex);
+ mutex_lock(&wm8994->fw_lock);
wm8994->enh_eq = fw;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&wm8994->fw_lock);
}
}
@@ -879,9 +879,9 @@ static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context)
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (fw && (wm8958_dsp2_fw(codec, "MBC+VSS", fw, true) == 0)) {
- mutex_lock(&codec->mutex);
+ mutex_lock(&wm8994->fw_lock);
wm8994->mbc_vss = fw;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&wm8994->fw_lock);
}
}
@@ -891,9 +891,9 @@ static void wm8958_mbc_loaded(const struct firmware *fw, void *context)
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (fw && (wm8958_dsp2_fw(codec, "MBC", fw, true) == 0)) {
- mutex_lock(&codec->mutex);
+ mutex_lock(&wm8994->fw_lock);
wm8994->mbc = fw;
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&wm8994->fw_lock);
}
}
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 4dc4e85116cd..031a1ae71d94 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -125,9 +125,10 @@ struct wm8960_priv {
struct snd_soc_dapm_widget *out3;
bool deemph;
int playback_fs;
+ struct wm8960_data pdata;
};
-#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0)
+#define wm8960_reset(c) regmap_write(c, WM8960_RESET, 0)
/* enumerated controls */
static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
@@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = {
static int wm8960_add_widgets(struct snd_soc_codec *codec)
{
- struct wm8960_data *pdata = codec->dev->platform_data;
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+ struct wm8960_data *pdata = &wm8960->pdata;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_dapm_widget *w;
@@ -942,56 +943,15 @@ static struct snd_soc_dai_driver wm8960_dai = {
.symmetric_rates = 1,
};
-static int wm8960_suspend(struct snd_soc_codec *codec)
-{
- struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-
- wm8960->set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8960_resume(struct snd_soc_codec *codec)
-{
- struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-
- wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int wm8960_probe(struct snd_soc_codec *codec)
{
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
- struct wm8960_data *pdata = dev_get_platdata(codec->dev);
- int ret;
-
- wm8960->set_bias_level = wm8960_set_bias_level_out3;
-
- if (!pdata) {
- dev_warn(codec->dev, "No platform data supplied\n");
- } else {
- if (pdata->capless)
- wm8960->set_bias_level = wm8960_set_bias_level_capless;
- }
-
- ret = wm8960_reset(codec);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to issue reset\n");
- return ret;
- }
-
- wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ struct wm8960_data *pdata = &wm8960->pdata;
- /* Latch the update bits */
- snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100);
- snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100);
+ if (pdata->capless)
+ wm8960->set_bias_level = wm8960_set_bias_level_capless;
+ else
+ wm8960->set_bias_level = wm8960_set_bias_level_out3;
snd_soc_add_codec_controls(codec, wm8960_snd_controls,
ARRAY_SIZE(wm8960_snd_controls));
@@ -1000,21 +960,10 @@ static int wm8960_probe(struct snd_soc_codec *codec)
return 0;
}
-/* power down chip */
-static int wm8960_remove(struct snd_soc_codec *codec)
-{
- struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-
- wm8960->set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8960 = {
.probe = wm8960_probe,
- .remove = wm8960_remove,
- .suspend = wm8960_suspend,
- .resume = wm8960_resume,
.set_bias_level = wm8960_set_bias_level,
+ .suspend_bias_off = true,
};
static const struct regmap_config wm8960_regmap = {
@@ -1029,6 +978,18 @@ static const struct regmap_config wm8960_regmap = {
.volatile_reg = wm8960_volatile,
};
+static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
+ struct wm8960_data *pdata)
+{
+ const struct device_node *np = i2c->dev.of_node;
+
+ if (of_property_read_bool(np, "wlf,capless"))
+ pdata->capless = true;
+
+ if (of_property_read_bool(np, "wlf,shared-lrclk"))
+ pdata->shared_lrclk = true;
+}
+
static int wm8960_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1045,7 +1006,18 @@ static int wm8960_i2c_probe(struct i2c_client *i2c,
if (IS_ERR(wm8960->regmap))
return PTR_ERR(wm8960->regmap);
- if (pdata && pdata->shared_lrclk) {
+ if (pdata)
+ memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data));
+ else if (i2c->dev.of_node)
+ wm8960_set_pdata_from_of(i2c, &wm8960->pdata);
+
+ ret = wm8960_reset(wm8960->regmap);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ if (wm8960->pdata.shared_lrclk) {
ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2,
0x4, 0x4);
if (ret != 0) {
@@ -1055,6 +1027,18 @@ static int wm8960_i2c_probe(struct i2c_client *i2c,
}
}
+ /* Latch the update bits */
+ regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_LADC, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_RADC, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_LDAC, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_RDAC, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_LOUT1, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_ROUT1, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_LOUT2, 0x100, 0x100);
+ regmap_update_bits(wm8960->regmap, WM8960_ROUT2, 0x100, 0x100);
+
i2c_set_clientdata(i2c, wm8960);
ret = snd_soc_register_codec(&i2c->dev,
@@ -1075,10 +1059,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
+static const struct of_device_id wm8960_of_match[] = {
+ { .compatible = "wlf,wm8960", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, wm8960_of_match);
+
static struct i2c_driver wm8960_i2c_driver = {
.driver = {
.name = "wm8960",
.owner = THIS_MODULE,
+ .of_match_table = wm8960_of_match,
},
.probe = wm8960_i2c_probe,
.remove = wm8960_i2c_remove,
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 41d23e920ad5..eeffd05384b4 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -835,7 +835,6 @@ static struct snd_soc_dai_driver wm8961_dai = {
static int wm8961_probe(struct snd_soc_codec *codec)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
u16 reg;
/* Enable class W */
@@ -871,50 +870,33 @@ static int wm8961_probe(struct snd_soc_codec *codec)
reg &= ~WM8961_MANUAL_MODE;
snd_soc_write(codec, WM8961_CLOCKING_3, reg);
- wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- snd_soc_add_codec_controls(codec, wm8961_snd_controls,
- ARRAY_SIZE(wm8961_snd_controls));
- snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets,
- ARRAY_SIZE(wm8961_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
-
- return 0;
-}
-
-static int wm8961_remove(struct snd_soc_codec *codec)
-{
- wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
#ifdef CONFIG_PM
-static int wm8961_suspend(struct snd_soc_codec *codec)
-{
- wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
static int wm8961_resume(struct snd_soc_codec *codec)
{
snd_soc_cache_sync(codec);
- wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
return 0;
}
#else
-#define wm8961_suspend NULL
#define wm8961_resume NULL
#endif
static struct snd_soc_codec_driver soc_codec_dev_wm8961 = {
.probe = wm8961_probe,
- .remove = wm8961_remove,
- .suspend = wm8961_suspend,
.resume = wm8961_resume,
.set_bias_level = wm8961_set_bias_level,
+ .suspend_bias_off = true,
+
+ .controls = wm8961_snd_controls,
+ .num_controls = ARRAY_SIZE(wm8961_snd_controls),
+ .dapm_widgets = wm8961_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8961_dapm_widgets),
+ .dapm_routes = audio_paths,
+ .num_dapm_routes = ARRAY_SIZE(audio_paths),
};
static const struct regmap_config wm8961_regmap = {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 9077411e62ce..1534d88a66e9 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -26,6 +26,7 @@
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
#include <linux/workqueue.h>
+#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
@@ -67,6 +68,7 @@ struct wm8962_priv {
int fll_fref;
int fll_fout;
+ struct mutex dsp2_ena_lock;
u16 dsp2_ena;
struct delayed_work mic_work;
@@ -1570,7 +1572,7 @@ static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol,
int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) &
WM8962_DSP2_ENA;
- mutex_lock(&codec->mutex);
+ mutex_lock(&wm8962->dsp2_ena_lock);
if (ucontrol->value.integer.value[0])
wm8962->dsp2_ena |= 1 << shift;
@@ -1590,7 +1592,7 @@ static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol,
}
out:
- mutex_unlock(&codec->mutex);
+ mutex_unlock(&wm8962->dsp2_ena_lock);
return ret;
}
@@ -3552,11 +3554,12 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
unsigned int reg;
int ret, i, irq_pol, trigger;
- wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv),
- GFP_KERNEL);
+ wm8962 = devm_kzalloc(&i2c->dev, sizeof(*wm8962), GFP_KERNEL);
if (wm8962 == NULL)
return -ENOMEM;
+ mutex_init(&wm8962->dsp2_ena_lock);
+
i2c_set_clientdata(i2c, wm8962);
INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 682e9eda1019..ff0e4646b934 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -568,18 +568,6 @@ static struct snd_soc_dai_driver wm8974_dai = {
.symmetric_rates = 1,
};
-static int wm8974_suspend(struct snd_soc_codec *codec)
-{
- wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8974_resume(struct snd_soc_codec *codec)
-{
- wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static const struct regmap_config wm8974_regmap = {
.reg_bits = 7,
.val_bits = 9,
@@ -599,24 +587,13 @@ static int wm8974_probe(struct snd_soc_codec *codec)
return ret;
}
- wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return ret;
-}
-
-/* power down chip */
-static int wm8974_remove(struct snd_soc_codec *codec)
-{
- wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm8974 = {
.probe = wm8974_probe,
- .remove = wm8974_remove,
- .suspend = wm8974_suspend,
- .resume = wm8974_resume,
.set_bias_level = wm8974_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8974_snd_controls,
.num_controls = ARRAY_SIZE(wm8974_snd_controls),
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index ee2ba574952b..cf7032911721 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -991,21 +991,11 @@ static int wm8978_probe(struct snd_soc_codec *codec)
for (i = 0; i < ARRAY_SIZE(update_reg); i++)
snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100);
- wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-/* power down chip */
-static int wm8978_remove(struct snd_soc_codec *codec)
-{
- wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm8978 = {
.probe = wm8978_probe,
- .remove = wm8978_remove,
.suspend = wm8978_suspend,
.resume = wm8978_resume,
.set_bias_level = wm8978_set_bias_level,
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index ac5defda8824..5d1cf08a72b8 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -967,29 +967,6 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-#ifdef CONFIG_PM
-static int wm8983_suspend(struct snd_soc_codec *codec)
-{
- wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8983_resume(struct snd_soc_codec *codec)
-{
- wm8983_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define wm8983_suspend NULL
-#define wm8983_resume NULL
-#endif
-
-static int wm8983_remove(struct snd_soc_codec *codec)
-{
- wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static int wm8983_probe(struct snd_soc_codec *codec)
{
int ret;
@@ -1055,10 +1032,8 @@ static struct snd_soc_dai_driver wm8983_dai = {
static struct snd_soc_codec_driver soc_codec_dev_wm8983 = {
.probe = wm8983_probe,
- .remove = wm8983_remove,
- .suspend = wm8983_suspend,
- .resume = wm8983_resume,
.set_bias_level = wm8983_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8983_snd_controls,
.num_controls = ARRAY_SIZE(wm8983_snd_controls),
.dapm_widgets = wm8983_dapm_widgets,
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index ee380190399f..0b3b54c9971d 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -961,29 +961,6 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-#ifdef CONFIG_PM
-static int wm8985_suspend(struct snd_soc_codec *codec)
-{
- wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8985_resume(struct snd_soc_codec *codec)
-{
- wm8985_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-#else
-#define wm8985_suspend NULL
-#define wm8985_resume NULL
-#endif
-
-static int wm8985_remove(struct snd_soc_codec *codec)
-{
- wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static int wm8985_probe(struct snd_soc_codec *codec)
{
size_t i;
@@ -1023,7 +1000,6 @@ static int wm8985_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8985_BIAS_CTRL, WM8985_BIASCUT,
WM8985_BIASCUT);
- wm8985_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
err_reg_enable:
@@ -1064,10 +1040,8 @@ static struct snd_soc_dai_driver wm8985_dai = {
static struct snd_soc_codec_driver soc_codec_dev_wm8985 = {
.probe = wm8985_probe,
- .remove = wm8985_remove,
- .suspend = wm8985_suspend,
- .resume = wm8985_resume,
.set_bias_level = wm8985_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8985_snd_controls,
.num_controls = ARRAY_SIZE(wm8985_snd_controls),
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index a5130d965146..e418199155a8 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -793,21 +793,6 @@ static struct snd_soc_dai_driver wm8988_dai = {
.symmetric_rates = 1,
};
-static int wm8988_suspend(struct snd_soc_codec *codec)
-{
- struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
-
- wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
- regcache_mark_dirty(wm8988->regmap);
- return 0;
-}
-
-static int wm8988_resume(struct snd_soc_codec *codec)
-{
- wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
static int wm8988_probe(struct snd_soc_codec *codec)
{
int ret = 0;
@@ -825,23 +810,13 @@ static int wm8988_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8988_ROUT2V, 0x0100, 0x0100);
snd_soc_update_bits(codec, WM8988_RINVOL, 0x0100, 0x0100);
- wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int wm8988_remove(struct snd_soc_codec *codec)
-{
- wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm8988 = {
.probe = wm8988_probe,
- .remove = wm8988_remove,
- .suspend = wm8988_suspend,
- .resume = wm8988_resume,
.set_bias_level = wm8988_set_bias_level,
+ .suspend_bias_off = true,
.controls = wm8988_snd_controls,
.num_controls = ARRAY_SIZE(wm8988_snd_controls),
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 03e43e3f395e..8a584229310a 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1271,18 +1271,6 @@ static struct snd_soc_dai_driver wm8990_dai = {
.ops = &wm8990_dai_ops,
};
-static int wm8990_suspend(struct snd_soc_codec *codec)
-{
- wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8990_resume(struct snd_soc_codec *codec)
-{
- wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
/*
* initialise the WM8990 driver
* register the mixer and dsp interfaces with the kernel
@@ -1309,19 +1297,11 @@ static int wm8990_probe(struct snd_soc_codec *codec)
return 0;
}
-/* power down chip */
-static int wm8990_remove(struct snd_soc_codec *codec)
-{
- wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm8990 = {
.probe = wm8990_probe,
- .remove = wm8990_remove,
- .suspend = wm8990_suspend,
- .resume = wm8990_resume,
.set_bias_level = wm8990_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = wm8990_snd_controls,
.num_controls = ARRAY_SIZE(wm8990_snd_controls),
.dapm_widgets = wm8990_dapm_widgets,
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index d0be89731cdb..b0ac2c3e31b9 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1227,32 +1227,6 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-static int wm8991_suspend(struct snd_soc_codec *codec)
-{
- wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8991_resume(struct snd_soc_codec *codec)
-{
- wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- return 0;
-}
-
-/* power down chip */
-static int wm8991_remove(struct snd_soc_codec *codec)
-{
- wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static int wm8991_probe(struct snd_soc_codec *codec)
-{
- wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
#define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
@@ -1293,11 +1267,9 @@ static struct snd_soc_dai_driver wm8991_dai = {
};
static struct snd_soc_codec_driver soc_codec_dev_wm8991 = {
- .probe = wm8991_probe,
- .remove = wm8991_remove,
- .suspend = wm8991_suspend,
- .resume = wm8991_resume,
.set_bias_level = wm8991_set_bias_level,
+ .suspend_bias_off = true,
+
.controls = wm8991_snd_controls,
.num_controls = ARRAY_SIZE(wm8991_snd_controls),
.dapm_widgets = wm8991_dapm_widgets,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 93b14eda355a..53c6fe359496 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1486,7 +1486,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
{
struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
wm8993->hubs_data.hp_startup_mode = 1;
wm8993->hubs_data.dcs_codes_l = -2;
@@ -1518,10 +1517,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
wm8993->pdata.micbias1_lvl,
wm8993->pdata.micbias2_lvl);
- ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- if (ret != 0)
- return ret;
-
snd_soc_add_codec_controls(codec, wm8993_snd_controls,
ARRAY_SIZE(wm8993_snd_controls));
if (wm8993->pdata.num_retune_configs != 0) {
@@ -1550,12 +1545,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
}
-static int wm8993_remove(struct snd_soc_codec *codec)
-{
- wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
#ifdef CONFIG_PM
static int wm8993_suspend(struct snd_soc_codec *codec)
{
@@ -1629,7 +1618,6 @@ static const struct regmap_config wm8993_regmap = {
static struct snd_soc_codec_driver soc_codec_dev_wm8993 = {
.probe = wm8993_probe,
- .remove = wm8993_remove,
.suspend = wm8993_suspend,
.resume = wm8993_resume,
.set_bias_level = wm8993_set_bias_level,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1fcb9f3f3097..36b767fa37a6 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -4391,8 +4391,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec)
struct wm8994 *control = wm8994->wm8994;
int i;
- wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i,
&wm8994->fll_locked[i]);
@@ -4457,6 +4455,8 @@ static int wm8994_probe(struct platform_device *pdev)
return -ENOMEM;
platform_set_drvdata(pdev, wm8994);
+ mutex_init(&wm8994->fw_lock);
+
wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent);
pm_runtime_enable(&pdev->dev);
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 6536f8d45ac6..dd73387b1cc4 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -13,6 +13,7 @@
#include <linux/firmware.h>
#include <linux/completion.h>
#include <linux/workqueue.h>
+#include <linux/mutex.h>
#include "wm_hubs.h"
@@ -156,6 +157,7 @@ struct wm8994_priv {
unsigned int aif1clk_disable:1;
unsigned int aif2clk_disable:1;
+ struct mutex fw_lock;
int dsp_active;
const struct firmware *cur_fw;
const struct firmware *mbc;
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 1288edeb8c7d..c280f0a3a424 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -2004,7 +2004,6 @@ static int wm8995_remove(struct snd_soc_codec *codec)
int i;
wm8995 = snd_soc_codec_get_drvdata(codec);
- wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF);
for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i)
regulator_unregister_notifier(wm8995->supplies[i].consumer,
@@ -2078,8 +2077,6 @@ static int wm8995_probe(struct snd_soc_codec *codec)
goto err_reg_enable;
}
- wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
/* Latch volume updates (right only; we always do left then right). */
snd_soc_update_bits(codec, WM8995_AIF1_DAC1_RIGHT_VOLUME,
WM8995_AIF1DAC1_VU_MASK, WM8995_AIF1DAC1_VU);
@@ -2102,13 +2099,6 @@ static int wm8995_probe(struct snd_soc_codec *codec)
wm8995_update_class_w(codec);
- snd_soc_add_codec_controls(codec, wm8995_snd_controls,
- ARRAY_SIZE(wm8995_snd_controls));
- snd_soc_dapm_new_controls(&codec->dapm, wm8995_dapm_widgets,
- ARRAY_SIZE(wm8995_dapm_widgets));
- snd_soc_dapm_add_routes(&codec->dapm, wm8995_intercon,
- ARRAY_SIZE(wm8995_intercon));
-
return 0;
err_reg_enable:
@@ -2205,6 +2195,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = {
.remove = wm8995_remove,
.set_bias_level = wm8995_set_bias_level,
.idle_bias_off = true,
+
+ .controls = wm8995_snd_controls,
+ .num_controls = ARRAY_SIZE(wm8995_snd_controls),
+ .dapm_widgets = wm8995_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8995_dapm_widgets),
+ .dapm_routes = wm8995_intercon,
+ .num_dapm_routes = ARRAY_SIZE(wm8995_intercon),
};
static struct regmap_config wm8995_regmap = {
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 0cdc9e2184ab..b1d946facd57 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1277,15 +1277,8 @@ static int wm9081_probe(struct snd_soc_codec *codec)
return 0;
}
-static int wm9081_remove(struct snd_soc_codec *codec)
-{
- wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm9081 = {
.probe = wm9081_probe,
- .remove = wm9081_remove,
.set_sysclk = wm9081_set_sysclk,
.set_bias_level = wm9081_set_bias_level,
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index a13f0725611a..6ffe8dc4f3fa 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -550,45 +550,15 @@ static int wm9090_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM9090_CLOCKING_1,
WM9090_TOCLK_ENA, WM9090_TOCLK_ENA);
- wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
wm9090_add_controls(codec);
return 0;
}
-#ifdef CONFIG_PM
-static int wm9090_suspend(struct snd_soc_codec *codec)
-{
- wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
-static int wm9090_resume(struct snd_soc_codec *codec)
-{
- wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-#else
-#define wm9090_suspend NULL
-#define wm9090_resume NULL
-#endif
-
-static int wm9090_remove(struct snd_soc_codec *codec)
-{
- wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_codec_dev_wm9090 = {
.probe = wm9090_probe,
- .remove = wm9090_remove,
- .suspend = wm9090_suspend,
- .resume = wm9090_resume,
.set_bias_level = wm9090_set_bias_level,
+ .suspend_bias_off = true,
};
static const struct regmap_config wm9090_regmap = {
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index c0b7f45dfa37..d3a800fa6f06 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -203,13 +203,14 @@ static const struct snd_soc_dapm_route wm9705_audio_map[] = {
/* We use a register cache to enhance read performance. */
static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
switch (reg) {
case AC97_RESET:
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
- return soc_ac97_ops->read(codec->ac97, reg);
+ return soc_ac97_ops->read(ac97, reg);
default:
reg = reg >> 1;
@@ -223,9 +224,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
- soc_ac97_ops->write(codec->ac97, reg, val);
+ soc_ac97_ops->write(ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9705_reg)))
cache[reg] = val;
@@ -263,7 +265,6 @@ static const struct snd_soc_dai_ops wm9705_dai_ops = {
static struct snd_soc_dai_driver wm9705_dai[] = {
{
.name = "wm9705-hifi",
- .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -294,36 +295,41 @@ static struct snd_soc_dai_driver wm9705_dai[] = {
static int wm9705_reset(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
if (soc_ac97_ops->reset) {
- soc_ac97_ops->reset(codec->ac97);
+ soc_ac97_ops->reset(ac97);
if (ac97_read(codec, 0) == wm9705_reg[0])
return 0; /* Success */
}
+ dev_err(codec->dev, "Failed to reset: AC97 link error\n");
+
return -EIO;
}
#ifdef CONFIG_PM
static int wm9705_soc_suspend(struct snd_soc_codec *codec)
{
- soc_ac97_ops->write(codec->ac97, AC97_POWERDOWN, 0xffff);
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ soc_ac97_ops->write(ac97, AC97_POWERDOWN, 0xffff);
return 0;
}
static int wm9705_soc_resume(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
int i, ret;
u16 *cache = codec->reg_cache;
ret = wm9705_reset(codec);
- if (ret < 0) {
- printk(KERN_ERR "could not reset AC97 codec\n");
+ if (ret < 0)
return ret;
- }
for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
- soc_ac97_ops->write(codec->ac97, i, cache[i>>1]);
+ soc_ac97_ops->write(ac97, i, cache[i>>1]);
}
return 0;
@@ -335,31 +341,34 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec)
static int wm9705_soc_probe(struct snd_soc_codec *codec)
{
+ struct snd_ac97 *ac97;
int ret = 0;
- ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0);
- if (ret < 0) {
- printk(KERN_ERR "wm9705: failed to register AC97 codec\n");
+ ac97 = snd_soc_new_ac97_codec(codec);
+ if (IS_ERR(ac97)) {
+ ret = PTR_ERR(ac97);
+ dev_err(codec->dev, "Failed to register AC97 codec\n");
return ret;
}
+ snd_soc_codec_set_drvdata(codec, ac97);
+
ret = wm9705_reset(codec);
if (ret)
goto reset_err;
- snd_soc_add_codec_controls(codec, wm9705_snd_ac97_controls,
- ARRAY_SIZE(wm9705_snd_ac97_controls));
-
return 0;
reset_err:
- snd_soc_free_ac97_codec(codec);
+ snd_soc_free_ac97_codec(ac97);
return ret;
}
static int wm9705_soc_remove(struct snd_soc_codec *codec)
{
- snd_soc_free_ac97_codec(codec);
+ struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_free_ac97_codec(ac97);
return 0;
}
@@ -374,6 +383,9 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9705 = {
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = wm9705_reg,
+
+ .controls = wm9705_snd_ac97_controls,
+ .num_controls = ARRAY_SIZE(wm9705_snd_ac97_controls),
.dapm_widgets = wm9705_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm9705_dapm_widgets),
.dapm_routes = wm9705_audio_map,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index c5eb746087b4..7c45971bb4ec 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -23,6 +23,12 @@
#include <sound/tlv.h>
#include "wm9712.h"
+struct wm9712_priv {
+ struct snd_ac97 *ac97;
+ unsigned int hp_mixer[2];
+ struct mutex lock;
+};
+
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg);
static int ac97_write(struct snd_soc_codec *codec,
@@ -48,12 +54,10 @@ static const u16 wm9712_reg[] = {
0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
- 0x0000, 0x0000 /* virtual hp mixers */
};
-/* virtual HP mixers regs */
-#define HPL_MIXER 0x80
-#define HPR_MIXER 0x82
+#define HPL_MIXER 0x0
+#define HPR_MIXER 0x1
static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
@@ -157,75 +161,108 @@ SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv),
SOC_SINGLE_TLV("Mic Boost Volume", AC97_MIC, 7, 1, 0, boost_tlv),
};
+static const unsigned int wm9712_mixer_mute_regs[] = {
+ AC97_VIDEO,
+ AC97_PCM,
+ AC97_LINE,
+ AC97_PHONE,
+ AC97_CD,
+ AC97_PC_BEEP,
+};
+
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
* This makes it impossible to determine the audio path.
*/
-static int mixer_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
+static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
- u16 l, r, beep, line, phone, mic, pcm, aux;
-
- l = ac97_read(w->codec, HPL_MIXER);
- r = ac97_read(w->codec, HPR_MIXER);
- beep = ac97_read(w->codec, AC97_PC_BEEP);
- mic = ac97_read(w->codec, AC97_VIDEO);
- phone = ac97_read(w->codec, AC97_PHONE);
- line = ac97_read(w->codec, AC97_LINE);
- pcm = ac97_read(w->codec, AC97_PCM);
- aux = ac97_read(w->codec, AC97_CD);
-
- if (l & 0x1 || r & 0x1)
- ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = ucontrol->value.enumerated.item[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int mixer, mask, shift, old;
+ struct snd_soc_dapm_update update;
+ bool change;
+
+ mixer = mc->shift >> 8;
+ shift = mc->shift & 0xff;
+ mask = 1 << shift;
+
+ mutex_lock(&wm9712->lock);
+ old = wm9712->hp_mixer[mixer];
+ if (ucontrol->value.enumerated.item[0])
+ wm9712->hp_mixer[mixer] |= mask;
else
- ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
+ wm9712->hp_mixer[mixer] &= ~mask;
+
+ change = old != wm9712->hp_mixer[mixer];
+ if (change) {
+ update.kcontrol = kcontrol;
+ update.reg = wm9712_mixer_mute_regs[shift];
+ update.mask = 0x8000;
+ if ((wm9712->hp_mixer[0] & mask) ||
+ (wm9712->hp_mixer[1] & mask))
+ update.val = 0x0;
+ else
+ update.val = 0x8000;
+
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, val,
+ &update);
+ }
- if (l & 0x2 || r & 0x2)
- ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
- else
- ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+ mutex_unlock(&wm9712->lock);
- if (l & 0x4 || r & 0x4)
- ac97_write(w->codec, AC97_LINE, line & 0x7fff);
- else
- ac97_write(w->codec, AC97_LINE, line | 0x8000);
+ return change;
+}
- if (l & 0x8 || r & 0x8)
- ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
- else
- ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int shift, mixer;
- if (l & 0x10 || r & 0x10)
- ac97_write(w->codec, AC97_CD, aux & 0x7fff);
- else
- ac97_write(w->codec, AC97_CD, aux | 0x8000);
+ mixer = mc->shift >> 8;
+ shift = mc->shift & 0xff;
- if (l & 0x20 || r & 0x20)
- ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
- else
- ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+ ucontrol->value.enumerated.item[0] =
+ (wm9712->hp_mixer[mixer] >> shift) & 1;
return 0;
}
+#define WM9712_HP_MIXER_CTRL(xname, xmixer, xshift) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = wm9712_hp_mixer_get, .put = wm9712_hp_mixer_put, \
+ .private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, \
+ (xmixer << 8) | xshift, 1, 0, 0) \
+}
+
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
- SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
- SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
- SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
- SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
- SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
- SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
+ WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPL_MIXER, 5),
+ WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 4),
+ WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPL_MIXER, 3),
+ WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPL_MIXER, 2),
+ WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 1),
+ WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPL_MIXER, 0),
};
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
- SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
- SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
- SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
- SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
- SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
- SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0),
+ WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPR_MIXER, 5),
+ WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 4),
+ WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPR_MIXER, 3),
+ WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPR_MIXER, 2),
+ WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 1),
+ WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPR_MIXER, 0),
};
/* Speaker Mixer */
@@ -299,12 +336,10 @@ SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
&wm9712_diff_sel_controls),
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
- &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
- mixer_event, SND_SOC_DAPM_POST_REG),
-SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
- &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
- mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_INT_PAGING, 9, 1,
+ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_INT_PAGING, 8, 1,
+ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls)),
SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
&wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
@@ -450,12 +485,13 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
reg == AC97_REC_GAIN)
- return soc_ac97_ops->read(codec->ac97, reg);
+ return soc_ac97_ops->read(wm9712->ac97, reg);
else {
reg = reg >> 1;
@@ -469,10 +505,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
- if (reg < 0x7c)
- soc_ac97_ops->write(codec->ac97, reg, val);
+ soc_ac97_ops->write(wm9712->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9712_reg)))
cache[reg] = val;
@@ -532,7 +568,6 @@ static const struct snd_soc_dai_ops wm9712_dai_ops_aux = {
static struct snd_soc_dai_driver wm9712_dai[] = {
{
.name = "wm9712-hifi",
- .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -581,40 +616,35 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec,
static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
{
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
+
if (try_warm && soc_ac97_ops->warm_reset) {
- soc_ac97_ops->warm_reset(codec->ac97);
+ soc_ac97_ops->warm_reset(wm9712->ac97);
if (ac97_read(codec, 0) == wm9712_reg[0])
return 1;
}
- soc_ac97_ops->reset(codec->ac97);
+ soc_ac97_ops->reset(wm9712->ac97);
if (soc_ac97_ops->warm_reset)
- soc_ac97_ops->warm_reset(codec->ac97);
+ soc_ac97_ops->warm_reset(wm9712->ac97);
if (ac97_read(codec, 0) != wm9712_reg[0])
goto err;
return 0;
err:
- printk(KERN_ERR "WM9712 AC97 reset failed\n");
+ dev_err(codec->dev, "Failed to reset: AC97 link error\n");
return -EIO;
}
-static int wm9712_soc_suspend(struct snd_soc_codec *codec)
-{
- wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
static int wm9712_soc_resume(struct snd_soc_codec *codec)
{
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
int i, ret;
u16 *cache = codec->reg_cache;
ret = wm9712_reset(codec, 1);
- if (ret < 0) {
- printk(KERN_ERR "could not reset AC97 codec\n");
+ if (ret < 0)
return ret;
- }
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -624,7 +654,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec)
if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
(i > 0x58 && i != 0x5c))
continue;
- soc_ac97_ops->write(codec->ac97, i, cache[i>>1]);
+ soc_ac97_ops->write(wm9712->ac97, i, cache[i>>1]);
}
}
@@ -633,52 +663,53 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec)
static int wm9712_soc_probe(struct snd_soc_codec *codec)
{
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
- ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0);
- if (ret < 0) {
- printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
+ wm9712->ac97 = snd_soc_new_ac97_codec(codec);
+ if (IS_ERR(wm9712->ac97)) {
+ ret = PTR_ERR(wm9712->ac97);
+ dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret);
return ret;
}
ret = wm9712_reset(codec, 0);
- if (ret < 0) {
- printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
+ if (ret < 0)
goto reset_err;
- }
/* set alc mux to none */
ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
- wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_codec_controls(codec, wm9712_snd_ac97_controls,
- ARRAY_SIZE(wm9712_snd_ac97_controls));
-
return 0;
reset_err:
- snd_soc_free_ac97_codec(codec);
+ snd_soc_free_ac97_codec(wm9712->ac97);
return ret;
}
static int wm9712_soc_remove(struct snd_soc_codec *codec)
{
- snd_soc_free_ac97_codec(codec);
+ struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_free_ac97_codec(wm9712->ac97);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm9712 = {
.probe = wm9712_soc_probe,
.remove = wm9712_soc_remove,
- .suspend = wm9712_soc_suspend,
.resume = wm9712_soc_resume,
.read = ac97_read,
.write = ac97_write,
.set_bias_level = wm9712_set_bias_level,
+ .suspend_bias_off = true,
.reg_cache_size = ARRAY_SIZE(wm9712_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = wm9712_reg,
+
+ .controls = wm9712_snd_ac97_controls,
+ .num_controls = ARRAY_SIZE(wm9712_snd_ac97_controls),
.dapm_widgets = wm9712_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets),
.dapm_routes = wm9712_audio_map,
@@ -687,6 +718,16 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9712 = {
static int wm9712_probe(struct platform_device *pdev)
{
+ struct wm9712_priv *wm9712;
+
+ wm9712 = devm_kzalloc(&pdev->dev, sizeof(*wm9712), GFP_KERNEL);
+ if (wm9712 == NULL)
+ return -ENOMEM;
+
+ mutex_init(&wm9712->lock);
+
+ platform_set_drvdata(pdev, wm9712);
+
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai));
}
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index bddee30a4bc7..5df7f6d12bef 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -30,7 +30,10 @@
#include "wm9713.h"
struct wm9713_priv {
+ struct snd_ac97 *ac97;
u32 pll_in; /* PLL input frequency */
+ unsigned int hp_mixer[2];
+ struct mutex lock;
};
static unsigned int ac97_read(struct snd_soc_codec *codec,
@@ -59,13 +62,10 @@ static const u16 wm9713_reg[] = {
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0006,
0x0001, 0x0000, 0x574d, 0x4c13,
- 0x0000, 0x0000, 0x0000
};
-/* virtual HP mixers regs */
-#define HPL_MIXER 0x80
-#define HPR_MIXER 0x82
-#define MICB_MUX 0x82
+#define HPL_MIXER 0
+#define HPR_MIXER 1
static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
@@ -110,7 +110,7 @@ SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
-SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
+SOC_ENUM_SINGLE_VIRT(2, wm9713_micb_select), /* mic selection 19 */
};
static const DECLARE_TLV_DB_SCALE(out_tlv, -4650, 150, 0);
@@ -234,6 +234,14 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
return 0;
}
+static const unsigned int wm9713_mixer_mute_regs[] = {
+ AC97_PC_BEEP,
+ AC97_MASTER_TONE,
+ AC97_PHONE,
+ AC97_REC_SEL,
+ AC97_PCM,
+ AC97_AUX,
+};
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
@@ -241,73 +249,95 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
* register map, thus we add a new (virtual) register to help determine the
* audio route within the device.
*/
-static int mixer_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
- u16 l, r, beep, tone, phone, rec, pcm, aux;
-
- l = ac97_read(w->codec, HPL_MIXER);
- r = ac97_read(w->codec, HPR_MIXER);
- beep = ac97_read(w->codec, AC97_PC_BEEP);
- tone = ac97_read(w->codec, AC97_MASTER_TONE);
- phone = ac97_read(w->codec, AC97_PHONE);
- rec = ac97_read(w->codec, AC97_REC_SEL);
- pcm = ac97_read(w->codec, AC97_PCM);
- aux = ac97_read(w->codec, AC97_AUX);
-
- if (event & SND_SOC_DAPM_PRE_REG)
- return 0;
- if ((l & 0x1) || (r & 0x1))
- ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
+ struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = ucontrol->value.enumerated.item[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int mixer, mask, shift, old;
+ struct snd_soc_dapm_update update;
+ bool change;
+
+ mixer = mc->shift >> 8;
+ shift = mc->shift & 0xff;
+ mask = (1 << shift);
+
+ mutex_lock(&wm9713->lock);
+ old = wm9713->hp_mixer[mixer];
+ if (ucontrol->value.enumerated.item[0])
+ wm9713->hp_mixer[mixer] |= mask;
else
- ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+ wm9713->hp_mixer[mixer] &= ~mask;
+
+ change = old != wm9713->hp_mixer[mixer];
+ if (change) {
+ update.kcontrol = kcontrol;
+ update.reg = wm9713_mixer_mute_regs[shift];
+ update.mask = 0x8000;
+ if ((wm9713->hp_mixer[0] & mask) ||
+ (wm9713->hp_mixer[1] & mask))
+ update.val = 0x0;
+ else
+ update.val = 0x8000;
+
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, val,
+ &update);
+ }
- if ((l & 0x2) || (r & 0x2))
- ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff);
- else
- ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000);
+ mutex_unlock(&wm9713->lock);
- if ((l & 0x4) || (r & 0x4))
- ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
- else
- ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+ return change;
+}
- if ((l & 0x8) || (r & 0x8))
- ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff);
- else
- ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000);
+static int wm9713_hp_mixer_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
+ struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int mixer, shift;
- if ((l & 0x10) || (r & 0x10))
- ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
- else
- ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+ mixer = mc->shift >> 8;
+ shift = mc->shift & 0xff;
- if ((l & 0x20) || (r & 0x20))
- ac97_write(w->codec, AC97_AUX, aux & 0x7fff);
- else
- ac97_write(w->codec, AC97_AUX, aux | 0x8000);
+ ucontrol->value.enumerated.item[0] =
+ (wm9713->hp_mixer[mixer] >> shift) & 1;
return 0;
}
+#define WM9713_HP_MIXER_CTRL(xname, xmixer, xshift) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = wm9713_hp_mixer_get, .put = wm9713_hp_mixer_put, \
+ .private_value = SOC_DOUBLE_VALUE(SND_SOC_NOPM, \
+ xshift, xmixer, 1, 0, 0) \
+}
+
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
-SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0),
-SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
-SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
-SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
-SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0),
-SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
+WM9713_HP_MIXER_CTRL("Beep Playback Switch", HPL_MIXER, 5),
+WM9713_HP_MIXER_CTRL("Voice Playback Switch", HPL_MIXER, 4),
+WM9713_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 3),
+WM9713_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 2),
+WM9713_HP_MIXER_CTRL("MonoIn Playback Switch", HPL_MIXER, 1),
+WM9713_HP_MIXER_CTRL("Bypass Playback Switch", HPL_MIXER, 0),
};
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
-SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0),
-SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
-SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
-SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
-SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0),
-SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0),
+WM9713_HP_MIXER_CTRL("Beep Playback Switch", HPR_MIXER, 5),
+WM9713_HP_MIXER_CTRL("Voice Playback Switch", HPR_MIXER, 4),
+WM9713_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 3),
+WM9713_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 2),
+WM9713_HP_MIXER_CTRL("MonoIn Playback Switch", HPR_MIXER, 1),
+WM9713_HP_MIXER_CTRL("Bypass Playback Switch", HPR_MIXER, 0),
};
/* headphone capture mux */
@@ -429,12 +459,10 @@ SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
&wm9713_mic_sel_mux_controls),
SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
&wm9713_micb_sel_mux_controls),
-SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
- &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls),
- mixer_event, SND_SOC_DAPM_POST_REG),
-SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
- &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls),
- mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
+ &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
+ &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
&wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
@@ -647,12 +675,13 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = {
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
+ struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
reg == AC97_CD)
- return soc_ac97_ops->read(codec->ac97, reg);
+ return soc_ac97_ops->read(wm9713->ac97, reg);
else {
reg = reg >> 1;
@@ -666,9 +695,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
+ struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
+
u16 *cache = codec->reg_cache;
- if (reg < 0x7c)
- soc_ac97_ops->write(codec->ac97, reg, val);
+ soc_ac97_ops->write(wm9713->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9713_reg)))
cache[reg] = val;
@@ -689,7 +719,8 @@ struct _pll_div {
* to allow rounding later */
#define FIXED_PLL_SIZE ((1 << 22) * 10)
-static void pll_factors(struct _pll_div *pll_div, unsigned int source)
+static void pll_factors(struct snd_soc_codec *codec,
+ struct _pll_div *pll_div, unsigned int source)
{
u64 Kpart;
unsigned int K, Ndiv, Nmod, target;
@@ -724,7 +755,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source)
Ndiv = target / source;
if ((Ndiv < 5) || (Ndiv > 12))
- printk(KERN_WARNING
+ dev_warn(codec->dev,
"WM9713 PLL N value %u out of recommended range!\n",
Ndiv);
@@ -768,7 +799,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
return 0;
}
- pll_factors(&pll_div, freq_in);
+ pll_factors(codec, &pll_div, freq_in);
if (pll_div.k == 0) {
reg = (pll_div.n << 12) | (pll_div.lf << 11) |
@@ -1049,7 +1080,6 @@ static const struct snd_soc_dai_ops wm9713_dai_ops_voice = {
static struct snd_soc_dai_driver wm9713_dai[] = {
{
.name = "wm9713-hifi",
- .ac97_control = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
@@ -1095,17 +1125,22 @@ static struct snd_soc_dai_driver wm9713_dai[] = {
int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
{
+ struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
+
if (try_warm && soc_ac97_ops->warm_reset) {
- soc_ac97_ops->warm_reset(codec->ac97);
+ soc_ac97_ops->warm_reset(wm9713->ac97);
if (ac97_read(codec, 0) == wm9713_reg[0])
return 1;
}
- soc_ac97_ops->reset(codec->ac97);
+ soc_ac97_ops->reset(wm9713->ac97);
if (soc_ac97_ops->warm_reset)
- soc_ac97_ops->warm_reset(codec->ac97);
- if (ac97_read(codec, 0) != wm9713_reg[0])
+ soc_ac97_ops->warm_reset(wm9713->ac97);
+ if (ac97_read(codec, 0) != wm9713_reg[0]) {
+ dev_err(codec->dev, "Failed to reset: AC97 link error\n");
return -EIO;
+ }
+
return 0;
}
EXPORT_SYMBOL_GPL(wm9713_reset);
@@ -1163,10 +1198,8 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec)
u16 *cache = codec->reg_cache;
ret = wm9713_reset(codec, 1);
- if (ret < 0) {
- printk(KERN_ERR "could not reset AC97 codec\n");
+ if (ret < 0)
return ret;
- }
wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1180,7 +1213,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec)
if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID ||
i == AC97_EXTENDED_MSTATUS || i > 0x66)
continue;
- soc_ac97_ops->write(codec->ac97, i, cache[i>>1]);
+ soc_ac97_ops->write(wm9713->ac97, i, cache[i>>1]);
}
}
@@ -1189,50 +1222,36 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec)
static int wm9713_soc_probe(struct snd_soc_codec *codec)
{
- struct wm9713_priv *wm9713;
+ struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
int ret = 0, reg;
- wm9713 = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL);
- if (wm9713 == NULL)
- return -ENOMEM;
- snd_soc_codec_set_drvdata(codec, wm9713);
-
- ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0);
- if (ret < 0)
- goto codec_err;
+ wm9713->ac97 = snd_soc_new_ac97_codec(codec);
+ if (IS_ERR(wm9713->ac97))
+ return PTR_ERR(wm9713->ac97);
/* do a cold reset for the controller and then try
* a warm reset followed by an optional cold reset for codec */
wm9713_reset(codec, 0);
ret = wm9713_reset(codec, 1);
- if (ret < 0) {
- printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n");
+ if (ret < 0)
goto reset_err;
- }
-
- wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* unmute the adc - move to kcontrol */
reg = ac97_read(codec, AC97_CD) & 0x7fff;
ac97_write(codec, AC97_CD, reg);
- snd_soc_add_codec_controls(codec, wm9713_snd_ac97_controls,
- ARRAY_SIZE(wm9713_snd_ac97_controls));
-
return 0;
reset_err:
- snd_soc_free_ac97_codec(codec);
-codec_err:
- kfree(wm9713);
+ snd_soc_free_ac97_codec(wm9713->ac97);
return ret;
}
static int wm9713_soc_remove(struct snd_soc_codec *codec)
{
struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
- snd_soc_free_ac97_codec(codec);
- kfree(wm9713);
+
+ snd_soc_free_ac97_codec(wm9713->ac97);
return 0;
}
@@ -1248,6 +1267,9 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = {
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = wm9713_reg,
+
+ .controls = wm9713_snd_ac97_controls,
+ .num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls),
.dapm_widgets = wm9713_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm9713_dapm_widgets),
.dapm_routes = wm9713_audio_map,
@@ -1256,6 +1278,16 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = {
static int wm9713_probe(struct platform_device *pdev)
{
+ struct wm9713_priv *wm9713;
+
+ wm9713 = devm_kzalloc(&pdev->dev, sizeof(*wm9713), GFP_KERNEL);
+ if (wm9713 == NULL)
+ return -ENOMEM;
+
+ mutex_init(&wm9713->lock);
+
+ platform_set_drvdata(pdev, wm9713);
+
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_wm9713, wm9713_dai, ARRAY_SIZE(wm9713_dai));
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 67124783558a..720d6e852986 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -21,6 +21,7 @@
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
+#include <linux/vmalloc.h>
#include <linux/workqueue.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -169,11 +170,12 @@ static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len,
if (buf == NULL)
return NULL;
- buf->buf = kmemdup(src, len, GFP_KERNEL | GFP_DMA);
+ buf->buf = vmalloc(len);
if (!buf->buf) {
- kfree(buf);
+ vfree(buf);
return NULL;
}
+ memcpy(buf->buf, src, len);
if (list)
list_add_tail(&buf->list, list);
@@ -188,7 +190,7 @@ static void wm_adsp_buf_free(struct list_head *list)
struct wm_adsp_buf,
list);
list_del(&buf->list);
- kfree(buf->buf);
+ vfree(buf->buf);
kfree(buf);
}
}
@@ -684,38 +686,24 @@ static int wm_adsp_load(struct wm_adsp *dsp)
}
if (reg) {
- size_t to_write = PAGE_SIZE;
- size_t remain = le32_to_cpu(region->len);
- const u8 *data = region->data;
-
- while (remain > 0) {
- if (remain < PAGE_SIZE)
- to_write = remain;
-
- buf = wm_adsp_buf_alloc(data,
- to_write,
- &buf_list);
- if (!buf) {
- adsp_err(dsp, "Out of memory\n");
- ret = -ENOMEM;
- goto out_fw;
- }
-
- ret = regmap_raw_write_async(regmap, reg,
- buf->buf,
- to_write);
- if (ret != 0) {
- adsp_err(dsp,
- "%s.%d: Failed to write %zd bytes at %d in %s: %d\n",
- file, regions,
- to_write, offset,
- region_name, ret);
- goto out_fw;
- }
+ buf = wm_adsp_buf_alloc(region->data,
+ le32_to_cpu(region->len),
+ &buf_list);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ ret = -ENOMEM;
+ goto out_fw;
+ }
- data += to_write;
- reg += to_write / 2;
- remain -= to_write;
+ ret = regmap_raw_write_async(regmap, reg, buf->buf,
+ le32_to_cpu(region->len));
+ if (ret != 0) {
+ adsp_err(dsp,
+ "%s.%d: Failed to write %d bytes at %d in %s: %d\n",
+ file, regions,
+ le32_to_cpu(region->len), offset,
+ region_name, ret);
+ goto out_fw;
}
}
@@ -1065,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
be32_to_cpu(adsp1_alg[i].zm));
region = kzalloc(sizeof(*region), GFP_KERNEL);
- if (!region)
- return -ENOMEM;
+ if (!region) {
+ ret = -ENOMEM;
+ goto out;
+ }
region->type = WMFW_ADSP1_DM;
region->alg = be32_to_cpu(adsp1_alg[i].alg.id);
region->base = be32_to_cpu(adsp1_alg[i].dm);
@@ -1083,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
}
region = kzalloc(sizeof(*region), GFP_KERNEL);
- if (!region)
- return -ENOMEM;
+ if (!region) {
+ ret = -ENOMEM;
+ goto out;
+ }
region->type = WMFW_ADSP1_ZM;
region->alg = be32_to_cpu(adsp1_alg[i].alg.id);
region->base = be32_to_cpu(adsp1_alg[i].zm);
@@ -1113,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
be32_to_cpu(adsp2_alg[i].zm));
region = kzalloc(sizeof(*region), GFP_KERNEL);
- if (!region)
- return -ENOMEM;
+ if (!region) {
+ ret = -ENOMEM;
+ goto out;
+ }
region->type = WMFW_ADSP2_XM;
region->alg = be32_to_cpu(adsp2_alg[i].alg.id);
region->base = be32_to_cpu(adsp2_alg[i].xm);
@@ -1131,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
}
region = kzalloc(sizeof(*region), GFP_KERNEL);
- if (!region)
- return -ENOMEM;
+ if (!region) {
+ ret = -ENOMEM;
+ goto out;
+ }
region->type = WMFW_ADSP2_YM;
region->alg = be32_to_cpu(adsp2_alg[i].alg.id);
region->base = be32_to_cpu(adsp2_alg[i].ym);
@@ -1149,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
}
region = kzalloc(sizeof(*region), GFP_KERNEL);
- if (!region)
- return -ENOMEM;
+ if (!region) {
+ ret = -ENOMEM;
+ goto out;
+ }
region->type = WMFW_ADSP2_ZM;
region->alg = be32_to_cpu(adsp2_alg[i].alg.id);
region->base = be32_to_cpu(adsp2_alg[i].zm);
@@ -1595,13 +1593,6 @@ static void wm_adsp2_boot_work(struct work_struct *work)
if (ret != 0)
goto err;
- ret = regmap_update_bits_async(dsp->regmap,
- dsp->base + ADSP2_CONTROL,
- ADSP2_CORE_ENA,
- ADSP2_CORE_ENA);
- if (ret != 0)
- goto err;
-
dsp->running = true;
return;
@@ -1651,8 +1642,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
ret = regmap_update_bits(dsp->regmap,
dsp->base + ADSP2_CONTROL,
- ADSP2_START,
- ADSP2_START);
+ ADSP2_CORE_ENA | ADSP2_START,
+ ADSP2_CORE_ENA | ADSP2_START);
if (ret != 0)
goto err;
break;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 0eed9b1b24e1..0dab382ba147 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -70,6 +70,7 @@ struct davinci_mcasp {
void __iomem *base;
u32 fifo_base;
struct device *dev;
+ struct snd_pcm_substream *substreams[2];
/* McASP specific data */
int tdm_slots;
@@ -80,6 +81,7 @@ struct davinci_mcasp {
u8 bclk_div;
u16 bclk_lrclk_ratio;
int streams;
+ u32 irq_request[2];
int sysclk_freq;
bool bclk_master;
@@ -90,6 +92,9 @@ struct davinci_mcasp {
bool dat_port;
+ /* Used for comstraint setting on the second stream */
+ u32 channels;
+
#ifdef CONFIG_PM_SLEEP
struct davinci_mcasp_context context;
#endif
@@ -154,9 +159,16 @@ static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp)
static void mcasp_start_rx(struct davinci_mcasp *mcasp)
{
+ if (mcasp->rxnumevt) { /* enable FIFO */
+ u32 reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+
+ mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+ mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
+ }
+
+ /* Start clocks */
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
-
/*
* When ASYNC == 0 the transmit and receive sections operate
* synchronously from the transmit clock and frame sync. We need to make
@@ -167,74 +179,69 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
}
+ /* Activate serializer(s) */
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0);
-
- mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
- mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0);
-
+ /* Release RX state machine */
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+ /* Release Frame Sync generator */
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
-
if (mcasp_is_synchronous(mcasp))
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+
+ /* enable receive IRQs */
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG,
+ mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]);
}
static void mcasp_start_tx(struct davinci_mcasp *mcasp)
{
- u8 offset = 0, i;
u32 cnt;
+ if (mcasp->txnumevt) { /* enable FIFO */
+ u32 reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+
+ mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+ mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
+ }
+
+ /* Start clocks */
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+ /* Activate serializer(s) */
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0);
- mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
- mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0);
- for (i = 0; i < mcasp->num_serializer; i++) {
- if (mcasp->serial_dir[i] == TX_MODE) {
- offset = i;
- break;
- }
- }
-
- /* wait for TX ready */
+ /* wait for XDATA to be cleared */
cnt = 0;
- while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(offset)) &
- TXSTATE) && (cnt < 100000))
+ while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) &
+ ~XRDATA) && (cnt < 100000))
cnt++;
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0);
+ /* Release TX state machine */
+ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
+ /* Release Frame Sync generator */
+ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+
+ /* enable transmit IRQs */
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG,
+ mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]);
}
static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
{
- u32 reg;
-
mcasp->streams++;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (mcasp->txnumevt) { /* enable FIFO */
- reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
- mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
- mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
- }
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
mcasp_start_tx(mcasp);
- } else {
- if (mcasp->rxnumevt) { /* enable FIFO */
- reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
- mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
- mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
- }
+ else
mcasp_start_rx(mcasp);
- }
}
static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
{
+ /* disable IRQ sources */
+ mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG,
+ mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]);
+
/*
* In synchronous mode stop the TX clocks if no other stream is
* running
@@ -244,12 +251,22 @@ static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0);
mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+
+ if (mcasp->rxnumevt) { /* disable FIFO */
+ u32 reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+
+ mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+ }
}
static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
{
u32 val = 0;
+ /* disable IRQ sources */
+ mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG,
+ mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]);
+
/*
* In synchronous mode keep TX clocks running if the capture stream is
* still running.
@@ -259,27 +276,92 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val);
mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+
+ if (mcasp->txnumevt) { /* disable FIFO */
+ u32 reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+
+ mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+ }
}
static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
{
- u32 reg;
-
mcasp->streams--;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (mcasp->txnumevt) { /* disable FIFO */
- reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
- mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
- }
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
mcasp_stop_tx(mcasp);
- } else {
- if (mcasp->rxnumevt) { /* disable FIFO */
- reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
- mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
- }
+ else
mcasp_stop_rx(mcasp);
+}
+
+static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data)
+{
+ struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+ struct snd_pcm_substream *substream;
+ u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK];
+ u32 handled_mask = 0;
+ u32 stat;
+
+ stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG);
+ if (stat & XUNDRN & irq_mask) {
+ dev_warn(mcasp->dev, "Transmit buffer underflow\n");
+ handled_mask |= XUNDRN;
+
+ substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream) {
+ snd_pcm_stream_lock_irq(substream);
+ if (snd_pcm_running(substream))
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stream_unlock_irq(substream);
+ }
}
+
+ if (!handled_mask)
+ dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n",
+ stat);
+
+ if (stat & XRERR)
+ handled_mask |= XRERR;
+
+ /* Ack the handled event only */
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask);
+
+ return IRQ_RETVAL(handled_mask);
+}
+
+static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
+{
+ struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+ struct snd_pcm_substream *substream;
+ u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE];
+ u32 handled_mask = 0;
+ u32 stat;
+
+ stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG);
+ if (stat & ROVRN & irq_mask) {
+ dev_warn(mcasp->dev, "Receive buffer overflow\n");
+ handled_mask |= ROVRN;
+
+ substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE];
+ if (substream) {
+ snd_pcm_stream_lock_irq(substream);
+ if (snd_pcm_running(substream))
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stream_unlock_irq(substream);
+ }
+ }
+
+ if (!handled_mask)
+ dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n",
+ stat);
+
+ if (stat & XRERR)
+ handled_mask |= XRERR;
+
+ /* Ack the handled event only */
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask);
+
+ return IRQ_RETVAL(handled_mask);
}
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -500,8 +582,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
* both left and right channels), so it has to be divided by number of
* tdm-slots (for I2S - divided by 2).
*/
- if (mcasp->bclk_lrclk_ratio)
- word_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
+ if (mcasp->bclk_lrclk_ratio) {
+ u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
+
+ /*
+ * When we have more bclk then it is needed for the data, we
+ * need to use the rotation to move the received samples to have
+ * correct alignment.
+ */
+ rx_rotate = (slot_length - word_length) / 4;
+ word_length = slot_length;
+ }
/* mapping of the XSSZ bit-field as described in the datasheet */
fmt = (word_length >> 1) - 1;
@@ -635,19 +726,29 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
return 0;
}
-static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
+static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
+ int channels)
{
int i, active_slots;
+ int total_slots;
+ int active_serializers;
u32 mask = 0;
u32 busel = 0;
- if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) {
- dev_err(mcasp->dev, "tdm slot %d not supported\n",
- mcasp->tdm_slots);
- return -EINVAL;
- }
+ total_slots = mcasp->tdm_slots;
+
+ /*
+ * If more than one serializer is needed, then use them with
+ * their specified tdm_slots count. Otherwise, one serializer
+ * can cope with the transaction using as many slots as channels
+ * in the stream, requires channels symmetry
+ */
+ active_serializers = (channels + total_slots - 1) / total_slots;
+ if (active_serializers == 1)
+ active_slots = channels;
+ else
+ active_slots = total_slots;
- active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots;
for (i = 0; i < active_slots; i++)
mask |= (1 << i);
@@ -659,12 +760,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
+ FSXMOD(total_slots), FSXMOD(0x1FF));
mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
+ FSRMOD(total_slots), FSRMOD(0x1FF));
return 0;
}
@@ -778,7 +879,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
ret = mcasp_dit_hw_param(mcasp, params_rate(params));
else
- ret = mcasp_i2s_hw_param(mcasp, substream->stream);
+ ret = mcasp_i2s_hw_param(mcasp, substream->stream,
+ channels);
if (ret)
return ret;
@@ -826,6 +928,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
davinci_config_channel_size(mcasp, word_length);
+ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
+ mcasp->channels = channels;
+
return 0;
}
@@ -854,7 +959,65 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
return ret;
}
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 max_channels = 0;
+ int i, dir;
+
+ mcasp->substreams[substream->stream] = substream;
+
+ if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+ return 0;
+
+ /*
+ * Limit the maximum allowed channels for the first stream:
+ * number of serializers for the direction * tdm slots per serializer
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = TX_MODE;
+ else
+ dir = RX_MODE;
+
+ for (i = 0; i < mcasp->num_serializer; i++) {
+ if (mcasp->serial_dir[i] == dir)
+ max_channels++;
+ }
+ max_channels *= mcasp->tdm_slots;
+ /*
+ * If the already active stream has less channels than the calculated
+ * limnit based on the seirializers * tdm_slots, we need to use that as
+ * a constraint for the second stream.
+ * Otherwise (first stream or less allowed channels) we use the
+ * calculated constraint.
+ */
+ if (mcasp->channels && mcasp->channels < max_channels)
+ max_channels = mcasp->channels;
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ 2, max_channels);
+ return 0;
+}
+
+static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+
+ mcasp->substreams[substream->stream] = NULL;
+
+ if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+ return;
+
+ if (!cpu_dai->active)
+ mcasp->channels = 0;
+}
+
static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+ .startup = davinci_mcasp_startup,
+ .shutdown = davinci_mcasp_shutdown,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
@@ -971,6 +1134,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
},
.ops = &davinci_mcasp_dai_ops,
+ .symmetric_samplebits = 1,
},
{
.name = "davinci-mcasp.1",
@@ -1194,6 +1358,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
struct resource *mem, *ioarea, *res, *dat;
struct davinci_mcasp_pdata *pdata;
struct davinci_mcasp *mcasp;
+ char *irq_name;
+ int irq;
int ret;
if (!pdev->dev.platform_data && !pdev->dev.of_node) {
@@ -1235,6 +1401,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
ret = pm_runtime_get_sync(&pdev->dev);
if (IS_ERR_VALUE(ret)) {
dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n");
+ pm_runtime_disable(&pdev->dev);
return ret;
}
@@ -1246,7 +1413,21 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
}
mcasp->op_mode = pdata->op_mode;
- mcasp->tdm_slots = pdata->tdm_slots;
+ /* sanity check for tdm slots parameter */
+ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) {
+ if (pdata->tdm_slots < 2) {
+ dev_err(&pdev->dev, "invalid tdm slots: %d\n",
+ pdata->tdm_slots);
+ mcasp->tdm_slots = 2;
+ } else if (pdata->tdm_slots > 32) {
+ dev_err(&pdev->dev, "invalid tdm slots: %d\n",
+ pdata->tdm_slots);
+ mcasp->tdm_slots = 32;
+ } else {
+ mcasp->tdm_slots = pdata->tdm_slots;
+ }
+ }
+
mcasp->num_serializer = pdata->num_serializer;
#ifdef CONFIG_PM_SLEEP
mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev,
@@ -1260,6 +1441,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
mcasp->dev = &pdev->dev;
+ irq = platform_get_irq_byname(pdev, "rx");
+ if (irq >= 0) {
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
+ dev_name(&pdev->dev));
+ ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+ davinci_mcasp_rx_irq_handler,
+ IRQF_ONESHOT, irq_name, mcasp);
+ if (ret) {
+ dev_err(&pdev->dev, "RX IRQ request failed\n");
+ goto err;
+ }
+
+ mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN;
+ }
+
+ irq = platform_get_irq_byname(pdev, "tx");
+ if (irq >= 0) {
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
+ dev_name(&pdev->dev));
+ ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+ davinci_mcasp_tx_irq_handler,
+ IRQF_ONESHOT, irq_name, mcasp);
+ if (ret) {
+ dev_err(&pdev->dev, "TX IRQ request failed\n");
+ goto err;
+ }
+
+ mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN;
+ }
+
dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
if (dat)
mcasp->dat_port = true;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 98fbc451892a..79dc511180bf 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -253,6 +253,13 @@
#define TXFSRST BIT(12) /* Frame Sync Generator Reset */
/*
+ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits
+ * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits
+ */
+#define XRERR BIT(8) /* Transmit/Receive error */
+#define XRDATA BIT(5) /* Transmit/Receive data ready */
+
+/*
* DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits
*/
#define MUTENA(val) (val)
@@ -279,6 +286,16 @@
#define TXDATADMADIS BIT(0)
/*
+ * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits
+ */
+#define ROVRN BIT(0)
+
+/*
+ * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits
+ */
+#define XUNDRN BIT(0)
+
+/*
* DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
*/
#define FIFO_ENABLE BIT(16)
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index e961388e6e9c..08f0229f8d68 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -338,31 +338,34 @@ static int dw_i2s_probe(struct platform_device *pdev)
return -EINVAL;
}
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- dev_err(&pdev->dev, "no i2s resource defined\n");
- return -ENODEV;
- }
-
- if (!devm_request_mem_region(&pdev->dev, res->start,
- resource_size(res), pdev->name)) {
- dev_err(&pdev->dev, "i2s region already claimed\n");
- return -EBUSY;
- }
-
dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
if (!dev) {
dev_warn(&pdev->dev, "kzalloc fail\n");
return -ENOMEM;
}
- dev->i2s_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!dev->i2s_base) {
- dev_err(&pdev->dev, "ioremap fail for i2s_region\n");
+ dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
+ if (!dw_i2s_dai) {
+ dev_err(&pdev->dev, "mem allocation failed for dai driver\n");
return -ENOMEM;
}
+ dw_i2s_dai->ops = &dw_i2s_dai_ops;
+ dw_i2s_dai->suspend = dw_i2s_suspend;
+ dw_i2s_dai->resume = dw_i2s_resume;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ dev_err(&pdev->dev, "no i2s resource defined\n");
+ return -ENODEV;
+ }
+
+ dev->i2s_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(dev->i2s_base)) {
+ dev_err(&pdev->dev, "ioremap fail for i2s_region\n");
+ return PTR_ERR(dev->i2s_base);
+ }
+
cap = pdata->cap;
dev->capability = cap;
dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
@@ -388,13 +391,6 @@ static int dw_i2s_probe(struct platform_device *pdev)
if (ret < 0)
goto err_clk_put;
- dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
- if (!dw_i2s_dai) {
- dev_err(&pdev->dev, "mem allocation failed for dai driver\n");
- ret = -ENOMEM;
- goto err_clk_disable;
- }
-
if (cap & DWC_I2S_PLAY) {
dev_dbg(&pdev->dev, " designware: play supported\n");
dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
@@ -411,10 +407,6 @@ static int dw_i2s_probe(struct platform_device *pdev)
dw_i2s_dai->capture.rates = pdata->snd_rates;
}
- dw_i2s_dai->ops = &dw_i2s_dai_ops;
- dw_i2s_dai->suspend = dw_i2s_suspend;
- dw_i2s_dai->resume = dw_i2s_resume;
-
dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component,
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index eb093d5b85c4..b175b0145a42 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -105,7 +105,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
int ret;
int int_port = 0, ext_port;
struct device_node *np = pdev->dev.of_node;
- struct device_node *ssi_np, *codec_np;
+ struct device_node *ssi_np = NULL, *codec_np = NULL;
eukrea_tlv320.dev = &pdev->dev;
if (np) {
@@ -217,8 +217,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
err:
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
- if (np)
- of_node_put(ssi_np);
+ of_node_put(ssi_np);
return ret;
}
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 007c772f3cef..3f6959c8e2f7 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -51,6 +51,7 @@ struct codec_priv {
* @sysclk_freq[2]: SYSCLK rates for set_sysclk()
* @sysclk_dir[2]: SYSCLK directions for set_sysclk()
* @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ * @slot_width: Slot width of each frame
*
* Note: [1] for tx and [0] for rx
*/
@@ -58,6 +59,7 @@ struct cpu_priv {
unsigned long sysclk_freq[2];
u32 sysclk_dir[2];
u32 sysclk_id[2];
+ u32 slot_width;
};
/**
@@ -125,7 +127,12 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
priv->sample_rate = params_rate(params);
priv->sample_format = params_format(params);
- if (priv->card.set_bias_level)
+ /*
+ * If codec-dai is DAI Master and all configurations are already in the
+ * set_bias_level(), bypass the remaining settings in hw_params().
+ * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
+ */
+ if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
return 0;
/* Specific configurations of DAIs starts from here */
@@ -137,6 +144,15 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+ if (cpu_priv->slot_width) {
+ ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
+ cpu_priv->slot_width);
+ if (ret) {
+ dev_err(dev, "failed to set TDM slot for cpu dai\n");
+ return ret;
+ }
+ }
+
return 0;
}
@@ -448,6 +464,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+ priv->cpu_priv.slot_width = 32;
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index a609aafc994d..b2b108805b24 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -151,14 +151,7 @@ static const struct snd_pcm_hardware fsl_dma_hardware = {
*/
static void fsl_dma_abort_stream(struct snd_pcm_substream *substream)
{
- unsigned long flags;
-
- snd_pcm_stream_lock_irqsave(substream, flags);
-
- if (snd_pcm_running(substream))
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-
- snd_pcm_stream_unlock_irqrestore(substream, flags);
+ snd_pcm_stop_xrun(substream);
}
/**
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a645e296199e..ca319d59f843 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -513,10 +513,15 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
u32 width = snd_pcm_format_width(params_format(params));
u32 channels = params_channels(params);
u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+ u32 slot_width = width;
u32 bclk, mask, val;
int ret;
- bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
+ /* Override slot_width if being specifially set */
+ if (esai_priv->slot_width)
+ slot_width = esai_priv->slot_width;
+
+ bclk = params_rate(params) * slot_width * esai_priv->slots;
ret = fsl_esai_set_bclk(dai, tx, bclk);
if (ret)
@@ -538,7 +543,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0);
- val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
+ val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
@@ -780,9 +785,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
return ret;
}
- /* Set a default slot size */
- esai_priv->slot_width = 32;
-
/* Set a default slot number */
esai_priv->slots = 2;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index e6955170dc42..b6b0d25f6ace 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -67,8 +67,6 @@
/**
* FSLSSI_I2S_FORMATS: audio formats supported by the SSI
*
- * This driver currently only supports the SSI running in I2S slave mode.
- *
* The SSI has a limitation in that the samples must be in the same byte
* order as the host CPU. This is because when multiple bytes are written
* to the STX register, the bytes and bits must be written in the same
@@ -1099,7 +1097,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = {
};
static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
- .ac97_control = 1,
+ .bus_control = true,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
@@ -1363,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
return PTR_ERR(ssi_private->regs);
}
- ssi_private->irq = irq_of_parse_and_map(np, 0);
+ ssi_private->irq = platform_get_irq(pdev, 0);
if (!ssi_private->irq) {
dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
return -ENXIO;
@@ -1389,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
if (ssi_private->soc->imx) {
ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem);
if (ret)
- goto error_irqmap;
+ return ret;
}
ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
@@ -1412,7 +1410,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev);
if (ret)
- goto error_asoc_register;
+ goto error_irq;
/*
* If codec-handle property is missing from SSI node, we assume
@@ -1460,10 +1458,6 @@ error_asoc_register:
if (ssi_private->soc->imx)
fsl_ssi_imx_clean(pdev, ssi_private);
-error_irqmap:
- if (ssi_private->use_dma)
- irq_dispose_mapping(ssi_private->irq);
-
return ret;
}
@@ -1480,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev)
if (ssi_private->soc->imx)
fsl_ssi_imx_clean(pdev, ssi_private);
- if (ssi_private->use_dma)
- irq_dispose_mapping(ssi_private->irq);
-
return 0;
}
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 1cb22dd034eb..1dab963a59f7 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -175,10 +175,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
fail:
if (data && !IS_ERR(data->codec_clk))
clk_put(data->codec_clk);
- if (ssi_np)
- of_node_put(ssi_np);
- if (codec_np)
- of_node_put(codec_np);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index e1dc40143600..0c9068ebe1e7 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -74,8 +74,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, data);
end:
- if (spdif_np)
- of_node_put(spdif_np);
+ of_node_put(spdif_np);
return ret;
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index ab2fdd76b693..60b0a5b1f1f1 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -382,7 +382,7 @@ static struct snd_soc_dai_driver imx_ssi_dai = {
static struct snd_soc_dai_driver imx_ac97_dai = {
.probe = imx_ssi_dai_probe,
- .ac97_control = 1,
+ .bus_control = true,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 3a3d17ce6ba4..48179ffe1543 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -281,10 +281,8 @@ static int imx_wm8962_probe(struct platform_device *pdev)
clk_fail:
clk_disable_unprepare(data->codec_clk);
fail:
- if (ssi_np)
- of_node_put(ssi_np);
- if (codec_np)
- of_node_put(codec_np);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index f2b5d756b1f3..0b82e209b6e3 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -327,9 +327,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
goto capture_alloc_err;
}
- if (rtd->codec->ac97)
- rtd->codec->ac97->private_data = psc_dma;
-
return 0;
capture_alloc_err:
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 24eafa2cfbf4..c6ed6ba965a9 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -237,7 +237,7 @@ static const struct snd_soc_dai_ops psc_ac97_digital_ops = {
static struct snd_soc_dai_driver psc_ac97_dai[] = {
{
.name = "mpc5200-psc-ac97.0",
- .ac97_control = 1,
+ .bus_control = true,
.probe = psc_ac97_probe,
.playback = {
.stream_name = "AC97 Playback",
@@ -257,7 +257,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = {
},
{
.name = "mpc5200-psc-ac97.1",
- .ac97_control = 1,
+ .bus_control = true,
.playback = {
.stream_name = "AC97 SPDIF",
.channels_min = 1,
@@ -282,7 +282,6 @@ static const struct snd_soc_component_driver psc_ac97_component = {
static int psc_ac97_of_probe(struct platform_device *op)
{
int rc;
- struct snd_ac97 ac97;
struct mpc52xx_psc __iomem *regs;
rc = mpc5200_audio_dma_create(op);
@@ -304,7 +303,6 @@ static int psc_ac97_of_probe(struct platform_device *op)
psc_dma = dev_get_drvdata(&op->dev);
regs = psc_dma->psc_regs;
- ac97.private_data = psc_dma;
psc_dma->imr = 0;
out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index d1b7293c133e..ece22d55ba82 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -29,7 +29,9 @@ struct simple_card_data {
} *dai_props;
unsigned int mclk_fs;
int gpio_hp_det;
+ int gpio_hp_det_invert;
int gpio_mic_det;
+ int gpio_mic_det_invert;
struct snd_soc_dai_link dai_link[]; /* dynamically allocated */
};
@@ -148,6 +150,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
simple_card_hp_jack_pins);
simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det;
+ simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert;
snd_soc_jack_add_gpios(&simple_card_hp_jack, 1,
&simple_card_hp_jack_gpio);
}
@@ -159,6 +162,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
ARRAY_SIZE(simple_card_mic_jack_pins),
simple_card_mic_jack_pins);
simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det;
+ simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert;
snd_soc_jack_add_gpios(&simple_card_mic_jack, 1,
&simple_card_mic_jack_gpio);
}
@@ -226,6 +230,52 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
return 0;
}
+static int asoc_simple_card_parse_daifmt(struct device_node *node,
+ struct simple_card_data *priv,
+ struct device_node *codec,
+ char *prefix, int idx)
+{
+ struct device *dev = simple_priv_to_dev(priv);
+ struct device_node *bitclkmaster = NULL;
+ struct device_node *framemaster = NULL;
+ struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
+ struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai;
+ struct asoc_simple_dai *codec_dai = &dai_props->codec_dai;
+ unsigned int daifmt;
+
+ daifmt = snd_soc_of_parse_daifmt(node, prefix,
+ &bitclkmaster, &framemaster);
+ daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+ if (strlen(prefix) && !bitclkmaster && !framemaster) {
+ /*
+ * No dai-link level and master setting was not found from
+ * sound node level, revert back to legacy DT parsing and
+ * take the settings from codec node.
+ */
+ dev_dbg(dev, "Revert to legacy daifmt parsing\n");
+
+ cpu_dai->fmt = codec_dai->fmt =
+ snd_soc_of_parse_daifmt(codec, NULL, NULL, NULL) |
+ (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK);
+ } else {
+ if (codec == bitclkmaster)
+ daifmt |= (codec == framemaster) ?
+ SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
+ else
+ daifmt |= (codec == framemaster) ?
+ SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
+
+ cpu_dai->fmt = daifmt;
+ codec_dai->fmt = daifmt;
+ }
+
+ of_node_put(bitclkmaster);
+ of_node_put(framemaster);
+
+ return 0;
+}
+
static int asoc_simple_card_dai_link_of(struct device_node *node,
struct simple_card_data *priv,
int idx,
@@ -234,10 +284,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
struct device *dev = simple_priv_to_dev(priv);
struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx);
struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
- struct device_node *np = NULL;
- struct device_node *bitclkmaster = NULL;
- struct device_node *framemaster = NULL;
- unsigned int daifmt;
+ struct device_node *cpu = NULL;
+ struct device_node *codec = NULL;
char *name;
char prop[128];
char *prefix = "";
@@ -247,85 +295,36 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
if (is_top_level_node)
prefix = "simple-audio-card,";
- daifmt = snd_soc_of_parse_daifmt(node, prefix,
- &bitclkmaster, &framemaster);
- daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
-
snprintf(prop, sizeof(prop), "%scpu", prefix);
- np = of_get_child_by_name(node, prop);
- if (!np) {
+ cpu = of_get_child_by_name(node, prop);
+
+ snprintf(prop, sizeof(prop), "%scodec", prefix);
+ codec = of_get_child_by_name(node, prop);
+
+ if (!cpu || !codec) {
ret = -EINVAL;
dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
goto dai_link_of_err;
}
- ret = asoc_simple_card_sub_parse_of(np, &dai_props->cpu_dai,
+ ret = asoc_simple_card_parse_daifmt(node, priv,
+ codec, prefix, idx);
+ if (ret < 0)
+ goto dai_link_of_err;
+
+ ret = asoc_simple_card_sub_parse_of(cpu, &dai_props->cpu_dai,
&dai_link->cpu_of_node,
&dai_link->cpu_dai_name,
&cpu_args);
if (ret < 0)
goto dai_link_of_err;
- dai_props->cpu_dai.fmt = daifmt;
- switch (((np == bitclkmaster) << 4) | (np == framemaster)) {
- case 0x11:
- dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBS_CFS;
- break;
- case 0x10:
- dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBS_CFM;
- break;
- case 0x01:
- dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBM_CFS;
- break;
- default:
- dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBM_CFM;
- break;
- }
-
- of_node_put(np);
- snprintf(prop, sizeof(prop), "%scodec", prefix);
- np = of_get_child_by_name(node, prop);
- if (!np) {
- ret = -EINVAL;
- dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
- goto dai_link_of_err;
- }
-
- ret = asoc_simple_card_sub_parse_of(np, &dai_props->codec_dai,
+ ret = asoc_simple_card_sub_parse_of(codec, &dai_props->codec_dai,
&dai_link->codec_of_node,
&dai_link->codec_dai_name, NULL);
if (ret < 0)
goto dai_link_of_err;
- if (strlen(prefix) && !bitclkmaster && !framemaster) {
- /*
- * No DAI link level and master setting was found
- * from sound node level, revert back to legacy DT
- * parsing and take the settings from codec node.
- */
- dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n",
- __func__);
- dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt =
- snd_soc_of_parse_daifmt(np, NULL, NULL, NULL) |
- (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK);
- } else {
- dai_props->codec_dai.fmt = daifmt;
- switch (((np == bitclkmaster) << 4) | (np == framemaster)) {
- case 0x11:
- dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBM_CFM;
- break;
- case 0x10:
- dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBM_CFS;
- break;
- case 0x01:
- dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBS_CFM;
- break;
- default:
- dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBS_CFS;
- break;
- }
- }
-
if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) {
ret = -EINVAL;
goto dai_link_of_err;
@@ -368,12 +367,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
dai_link->cpu_dai_name = NULL;
dai_link_of_err:
- if (np)
- of_node_put(np);
- if (bitclkmaster)
- of_node_put(bitclkmaster);
- if (framemaster)
- of_node_put(framemaster);
+ of_node_put(cpu);
+ of_node_put(codec);
+
return ret;
}
@@ -381,6 +377,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
struct simple_card_data *priv)
{
struct device *dev = simple_priv_to_dev(priv);
+ enum of_gpio_flags flags;
u32 val;
int ret;
@@ -436,13 +433,15 @@ static int asoc_simple_card_parse_of(struct device_node *node,
return ret;
}
- priv->gpio_hp_det = of_get_named_gpio(node,
- "simple-audio-card,hp-det-gpio", 0);
+ priv->gpio_hp_det = of_get_named_gpio_flags(node,
+ "simple-audio-card,hp-det-gpio", 0, &flags);
+ priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW);
if (priv->gpio_hp_det == -EPROBE_DEFER)
return -EPROBE_DEFER;
- priv->gpio_mic_det = of_get_named_gpio(node,
- "simple-audio-card,mic-det-gpio", 0);
+ priv->gpio_mic_det = of_get_named_gpio_flags(node,
+ "simple-audio-card,mic-det-gpio", 0, &flags);
+ priv->gpio_mic_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW);
if (priv->gpio_mic_det == -EPROBE_DEFER)
return -EPROBE_DEFER;
@@ -457,18 +456,13 @@ static int asoc_simple_card_unref(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
struct snd_soc_dai_link *dai_link;
- struct device_node *np;
int num_links;
for (num_links = 0, dai_link = card->dai_link;
num_links < card->num_links;
num_links++, dai_link++) {
- np = (struct device_node *) dai_link->cpu_of_node;
- if (np)
- of_node_put(np);
- np = (struct device_node *) dai_link->codec_of_node;
- if (np)
- of_node_put(np);
+ of_node_put(dai_link->cpu_of_node);
+ of_node_put(dai_link->codec_of_node);
}
return 0;
}
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index f5b4a9c79cdf..e989ecf046c9 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -3,6 +3,7 @@ config SND_MFLD_MACHINE
depends on INTEL_SCU_IPC
select SND_SOC_SN95031
select SND_SST_MFLD_PLATFORM
+ select SND_SST_IPC_PCI
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
@@ -12,10 +13,23 @@ config SND_MFLD_MACHINE
config SND_SST_MFLD_PLATFORM
tristate
+config SND_SST_IPC
+ tristate
+
+config SND_SST_IPC_PCI
+ tristate
+ select SND_SST_IPC
+
+config SND_SST_IPC_ACPI
+ tristate
+ select SND_SST_IPC
+ depends on ACPI
+
config SND_SOC_INTEL_SST
tristate "ASoC support for Intel(R) Smart Sound Technology"
select SND_SOC_INTEL_SST_ACPI if ACPI
depends on (X86 || COMPILE_TEST)
+ depends on DW_DMAC_CORE
help
This adds support for Intel(R) Smart Sound Technology (SST).
Say Y if you have such a device
@@ -32,7 +46,8 @@ config SND_SOC_INTEL_BAYTRAIL
config SND_SOC_INTEL_HASWELL_MACH
tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
- depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\
+ I2C_DESIGNWARE_PLATFORM
select SND_SOC_INTEL_HASWELL
select SND_SOC_RT5640
help
@@ -61,7 +76,8 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
config SND_SOC_INTEL_BROADWELL_MACH
tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
- depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\
+ I2C_DESIGNWARE_PLATFORM
select SND_SOC_INTEL_HASWELL
select SND_COMPRESS_OFFLOAD
select SND_SOC_RT286
@@ -70,3 +86,27 @@ config SND_SOC_INTEL_BROADWELL_MACH
Ultrabook platforms.
Say Y if you have such a device
If unsure select "N".
+
+config SND_SOC_INTEL_BYTCR_RT5640_MACH
+ tristate "ASoC Audio DSP Support for MID BYT Platform"
+ depends on X86
+ select SND_SOC_RT5640
+ select SND_SST_MFLD_PLATFORM
+ select SND_SST_IPC_ACPI
+ help
+ This adds support for ASoC machine driver for Intel(R) MID Baytrail platform
+ used as alsa device in audio substem in Intel(R) MID devices
+ Say Y if you have such a device
+ If unsure select "N".
+
+config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
+ tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
+ depends on X86_INTEL_LPSS
+ select SND_SOC_RT5670
+ select SND_SST_MFLD_PLATFORM
+ select SND_SST_IPC_ACPI
+ help
+ This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
+ platforms with RT5672 audio codec.
+ Say Y if you have such a device
+ If unsure select "N".
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index f841786dad15..e928ec385300 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -26,8 +26,15 @@ snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-broadwell-objs := broadwell.o
+snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
+snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
+
+# DSP driver
+obj-$(CONFIG_SND_SST_IPC) += sst/
diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c
index 0e550f14028f..c256764e3c4b 100644
--- a/sound/soc/intel/broadwell.c
+++ b/sound/soc/intel/broadwell.c
@@ -19,6 +19,7 @@
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
+#include <sound/jack.h>
#include <sound/pcm_params.h>
#include "sst-dsp.h"
@@ -26,8 +27,26 @@
#include "../codecs/rt286.h"
+static struct snd_soc_jack broadwell_headset;
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin broadwell_headset_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static const struct snd_kcontrol_new broadwell_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+};
+
static const struct snd_soc_dapm_widget broadwell_widgets[] = {
- SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_MIC("DMIC1", NULL),
@@ -42,7 +61,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
{"Speaker", NULL, "SPOL"},
/* HP jack connectors - unknown if we have jack deteck */
- {"Headphones", NULL, "HPO Pin"},
+ {"Headphone Jack", NULL, "HPO Pin"},
/* other jacks */
{"MIC1", NULL, "Mic Jack"},
@@ -57,6 +76,27 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
};
+static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int ret = 0;
+ ret = snd_soc_jack_new(codec, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset);
+
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_pins(&broadwell_headset,
+ ARRAY_SIZE(broadwell_headset_pins),
+ broadwell_headset_pins);
+ if (ret)
+ return ret;
+
+ rt286_mic_detect(codec, &broadwell_headset);
+ return 0;
+}
+
+
static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
@@ -116,7 +156,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
}
/* always connected - check HP for jack detect */
- snd_soc_dapm_enable_pin(dapm, "Headphones");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
snd_soc_dapm_enable_pin(dapm, "Line Jack");
@@ -131,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
/* Front End DAI links */
{
.name = "System PCM",
- .stream_name = "System Playback",
+ .stream_name = "System Playback/Capture",
.cpu_dai_name = "System Pin",
.platform_name = "haswell-pcm-audio",
.dynamic = 1,
@@ -140,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.init = broadwell_rtd_init,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_playback = 1,
+ .dpcm_capture = 1,
},
{
.name = "Offload0",
@@ -174,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
},
- {
- .name = "Capture PCM",
- .stream_name = "Capture",
- .cpu_dai_name = "Capture Pin",
- .platform_name = "haswell-pcm-audio",
- .dynamic = 1,
- .codec_name = "snd-soc-dummy",
- .codec_dai_name = "snd-soc-dummy-dai",
- .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .dpcm_capture = 1,
- },
-
/* Back End DAI links */
{
/* SSP0 - Codec */
@@ -196,6 +225,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.no_pcm = 1,
.codec_name = "i2c-INT343A:00",
.codec_dai_name = "rt286-aif1",
+ .init = broadwell_rt286_codec_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ignore_suspend = 1,
@@ -213,6 +243,8 @@ static struct snd_soc_card broadwell_rt286 = {
.owner = THIS_MODULE,
.dai_link = broadwell_rt286_dais,
.num_links = ARRAY_SIZE(broadwell_rt286_dais),
+ .controls = broadwell_controls,
+ .num_controls = ARRAY_SIZE(broadwell_controls),
.dapm_widgets = broadwell_widgets,
.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
.dapm_routes = broadwell_rt286_map,
diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c
new file mode 100644
index 000000000000..f5d0fc1ab10c
--- /dev/null
+++ b/sound/soc/intel/bytcr_dpcm_rt5640.c
@@ -0,0 +1,230 @@
+/*
+ * byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform
+ *
+ * Copyright (C) 2014 Intel Corp
+ * Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/input.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../codecs/rt5640.h"
+#include "sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget byt_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_audio_map[] = {
+ {"IN2P", NULL, "Headset Mic"},
+ {"IN2N", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "MICBIAS1"},
+ {"IN1P", NULL, "MICBIAS1"},
+ {"LDO2", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Ext Spk", NULL, "SPOLP"},
+ {"Ext Spk", NULL, "SPOLN"},
+ {"Ext Spk", NULL, "SPORP"},
+ {"Ext Spk", NULL, "SPORN"},
+
+ {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"ssp2 Tx", NULL, "codec_out0"},
+ {"ssp2 Tx", NULL, "codec_out1"},
+ {"codec_in0", NULL, "ssp2 Rx"},
+ {"codec_in1", NULL, "ssp2 Rx"},
+ {"ssp2 Rx", NULL, "AIF1 Capture"},
+};
+
+static const struct snd_kcontrol_new byt_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int byt_aif1_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ snd_soc_dai_set_bclk_ratio(codec_dai, 50);
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+ params_rate(params) * 512,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec clock %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+ params_rate(params) * 50,
+ params_rate(params) * 512);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_pcm_stream byt_dai_params = {
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The DSP will covert the FE rate to 48k, stereo, 24bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP2 to 24-bit */
+ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S24_LE);
+ return 0;
+}
+
+static unsigned int rates_48000[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+ .count = ARRAY_SIZE(rates_48000),
+ .list = rates_48000,
+};
+
+static int byt_aif1_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_48000);
+}
+
+static struct snd_soc_ops byt_aif1_ops = {
+ .startup = byt_aif1_startup,
+};
+
+static struct snd_soc_ops byt_be_ssp2_ops = {
+ .hw_params = byt_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link byt_dailink[] = {
+ [MERR_DPCM_AUDIO] = {
+ .name = "Baytrail Audio Port",
+ .stream_name = "Baytrail Audio",
+ .cpu_dai_name = "media-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ .ignore_suspend = 1,
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &byt_aif1_ops,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Baytrail Compressed Port",
+ .stream_name = "Baytrail Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+ /* back ends */
+ {
+ .name = "SSP2-Codec",
+ .be_id = 1,
+ .cpu_dai_name = "ssp2-port",
+ .platform_name = "sst-mfld-platform",
+ .no_pcm = 1,
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .be_hw_params_fixup = byt_codec_fixup,
+ .ignore_suspend = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &byt_be_ssp2_ops,
+ },
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_byt = {
+ .name = "baytrailcraudio",
+ .dai_link = byt_dailink,
+ .num_links = ARRAY_SIZE(byt_dailink),
+ .dapm_widgets = byt_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets),
+ .dapm_routes = byt_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(byt_audio_map),
+ .controls = byt_mc_controls,
+ .num_controls = ARRAY_SIZE(byt_mc_controls),
+};
+
+static int snd_byt_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+
+ /* register the soc card */
+ snd_soc_card_byt.dev = &pdev->dev;
+
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt);
+ if (ret_val) {
+ dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, &snd_soc_card_byt);
+ return ret_val;
+}
+
+static struct platform_driver snd_byt_mc_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "bytt100_rt5640",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = snd_byt_mc_probe,
+};
+
+module_platform_driver(snd_byt_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytrt5640-audio");
diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c
new file mode 100644
index 000000000000..9b8b561171b7
--- /dev/null
+++ b/sound/soc/intel/cht_bsw_rt5672.c
@@ -0,0 +1,285 @@
+/*
+ * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
+ * Cherrytrail and Braswell, with RT5672 codec.
+ *
+ * Copyright (C) 2014 Intel Corp
+ * Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
+ * Mengdong Lin <mengdong.lin@intel.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../codecs/rt5670.h"
+#include "sst-atom-controls.h"
+
+/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
+#define CHT_PLAT_CLK_3_HZ 19200000
+#define CHT_CODEC_DAI "rt5670-aif1"
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+ int i;
+
+ for (i = 0; i < card->num_rtd; i++) {
+ struct snd_soc_pcm_runtime *rtd;
+
+ rtd = card->rtd + i;
+ if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+ strlen(CHT_CODEC_DAI)))
+ return rtd->codec_dai;
+ }
+ return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct snd_soc_dai *codec_dai;
+
+ codec_dai = cht_get_codec_dai(card);
+ if (!codec_dai) {
+ dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+ return -EIO;
+ }
+
+ if (!SND_SOC_DAPM_EVENT_OFF(event))
+ return 0;
+
+ /* Set codec sysclk source to its internal clock because codec PLL will
+ * be off when idle and MCLK will also be off by ACPI when codec is
+ * runtime suspended. Codec needs clock for jack detection and button
+ * press.
+ */
+ snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
+ 0, SND_SOC_CLOCK_IN);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+ platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+ {"IN1P", NULL, "Headset Mic"},
+ {"IN1N", NULL, "Headset Mic"},
+ {"DMIC L1", NULL, "Int Mic"},
+ {"DMIC R1", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Ext Spk", NULL, "SPOLP"},
+ {"Ext Spk", NULL, "SPOLN"},
+ {"Ext Spk", NULL, "SPORP"},
+ {"Ext Spk", NULL, "SPORN"},
+ {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"ssp2 Tx", NULL, "codec_out0"},
+ {"ssp2 Tx", NULL, "codec_out1"},
+ {"codec_in0", NULL, "ssp2 Rx"},
+ {"codec_in1", NULL, "ssp2 Rx"},
+ {"ssp2 Rx", NULL, "AIF1 Capture"},
+ {"Headphone", NULL, "Platform Clock"},
+ {"Headset Mic", NULL, "Platform Clock"},
+ {"Int Mic", NULL, "Platform Clock"},
+ {"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ /* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
+ CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ /* set codec sysclk source to PLL */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
+ params_rate(params) * 512,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+ return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_dai *codec_dai = runtime->codec_dai;
+
+ /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+ if (ret < 0) {
+ dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The DSP will covert the FE rate to 48k, stereo, 24bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP2 to 24-bit */
+ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S24_LE);
+ return 0;
+}
+
+static unsigned int rates_48000[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+ .count = ARRAY_SIZE(rates_48000),
+ .list = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+ .startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+ .hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+ /* Front End DAI links */
+ [MERR_DPCM_AUDIO] = {
+ .name = "Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "media-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ .ignore_suspend = 1,
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_aif1_ops,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Compressed Port",
+ .stream_name = "Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP2 - Codec */
+ .name = "SSP2-Codec",
+ .be_id = 1,
+ .cpu_dai_name = "ssp2-port",
+ .platform_name = "sst-mfld-platform",
+ .no_pcm = 1,
+ .codec_dai_name = "rt5670-aif1",
+ .codec_name = "i2c-10EC5670:00",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .init = cht_codec_init,
+ .be_hw_params_fixup = cht_codec_fixup,
+ .ignore_suspend = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_be_ssp2_ops,
+ },
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+ .name = "cherrytrailcraudio",
+ .dai_link = cht_dailink,
+ .num_links = ARRAY_SIZE(cht_dailink),
+ .dapm_widgets = cht_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+ .dapm_routes = cht_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+ .controls = cht_mc_controls,
+ .num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+
+ /* register the soc card */
+ snd_soc_card_cht.dev = &pdev->dev;
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+ if (ret_val) {
+ dev_err(&pdev->dev,
+ "snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, &snd_soc_card_cht);
+ return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "cht-bsw-rt5672",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5672");
diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c
index 3981982674ac..cb8a482b5f30 100644
--- a/sound/soc/intel/haswell.c
+++ b/sound/soc/intel/haswell.c
@@ -109,7 +109,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
/* Front End DAI links */
{
.name = "System",
- .stream_name = "System Playback",
+ .stream_name = "System Playback/Capture",
.cpu_dai_name = "System Pin",
.platform_name = "haswell-pcm-audio",
.dynamic = 1,
@@ -118,6 +118,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
.init = haswell_rtd_init,
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_playback = 1,
+ .dpcm_capture = 1,
},
{
.name = "Offload0",
@@ -152,17 +153,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dpcm_capture = 1,
},
- {
- .name = "Capture",
- .stream_name = "Capture",
- .cpu_dai_name = "Capture Pin",
- .platform_name = "haswell-pcm-audio",
- .dynamic = 1,
- .codec_name = "snd-soc-dummy",
- .codec_dai_name = "snd-soc-dummy-dai",
- .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
- .dpcm_capture = 1,
- },
/* Back End DAI links */
{
diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
index 7104a34181a9..90aa5c0476f3 100644
--- a/sound/soc/intel/sst-atom-controls.c
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -15,6 +15,9 @@
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
+ * In the dpcm driver modelling when a particular FE/BE/Mixer/Pipe is active
+ * we forward the settings and parameters, rest we keep the values in
+ * driver and forward when DAPM enables them
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
@@ -81,6 +84,183 @@ static int sst_fill_and_send_cmd(struct sst_data *drv,
return ret;
}
+/**
+ * tx map value is a bitfield where each bit represents a FW channel
+ *
+ * 3 2 1 0 # 0 = codec0, 1 = codec1
+ * RLRLRLRL # 3, 4 = reserved
+ *
+ * e.g. slot 0 rx map = 00001100b -> data from slot 0 goes into codec_in1 L,R
+ */
+static u8 sst_ssp_tx_map[SST_MAX_TDM_SLOTS] = {
+ 0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default rx map */
+};
+
+/**
+ * rx map value is a bitfield where each bit represents a slot
+ *
+ * 76543210 # 0 = slot 0, 1 = slot 1
+ *
+ * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot 0, 2
+ */
+static u8 sst_ssp_rx_map[SST_MAX_TDM_SLOTS] = {
+ 0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default tx map */
+};
+
+/**
+ * NOTE: this is invoked with lock held
+ */
+static int sst_send_slot_map(struct sst_data *drv)
+{
+ struct sst_param_sba_ssp_slot_map cmd;
+
+ SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+ cmd.header.command_id = SBA_SET_SSP_SLOT_MAP;
+ cmd.header.length = sizeof(struct sst_param_sba_ssp_slot_map)
+ - sizeof(struct sst_dsp_header);
+
+ cmd.param_id = SBA_SET_SSP_SLOT_MAP;
+ cmd.param_len = sizeof(cmd.rx_slot_map) + sizeof(cmd.tx_slot_map)
+ + sizeof(cmd.ssp_index);
+ cmd.ssp_index = SSP_CODEC;
+
+ memcpy(cmd.rx_slot_map, &sst_ssp_tx_map[0], sizeof(cmd.rx_slot_map));
+ memcpy(cmd.tx_slot_map, &sst_ssp_rx_map[0], sizeof(cmd.tx_slot_map));
+
+ return sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS,
+ SST_FLAG_BLOCKED, SST_TASK_SBA, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+}
+
+int sst_slot_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct sst_enum *e = (struct sst_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = e->max;
+
+ if (uinfo->value.enumerated.item > e->max - 1)
+ uinfo->value.enumerated.item = e->max - 1;
+ strcpy(uinfo->value.enumerated.name,
+ e->texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+/**
+ * sst_slot_get - get the status of the interleaver/deinterleaver control
+ *
+ * Searches the map where the control status is stored, and gets the
+ * channel/slot which is currently set for this enumerated control. Since it is
+ * an enumerated control, there is only one possible value.
+ */
+static int sst_slot_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct sst_enum *e = (void *)kcontrol->private_value;
+ struct snd_soc_component *c = snd_kcontrol_chip(kcontrol);
+ struct sst_data *drv = snd_soc_component_get_drvdata(c);
+ unsigned int ctl_no = e->reg;
+ unsigned int is_tx = e->tx;
+ unsigned int val, mux;
+ u8 *map = is_tx ? sst_ssp_rx_map : sst_ssp_tx_map;
+
+ mutex_lock(&drv->lock);
+ val = 1 << ctl_no;
+ /* search which slot/channel has this bit set - there should be only one */
+ for (mux = e->max; mux > 0; mux--)
+ if (map[mux - 1] & val)
+ break;
+
+ ucontrol->value.enumerated.item[0] = mux;
+ mutex_unlock(&drv->lock);
+
+ dev_dbg(c->dev, "%s - %s map = %#x\n",
+ is_tx ? "tx channel" : "rx slot",
+ e->texts[mux], mux ? map[mux - 1] : -1);
+ return 0;
+}
+
+/* sst_check_and_send_slot_map - helper for checking power state and sending
+ * slot map cmd
+ *
+ * called with lock held
+ */
+static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol *kcontrol)
+{
+ struct sst_enum *e = (void *)kcontrol->private_value;
+ int ret = 0;
+
+ if (e->w && e->w->power)
+ ret = sst_send_slot_map(drv);
+ else
+ dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n",
+ kcontrol->id.name);
+ return ret;
+}
+
+/**
+ * sst_slot_put - set the status of interleaver/deinterleaver control
+ *
+ * (de)interleaver controls are defined in opposite sense to be user-friendly
+ *
+ * Instead of the enum value being the value written to the register, it is the
+ * register address; and the kcontrol number (register num) is the value written
+ * to the register. This is so that there can be only one value for each
+ * slot/channel since there is only one control for each slot/channel.
+ *
+ * This means that whenever an enum is set, we need to clear the bit
+ * for that kcontrol_no for all the interleaver OR deinterleaver registers
+ */
+static int sst_slot_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol);
+ struct sst_data *drv = snd_soc_component_get_drvdata(c);
+ struct sst_enum *e = (void *)kcontrol->private_value;
+ int i, ret = 0;
+ unsigned int ctl_no = e->reg;
+ unsigned int is_tx = e->tx;
+ unsigned int slot_channel_no;
+ unsigned int val, mux;
+ u8 *map;
+
+ map = is_tx ? sst_ssp_rx_map : sst_ssp_tx_map;
+
+ val = 1 << ctl_no;
+ mux = ucontrol->value.enumerated.item[0];
+ if (mux > e->max - 1)
+ return -EINVAL;
+
+ mutex_lock(&drv->lock);
+ /* first clear all registers of this bit */
+ for (i = 0; i < e->max; i++)
+ map[i] &= ~val;
+
+ if (mux == 0) {
+ /* kctl set to 'none' and we reset the bits so send IPC */
+ ret = sst_check_and_send_slot_map(drv, kcontrol);
+
+ mutex_unlock(&drv->lock);
+ return ret;
+ }
+
+ /* offset by one to take "None" into account */
+ slot_channel_no = mux - 1;
+ map[slot_channel_no] |= val;
+
+ dev_dbg(c->dev, "%s %s map = %#x\n",
+ is_tx ? "tx channel" : "rx slot",
+ e->texts[mux], map[slot_channel_no]);
+
+ ret = sst_check_and_send_slot_map(drv, kcontrol);
+
+ mutex_unlock(&drv->lock);
+ return ret;
+}
+
static int sst_send_algo_cmd(struct sst_data *drv,
struct sst_algo_control *bc)
{
@@ -104,6 +284,34 @@ static int sst_send_algo_cmd(struct sst_data *drv,
return ret;
}
+/**
+ * sst_find_and_send_pipe_algo - send all the algo parameters for a pipe
+ *
+ * The algos which are in each pipeline are sent to the firmware one by one
+ *
+ * Called with lock held
+ */
+static int sst_find_and_send_pipe_algo(struct sst_data *drv,
+ const char *pipe, struct sst_ids *ids)
+{
+ int ret = 0;
+ struct sst_algo_control *bc;
+ struct sst_module *algo = NULL;
+
+ dev_dbg(&drv->pdev->dev, "Enter: widget=%s\n", pipe);
+
+ list_for_each_entry(algo, &ids->algo_list, node) {
+ bc = (void *)algo->kctl->private_value;
+
+ dev_dbg(&drv->pdev->dev, "Found algo control name=%s pipe=%s\n",
+ algo->kctl->id.name, pipe);
+ ret = sst_send_algo_cmd(drv, bc);
+ if (ret)
+ return ret;
+ }
+ return ret;
+}
+
static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -162,6 +370,743 @@ static int sst_algo_control_set(struct snd_kcontrol *kcontrol,
return ret;
}
+static int sst_gain_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = mc->stereo ? 2 : 1;
+ uinfo->value.integer.min = mc->min;
+ uinfo->value.integer.max = mc->max;
+
+ return 0;
+}
+
+/**
+ * sst_send_gain_cmd - send the gain algorithm IPC to the FW
+ * @gv: the stored value of gain (also contains rampduration)
+ * @mute: flag that indicates whether this was called from the
+ * digital_mute callback or directly. If called from the
+ * digital_mute callback, module will be muted/unmuted based on this
+ * flag. The flag is always 0 if called directly.
+ *
+ * Called with sst_data.lock held
+ *
+ * The user-set gain value is sent only if the user-controllable 'mute' control
+ * is OFF (indicated by gv->mute). Otherwise, the mute value (MIN value) is
+ * sent.
+ */
+static int sst_send_gain_cmd(struct sst_data *drv, struct sst_gain_value *gv,
+ u16 task_id, u16 loc_id, u16 module_id, int mute)
+{
+ struct sst_cmd_set_gain_dual cmd;
+
+ dev_dbg(&drv->pdev->dev, "Enter\n");
+
+ cmd.header.command_id = MMX_SET_GAIN;
+ SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+ cmd.gain_cell_num = 1;
+
+ if (mute || gv->mute) {
+ cmd.cell_gains[0].cell_gain_left = SST_GAIN_MIN_VALUE;
+ cmd.cell_gains[0].cell_gain_right = SST_GAIN_MIN_VALUE;
+ } else {
+ cmd.cell_gains[0].cell_gain_left = gv->l_gain;
+ cmd.cell_gains[0].cell_gain_right = gv->r_gain;
+ }
+
+ SST_FILL_DESTINATION(2, cmd.cell_gains[0].dest,
+ loc_id, module_id);
+ cmd.cell_gains[0].gain_time_constant = gv->ramp_duration;
+
+ cmd.header.length = sizeof(struct sst_cmd_set_gain_dual)
+ - sizeof(struct sst_dsp_header);
+
+ /* we are with lock held, so call the unlocked api to send */
+ return sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS,
+ SST_FLAG_BLOCKED, task_id, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+}
+
+static int sst_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+ struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value;
+ struct sst_gain_value *gv = mc->gain_val;
+
+ switch (mc->type) {
+ case SST_GAIN_TLV:
+ ucontrol->value.integer.value[0] = gv->l_gain;
+ ucontrol->value.integer.value[1] = gv->r_gain;
+ break;
+
+ case SST_GAIN_MUTE:
+ ucontrol->value.integer.value[0] = gv->mute ? 1 : 0;
+ break;
+
+ case SST_GAIN_RAMP_DURATION:
+ ucontrol->value.integer.value[0] = gv->ramp_duration;
+ break;
+
+ default:
+ dev_err(component->dev, "Invalid Input- gain type:%d\n",
+ mc->type);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int sst_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int ret = 0;
+ struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol);
+ struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+ struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value;
+ struct sst_gain_value *gv = mc->gain_val;
+
+ mutex_lock(&drv->lock);
+
+ switch (mc->type) {
+ case SST_GAIN_TLV:
+ gv->l_gain = ucontrol->value.integer.value[0];
+ gv->r_gain = ucontrol->value.integer.value[1];
+ dev_dbg(cmpnt->dev, "%s: Volume %d, %d\n",
+ mc->pname, gv->l_gain, gv->r_gain);
+ break;
+
+ case SST_GAIN_MUTE:
+ gv->mute = !!ucontrol->value.integer.value[0];
+ dev_dbg(cmpnt->dev, "%s: Mute %d\n", mc->pname, gv->mute);
+ break;
+
+ case SST_GAIN_RAMP_DURATION:
+ gv->ramp_duration = ucontrol->value.integer.value[0];
+ dev_dbg(cmpnt->dev, "%s: Ramp Delay%d\n",
+ mc->pname, gv->ramp_duration);
+ break;
+
+ default:
+ mutex_unlock(&drv->lock);
+ dev_err(cmpnt->dev, "Invalid Input- gain type:%d\n",
+ mc->type);
+ return -EINVAL;
+ }
+
+ if (mc->w && mc->w->power)
+ ret = sst_send_gain_cmd(drv, gv, mc->task_id,
+ mc->pipe_id | mc->instance_id, mc->module_id, 0);
+ mutex_unlock(&drv->lock);
+
+ return ret;
+}
+
+static int sst_set_pipe_gain(struct sst_ids *ids,
+ struct sst_data *drv, int mute);
+
+static int sst_send_pipe_module_params(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol)
+{
+ struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+ struct sst_data *drv = snd_soc_component_get_drvdata(c);
+ struct sst_ids *ids = w->priv;
+
+ mutex_lock(&drv->lock);
+ sst_find_and_send_pipe_algo(drv, w->name, ids);
+ sst_set_pipe_gain(ids, drv, 0);
+ mutex_unlock(&drv->lock);
+
+ return 0;
+}
+
+static int sst_generic_modules_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ return sst_send_pipe_module_params(w, k);
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(sst_gain_tlv_common, SST_GAIN_MIN_VALUE * 10, 10, 0);
+
+/* Look up table to convert MIXER SW bit regs to SWM inputs */
+static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = {
+ [SST_IP_CODEC0] = SST_SWM_IN_CODEC0,
+ [SST_IP_CODEC1] = SST_SWM_IN_CODEC1,
+ [SST_IP_LOOP0] = SST_SWM_IN_SPROT_LOOP,
+ [SST_IP_LOOP1] = SST_SWM_IN_MEDIA_LOOP1,
+ [SST_IP_LOOP2] = SST_SWM_IN_MEDIA_LOOP2,
+ [SST_IP_PCM0] = SST_SWM_IN_PCM0,
+ [SST_IP_PCM1] = SST_SWM_IN_PCM1,
+ [SST_IP_MEDIA0] = SST_SWM_IN_MEDIA0,
+ [SST_IP_MEDIA1] = SST_SWM_IN_MEDIA1,
+ [SST_IP_MEDIA2] = SST_SWM_IN_MEDIA2,
+ [SST_IP_MEDIA3] = SST_SWM_IN_MEDIA3,
+};
+
+/**
+ * fill_swm_input - fill in the SWM input ids given the register
+ *
+ * The register value is a bit-field inicated which mixer inputs are ON. Use the
+ * lookup table to get the input-id and fill it in the structure.
+ */
+static int fill_swm_input(struct snd_soc_component *cmpnt,
+ struct swm_input_ids *swm_input, unsigned int reg)
+{
+ uint i, is_set, nb_inputs = 0;
+ u16 input_loc_id;
+
+ dev_dbg(cmpnt->dev, "reg: %#x\n", reg);
+ for (i = 0; i < SST_SWM_INPUT_COUNT; i++) {
+ is_set = reg & BIT(i);
+ if (!is_set)
+ continue;
+
+ input_loc_id = swm_mixer_input_ids[i];
+ SST_FILL_DESTINATION(2, swm_input->input_id,
+ input_loc_id, SST_DEFAULT_MODULE_ID);
+ nb_inputs++;
+ swm_input++;
+ dev_dbg(cmpnt->dev, "input id: %#x, nb_inputs: %d\n",
+ input_loc_id, nb_inputs);
+
+ if (nb_inputs == SST_CMD_SWM_MAX_INPUTS) {
+ dev_warn(cmpnt->dev, "SET_SWM cmd max inputs reached");
+ break;
+ }
+ }
+ return nb_inputs;
+}
+
+
+/**
+ * called with lock held
+ */
+static int sst_set_pipe_gain(struct sst_ids *ids,
+ struct sst_data *drv, int mute)
+{
+ int ret = 0;
+ struct sst_gain_mixer_control *mc;
+ struct sst_gain_value *gv;
+ struct sst_module *gain = NULL;
+
+ list_for_each_entry(gain, &ids->gain_list, node) {
+ struct snd_kcontrol *kctl = gain->kctl;
+
+ dev_dbg(&drv->pdev->dev, "control name=%s\n", kctl->id.name);
+ mc = (void *)kctl->private_value;
+ gv = mc->gain_val;
+
+ ret = sst_send_gain_cmd(drv, gv, mc->task_id,
+ mc->pipe_id | mc->instance_id, mc->module_id, mute);
+ if (ret)
+ return ret;
+ }
+ return ret;
+}
+
+static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct sst_cmd_set_swm cmd;
+ struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+ struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+ struct sst_ids *ids = w->priv;
+ bool set_mixer = false;
+ struct soc_mixer_control *mc;
+ int val = 0;
+ int i = 0;
+
+ dev_dbg(cmpnt->dev, "widget = %s\n", w->name);
+ /*
+ * Identify which mixer input is on and send the bitmap of the
+ * inputs as an IPC to the DSP.
+ */
+ for (i = 0; i < w->num_kcontrols; i++) {
+ if (dapm_kcontrol_get_value(w->kcontrols[i])) {
+ mc = (struct soc_mixer_control *)(w->kcontrols[i])->private_value;
+ val |= 1 << mc->shift;
+ }
+ }
+ dev_dbg(cmpnt->dev, "val = %#x\n", val);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ case SND_SOC_DAPM_POST_PMD:
+ set_mixer = true;
+ break;
+ case SND_SOC_DAPM_POST_REG:
+ if (w->power)
+ set_mixer = true;
+ break;
+ default:
+ set_mixer = false;
+ }
+
+ if (set_mixer == false)
+ return 0;
+
+ if (SND_SOC_DAPM_EVENT_ON(event) ||
+ event == SND_SOC_DAPM_POST_REG)
+ cmd.switch_state = SST_SWM_ON;
+ else
+ cmd.switch_state = SST_SWM_OFF;
+
+ SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+ /* MMX_SET_SWM == SBA_SET_SWM */
+ cmd.header.command_id = SBA_SET_SWM;
+
+ SST_FILL_DESTINATION(2, cmd.output_id,
+ ids->location_id, SST_DEFAULT_MODULE_ID);
+ cmd.nb_inputs = fill_swm_input(cmpnt, &cmd.input[0], val);
+ cmd.header.length = offsetof(struct sst_cmd_set_swm, input)
+ - sizeof(struct sst_dsp_header)
+ + (cmd.nb_inputs * sizeof(cmd.input[0]));
+
+ return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+ ids->task_id, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+}
+
+/* SBA mixers - 16 inputs */
+#define SST_SBA_DECLARE_MIX_CONTROLS(kctl_name) \
+ static const struct snd_kcontrol_new kctl_name[] = { \
+ SOC_DAPM_SINGLE("codec_in0 Switch", SND_SOC_NOPM, SST_IP_CODEC0, 1, 0), \
+ SOC_DAPM_SINGLE("codec_in1 Switch", SND_SOC_NOPM, SST_IP_CODEC1, 1, 0), \
+ SOC_DAPM_SINGLE("sprot_loop_in Switch", SND_SOC_NOPM, SST_IP_LOOP0, 1, 0), \
+ SOC_DAPM_SINGLE("media_loop1_in Switch", SND_SOC_NOPM, SST_IP_LOOP1, 1, 0), \
+ SOC_DAPM_SINGLE("media_loop2_in Switch", SND_SOC_NOPM, SST_IP_LOOP2, 1, 0), \
+ SOC_DAPM_SINGLE("pcm0_in Switch", SND_SOC_NOPM, SST_IP_PCM0, 1, 0), \
+ SOC_DAPM_SINGLE("pcm1_in Switch", SND_SOC_NOPM, SST_IP_PCM1, 1, 0), \
+ }
+
+#define SST_SBA_MIXER_GRAPH_MAP(mix_name) \
+ { mix_name, "codec_in0 Switch", "codec_in0" }, \
+ { mix_name, "codec_in1 Switch", "codec_in1" }, \
+ { mix_name, "sprot_loop_in Switch", "sprot_loop_in" }, \
+ { mix_name, "media_loop1_in Switch", "media_loop1_in" }, \
+ { mix_name, "media_loop2_in Switch", "media_loop2_in" }, \
+ { mix_name, "pcm0_in Switch", "pcm0_in" }, \
+ { mix_name, "pcm1_in Switch", "pcm1_in" }
+
+#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name) \
+ static const struct snd_kcontrol_new kctl_name[] = { \
+ SOC_DAPM_SINGLE("media0_in Switch", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0), \
+ SOC_DAPM_SINGLE("media1_in Switch", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0), \
+ SOC_DAPM_SINGLE("media2_in Switch", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0), \
+ SOC_DAPM_SINGLE("media3_in Switch", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0), \
+ }
+
+SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media0_controls);
+SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media1_controls);
+
+/* 18 SBA mixers */
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm1_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls);
+
+/*
+ * sst_handle_vb_timer - Start/Stop the DSP scheduler
+ *
+ * The DSP expects first cmd to be SBA_VB_START, so at first startup send
+ * that.
+ * DSP expects last cmd to be SBA_VB_IDLE, so at last shutdown send that.
+ *
+ * Do refcount internally so that we send command only at first start
+ * and last end. Since SST driver does its own ref count, invoke sst's
+ * power ops always!
+ */
+int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable)
+{
+ int ret = 0;
+ struct sst_cmd_generic cmd;
+ struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
+ static int timer_usage;
+
+ if (enable)
+ cmd.header.command_id = SBA_VB_START;
+ else
+ cmd.header.command_id = SBA_IDLE;
+ dev_dbg(dai->dev, "enable=%u, usage=%d\n", enable, timer_usage);
+
+ SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+ cmd.header.length = 0;
+
+ if (enable) {
+ ret = sst->ops->power(sst->dev, true);
+ if (ret < 0)
+ return ret;
+ }
+
+ mutex_lock(&drv->lock);
+ if (enable)
+ timer_usage++;
+ else
+ timer_usage--;
+
+ /*
+ * Send the command only if this call is the first enable or last
+ * disable
+ */
+ if ((enable && (timer_usage == 1)) ||
+ (!enable && (timer_usage == 0))) {
+ ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_CMD,
+ SST_FLAG_BLOCKED, SST_TASK_SBA, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+ if (ret && enable) {
+ timer_usage--;
+ enable = false;
+ }
+ }
+ mutex_unlock(&drv->lock);
+
+ if (!enable)
+ sst->ops->power(sst->dev, false);
+ return ret;
+}
+
+/**
+ * sst_ssp_config - contains SSP configuration for media UC
+ */
+static const struct sst_ssp_config sst_ssp_configs = {
+ .ssp_id = SSP_CODEC,
+ .bits_per_slot = 24,
+ .slots = 4,
+ .ssp_mode = SSP_MODE_MASTER,
+ .pcm_mode = SSP_PCM_MODE_NETWORK,
+ .duplex = SSP_DUPLEX,
+ .ssp_protocol = SSP_MODE_PCM,
+ .fs_width = 1,
+ .fs_frequency = SSP_FS_48_KHZ,
+ .active_slot_map = 0xF,
+ .start_delay = 0,
+};
+
+int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable)
+{
+ struct sst_cmd_sba_hw_set_ssp cmd;
+ struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
+ const struct sst_ssp_config *config;
+
+ dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id);
+
+ SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+ cmd.header.command_id = SBA_HW_SET_SSP;
+ cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp)
+ - sizeof(struct sst_dsp_header);
+
+ config = &sst_ssp_configs;
+ dev_dbg(dai->dev, "ssp_id: %u\n", config->ssp_id);
+
+ if (enable)
+ cmd.switch_state = SST_SWITCH_ON;
+ else
+ cmd.switch_state = SST_SWITCH_OFF;
+
+ cmd.selection = config->ssp_id;
+ cmd.nb_bits_per_slots = config->bits_per_slot;
+ cmd.nb_slots = config->slots;
+ cmd.mode = config->ssp_mode | (config->pcm_mode << 1);
+ cmd.duplex = config->duplex;
+ cmd.active_tx_slot_map = config->active_slot_map;
+ cmd.active_rx_slot_map = config->active_slot_map;
+ cmd.frame_sync_frequency = config->fs_frequency;
+ cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH;
+ cmd.data_polarity = 1;
+ cmd.frame_sync_width = config->fs_width;
+ cmd.ssp_protocol = config->ssp_protocol;
+ cmd.start_delay = config->start_delay;
+ cmd.reserved1 = cmd.reserved2 = 0xFF;
+
+ return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+ SST_TASK_SBA, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+}
+
+static int sst_set_be_modules(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ int ret = 0;
+ struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+ struct sst_data *drv = snd_soc_component_get_drvdata(c);
+
+ dev_dbg(c->dev, "Enter: widget=%s\n", w->name);
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ ret = sst_send_slot_map(drv);
+ if (ret)
+ return ret;
+ ret = sst_send_pipe_module_params(w, k);
+ }
+ return ret;
+}
+
+static int sst_set_media_path(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ int ret = 0;
+ struct sst_cmd_set_media_path cmd;
+ struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+ struct sst_data *drv = snd_soc_component_get_drvdata(c);
+ struct sst_ids *ids = w->priv;
+
+ dev_dbg(c->dev, "widget=%s\n", w->name);
+ dev_dbg(c->dev, "task=%u, location=%#x\n",
+ ids->task_id, ids->location_id);
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ cmd.switch_state = SST_PATH_ON;
+ else
+ cmd.switch_state = SST_PATH_OFF;
+
+ SST_FILL_DESTINATION(2, cmd.header.dst,
+ ids->location_id, SST_DEFAULT_MODULE_ID);
+
+ /* MMX_SET_MEDIA_PATH == SBA_SET_MEDIA_PATH */
+ cmd.header.command_id = MMX_SET_MEDIA_PATH;
+ cmd.header.length = sizeof(struct sst_cmd_set_media_path)
+ - sizeof(struct sst_dsp_header);
+
+ ret = sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+ ids->task_id, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+ if (ret)
+ return ret;
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ ret = sst_send_pipe_module_params(w, k);
+ return ret;
+}
+
+static int sst_set_media_loop(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ int ret = 0;
+ struct sst_cmd_sba_set_media_loop_map cmd;
+ struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+ struct sst_data *drv = snd_soc_component_get_drvdata(c);
+ struct sst_ids *ids = w->priv;
+
+ dev_dbg(c->dev, "Enter:widget=%s\n", w->name);
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ cmd.switch_state = SST_SWITCH_ON;
+ else
+ cmd.switch_state = SST_SWITCH_OFF;
+
+ SST_FILL_DESTINATION(2, cmd.header.dst,
+ ids->location_id, SST_DEFAULT_MODULE_ID);
+
+ cmd.header.command_id = SBA_SET_MEDIA_LOOP_MAP;
+ cmd.header.length = sizeof(struct sst_cmd_sba_set_media_loop_map)
+ - sizeof(struct sst_dsp_header);
+ cmd.param.part.cfg.rate = 2; /* 48khz */
+
+ cmd.param.part.cfg.format = ids->format; /* stereo/Mono */
+ cmd.param.part.cfg.s_length = 1; /* 24bit left justified */
+ cmd.map = 0; /* Algo sequence: Gain - DRP - FIR - IIR */
+
+ ret = sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+ SST_TASK_SBA, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+ if (ret)
+ return ret;
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ ret = sst_send_pipe_module_params(w, k);
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget sst_dapm_widgets[] = {
+ SST_AIF_IN("codec_in0", sst_set_be_modules),
+ SST_AIF_IN("codec_in1", sst_set_be_modules),
+ SST_AIF_OUT("codec_out0", sst_set_be_modules),
+ SST_AIF_OUT("codec_out1", sst_set_be_modules),
+
+ /* Media Paths */
+ /* MediaX IN paths are set via ALLOC, so no SET_MEDIA_PATH command */
+ SST_PATH_INPUT("media0_in", SST_TASK_MMX, SST_SWM_IN_MEDIA0, sst_generic_modules_event),
+ SST_PATH_INPUT("media1_in", SST_TASK_MMX, SST_SWM_IN_MEDIA1, NULL),
+ SST_PATH_INPUT("media2_in", SST_TASK_MMX, SST_SWM_IN_MEDIA2, sst_set_media_path),
+ SST_PATH_INPUT("media3_in", SST_TASK_MMX, SST_SWM_IN_MEDIA3, NULL),
+ SST_PATH_OUTPUT("media0_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA0, sst_set_media_path),
+ SST_PATH_OUTPUT("media1_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA1, sst_set_media_path),
+
+ /* SBA PCM Paths */
+ SST_PATH_INPUT("pcm0_in", SST_TASK_SBA, SST_SWM_IN_PCM0, sst_set_media_path),
+ SST_PATH_INPUT("pcm1_in", SST_TASK_SBA, SST_SWM_IN_PCM1, sst_set_media_path),
+ SST_PATH_OUTPUT("pcm0_out", SST_TASK_SBA, SST_SWM_OUT_PCM0, sst_set_media_path),
+ SST_PATH_OUTPUT("pcm1_out", SST_TASK_SBA, SST_SWM_OUT_PCM1, sst_set_media_path),
+ SST_PATH_OUTPUT("pcm2_out", SST_TASK_SBA, SST_SWM_OUT_PCM2, sst_set_media_path),
+
+ /* SBA Loops */
+ SST_PATH_INPUT("sprot_loop_in", SST_TASK_SBA, SST_SWM_IN_SPROT_LOOP, NULL),
+ SST_PATH_INPUT("media_loop1_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP1, NULL),
+ SST_PATH_INPUT("media_loop2_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP2, NULL),
+ SST_PATH_MEDIA_LOOP_OUTPUT("sprot_loop_out", SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, SST_FMT_MONO, sst_set_media_loop),
+ SST_PATH_MEDIA_LOOP_OUTPUT("media_loop1_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, SST_FMT_MONO, sst_set_media_loop),
+ SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop),
+
+ /* Media Mixers */
+ SST_SWM_MIXER("media0_out mix 0", SND_SOC_NOPM, SST_TASK_MMX, SST_SWM_OUT_MEDIA0,
+ sst_mix_media0_controls, sst_swm_mixer_event),
+ SST_SWM_MIXER("media1_out mix 0", SND_SOC_NOPM, SST_TASK_MMX, SST_SWM_OUT_MEDIA1,
+ sst_mix_media1_controls, sst_swm_mixer_event),
+
+ /* SBA PCM mixers */
+ SST_SWM_MIXER("pcm0_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM0,
+ sst_mix_pcm0_controls, sst_swm_mixer_event),
+ SST_SWM_MIXER("pcm1_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM1,
+ sst_mix_pcm1_controls, sst_swm_mixer_event),
+ SST_SWM_MIXER("pcm2_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM2,
+ sst_mix_pcm2_controls, sst_swm_mixer_event),
+
+ /* SBA Loop mixers */
+ SST_SWM_MIXER("sprot_loop_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP,
+ sst_mix_sprot_l0_controls, sst_swm_mixer_event),
+ SST_SWM_MIXER("media_loop1_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1,
+ sst_mix_media_l1_controls, sst_swm_mixer_event),
+ SST_SWM_MIXER("media_loop2_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2,
+ sst_mix_media_l2_controls, sst_swm_mixer_event),
+
+ /* SBA Backend mixers */
+ SST_SWM_MIXER("codec_out0 mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_CODEC0,
+ sst_mix_codec0_controls, sst_swm_mixer_event),
+ SST_SWM_MIXER("codec_out1 mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_CODEC1,
+ sst_mix_codec1_controls, sst_swm_mixer_event),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"media0_in", NULL, "Compress Playback"},
+ {"media1_in", NULL, "Headset Playback"},
+ {"media2_in", NULL, "pcm0_out"},
+
+ {"media0_out mix 0", "media0_in Switch", "media0_in"},
+ {"media0_out mix 0", "media1_in Switch", "media1_in"},
+ {"media0_out mix 0", "media2_in Switch", "media2_in"},
+ {"media0_out mix 0", "media3_in Switch", "media3_in"},
+ {"media1_out mix 0", "media0_in Switch", "media0_in"},
+ {"media1_out mix 0", "media1_in Switch", "media1_in"},
+ {"media1_out mix 0", "media2_in Switch", "media2_in"},
+ {"media1_out mix 0", "media3_in Switch", "media3_in"},
+
+ {"media0_out", NULL, "media0_out mix 0"},
+ {"media1_out", NULL, "media1_out mix 0"},
+ {"pcm0_in", NULL, "media0_out"},
+ {"pcm1_in", NULL, "media1_out"},
+
+ {"Headset Capture", NULL, "pcm1_out"},
+ {"Headset Capture", NULL, "pcm2_out"},
+ {"pcm0_out", NULL, "pcm0_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("pcm0_out mix 0"),
+ {"pcm1_out", NULL, "pcm1_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("pcm1_out mix 0"),
+ {"pcm2_out", NULL, "pcm2_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("pcm2_out mix 0"),
+
+ {"media_loop1_in", NULL, "media_loop1_out"},
+ {"media_loop1_out", NULL, "media_loop1_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("media_loop1_out mix 0"),
+ {"media_loop2_in", NULL, "media_loop2_out"},
+ {"media_loop2_out", NULL, "media_loop2_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("media_loop2_out mix 0"),
+ {"sprot_loop_in", NULL, "sprot_loop_out"},
+ {"sprot_loop_out", NULL, "sprot_loop_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("sprot_loop_out mix 0"),
+
+ {"codec_out0", NULL, "codec_out0 mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("codec_out0 mix 0"),
+ {"codec_out1", NULL, "codec_out1 mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("codec_out1 mix 0"),
+
+};
+static const char * const slot_names[] = {
+ "none",
+ "slot 0", "slot 1", "slot 2", "slot 3",
+ "slot 4", "slot 5", "slot 6", "slot 7", /* not supported by FW */
+};
+
+static const char * const channel_names[] = {
+ "none",
+ "codec_out0_0", "codec_out0_1", "codec_out1_0", "codec_out1_1",
+ "codec_out2_0", "codec_out2_1", "codec_out3_0", "codec_out3_1", /* not supported by FW */
+};
+
+#define SST_INTERLEAVER(xpname, slot_name, slotno) \
+ SST_SSP_SLOT_CTL(xpname, "tx interleaver", slot_name, slotno, true, \
+ channel_names, sst_slot_get, sst_slot_put)
+
+#define SST_DEINTERLEAVER(xpname, channel_name, channel_no) \
+ SST_SSP_SLOT_CTL(xpname, "rx deinterleaver", channel_name, channel_no, false, \
+ slot_names, sst_slot_get, sst_slot_put)
+
+static const struct snd_kcontrol_new sst_slot_controls[] = {
+ SST_INTERLEAVER("codec_out", "slot 0", 0),
+ SST_INTERLEAVER("codec_out", "slot 1", 1),
+ SST_INTERLEAVER("codec_out", "slot 2", 2),
+ SST_INTERLEAVER("codec_out", "slot 3", 3),
+ SST_DEINTERLEAVER("codec_in", "codec_in0_0", 0),
+ SST_DEINTERLEAVER("codec_in", "codec_in0_1", 1),
+ SST_DEINTERLEAVER("codec_in", "codec_in1_0", 2),
+ SST_DEINTERLEAVER("codec_in", "codec_in1_1", 3),
+};
+
+/* Gain helper with min/max set */
+#define SST_GAIN(name, path_id, task_id, instance, gain_var) \
+ SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \
+ SST_GAIN_TC_MIN, SST_GAIN_TC_MAX, \
+ sst_gain_get, sst_gain_put, \
+ SST_MODULE_ID_GAIN_CELL, path_id, instance, task_id, \
+ sst_gain_tlv_common, gain_var)
+
+#define SST_VOLUME(name, path_id, task_id, instance, gain_var) \
+ SST_GAIN_KCONTROLS(name, "Volume", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \
+ SST_GAIN_TC_MIN, SST_GAIN_TC_MAX, \
+ sst_gain_get, sst_gain_put, \
+ SST_MODULE_ID_VOLUME, path_id, instance, task_id, \
+ sst_gain_tlv_common, gain_var)
+
+static struct sst_gain_value sst_gains[];
+
+static const struct snd_kcontrol_new sst_gain_controls[] = {
+ SST_GAIN("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[0]),
+ SST_GAIN("media1_in", SST_PATH_INDEX_MEDIA1_IN, SST_TASK_MMX, 0, &sst_gains[1]),
+ SST_GAIN("media2_in", SST_PATH_INDEX_MEDIA2_IN, SST_TASK_MMX, 0, &sst_gains[2]),
+ SST_GAIN("media3_in", SST_PATH_INDEX_MEDIA3_IN, SST_TASK_MMX, 0, &sst_gains[3]),
+
+ SST_GAIN("pcm0_in", SST_PATH_INDEX_PCM0_IN, SST_TASK_SBA, 0, &sst_gains[4]),
+ SST_GAIN("pcm1_in", SST_PATH_INDEX_PCM1_IN, SST_TASK_SBA, 0, &sst_gains[5]),
+ SST_GAIN("pcm1_out", SST_PATH_INDEX_PCM1_OUT, SST_TASK_SBA, 0, &sst_gains[6]),
+ SST_GAIN("pcm2_out", SST_PATH_INDEX_PCM2_OUT, SST_TASK_SBA, 0, &sst_gains[7]),
+
+ SST_GAIN("codec_in0", SST_PATH_INDEX_CODEC_IN0, SST_TASK_SBA, 0, &sst_gains[8]),
+ SST_GAIN("codec_in1", SST_PATH_INDEX_CODEC_IN1, SST_TASK_SBA, 0, &sst_gains[9]),
+ SST_GAIN("codec_out0", SST_PATH_INDEX_CODEC_OUT0, SST_TASK_SBA, 0, &sst_gains[10]),
+ SST_GAIN("codec_out1", SST_PATH_INDEX_CODEC_OUT1, SST_TASK_SBA, 0, &sst_gains[11]),
+ SST_GAIN("media_loop1_out", SST_PATH_INDEX_MEDIA_LOOP1_OUT, SST_TASK_SBA, 0, &sst_gains[12]),
+ SST_GAIN("media_loop2_out", SST_PATH_INDEX_MEDIA_LOOP2_OUT, SST_TASK_SBA, 0, &sst_gains[13]),
+ SST_GAIN("sprot_loop_out", SST_PATH_INDEX_SPROT_LOOP_OUT, SST_TASK_SBA, 0, &sst_gains[14]),
+ SST_VOLUME("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[15]),
+};
+
+#define SST_GAIN_NUM_CONTROLS 3
+/* the SST_GAIN macro above will create three alsa controls for each
+ * instance invoked, gain, mute and ramp duration, which use the same gain
+ * cell sst_gain to keep track of data
+ * To calculate number of gain cell instances we need to device by 3 in
+ * below caulcation for gain cell memory.
+ * This gets rid of static number and issues while adding new controls
+ */
+static struct sst_gain_value sst_gains[ARRAY_SIZE(sst_gain_controls)/SST_GAIN_NUM_CONTROLS];
+
static const struct snd_kcontrol_new sst_algo_controls[] = {
SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24,
SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR),
@@ -198,21 +1143,280 @@ static int sst_algo_control_init(struct device *dev)
return 0;
}
-int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
+static bool is_sst_dapm_widget(struct snd_soc_dapm_widget *w)
+{
+ switch (w->id) {
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_mixer:
+ return true;
+ default:
+ return false;
+ }
+}
+
+/**
+ * sst_send_pipe_gains - send gains for the front-end DAIs
+ *
+ * The gains in the pipes connected to the front-ends are muted/unmuted
+ * automatically via the digital_mute() DAPM callback. This function sends the
+ * gains for the front-end pipes.
+ */
+int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute)
+{
+ struct sst_data *drv = snd_soc_dai_get_drvdata(dai);
+ struct snd_soc_dapm_widget *w;
+ struct snd_soc_dapm_path *p = NULL;
+
+ dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dev_dbg(dai->dev, "Stream name=%s\n",
+ dai->playback_widget->name);
+ w = dai->playback_widget;
+ list_for_each_entry(p, &w->sinks, list_source) {
+ if (p->connected && !p->connected(w, p->sink))
+ continue;
+
+ if (p->connect && p->sink->power &&
+ is_sst_dapm_widget(p->sink)) {
+ struct sst_ids *ids = p->sink->priv;
+
+ dev_dbg(dai->dev, "send gains for widget=%s\n",
+ p->sink->name);
+ mutex_lock(&drv->lock);
+ sst_set_pipe_gain(ids, drv, mute);
+ mutex_unlock(&drv->lock);
+ }
+ }
+ } else {
+ dev_dbg(dai->dev, "Stream name=%s\n",
+ dai->capture_widget->name);
+ w = dai->capture_widget;
+ list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connected && !p->connected(w, p->sink))
+ continue;
+
+ if (p->connect && p->source->power &&
+ is_sst_dapm_widget(p->source)) {
+ struct sst_ids *ids = p->source->priv;
+
+ dev_dbg(dai->dev, "send gain for widget=%s\n",
+ p->source->name);
+ mutex_lock(&drv->lock);
+ sst_set_pipe_gain(ids, drv, mute);
+ mutex_unlock(&drv->lock);
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * sst_fill_module_list - populate the list of modules/gains for a pipe
+ *
+ *
+ * Fills the widget pointer in the kcontrol private data, and also fills the
+ * kcontrol pointer in the widget private data.
+ *
+ * Widget pointer is used to send the algo/gain in the .put() handler if the
+ * widget is powerd on.
+ *
+ * Kcontrol pointer is used to send the algo/gain in the widget power ON/OFF
+ * event handler. Each widget (pipe) has multiple algos stored in the algo_list.
+ */
+static int sst_fill_module_list(struct snd_kcontrol *kctl,
+ struct snd_soc_dapm_widget *w, int type)
{
+ struct sst_module *module = NULL;
+ struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+ struct sst_ids *ids = w->priv;
int ret = 0;
+
+ module = devm_kzalloc(c->dev, sizeof(*module), GFP_KERNEL);
+ if (!module)
+ return -ENOMEM;
+
+ if (type == SST_MODULE_GAIN) {
+ struct sst_gain_mixer_control *mc = (void *)kctl->private_value;
+
+ mc->w = w;
+ module->kctl = kctl;
+ list_add_tail(&module->node, &ids->gain_list);
+ } else if (type == SST_MODULE_ALGO) {
+ struct sst_algo_control *bc = (void *)kctl->private_value;
+
+ bc->w = w;
+ module->kctl = kctl;
+ list_add_tail(&module->node, &ids->algo_list);
+ } else {
+ dev_err(c->dev, "invoked for unknown type %d module %s",
+ type, kctl->id.name);
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+/**
+ * sst_fill_widget_module_info - fill list of gains/algos for the pipe
+ * @widget: pipe modelled as a DAPM widget
+ *
+ * Fill the list of gains/algos for the widget by looking at all the card
+ * controls and comparing the name of the widget with the first part of control
+ * name. First part of control name contains the pipe name (widget name).
+ */
+static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w,
+ struct snd_soc_platform *platform)
+{
+ struct snd_kcontrol *kctl;
+ int index, ret = 0;
+ struct snd_card *card = platform->component.card->snd_card;
+ char *idx;
+
+ down_read(&card->controls_rwsem);
+
+ list_for_each_entry(kctl, &card->controls, list) {
+ idx = strstr(kctl->id.name, " ");
+ if (idx == NULL)
+ continue;
+ index = strlen(kctl->id.name) - strlen(idx);
+
+ if (strstr(kctl->id.name, "Volume") &&
+ !strncmp(kctl->id.name, w->name, index))
+ ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN);
+
+ else if (strstr(kctl->id.name, "params") &&
+ !strncmp(kctl->id.name, w->name, index))
+ ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO);
+
+ else if (strstr(kctl->id.name, "Switch") &&
+ !strncmp(kctl->id.name, w->name, index) &&
+ strstr(kctl->id.name, "Gain")) {
+ struct sst_gain_mixer_control *mc =
+ (void *)kctl->private_value;
+
+ mc->w = w;
+
+ } else if (strstr(kctl->id.name, "interleaver") &&
+ !strncmp(kctl->id.name, w->name, index)) {
+ struct sst_enum *e = (void *)kctl->private_value;
+
+ e->w = w;
+
+ } else if (strstr(kctl->id.name, "deinterleaver") &&
+ !strncmp(kctl->id.name, w->name, index)) {
+
+ struct sst_enum *e = (void *)kctl->private_value;
+
+ e->w = w;
+ }
+
+ if (ret < 0) {
+ up_read(&card->controls_rwsem);
+ return ret;
+ }
+ }
+
+ up_read(&card->controls_rwsem);
+ return 0;
+}
+
+/**
+ * sst_fill_linked_widgets - fill the parent pointer for the linked widget
+ */
+static void sst_fill_linked_widgets(struct snd_soc_platform *platform,
+ struct sst_ids *ids)
+{
+ struct snd_soc_dapm_widget *w;
+ unsigned int len = strlen(ids->parent_wname);
+
+ list_for_each_entry(w, &platform->component.card->widgets, list) {
+ if (!strncmp(ids->parent_wname, w->name, len)) {
+ ids->parent_w = w;
+ break;
+ }
+ }
+}
+
+/**
+ * sst_map_modules_to_pipe - fill algo/gains list for all pipes
+ */
+static int sst_map_modules_to_pipe(struct snd_soc_platform *platform)
+{
+ struct snd_soc_dapm_widget *w;
+ int ret = 0;
+
+ list_for_each_entry(w, &platform->component.card->widgets, list) {
+ if (is_sst_dapm_widget(w) && (w->priv)) {
+ struct sst_ids *ids = w->priv;
+
+ dev_dbg(platform->dev, "widget type=%d name=%s\n",
+ w->id, w->name);
+ INIT_LIST_HEAD(&ids->algo_list);
+ INIT_LIST_HEAD(&ids->gain_list);
+ ret = sst_fill_widget_module_info(w, platform);
+
+ if (ret < 0)
+ return ret;
+
+ /* fill linked widgets */
+ if (ids->parent_wname != NULL)
+ sst_fill_linked_widgets(platform, ids);
+ }
+ }
+ return 0;
+}
+
+int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
+{
+ int i, ret = 0;
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(&platform->component);
struct sst_data *drv = snd_soc_platform_get_drvdata(platform);
+ unsigned int gains = ARRAY_SIZE(sst_gain_controls)/3;
drv->byte_stream = devm_kzalloc(platform->dev,
SST_MAX_BIN_BYTES, GFP_KERNEL);
if (!drv->byte_stream)
return -ENOMEM;
- /*Initialize algo control params*/
+ snd_soc_dapm_new_controls(dapm, sst_dapm_widgets,
+ ARRAY_SIZE(sst_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon,
+ ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_widgets(dapm->card);
+
+ for (i = 0; i < gains; i++) {
+ sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT;
+ sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT;
+ sst_gains[i].r_gain = SST_GAIN_VOLUME_DEFAULT;
+ sst_gains[i].ramp_duration = SST_GAIN_RAMP_DURATION_DEFAULT;
+ }
+
+ ret = snd_soc_add_platform_controls(platform, sst_gain_controls,
+ ARRAY_SIZE(sst_gain_controls));
+ if (ret)
+ return ret;
+
+ /* Initialize algo control params */
ret = sst_algo_control_init(platform->dev);
if (ret)
return ret;
ret = snd_soc_add_platform_controls(platform, sst_algo_controls,
ARRAY_SIZE(sst_algo_controls));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_add_platform_controls(platform, sst_slot_controls,
+ ARRAY_SIZE(sst_slot_controls));
+ if (ret)
+ return ret;
+
+ ret = sst_map_modules_to_pipe(platform);
+
return ret;
}
diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index a73e894b175c..dfebfdd5eb2a 100644
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -23,6 +23,9 @@
#ifndef __SST_ATOM_CONTROLS_H__
#define __SST_ATOM_CONTROLS_H__
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
enum {
MERR_DPCM_AUDIO = 0,
MERR_DPCM_COMPR,
@@ -360,16 +363,416 @@ struct sst_dsp_header {
struct sst_cmd_generic {
struct sst_dsp_header header;
} __packed;
+
+struct swm_input_ids {
+ struct sst_destination_id input_id;
+} __packed;
+
+struct sst_cmd_set_swm {
+ struct sst_dsp_header header;
+ struct sst_destination_id output_id;
+ u16 switch_state;
+ u16 nb_inputs;
+ struct swm_input_ids input[SST_CMD_SWM_MAX_INPUTS];
+} __packed;
+
+struct sst_cmd_set_media_path {
+ struct sst_dsp_header header;
+ u16 switch_state;
+} __packed;
+
+struct pcm_cfg {
+ u8 s_length:2;
+ u8 rate:3;
+ u8 format:3;
+} __packed;
+
+struct sst_cmd_set_speech_path {
+ struct sst_dsp_header header;
+ u16 switch_state;
+ struct {
+ u16 rsvd:8;
+ struct pcm_cfg cfg;
+ } config;
+} __packed;
+
+struct gain_cell {
+ struct sst_destination_id dest;
+ s16 cell_gain_left;
+ s16 cell_gain_right;
+ u16 gain_time_constant;
+} __packed;
+
+#define NUM_GAIN_CELLS 1
+struct sst_cmd_set_gain_dual {
+ struct sst_dsp_header header;
+ u16 gain_cell_num;
+ struct gain_cell cell_gains[NUM_GAIN_CELLS];
+} __packed;
struct sst_cmd_set_params {
struct sst_destination_id dst;
u16 command_id;
char params[0];
} __packed;
+
+
+struct sst_cmd_sba_vb_start {
+ struct sst_dsp_header header;
+} __packed;
+
+union sba_media_loop_params {
+ struct {
+ u16 rsvd:8;
+ struct pcm_cfg cfg;
+ } part;
+ u16 full;
+} __packed;
+
+struct sst_cmd_sba_set_media_loop_map {
+ struct sst_dsp_header header;
+ u16 switch_state;
+ union sba_media_loop_params param;
+ u16 map;
+} __packed;
+
+struct sst_cmd_tone_stop {
+ struct sst_dsp_header header;
+ u16 switch_state;
+} __packed;
+
+enum sst_ssp_mode {
+ SSP_MODE_MASTER = 0,
+ SSP_MODE_SLAVE = 1,
+};
+
+enum sst_ssp_pcm_mode {
+ SSP_PCM_MODE_NORMAL = 0,
+ SSP_PCM_MODE_NETWORK = 1,
+};
+
+enum sst_ssp_duplex {
+ SSP_DUPLEX = 0,
+ SSP_RX = 1,
+ SSP_TX = 2,
+};
+
+enum sst_ssp_fs_frequency {
+ SSP_FS_8_KHZ = 0,
+ SSP_FS_16_KHZ = 1,
+ SSP_FS_44_1_KHZ = 2,
+ SSP_FS_48_KHZ = 3,
+};
+
+enum sst_ssp_fs_polarity {
+ SSP_FS_ACTIVE_LOW = 0,
+ SSP_FS_ACTIVE_HIGH = 1,
+};
+
+enum sst_ssp_protocol {
+ SSP_MODE_PCM = 0,
+ SSP_MODE_I2S = 1,
+};
+
+enum sst_ssp_port_id {
+ SSP_MODEM = 0,
+ SSP_BT = 1,
+ SSP_FM = 2,
+ SSP_CODEC = 3,
+};
+
+struct sst_cmd_sba_hw_set_ssp {
+ struct sst_dsp_header header;
+ u16 selection; /* 0:SSP0(def), 1:SSP1, 2:SSP2 */
+
+ u16 switch_state;
+
+ u16 nb_bits_per_slots:6; /* 0-32 bits, 24 (def) */
+ u16 nb_slots:4; /* 0-8: slots per frame */
+ u16 mode:3; /* 0:Master, 1: Slave */
+ u16 duplex:3;
+
+ u16 active_tx_slot_map:8; /* Bit map, 0:off, 1:on */
+ u16 reserved1:8;
+
+ u16 active_rx_slot_map:8; /* Bit map 0: Off, 1:On */
+ u16 reserved2:8;
+
+ u16 frame_sync_frequency;
+
+ u16 frame_sync_polarity:8;
+ u16 data_polarity:8;
+
+ u16 frame_sync_width; /* 1 to N clocks */
+ u16 ssp_protocol:8;
+ u16 start_delay:8; /* Start delay in terms of clock ticks */
+} __packed;
+
+#define SST_MAX_TDM_SLOTS 8
+
+struct sst_param_sba_ssp_slot_map {
+ struct sst_dsp_header header;
+
+ u16 param_id;
+ u16 param_len;
+ u16 ssp_index;
+
+ u8 rx_slot_map[SST_MAX_TDM_SLOTS];
+ u8 tx_slot_map[SST_MAX_TDM_SLOTS];
+} __packed;
+
+enum {
+ SST_PROBE_EXTRACTOR = 0,
+ SST_PROBE_INJECTOR = 1,
+};
+
+/**** widget defines *****/
+
+#define SST_MODULE_GAIN 1
+#define SST_MODULE_ALGO 2
+
+#define SST_FMT_MONO 0
+#define SST_FMT_STEREO 3
+
+/* physical SSP numbers */
+enum {
+ SST_SSP0 = 0,
+ SST_SSP1,
+ SST_SSP2,
+ SST_SSP_LAST = SST_SSP2,
+};
+
+#define SST_NUM_SSPS (SST_SSP_LAST + 1) /* physical SSPs */
+#define SST_MAX_SSP_MUX 2 /* single SSP muxed between pipes */
+#define SST_MAX_SSP_DOMAINS 2 /* domains present in each pipe */
+
+struct sst_module {
+ struct snd_kcontrol *kctl;
+ struct list_head node;
+};
+
+struct sst_ssp_config {
+ u8 ssp_id;
+ u8 bits_per_slot;
+ u8 slots;
+ u8 ssp_mode;
+ u8 pcm_mode;
+ u8 duplex;
+ u8 ssp_protocol;
+ u8 fs_frequency;
+ u8 active_slot_map;
+ u8 start_delay;
+ u16 fs_width;
+};
+
+struct sst_ssp_cfg {
+ const u8 ssp_number;
+ const int *mux_shift;
+ const int (*domain_shift)[SST_MAX_SSP_MUX];
+ const struct sst_ssp_config (*ssp_config)[SST_MAX_SSP_MUX][SST_MAX_SSP_DOMAINS];
+};
+
+struct sst_ids {
+ u16 location_id;
+ u16 module_id;
+ u8 task_id;
+ u8 format;
+ u8 reg;
+ const char *parent_wname;
+ struct snd_soc_dapm_widget *parent_w;
+ struct list_head algo_list;
+ struct list_head gain_list;
+ const struct sst_pcm_format *pcm_fmt;
+};
+
+
+#define SST_AIF_IN(wname, wevent) \
+{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = NULL, \
+ .reg = SND_SOC_NOPM, .shift = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \
+}
+
+#define SST_AIF_OUT(wname, wevent) \
+{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = NULL, \
+ .reg = SND_SOC_NOPM, .shift = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \
+}
+
+#define SST_INPUT(wname, wevent) \
+{ .id = snd_soc_dapm_input, .name = wname, .sname = NULL, \
+ .reg = SND_SOC_NOPM, .shift = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \
+}
+
+#define SST_OUTPUT(wname, wevent) \
+{ .id = snd_soc_dapm_output, .name = wname, .sname = NULL, \
+ .reg = SND_SOC_NOPM, .shift = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \
+}
+
+#define SST_DAPM_OUTPUT(wname, wloc_id, wtask_id, wformat, wevent) \
+{ .id = snd_soc_dapm_output, .name = wname, .sname = NULL, \
+ .reg = SND_SOC_NOPM, .shift = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .priv = (void *)&(struct sst_ids) { .location_id = wloc_id, .task_id = wtask_id,\
+ .pcm_fmt = wformat, } \
+}
+
+#define SST_PATH(wname, wtask, wloc_id, wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \
+ .kcontrol_news = NULL, .num_kcontrols = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = wflags, \
+ .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, } \
+}
+
+#define SST_LINKED_PATH(wname, wtask, wloc_id, linked_wname, wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \
+ .kcontrol_news = NULL, .num_kcontrols = 0, \
+ .on_val = 1, .off_val = 0, \
+ .event = wevent, .event_flags = wflags, \
+ .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \
+ .parent_wname = linked_wname} \
+}
+
+#define SST_PATH_MEDIA_LOOP(wname, wtask, wloc_id, wformat, wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \
+ .kcontrol_news = NULL, .num_kcontrols = 0, \
+ .event = wevent, .event_flags = wflags, \
+ .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \
+ .format = wformat,} \
+}
+
+/* output is triggered before input */
+#define SST_PATH_INPUT(name, task_id, loc_id, event) \
+ SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
+
+#define SST_PATH_LINKED_INPUT(name, task_id, loc_id, linked_wname, event) \
+ SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event, \
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
+
+#define SST_PATH_OUTPUT(name, task_id, loc_id, event) \
+ SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)
+
+#define SST_PATH_LINKED_OUTPUT(name, task_id, loc_id, linked_wname, event) \
+ SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event, \
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)
+
+#define SST_PATH_MEDIA_LOOP_OUTPUT(name, task_id, loc_id, format, event) \
+ SST_PATH_MEDIA_LOOP(name, task_id, loc_id, format, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)
+
+
+#define SST_SWM_MIXER(wname, wreg, wtask, wloc_id, wcontrols, wevent) \
+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols),\
+ .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD | \
+ SND_SOC_DAPM_POST_REG, \
+ .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \
+ .reg = wreg } \
+}
+
+enum sst_gain_kcontrol_type {
+ SST_GAIN_TLV,
+ SST_GAIN_MUTE,
+ SST_GAIN_RAMP_DURATION,
+};
+
+struct sst_gain_mixer_control {
+ bool stereo;
+ enum sst_gain_kcontrol_type type;
+ struct sst_gain_value *gain_val;
+ int max;
+ int min;
+ u16 instance_id;
+ u16 module_id;
+ u16 pipe_id;
+ u16 task_id;
+ char pname[44];
+ struct snd_soc_dapm_widget *w;
+};
+
+struct sst_gain_value {
+ u16 ramp_duration;
+ s16 l_gain;
+ s16 r_gain;
+ bool mute;
+};
+#define SST_GAIN_VOLUME_DEFAULT (-1440)
+#define SST_GAIN_RAMP_DURATION_DEFAULT 5 /* timeconstant */
+#define SST_GAIN_MUTE_DEFAULT true
+
+#define SST_GAIN_KCONTROL_TLV(xname, xhandler_get, xhandler_put, \
+ xmod, xpipe, xinstance, xtask, tlv_array, xgain_val, \
+ xmin, xmax, xpname) \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = sst_gain_ctl_info,\
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct sst_gain_mixer_control) \
+ { .stereo = true, .max = xmax, .min = xmin, .type = SST_GAIN_TLV, \
+ .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\
+ .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname}
+
+#define SST_GAIN_KCONTROL_INT(xname, xhandler_get, xhandler_put, \
+ xmod, xpipe, xinstance, xtask, xtype, xgain_val, \
+ xmin, xmax, xpname) \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = sst_gain_ctl_info, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct sst_gain_mixer_control) \
+ { .stereo = false, .max = xmax, .min = xmin, .type = xtype, \
+ .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\
+ .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname}
+
+#define SST_GAIN_KCONTROL_BOOL(xname, xhandler_get, xhandler_put,\
+ xmod, xpipe, xinstance, xtask, xgain_val, xpname) \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_bool_ext, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct sst_gain_mixer_control) \
+ { .stereo = false, .type = SST_GAIN_MUTE, \
+ .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\
+ .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname}
#define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \
xpname " " xmname " " #xinstance " " xtype
#define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \
xpname " " xmname " " #xinstance " " xtype " " xsubmodule
+
+/*
+ * 3 Controls for each Gain module
+ * e.g. - pcm0_in Gain 0 Volume
+ * - pcm0_in Gain 0 Ramp Delay
+ * - pcm0_in Gain 0 Switch
+ */
+#define SST_GAIN_KCONTROLS(xpname, xmname, xmin_gain, xmax_gain, xmin_tc, xmax_tc, \
+ xhandler_get, xhandler_put, \
+ xmod, xpipe, xinstance, xtask, tlv_array, xgain_val) \
+ { SST_GAIN_KCONTROL_INT(SST_CONTROL_NAME(xpname, xmname, xinstance, "Ramp Delay"), \
+ xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, SST_GAIN_RAMP_DURATION, \
+ xgain_val, xmin_tc, xmax_tc, xpname) }, \
+ { SST_GAIN_KCONTROL_BOOL(SST_CONTROL_NAME(xpname, xmname, xinstance, "Switch"), \
+ xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, \
+ xgain_val, xpname) } ,\
+ { SST_GAIN_KCONTROL_TLV(SST_CONTROL_NAME(xpname, xmname, xinstance, "Volume"), \
+ xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, tlv_array, \
+ xgain_val, xmin_gain, xmax_gain, xpname) }
+
+#define SST_GAIN_TC_MIN 5
+#define SST_GAIN_TC_MAX 5000
+#define SST_GAIN_MIN_VALUE -1440 /* in 0.1 DB units */
+#define SST_GAIN_MAX_VALUE 360
+
enum sst_algo_kcontrol_type {
SST_ALGO_PARAMS,
SST_ALGO_BYPASS,
@@ -439,4 +842,29 @@ struct sst_enum {
struct snd_soc_dapm_widget *w;
};
+/* only 4 slots/channels supported atm */
+#define SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts) \
+ (struct sst_enum){ .reg = s_ch_no, .tx = is_tx, .max = 4+1, .texts = xtexts, }
+
+#define SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name) \
+ xpname " " xmname " " s_ch_name
+
+#define SST_SSP_SLOT_CTL(xpname, xmname, s_ch_name, s_ch_no, is_tx, xtexts, xget, xput) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name), \
+ .info = sst_slot_enum_info, \
+ .get = xget, .put = xput, \
+ .private_value = (unsigned long)&SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts), \
+}
+
+#define SST_MUX_CTL_NAME(xpname, xinstance) \
+ xpname " " #xinstance
+
+#define SST_SSP_MUX_ENUM(xreg, xshift, xtexts) \
+ (struct soc_enum) SOC_ENUM_DOUBLE(xreg, xshift, xshift, ARRAY_SIZE(xtexts), xtexts)
+
+#define SST_SSP_MUX_CTL(xpname, xinstance, xreg, xshift, xtexts) \
+ SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \
+ SST_SSP_MUX_ENUM(xreg, xshift, xtexts))
+
#endif
diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c
index fc588764ffa3..5a9e56700f31 100644
--- a/sound/soc/intel/sst-baytrail-dsp.c
+++ b/sound/soc/intel/sst-baytrail-dsp.c
@@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
{
struct dma_block_info *block;
struct sst_module *mod;
- struct sst_module_data block_data;
struct sst_module_template template;
int count;
memset(&template, 0, sizeof(template));
template.id = module->type;
template.entry = module->entry_point;
- template.p.type = SST_MEM_DRAM;
- template.p.data_type = SST_DATA_P;
- template.s.type = SST_MEM_DRAM;
- template.s.data_type = SST_DATA_S;
mod = sst_module_new(fw, &template, NULL);
if (mod == NULL)
@@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
switch (block->type) {
case SST_BYT_IRAM:
- block_data.offset = block->ram_offset +
+ mod->offset = block->ram_offset +
dsp->addr.iram_offset;
- block_data.type = SST_MEM_IRAM;
+ mod->type = SST_MEM_IRAM;
break;
case SST_BYT_DRAM:
- block_data.offset = block->ram_offset +
+ mod->offset = block->ram_offset +
dsp->addr.dram_offset;
- block_data.type = SST_MEM_DRAM;
+ mod->type = SST_MEM_DRAM;
break;
case SST_BYT_CACHE:
- block_data.offset = block->ram_offset +
+ mod->offset = block->ram_offset +
(dsp->addr.fw_ext - dsp->addr.lpe);
- block_data.type = SST_MEM_CACHE;
+ mod->type = SST_MEM_CACHE;
break;
default:
dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n",
@@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
return -EINVAL;
}
- block_data.size = block->size;
- block_data.data_type = SST_DATA_M;
- block_data.data = (void *)block + sizeof(*block);
+ mod->size = block->size;
+ mod->data = (void *)block + sizeof(*block);
- sst_module_insert_fixed_block(mod, &block_data);
+ sst_module_alloc_blocks(mod);
block = (void *)block + sizeof(*block) + block->size;
}
diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h
index ffb308bd81ce..b9da030e312d 100644
--- a/sound/soc/intel/sst-dsp-priv.h
+++ b/sound/soc/intel/sst-dsp-priv.h
@@ -26,6 +26,9 @@ struct sst_mem_block;
struct sst_module;
struct sst_fw;
+/* do we need to remove or keep */
+#define DSP_DRAM_ADDR_OFFSET 0x400000
+
/*
* DSP Operations exported by platform Audio DSP driver.
*/
@@ -33,6 +36,9 @@ struct sst_ops {
/* DSP core boot / reset */
void (*boot)(struct sst_dsp *);
void (*reset)(struct sst_dsp *);
+ int (*wake)(struct sst_dsp *);
+ void (*sleep)(struct sst_dsp *);
+ void (*stall)(struct sst_dsp *);
/* Shim IO */
void (*write)(void __iomem *addr, u32 offset, u32 value);
@@ -67,6 +73,8 @@ struct sst_addr {
u32 shim_offset;
u32 iram_offset;
u32 dram_offset;
+ u32 dsp_iram_offset;
+ u32 dsp_dram_offset;
void __iomem *lpe;
void __iomem *shim;
void __iomem *pci_cfg;
@@ -84,15 +92,6 @@ struct sst_mailbox {
};
/*
- * Audio DSP Firmware data types.
- */
-enum sst_data_type {
- SST_DATA_M = 0, /* module block data */
- SST_DATA_P = 1, /* peristant data (text, data) */
- SST_DATA_S = 2, /* scratch data (usually buffers) */
-};
-
-/*
* Audio DSP memory block types.
*/
enum sst_mem_type {
@@ -125,23 +124,6 @@ struct sst_fw {
};
/*
- * Audio DSP Generic Module data.
- *
- * This is used to dsecribe any sections of persistent (text and data) and
- * scratch (buffers) of module data in ADSP memory space.
- */
-struct sst_module_data {
-
- enum sst_mem_type type; /* destination memory type */
- enum sst_data_type data_type; /* type of module data */
-
- u32 size; /* size in bytes */
- int32_t offset; /* offset in FW file */
- u32 data_offset; /* offset in ADSP memory space */
- void *data; /* module data */
-};
-
-/*
* Audio DSP Generic Module Template.
*
* Used to define and register a new FW module. This data is extracted from
@@ -150,15 +132,52 @@ struct sst_module_data {
struct sst_module_template {
u32 id;
u32 entry; /* entry point */
- struct sst_module_data s; /* scratch data */
- struct sst_module_data p; /* peristant data */
+ u32 scratch_size;
+ u32 persistent_size;
+};
+
+/*
+ * Block Allocator - Used to allocate blocks of DSP memory.
+ */
+struct sst_block_allocator {
+ u32 id;
+ u32 offset;
+ int size;
+ enum sst_mem_type type;
+};
+
+/*
+ * Runtime Module Instance - A module object can be instanciated multiple
+ * times within the DSP FW.
+ */
+struct sst_module_runtime {
+ struct sst_dsp *dsp;
+ int id;
+ struct sst_module *module; /* parent module we belong too */
+
+ u32 persistent_offset; /* private memory offset */
+ void *private;
+
+ struct list_head list;
+ struct list_head block_list; /* list of blocks used */
+};
+
+/*
+ * Runtime Module Context - The runtime context must be manually stored by the
+ * driver prior to enter S3 and restored after leaving S3. This should really be
+ * part of the memory context saved by the enter D3 message IPC ???
+ */
+struct sst_module_runtime_context {
+ dma_addr_t dma_buffer;
+ u32 *buffer;
};
/*
* Audio DSP Generic Module.
*
* Each Firmware file can consist of 1..N modules. A module can span multiple
- * ADSP memory blocks. The simplest FW will be a file with 1 module.
+ * ADSP memory blocks. The simplest FW will be a file with 1 module. A module
+ * can be instanciated multiple times in the DSP.
*/
struct sst_module {
struct sst_dsp *dsp;
@@ -167,10 +186,13 @@ struct sst_module {
/* module configuration */
u32 id;
u32 entry; /* module entry point */
- u32 offset; /* module offset in firmware file */
+ s32 offset; /* module offset in firmware file */
u32 size; /* module size */
- struct sst_module_data s; /* scratch data */
- struct sst_module_data p; /* peristant data */
+ u32 scratch_size; /* global scratch memory required */
+ u32 persistent_size; /* private memory required */
+ enum sst_mem_type type; /* destination memory type */
+ u32 data_offset; /* offset in ADSP memory space */
+ void *data; /* module data */
/* runtime */
u32 usage_count; /* can be unloaded if count == 0 */
@@ -180,6 +202,7 @@ struct sst_module {
struct list_head block_list; /* Module list of blocks in use */
struct list_head list; /* DSP list of modules */
struct list_head list_fw; /* FW list of modules */
+ struct list_head runtime_list; /* list of runtime module objects*/
};
/*
@@ -208,7 +231,6 @@ struct sst_mem_block {
struct sst_block_ops *ops; /* block operations, if any */
/* block status */
- enum sst_data_type data_type; /* data type held in this block */
u32 bytes_used; /* bytes in use by modules */
void *private; /* generic core does not touch this */
int users; /* number of modules using this block */
@@ -253,6 +275,11 @@ struct sst_dsp {
struct list_head module_list;
struct list_head fw_list;
+ /* scratch buffer */
+ struct list_head scratch_block_list;
+ u32 scratch_offset;
+ u32 scratch_size;
+
/* platform data */
struct sst_pdata *pdata;
@@ -290,18 +317,33 @@ void sst_fw_unload(struct sst_fw *sst_fw);
/* Create/Free firmware modules */
struct sst_module *sst_module_new(struct sst_fw *sst_fw,
struct sst_module_template *template, void *private);
-void sst_module_free(struct sst_module *sst_module);
-int sst_module_insert(struct sst_module *sst_module);
-int sst_module_remove(struct sst_module *sst_module);
-int sst_module_insert_fixed_block(struct sst_module *module,
- struct sst_module_data *data);
+void sst_module_free(struct sst_module *module);
struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id);
-
-/* allocate/free pesistent/scratch memory regions managed by drv */
-struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp);
-void sst_mem_block_free_scratch(struct sst_dsp *dsp,
- struct sst_module *scratch);
-int sst_block_module_remove(struct sst_module *module);
+int sst_module_alloc_blocks(struct sst_module *module);
+int sst_module_free_blocks(struct sst_module *module);
+
+/* Create/Free firmware module runtime instances */
+struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module,
+ int id, void *private);
+void sst_module_runtime_free(struct sst_module_runtime *runtime);
+struct sst_module_runtime *sst_module_runtime_get_from_id(
+ struct sst_module *module, u32 id);
+int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime,
+ int offset);
+int sst_module_runtime_free_blocks(struct sst_module_runtime *runtime);
+int sst_module_runtime_save(struct sst_module_runtime *runtime,
+ struct sst_module_runtime_context *context);
+int sst_module_runtime_restore(struct sst_module_runtime *runtime,
+ struct sst_module_runtime_context *context);
+
+/* generic block allocation */
+int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba,
+ struct list_head *block_list);
+int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list);
+
+/* scratch allocation */
+int sst_block_alloc_scratch(struct sst_dsp *dsp);
+void sst_block_free_scratch(struct sst_dsp *dsp);
/* Register the DSPs