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authorLinus Torvalds <torvalds@linux-foundation.org>2013-05-03 09:10:23 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2013-05-03 09:10:23 -0700
commit9992ba72327fa0d8bdc9fb624e80f5cce338a711 (patch)
treee0bf31ae53cb19c44674df7e0d0343a26037ad34 /sound
parent00fdffb5131125dce0702bf61e24a806ec3aed80 (diff)
parent4ca231b2e6ed171107c5b21f9e92d1965fd6fd9e (diff)
downloadkernel-9992ba72327fa0d8bdc9fb624e80f5cce338a711.tar.gz
Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "Mostly many small changes spread as seen in diffstat in sound/* directory by this update. A significant change in the subsystem level is the introduction of snd_soc_component, which will help more generic handling of SoC and off-SoC components. Also, snd_BUG_ON() macro is enabled unconditionally now due to its misuses, so people might hit kernel warnings (it's a good thing for us). - compress-offload: support for capture by Charles Keepax - HD-audio: codec delay support by Dylan Reid - HD-audio: improvements/fixes in generic parser: better headphone mic and headset mic support, jack_modes hint consolidation, proper beep attach/detachment, generalized power filter controls by David Henningsson, et al - HD-audio: Improved management of HDMI codec pins/converters - HD-audio: Better pin/DAC assignment for VIA codecs - HD-audio: Haswell HDMI workarounds - HD-audio: ALC268 codec support, a few new quirks for Chromebooks - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency fix by Clemens Ladisch - USB: support for DSD formats by Daniel Mack - USB: A few UAC2 device endian/cock fixes by Eldad Zack - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller, Yamaha THRxx devices - HDSPM: updates for TCO controls by Adrian Knoth - ASoC: Add a snd_soc_component object type for generic handling of SoC and off-SoC components by Kuninori Morimoto, - dmaengine: a large set of cleanups and conversions by Lars-Peter Clausen - ASoC DAPM: performance optimizations from Ryo Tsutsui - ASoC DAPM: support for mixer control sharing by Stephen Warren - ASoC: multiplatform ARM cleanups from Arnd Bergmann - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack" * tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits) ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats ALSA: pcm_format_to_bits strong-typed conversion ALSA: compress: fix the states to check for allowing read ALSA: hda - Move Thinkpad X220 to use auto parser ALSA: USB: adjust for changed 3.8 USB API ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources sound: oss/dmabuf: use dma_map_single ALSA: ali5451: use mdelay instead of large udelay constants ALSA: hda - Add the support for ALC286 codec ALSA: usb-audio: USB quirk for Yamaha THR10C ALSA: usb-audio: USB quirk for Yamaha THR5A ALSA: usb-audio: USB quirk for Yamaha THR10 ALSA: usb-audio: Fix autopm error during probing ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT ALSA: sound kconfig typo ALSA: emu10k1: Fix dock firmware loading ASoC: ux500: forward declare msp_i2s_platform_data ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers ...
Diffstat (limited to 'sound')
-rw-r--r--sound/aoa/soundbus/i2sbus/pcm.c2
-rw-r--r--sound/atmel/ac97c.c4
-rw-r--r--sound/core/compress_offload.c128
-rw-r--r--sound/core/control.c41
-rw-r--r--sound/core/device.c8
-rw-r--r--sound/core/hwdep.c2
-rw-r--r--sound/core/info.c41
-rw-r--r--sound/core/init.c16
-rw-r--r--sound/core/isadma.c2
-rw-r--r--sound/core/jack.c6
-rw-r--r--sound/core/memalloc.c20
-rw-r--r--sound/core/memory.c4
-rw-r--r--sound/core/pcm.c8
-rw-r--r--sound/core/pcm_lib.c54
-rw-r--r--sound/core/pcm_memory.c19
-rw-r--r--sound/core/pcm_misc.c36
-rw-r--r--sound/core/pcm_native.c13
-rw-r--r--sound/core/rawmidi.c14
-rw-r--r--sound/core/sound.c7
-rw-r--r--sound/core/vmaster.c7
-rw-r--r--sound/drivers/Kconfig2
-rw-r--r--sound/drivers/aloop.c2
-rw-r--r--sound/drivers/mpu401/mpu401_uart.c6
-rw-r--r--sound/oss/dmabuf.c3
-rw-r--r--sound/oss/sb_common.c3
-rw-r--r--sound/oss/uart401.c11
-rw-r--r--sound/pci/ac97/ac97_codec.c16
-rw-r--r--sound/pci/ac97/ac97_pcm.c10
-rw-r--r--sound/pci/ali5451/ali5451.c8
-rw-r--r--sound/pci/asihpi/asihpi.c6
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c39
-rw-r--r--sound/pci/hda/hda_auto_parser.c68
-rw-r--r--sound/pci/hda/hda_auto_parser.h29
-rw-r--r--sound/pci/hda/hda_beep.c39
-rw-r--r--sound/pci/hda/hda_beep.h1
-rw-r--r--sound/pci/hda/hda_codec.c28
-rw-r--r--sound/pci/hda/hda_codec.h3
-rw-r--r--sound/pci/hda/hda_generic.c433
-rw-r--r--sound/pci/hda/hda_generic.h30
-rw-r--r--sound/pci/hda/hda_intel.c39
-rw-r--r--sound/pci/hda/hda_jack.c43
-rw-r--r--sound/pci/hda/hda_local.h4
-rw-r--r--sound/pci/hda/patch_analog.c33
-rw-r--r--sound/pci/hda/patch_ca0132.c82
-rw-r--r--sound/pci/hda/patch_cirrus.c19
-rw-r--r--sound/pci/hda/patch_conexant.c21
-rw-r--r--sound/pci/hda/patch_hdmi.c236
-rw-r--r--sound/pci/hda/patch_realtek.c715
-rw-r--r--sound/pci/hda/patch_sigmatel.c39
-rw-r--r--sound/pci/hda/patch_via.c20
-rw-r--r--sound/pci/rme9652/hdspm.c382
-rw-r--r--sound/soc/Kconfig4
-rw-r--r--sound/soc/Makefile4
-rw-r--r--sound/soc/atmel/atmel-pcm-dma.c29
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c54
-rw-r--r--sound/soc/au1x/ac97c.c9
-rw-r--r--sound/soc/au1x/i2sc.c9
-rw-r--r--sound/soc/au1x/psc-ac97.c9
-rw-r--r--sound/soc/au1x/psc-i2s.c9
-rw-r--r--sound/soc/blackfin/bf5xx-ac97.c9
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c9
-rw-r--r--sound/soc/blackfin/bf5xx-tdm.c9
-rw-r--r--sound/soc/blackfin/bf6xx-i2s.c9
-rw-r--r--sound/soc/cirrus/edb93xx.c1
-rw-r--r--sound/soc/cirrus/ep93xx-ac97.c18
-rw-r--r--sound/soc/cirrus/ep93xx-i2s.c25
-rw-r--r--sound/soc/cirrus/ep93xx-pcm.c38
-rw-r--r--sound/soc/cirrus/ep93xx-pcm.h20
-rw-r--r--sound/soc/cirrus/simone.c2
-rw-r--r--sound/soc/cirrus/snappercl15.c1
-rw-r--r--sound/soc/codecs/Kconfig10
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/adau1373.c5
-rw-r--r--sound/soc/codecs/ak4104.c55
-rw-r--r--sound/soc/codecs/ak5386.c152
-rw-r--r--sound/soc/codecs/arizona.c497
-rw-r--r--sound/soc/codecs/arizona.h31
-rw-r--r--sound/soc/codecs/cs4271.c166
-rw-r--r--sound/soc/codecs/cs42l73.c6
-rw-r--r--sound/soc/codecs/max98088.c32
-rw-r--r--sound/soc/codecs/max98090.c45
-rw-r--r--sound/soc/codecs/si476x.c47
-rw-r--r--sound/soc/codecs/tas5086.c591
-rw-r--r--sound/soc/codecs/wm0010.c6
-rw-r--r--sound/soc/codecs/wm2000.c4
-rw-r--r--sound/soc/codecs/wm2000.h2
-rw-r--r--sound/soc/codecs/wm2200.c2
-rw-r--r--sound/soc/codecs/wm5102.c121
-rw-r--r--sound/soc/codecs/wm5102.h6
-rw-r--r--sound/soc/codecs/wm5110.c46
-rw-r--r--sound/soc/codecs/wm5110.h6
-rw-r--r--sound/soc/codecs/wm8903.c4
-rw-r--r--sound/soc/codecs/wm8960.c10
-rw-r--r--sound/soc/codecs/wm8994.c68
-rw-r--r--sound/soc/codecs/wm8994.h3
-rw-r--r--sound/soc/codecs/wm_adsp.c124
-rw-r--r--sound/soc/codecs/wm_adsp.h5
-rw-r--r--sound/soc/codecs/wm_hubs.c9
-rw-r--r--sound/soc/davinci/davinci-i2s.c15
-rw-r--r--sound/soc/davinci/davinci-mcasp.c110
-rw-r--r--sound/soc/davinci/davinci-mcasp.h2
-rw-r--r--sound/soc/davinci/davinci-pcm.c9
-rw-r--r--sound/soc/davinci/davinci-vcif.c11
-rw-r--r--sound/soc/dwc/designware_i2s.c9
-rw-r--r--sound/soc/fsl/Kconfig2
-rw-r--r--sound/soc/fsl/fsl_ssi.c64
-rw-r--r--sound/soc/fsl/fsl_ssi.h8
-rw-r--r--sound/soc/fsl/imx-audmux.c3
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c120
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c6
-rw-r--r--sound/soc/fsl/imx-pcm.c6
-rw-r--r--sound/soc/fsl/imx-pcm.h24
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c2
-rw-r--r--sound/soc/fsl/imx-ssi.c62
-rw-r--r--sound/soc/fsl/imx-ssi.h7
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c8
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c9
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c9
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c11
-rw-r--r--sound/soc/mid-x86/sst_platform.c20
-rw-r--r--sound/soc/mid-x86/sst_platform.h2
-rw-r--r--sound/soc/mxs/Kconfig2
-rw-r--r--sound/soc/mxs/mxs-pcm.c176
-rw-r--r--sound/soc/mxs/mxs-pcm.h4
-rw-r--r--sound/soc/mxs/mxs-saif.c20
-rw-r--r--sound/soc/nuc900/nuc900-ac97.c9
-rw-r--r--sound/soc/omap/am3517evm.c1
-rw-r--r--sound/soc/omap/ams-delta.c1
-rw-r--r--sound/soc/omap/mcbsp.c14
-rw-r--r--sound/soc/omap/mcbsp.h7
-rw-r--r--sound/soc/omap/n810.c1
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c1
-rw-r--r--sound/soc/omap/omap-dmic.c47
-rw-r--r--sound/soc/omap/omap-hdmi.c33
-rw-r--r--sound/soc/omap/omap-mcbsp.c27
-rw-r--r--sound/soc/omap/omap-mcpdm.c118
-rw-r--r--sound/soc/omap/omap-pcm.c86
-rw-r--r--sound/soc/omap/omap-pcm.h40
-rw-r--r--sound/soc/omap/omap-twl4030.c1
-rw-r--r--sound/soc/omap/omap3pandora.c9
-rw-r--r--sound/soc/omap/osk5912.c1
-rw-r--r--sound/soc/omap/rx51.c1
-rw-r--r--sound/soc/pxa/mmp-pcm.c34
-rw-r--r--sound/soc/pxa/mmp-sspa.c9
-rw-r--r--sound/soc/pxa/pxa-ssp.c9
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c11
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c9
-rw-r--r--sound/soc/s6000/s6000-i2s.c9
-rw-r--r--sound/soc/samsung/Kconfig5
-rw-r--r--sound/soc/samsung/ac97.c14
-rw-r--r--sound/soc/samsung/goni_wm8994.c11
-rw-r--r--sound/soc/samsung/h1940_uda1380.c2
-rw-r--r--sound/soc/samsung/i2s.c16
-rw-r--r--sound/soc/samsung/idma.c11
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c3
-rw-r--r--sound/soc/samsung/pcm.c11
-rw-r--r--sound/soc/samsung/regs-ac97.h67
-rw-r--r--sound/soc/samsung/regs-iis.h70
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c2
-rw-r--r--sound/soc/samsung/s3c-i2s-v2.c9
-rw-r--r--sound/soc/samsung/s3c-i2s-v2.h7
-rw-r--r--sound/soc/samsung/s3c2412-i2s.c12
-rw-r--r--sound/soc/samsung/s3c24xx-i2s.c13
-rw-r--r--sound/soc/samsung/s3c24xx_uda134x.c2
-rw-r--r--sound/soc/samsung/spdif.c11
-rw-r--r--sound/soc/sh/fsi.c14
-rw-r--r--sound/soc/sh/hac.c10
-rw-r--r--sound/soc/sh/migor.c2
-rw-r--r--sound/soc/sh/siu_dai.c11
-rw-r--r--sound/soc/sh/ssi.c10
-rw-r--r--sound/soc/soc-compress.c38
-rw-r--r--sound/soc/soc-core.c213
-rw-r--r--sound/soc/soc-dapm.c320
-rw-r--r--sound/soc/soc-dmaengine-pcm.c150
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c300
-rw-r--r--sound/soc/soc-io.c5
-rw-r--r--sound/soc/soc-utils.c25
-rw-r--r--sound/soc/spear/spdif_in.c9
-rw-r--r--sound/soc/spear/spdif_out.c9
-rw-r--r--sound/soc/spear/spear_pcm.c23
-rw-r--r--sound/soc/tegra/Kconfig2
-rw-r--r--sound/soc/tegra/tegra20_ac97.c28
-rw-r--r--sound/soc/tegra/tegra20_ac97.h4
-rw-r--r--sound/soc/tegra/tegra20_i2s.c28
-rw-r--r--sound/soc/tegra/tegra20_i2s.h4
-rw-r--r--sound/soc/tegra/tegra20_spdif.c22
-rw-r--r--sound/soc/tegra/tegra20_spdif.h4
-rw-r--r--sound/soc/tegra/tegra30_ahub.c72
-rw-r--r--sound/soc/tegra/tegra30_ahub.h24
-rw-r--r--sound/soc/tegra/tegra30_i2s.c28
-rw-r--r--sound/soc/tegra/tegra30_i2s.h4
-rw-r--r--sound/soc/tegra/tegra_alc5632.c15
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.c34
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h1
-rw-r--r--sound/soc/tegra/tegra_pcm.c187
-rw-r--r--sound/soc/tegra/tegra_pcm.h7
-rw-r--r--sound/soc/tegra/tegra_wm8753.c15
-rw-r--r--sound/soc/tegra/tegra_wm8903.c179
-rw-r--r--sound/soc/tegra/tegra_wm9712.c7
-rw-r--r--sound/soc/tegra/trimslice.c56
-rw-r--r--sound/soc/txx9/txx9aclc-ac97.c9
-rw-r--r--sound/soc/ux500/Kconfig2
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c13
-rw-r--r--sound/soc/ux500/ux500_msp_dai.h5
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.h1
-rw-r--r--sound/soc/ux500/ux500_pcm.c229
-rw-r--r--sound/soc/ux500/ux500_pcm.h14
-rw-r--r--sound/sound_core.c22
-rw-r--r--sound/spi/at73c213.c22
-rw-r--r--sound/usb/6fire/pcm.c1
-rw-r--r--sound/usb/caiaq/audio.c462
-rw-r--r--sound/usb/caiaq/audio.h4
-rw-r--r--sound/usb/caiaq/control.c67
-rw-r--r--sound/usb/caiaq/control.h2
-rw-r--r--sound/usb/caiaq/device.c248
-rw-r--r--sound/usb/caiaq/device.h18
-rw-r--r--sound/usb/caiaq/input.c336
-rw-r--r--sound/usb/caiaq/input.h6
-rw-r--r--sound/usb/caiaq/midi.c63
-rw-r--r--sound/usb/caiaq/midi.h5
-rw-r--r--sound/usb/card.c15
-rw-r--r--sound/usb/card.h9
-rw-r--r--sound/usb/clock.c189
-rw-r--r--sound/usb/clock.h3
-rw-r--r--sound/usb/endpoint.c32
-rw-r--r--sound/usb/endpoint.h2
-rw-r--r--sound/usb/format.c30
-rw-r--r--sound/usb/format.h2
-rw-r--r--sound/usb/helper.c10
-rw-r--r--sound/usb/midi.c25
-rw-r--r--sound/usb/misc/ua101.c3
-rw-r--r--sound/usb/pcm.c158
-rw-r--r--sound/usb/proc.c9
-rw-r--r--sound/usb/quirks-table.h121
-rw-r--r--sound/usb/quirks.c60
-rw-r--r--sound/usb/quirks.h5
-rw-r--r--sound/usb/stream.c23
-rw-r--r--sound/usb/usbaudio.h1
-rw-r--r--sound/usb/usx2y/usb_stream.c1
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c1
-rw-r--r--sound/usb/usx2y/usx2yhwdeppcm.c1
241 files changed, 7059 insertions, 3622 deletions
diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c
index 19491ed9292f..7b74a4ba75f8 100644
--- a/sound/aoa/soundbus/i2sbus/pcm.c
+++ b/sound/aoa/soundbus/i2sbus/pcm.c
@@ -179,7 +179,7 @@ static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in)
*/
if (other->active) {
/* FIXME: is this guaranteed by the alsa api? */
- hw->formats &= (1ULL << i2sdev->format);
+ hw->formats &= pcm_format_to_bits(i2sdev->format);
/* see above, restrict rates to the one we already have */
hw->rate_min = i2sdev->rate;
hw->rate_max = i2sdev->rate;
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 79d6bda58753..6b7e2b5a72de 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -182,7 +182,7 @@ static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream)
runtime->hw.rate_max = chip->cur_rate;
}
if (chip->cur_format)
- runtime->hw.formats = (1ULL << chip->cur_format);
+ runtime->hw.formats = pcm_format_to_bits(chip->cur_format);
mutex_unlock(&opened_mutex);
chip->playback_substream = substream;
return 0;
@@ -201,7 +201,7 @@ static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream)
runtime->hw.rate_max = chip->cur_rate;
}
if (chip->cur_format)
- runtime->hw.formats = (1ULL << chip->cur_format);
+ runtime->hw.formats = pcm_format_to_bits(chip->cur_format);
mutex_unlock(&opened_mutex);
chip->capture_substream = substream;
return 0;
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index c84abc886e90..99db892d7299 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -28,11 +28,13 @@
#include <linux/file.h>
#include <linux/fs.h>
#include <linux/list.h>
+#include <linux/math64.h>
#include <linux/mm.h>
#include <linux/mutex.h>
#include <linux/poll.h>
#include <linux/slab.h>
#include <linux/sched.h>
+#include <linux/types.h>
#include <linux/uio.h>
#include <linux/uaccess.h>
#include <linux/module.h>
@@ -152,26 +154,23 @@ static int snd_compr_update_tstamp(struct snd_compr_stream *stream,
stream->ops->pointer(stream, tstamp);
pr_debug("dsp consumed till %d total %d bytes\n",
tstamp->byte_offset, tstamp->copied_total);
- stream->runtime->hw_pointer = tstamp->byte_offset;
- stream->runtime->total_bytes_transferred = tstamp->copied_total;
+ if (stream->direction == SND_COMPRESS_PLAYBACK)
+ stream->runtime->total_bytes_transferred = tstamp->copied_total;
+ else
+ stream->runtime->total_bytes_available = tstamp->copied_total;
return 0;
}
static size_t snd_compr_calc_avail(struct snd_compr_stream *stream,
struct snd_compr_avail *avail)
{
- long avail_calc; /*this needs to be signed variable */
-
memset(avail, 0, sizeof(*avail));
snd_compr_update_tstamp(stream, &avail->tstamp);
/* Still need to return avail even if tstamp can't be filled in */
- /* FIXME: This needs to be different for capture stream,
- available is # of compressed data, for playback it's
- remainder of buffer */
-
if (stream->runtime->total_bytes_available == 0 &&
- stream->runtime->state == SNDRV_PCM_STATE_SETUP) {
+ stream->runtime->state == SNDRV_PCM_STATE_SETUP &&
+ stream->direction == SND_COMPRESS_PLAYBACK) {
pr_debug("detected init and someone forgot to do a write\n");
return stream->runtime->buffer_size;
}
@@ -180,26 +179,22 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream,
stream->runtime->total_bytes_transferred);
if (stream->runtime->total_bytes_available ==
stream->runtime->total_bytes_transferred) {
- pr_debug("both pointers are same, returning full avail\n");
- return stream->runtime->buffer_size;
+ if (stream->direction == SND_COMPRESS_PLAYBACK) {
+ pr_debug("both pointers are same, returning full avail\n");
+ return stream->runtime->buffer_size;
+ } else {
+ pr_debug("both pointers are same, returning no avail\n");
+ return 0;
+ }
}
- /* FIXME: this routine isn't consistent, in one test we use
- * cumulative values and in the other byte offsets. Do we
- * really need the byte offsets if the cumulative values have
- * been updated? In the PCM interface app_ptr and hw_ptr are
- * already cumulative */
+ avail->avail = stream->runtime->total_bytes_available -
+ stream->runtime->total_bytes_transferred;
+ if (stream->direction == SND_COMPRESS_PLAYBACK)
+ avail->avail = stream->runtime->buffer_size - avail->avail;
- avail_calc = stream->runtime->buffer_size -
- (stream->runtime->app_pointer - stream->runtime->hw_pointer);
- pr_debug("calc avail as %ld, app_ptr %lld, hw+ptr %lld\n", avail_calc,
- stream->runtime->app_pointer,
- stream->runtime->hw_pointer);
- if (avail_calc >= stream->runtime->buffer_size)
- avail_calc -= stream->runtime->buffer_size;
- pr_debug("ret avail as %ld\n", avail_calc);
- avail->avail = avail_calc;
- return avail_calc;
+ pr_debug("ret avail as %lld\n", avail->avail);
+ return avail->avail;
}
static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream)
@@ -230,21 +225,24 @@ static int snd_compr_write_data(struct snd_compr_stream *stream,
void *dstn;
size_t copy;
struct snd_compr_runtime *runtime = stream->runtime;
+ /* 64-bit Modulus */
+ u64 app_pointer = div64_u64(runtime->total_bytes_available,
+ runtime->buffer_size);
+ app_pointer = runtime->total_bytes_available -
+ (app_pointer * runtime->buffer_size);
- dstn = runtime->buffer + runtime->app_pointer;
+ dstn = runtime->buffer + app_pointer;
pr_debug("copying %ld at %lld\n",
- (unsigned long)count, runtime->app_pointer);
- if (count < runtime->buffer_size - runtime->app_pointer) {
+ (unsigned long)count, app_pointer);
+ if (count < runtime->buffer_size - app_pointer) {
if (copy_from_user(dstn, buf, count))
return -EFAULT;
- runtime->app_pointer += count;
} else {
- copy = runtime->buffer_size - runtime->app_pointer;
+ copy = runtime->buffer_size - app_pointer;
if (copy_from_user(dstn, buf, copy))
return -EFAULT;
if (copy_from_user(runtime->buffer, buf + copy, count - copy))
return -EFAULT;
- runtime->app_pointer = count - copy;
}
/* if DSP cares, let it know data has been written */
if (stream->ops->ack)
@@ -278,10 +276,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf,
if (avail > count)
avail = count;
- if (stream->ops->copy)
- retval = stream->ops->copy(stream, buf, avail);
- else
+ if (stream->ops->copy) {
+ char __user* cbuf = (char __user*)buf;
+ retval = stream->ops->copy(stream, cbuf, avail);
+ } else {
retval = snd_compr_write_data(stream, buf, avail);
+ }
if (retval > 0)
stream->runtime->total_bytes_available += retval;
@@ -300,7 +300,49 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf,
static ssize_t snd_compr_read(struct file *f, char __user *buf,
size_t count, loff_t *offset)
{
- return -ENXIO;
+ struct snd_compr_file *data = f->private_data;
+ struct snd_compr_stream *stream;
+ size_t avail;
+ int retval;
+
+ if (snd_BUG_ON(!data))
+ return -EFAULT;
+
+ stream = &data->stream;
+ mutex_lock(&stream->device->lock);
+
+ /* read is allowed when stream is running, paused, draining and setup
+ * (yes setup is state which we transition to after stop, so if user
+ * wants to read data after stop we allow that)
+ */
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_XRUN:
+ case SNDRV_PCM_STATE_SUSPENDED:
+ case SNDRV_PCM_STATE_DISCONNECTED:
+ retval = -EBADFD;
+ goto out;
+ }
+
+ avail = snd_compr_get_avail(stream);
+ pr_debug("avail returned %ld\n", (unsigned long)avail);
+ /* calculate how much we can read from buffer */
+ if (avail > count)
+ avail = count;
+
+ if (stream->ops->copy) {
+ retval = stream->ops->copy(stream, buf, avail);
+ } else {
+ retval = -ENXIO;
+ goto out;
+ }
+ if (retval > 0)
+ stream->runtime->total_bytes_transferred += retval;
+
+out:
+ mutex_unlock(&stream->device->lock);
+ return retval;
}
static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma)
@@ -375,6 +417,7 @@ snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg)
if (!stream->ops->get_caps)
return -ENXIO;
+ memset(&caps, 0, sizeof(caps));
retval = stream->ops->get_caps(stream, &caps);
if (retval)
goto out;
@@ -393,7 +436,7 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg)
if (!stream->ops->get_codec_caps)
return -ENXIO;
- caps = kmalloc(sizeof(*caps), GFP_KERNEL);
+ caps = kzalloc(sizeof(*caps), GFP_KERNEL);
if (!caps)
return -ENOMEM;
@@ -485,9 +528,14 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
retval = stream->ops->set_params(stream, params);
if (retval)
goto out;
- stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+
stream->metadata_set = false;
stream->next_track = false;
+
+ if (stream->direction == SND_COMPRESS_PLAYBACK)
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ else
+ stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
} else {
return -EPERM;
}
@@ -505,7 +553,7 @@ snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg)
if (!stream->ops->get_params)
return -EBADFD;
- params = kmalloc(sizeof(*params), GFP_KERNEL);
+ params = kzalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
retval = stream->ops->get_params(stream, params);
@@ -622,8 +670,6 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
if (!retval) {
stream->runtime->state = SNDRV_PCM_STATE_SETUP;
wake_up(&stream->runtime->sleep);
- stream->runtime->hw_pointer = 0;
- stream->runtime->app_pointer = 0;
stream->runtime->total_bytes_available = 0;
stream->runtime->total_bytes_transferred = 0;
}
diff --git a/sound/core/control.c b/sound/core/control.c
index 8c7c2c9bba61..d8aa206e8bde 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -190,7 +190,7 @@ EXPORT_SYMBOL(snd_ctl_notify);
* Allocates a new struct snd_kcontrol instance and copies the given template
* to the new instance. It does not copy volatile data (access).
*
- * Returns the pointer of the new instance, or NULL on failure.
+ * Return: The pointer of the new instance, or %NULL on failure.
*/
static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control,
unsigned int access)
@@ -224,7 +224,7 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control,
* template. When the access field of ncontrol is 0, it's assumed as
* READWRITE access. When the count field is 0, it's assumes as one.
*
- * Returns the pointer of the newly generated instance, or NULL on failure.
+ * Return: The pointer of the newly generated instance, or %NULL on failure.
*/
struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol,
void *private_data)
@@ -322,9 +322,10 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count)
* snd_ctl_new1() to the given card. Assigns also an unique
* numid used for fast search.
*
- * Returns zero if successful, or a negative error code on failure.
- *
* It frees automatically the control which cannot be added.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
+ *
*/
int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol)
{
@@ -380,9 +381,9 @@ EXPORT_SYMBOL(snd_ctl_add);
* and the add_on_replace flag is set, the control is added. If the
* control exists, it is destroyed first.
*
- * Returns zero if successful, or a negative error code on failure.
- *
* It frees automatically the control which cannot be added or replaced.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol,
bool add_on_replace)
@@ -442,8 +443,8 @@ EXPORT_SYMBOL(snd_ctl_replace);
* Removes the control from the card and then releases the instance.
* You don't need to call snd_ctl_free_one(). You must be in
* the write lock - down_write(&card->controls_rwsem).
- *
- * Returns 0 if successful, or a negative error code on failure.
+ *
+ * Return: 0 if successful, or a negative error code on failure.
*/
int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol)
{
@@ -470,8 +471,8 @@ EXPORT_SYMBOL(snd_ctl_remove);
*
* Finds the control instance with the given id, removes it from the
* card list and releases it.
- *
- * Returns 0 if successful, or a negative error code on failure.
+ *
+ * Return: 0 if successful, or a negative error code on failure.
*/
int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id)
{
@@ -498,8 +499,8 @@ EXPORT_SYMBOL(snd_ctl_remove_id);
*
* Finds the control instance with the given id, removes it from the
* card list and releases it.
- *
- * Returns 0 if successful, or a negative error code on failure.
+ *
+ * Return: 0 if successful, or a negative error code on failure.
*/
static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file,
struct snd_ctl_elem_id *id)
@@ -541,7 +542,7 @@ error:
* Finds the control instance with the given id, and activate or
* inactivate the control together with notification, if changed.
*
- * Returns 0 if unchanged, 1 if changed, or a negative error code on failure.
+ * Return: 0 if unchanged, 1 if changed, or a negative error code on failure.
*/
int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
int active)
@@ -587,7 +588,7 @@ EXPORT_SYMBOL_GPL(snd_ctl_activate_id);
* Finds the control with the old id from the card, and replaces the
* id with the new one.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id,
struct snd_ctl_elem_id *dst_id)
@@ -616,10 +617,11 @@ EXPORT_SYMBOL(snd_ctl_rename_id);
*
* Finds the control instance with the given number-id from the card.
*
- * Returns the pointer of the instance if found, or NULL if not.
- *
* The caller must down card->controls_rwsem before calling this function
* (if the race condition can happen).
+ *
+ * Return: The pointer of the instance if found, or %NULL if not.
+ *
*/
struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid)
{
@@ -643,10 +645,11 @@ EXPORT_SYMBOL(snd_ctl_find_numid);
*
* Finds the control instance with the given id from the card.
*
- * Returns the pointer of the instance if found, or NULL if not.
- *
* The caller must down card->controls_rwsem before calling this function
* (if the race condition can happen).
+ *
+ * Return: The pointer of the instance if found, or %NULL if not.
+ *
*/
struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card,
struct snd_ctl_elem_id *id)
@@ -1710,6 +1713,8 @@ EXPORT_SYMBOL(snd_ctl_boolean_stereo_info);
* Sets all required fields in @info to their appropriate values.
* If the control's accessibility is not the default (readable and writable),
* the caller has to fill @info->access.
+ *
+ * Return: Zero.
*/
int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
unsigned int items, const char *const names[])
diff --git a/sound/core/device.c b/sound/core/device.c
index f03cb5444a5a..df88defed176 100644
--- a/sound/core/device.c
+++ b/sound/core/device.c
@@ -39,7 +39,7 @@
* The data pointer plays a role as the identifier, too, so the
* pointer address must be unique and unchanged.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_device_new(struct snd_card *card, snd_device_type_t type,
void *device_data, struct snd_device_ops *ops)
@@ -73,7 +73,7 @@ EXPORT_SYMBOL(snd_device_new);
* callbacks, dev_disconnect and dev_free, corresponding to the state.
* Then release the device.
*
- * Returns zero if successful, or a negative error code on failure or if the
+ * Return: Zero if successful, or a negative error code on failure or if the
* device not found.
*/
int snd_device_free(struct snd_card *card, void *device_data)
@@ -116,7 +116,7 @@ EXPORT_SYMBOL(snd_device_free);
*
* Usually called from snd_card_disconnect().
*
- * Returns zero if successful, or a negative error code on failure or if the
+ * Return: Zero if successful, or a negative error code on failure or if the
* device not found.
*/
int snd_device_disconnect(struct snd_card *card, void *device_data)
@@ -151,7 +151,7 @@ int snd_device_disconnect(struct snd_card *card, void *device_data)
* but it can be called later if any new devices are created after
* invocation of snd_card_register().
*
- * Returns zero if successful, or a negative error code on failure or if the
+ * Return: Zero if successful, or a negative error code on failure or if the
* device not found.
*/
int snd_device_register(struct snd_card *card, void *device_data)
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index 3f7f6628cf7b..d105073298cb 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -356,7 +356,7 @@ static const struct file_operations snd_hwdep_f_ops =
* The callbacks (hwdep->ops) must be set on the returned instance
* after this call manually by the caller.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_hwdep_new(struct snd_card *card, char *id, int device,
struct snd_hwdep **rhwdep)
diff --git a/sound/core/info.c b/sound/core/info.c
index 3c9bd6b10a96..e79baa11b60e 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -89,7 +89,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer,
char *nbuf;
nsize = PAGE_ALIGN(nsize);
- nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL);
+ nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL | __GFP_ZERO);
if (! nbuf)
return -ENOMEM;
@@ -105,7 +105,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer,
*
* Outputs the string on the procfs buffer just like printf().
*
- * Returns the size of output string.
+ * Return: The size of output string, or a negative error code.
*/
int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...)
{
@@ -344,7 +344,7 @@ static int snd_info_entry_open(struct inode *inode, struct file *file)
goto __nomem;
data->rbuffer = buffer;
buffer->len = PAGE_SIZE;
- buffer->buffer = kmalloc(buffer->len, GFP_KERNEL);
+ buffer->buffer = kzalloc(buffer->len, GFP_KERNEL);
if (buffer->buffer == NULL)
goto __nomem;
}
@@ -683,32 +683,27 @@ int snd_info_card_free(struct snd_card *card)
*
* Reads one line from the buffer and stores the string.
*
- * Returns zero if successful, or 1 if error or EOF.
+ * Return: Zero if successful, or 1 if error or EOF.
*/
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
{
int c = -1;
+ if (snd_BUG_ON(!buffer || !buffer->buffer))
+ return 1;
if (len <= 0 || buffer->stop || buffer->error)
return 1;
- while (--len > 0) {
+ while (!buffer->stop) {
c = buffer->buffer[buffer->curr++];
- if (c == '\n') {
- if (buffer->curr >= buffer->size)
- buffer->stop = 1;
- break;
- }
- *line++ = c;
- if (buffer->curr >= buffer->size) {
+ if (buffer->curr >= buffer->size)
buffer->stop = 1;
+ if (c == '\n')
break;
+ if (len) {
+ len--;
+ *line++ = c;
}
}
- while (c != '\n' && !buffer->stop) {
- c = buffer->buffer[buffer->curr++];
- if (buffer->curr >= buffer->size)
- buffer->stop = 1;
- }
*line = '\0';
return 0;
}
@@ -724,7 +719,7 @@ EXPORT_SYMBOL(snd_info_get_line);
* Parses the original string and copy a token to the given
* string buffer.
*
- * Returns the updated pointer of the original string so that
+ * Return: The updated pointer of the original string so that
* it can be used for the next call.
*/
const char *snd_info_get_str(char *dest, const char *src, int len)
@@ -763,7 +758,7 @@ EXPORT_SYMBOL(snd_info_get_str);
* Usually called from other functions such as
* snd_info_create_card_entry().
*
- * Returns the pointer of the new instance, or NULL on failure.
+ * Return: The pointer of the new instance, or %NULL on failure.
*/
static struct snd_info_entry *snd_info_create_entry(const char *name)
{
@@ -792,7 +787,7 @@ static struct snd_info_entry *snd_info_create_entry(const char *name)
*
* Creates a new info entry and assigns it to the given module.
*
- * Returns the pointer of the new instance, or NULL on failure.
+ * Return: The pointer of the new instance, or %NULL on failure.
*/
struct snd_info_entry *snd_info_create_module_entry(struct module * module,
const char *name,
@@ -816,7 +811,7 @@ EXPORT_SYMBOL(snd_info_create_module_entry);
*
* Creates a new info entry and assigns it to the given card.
*
- * Returns the pointer of the new instance, or NULL on failure.
+ * Return: The pointer of the new instance, or %NULL on failure.
*/
struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
const char *name,
@@ -882,7 +877,7 @@ static int snd_info_dev_register_entry(struct snd_device *device)
* For releasing this entry, use snd_device_free() instead of
* snd_info_free_entry().
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_card_proc_new(struct snd_card *card, const char *name,
struct snd_info_entry **entryp)
@@ -938,7 +933,7 @@ EXPORT_SYMBOL(snd_info_free_entry);
*
* Registers the proc info entry.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_info_register(struct snd_info_entry * entry)
{
diff --git a/sound/core/init.c b/sound/core/init.c
index 7b012d15c2cf..6ef06400dfc8 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -144,7 +144,7 @@ static inline int init_info_for_card(struct snd_card *card)
* space for the driver to use freely. The allocated struct is stored
* in the given card_ret pointer.
*
- * Returns zero if successful or a negative error code.
+ * Return: Zero if successful or a negative error code.
*/
int snd_card_create(int idx, const char *xid,
struct module *module, int extra_size,
@@ -337,7 +337,7 @@ static const struct file_operations snd_shutdown_f_ops =
*
* Disconnects all APIs from the file-operations (user space).
*
- * Returns zero, otherwise a negative error code.
+ * Return: Zero, otherwise a negative error code.
*
* Note: The current implementation replaces all active file->f_op with special
* dummy file operations (they do nothing except release).
@@ -415,7 +415,7 @@ EXPORT_SYMBOL(snd_card_disconnect);
* devices automatically. That is, you don't have to release the devices
* by yourself.
*
- * Returns zero. Frees all associated devices and frees the control
+ * Return: Zero. Frees all associated devices and frees the control
* interface associated to given soundcard.
*/
static int snd_card_do_free(struct snd_card *card)
@@ -677,7 +677,7 @@ static struct device_attribute card_number_attrs =
* external accesses. Thus, you should call this function at the end
* of the initialization of the card.
*
- * Returns zero otherwise a negative error code if the registration failed.
+ * Return: Zero otherwise a negative error code if the registration failed.
*/
int snd_card_register(struct snd_card *card)
{
@@ -849,7 +849,7 @@ int __exit snd_card_info_done(void)
* This function adds the component id string to the supported list.
* The component can be referred from the alsa-lib.
*
- * Returns zero otherwise a negative error code.
+ * Return: Zero otherwise a negative error code.
*/
int snd_component_add(struct snd_card *card, const char *component)
@@ -883,7 +883,7 @@ EXPORT_SYMBOL(snd_component_add);
* This linked-list is used to keep tracking the connection state,
* and to avoid the release of busy resources by hotplug.
*
- * Returns zero or a negative error code.
+ * Return: zero or a negative error code.
*/
int snd_card_file_add(struct snd_card *card, struct file *file)
{
@@ -920,7 +920,7 @@ EXPORT_SYMBOL(snd_card_file_add);
* called beforehand, it processes the pending release of
* resources.
*
- * Returns zero or a negative error code.
+ * Return: Zero or a negative error code.
*/
int snd_card_file_remove(struct snd_card *card, struct file *file)
{
@@ -959,6 +959,8 @@ EXPORT_SYMBOL(snd_card_file_remove);
*
* Waits until the power-state is changed.
*
+ * Return: Zero if successful, or a negative error code.
+ *
* Note: the power lock must be active before call.
*/
int snd_power_wait(struct snd_card *card, unsigned int power_state)
diff --git a/sound/core/isadma.c b/sound/core/isadma.c
index c0f1208bb7df..e2b386156a4c 100644
--- a/sound/core/isadma.c
+++ b/sound/core/isadma.c
@@ -81,7 +81,7 @@ EXPORT_SYMBOL(snd_dma_disable);
* @dma: the dma number
* @size: the dma transfer size
*
- * Returns the current pointer in DMA tranfer buffer in bytes
+ * Return: The current pointer in DMA transfer buffer in bytes.
*/
unsigned int snd_dma_pointer(unsigned long dma, unsigned int size)
{
diff --git a/sound/core/jack.c b/sound/core/jack.c
index a06b1651fcba..b35fe7345c20 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -98,8 +98,8 @@ static int snd_jack_dev_register(struct snd_device *device)
*
* Creates a new jack object.
*
- * Returns zero if successful, or a negative error code on failure.
- * On success jjack will be initialised.
+ * Return: Zero if successful, or a negative error code on failure.
+ * On success @jjack will be initialised.
*/
int snd_jack_new(struct snd_card *card, const char *id, int type,
struct snd_jack **jjack)
@@ -189,6 +189,8 @@ EXPORT_SYMBOL(snd_jack_set_parent);
* using this abstraction.
*
* This function may only be called prior to registration of the jack.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_jack_set_key(struct snd_jack *jack, enum snd_jack_types type,
int keytype)
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 691569238435..bdf826f4fe0c 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -81,7 +81,7 @@ static inline void dec_snd_pages(int order)
*
* Allocates the physically contiguous pages with the given size.
*
- * Returns the pointer of the buffer, or NULL if no enoguh memory.
+ * Return: The pointer of the buffer, or %NULL if no enough memory.
*/
void *snd_malloc_pages(size_t size, gfp_t gfp_flags)
{
@@ -175,9 +175,9 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
*
* Calls the memory-allocator function for the corresponding
* buffer type.
- *
- * Returns zero if the buffer with the given size is allocated successfully,
- * other a negative value at error.
+ *
+ * Return: Zero if the buffer with the given size is allocated successfully,
+ * otherwise a negative value on error.
*/
int snd_dma_alloc_pages(int type, struct device *device, size_t size,
struct snd_dma_buffer *dmab)
@@ -229,9 +229,9 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
* buffer type. When no space is left, this function reduces the size and
* tries to allocate again. The size actually allocated is stored in
* res_size argument.
- *
- * Returns zero if the buffer with the given size is allocated successfully,
- * other a negative value at error.
+ *
+ * Return: Zero if the buffer with the given size is allocated successfully,
+ * otherwise a negative value on error.
*/
int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size,
struct snd_dma_buffer *dmab)
@@ -292,7 +292,7 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
* Looks for the reserved-buffer list and re-uses if the same buffer
* is found in the list. When the buffer is found, it's removed from the free list.
*
- * Returns the size of buffer if the buffer is found, or zero if not found.
+ * Return: The size of buffer if the buffer is found, or zero if not found.
*/
size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id)
{
@@ -326,8 +326,8 @@ size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id)
* @id: the buffer id
*
* Reserves the given buffer as a reserved buffer.
- *
- * Returns zero if successful, or a negative code at error.
+ *
+ * Return: Zero if successful, or a negative code on error.
*/
int snd_dma_reserve_buf(struct snd_dma_buffer *dmab, unsigned int id)
{
diff --git a/sound/core/memory.c b/sound/core/memory.c
index 66a278d0b04e..36c0f1a2e189 100644
--- a/sound/core/memory.c
+++ b/sound/core/memory.c
@@ -33,7 +33,7 @@
*
* Copies the data from mmio-space to user-space.
*
- * Returns zero if successful, or non-zero on failure.
+ * Return: Zero if successful, or non-zero on failure.
*/
int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count)
{
@@ -66,7 +66,7 @@ EXPORT_SYMBOL(copy_to_user_fromio);
*
* Copies the data from user-space to mmio-space.
*
- * Returns zero if successful, or non-zero on failure.
+ * Return: Zero if successful, or non-zero on failure.
*/
int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count)
{
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 61798f85d030..17f45e8aa89c 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = {
FORMAT(G723_24_1B),
FORMAT(G723_40),
FORMAT(G723_40_1B),
+ FORMAT(DSD_U8),
+ FORMAT(DSD_U16_LE),
};
const char *snd_pcm_format_name(snd_pcm_format_t format)
@@ -637,7 +639,7 @@ static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substrea
* calling this, i.e. zero must be given to the argument of
* snd_pcm_new().
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
{
@@ -759,7 +761,7 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device,
* The pcm operators have to be set afterwards to the new instance
* via snd_pcm_set_ops().
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count, struct snd_pcm **rpcm)
@@ -787,7 +789,7 @@ EXPORT_SYMBOL(snd_pcm_new);
* The pcm operators have to be set afterwards to the new instance
* via snd_pcm_set_ops().
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index c4840ff75d00..41b3dfe68698 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -666,7 +666,8 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b,
* The interval is changed to the range satisfying both intervals.
* The interval status (min, max, integer, etc.) are evaluated.
*
- * Returns non-zero if the value is changed, zero if not changed.
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
*/
int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v)
{
@@ -865,7 +866,8 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
* @nump: pointer to store the resultant numerator
* @denp: pointer to store the resultant denominator
*
- * Returns non-zero if the value is changed, zero if not changed.
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
*/
int snd_interval_ratnum(struct snd_interval *i,
unsigned int rats_count, struct snd_ratnum *rats,
@@ -983,7 +985,8 @@ EXPORT_SYMBOL(snd_interval_ratnum);
* @nump: pointer to store the resultant numerator
* @denp: pointer to store the resultant denominator
*
- * Returns non-zero if the value is changed, zero if not changed.
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
*/
static int snd_interval_ratden(struct snd_interval *i,
unsigned int rats_count, struct snd_ratden *rats,
@@ -1082,7 +1085,8 @@ static int snd_interval_ratden(struct snd_interval *i,
* When mask is non-zero, only the elements corresponding to bit 1 are
* evaluated.
*
- * Returns non-zero if the value is changed, zero if not changed.
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
*/
int snd_interval_list(struct snd_interval *i, unsigned int count,
const unsigned int *list, unsigned int mask)
@@ -1142,7 +1146,7 @@ static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned
* @private: the private data pointer passed to function
* @dep: the dependent variables
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond,
int var,
@@ -1200,6 +1204,8 @@ EXPORT_SYMBOL(snd_pcm_hw_rule_add);
* @mask: the bitmap mask
*
* Apply the constraint of the given bitmap mask to a 32-bit mask parameter.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
u_int32_t mask)
@@ -1220,6 +1226,8 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param
* @mask: the 64bit bitmap mask
*
* Apply the constraint of the given bitmap mask to a 64-bit mask parameter.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
u_int64_t mask)
@@ -1240,6 +1248,9 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par
* @var: hw_params variable to apply the integer constraint
*
* Apply the constraint of integer to an interval parameter.
+ *
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
*/
int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var)
{
@@ -1257,6 +1268,9 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_integer);
* @max: the maximal value
*
* Apply the min/max range constraint to an interval parameter.
+ *
+ * Return: Positive if the value is changed, zero if it's not changed, or a
+ * negative error code.
*/
int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
unsigned int min, unsigned int max)
@@ -1288,6 +1302,8 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
* @l: list
*
* Apply the list of constraints to an interval parameter.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
unsigned int cond,
@@ -1322,6 +1338,8 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
* @cond: condition bits
* @var: hw_params variable to apply the ratnums constraint
* @r: struct snd_ratnums constriants
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime,
unsigned int cond,
@@ -1355,6 +1373,8 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
* @cond: condition bits
* @var: hw_params variable to apply the ratdens constraint
* @r: struct snd_ratdens constriants
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime,
unsigned int cond,
@@ -1386,6 +1406,8 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params,
* @cond: condition bits
* @width: sample bits width
* @msbits: msbits width
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime,
unsigned int cond,
@@ -1414,6 +1436,8 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params,
* @cond: condition bits
* @var: hw_params variable to apply the step constraint
* @step: step size
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime,
unsigned int cond,
@@ -1444,6 +1468,8 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm
* @runtime: PCM runtime instance
* @cond: condition bits
* @var: hw_params variable to apply the power-of-2 constraint
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
unsigned int cond,
@@ -1470,6 +1496,8 @@ static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params,
* snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling
* @runtime: PCM runtime instance
* @base_rate: the rate at which the hardware does not resample
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
unsigned int base_rate)
@@ -1519,8 +1547,8 @@ EXPORT_SYMBOL(_snd_pcm_hw_params_any);
* @var: parameter to retrieve
* @dir: pointer to the direction (-1,0,1) or %NULL
*
- * Return the value for field @var if it's fixed in configuration space
- * defined by @params. Return -%EINVAL otherwise.
+ * Return: The value for field @var if it's fixed in configuration space
+ * defined by @params. -%EINVAL otherwise.
*/
int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var, int *dir)
@@ -1591,7 +1619,8 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params,
*
* Inside configuration space defined by @params remove from @var all
* values > minimum. Reduce configuration space accordingly.
- * Return the minimum.
+ *
+ * Return: The minimum, or a negative error code on failure.
*/
int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm,
struct snd_pcm_hw_params *params,
@@ -1637,7 +1666,8 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params,
*
* Inside configuration space defined by @params remove from @var all
* values < maximum. Reduce configuration space accordingly.
- * Return the maximum.
+ *
+ * Return: The maximum, or a negative error code on failure.
*/
int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
struct snd_pcm_hw_params *params,
@@ -1665,6 +1695,8 @@ EXPORT_SYMBOL(snd_pcm_hw_param_last);
* The configuration chosen is that obtained fixing in this order:
* first access, first format, first subformat, min channels,
* min rate, min period time, max buffer size, min tick time
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm,
struct snd_pcm_hw_params *params)
@@ -1771,7 +1803,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream,
* Processes the generic ioctl commands for PCM.
* Can be passed as the ioctl callback for PCM ops.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream,
unsigned int cmd, void *arg)
@@ -2510,7 +2542,7 @@ static void pcm_chmap_ctl_private_free(struct snd_kcontrol *kcontrol)
* @info_ret: store struct snd_pcm_chmap instance if non-NULL
*
* Create channel-mapping control elements assigned to the given PCM stream(s).
- * Returns zero if succeed, or a negative error value.
+ * Return: Zero if successful, or a negative error value.
*/
int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
const struct snd_pcm_chmap_elem *chmap,
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 69e01c4fc32d..0af622c34e19 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -95,7 +95,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream
*
* Releases the pre-allocated buffer of the given substream.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream)
{
@@ -115,7 +115,7 @@ int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream)
*
* Releases all the pre-allocated buffers on the given pcm.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm)
{
@@ -265,7 +265,7 @@ static int snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream,
* destruction time. The dma_buf_id must be unique for all systems
* (in the same DMA buffer type) e.g. using snd_dma_pci_buf_id().
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
int type, struct device *data,
@@ -289,7 +289,7 @@ EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages);
* Do pre-allocation to all substreams of the given pcm for the
* specified DMA type.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
int type, void *data,
@@ -313,8 +313,9 @@ EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all);
* @substream: the pcm substream instance
* @offset: the buffer offset
*
- * Returns the page struct at the given buffer offset.
* Used as the page callback of PCM ops.
+ *
+ * Return: The page struct at the given buffer offset. %NULL on failure.
*/
struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned long offset)
{
@@ -337,7 +338,7 @@ EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page);
* Allocates the DMA buffer on the BUS type given earlier to
* snd_pcm_lib_preallocate_xxx_pages().
*
- * Returns 1 if the buffer is changed, 0 if not changed, or a negative
+ * Return: 1 if the buffer is changed, 0 if not changed, or a negative
* code on failure.
*/
int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size)
@@ -390,7 +391,7 @@ EXPORT_SYMBOL(snd_pcm_lib_malloc_pages);
*
* Releases the DMA buffer allocated via snd_pcm_lib_malloc_pages().
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream)
{
@@ -437,6 +438,8 @@ EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer);
* snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer
* @substream: the substream with a buffer allocated by
* snd_pcm_lib_alloc_vmalloc_buffer()
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream)
{
@@ -458,6 +461,8 @@ EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer);
* @offset: offset in the buffer
*
* This function is to be used as the page callback in the PCM ops.
+ *
+ * Return: The page struct, or %NULL on failure.
*/
struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream,
unsigned long offset)
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index d4fc1bfbe457..43f24cce3dec 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -140,6 +140,14 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
.width = 5, .phys = 5, .le = -1, .signd = -1,
.silence = {},
},
+ [SNDRV_PCM_FORMAT_DSD_U8] = {
+ .width = 8, .phys = 8, .le = 1, .signd = 0,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_DSD_U16_LE] = {
+ .width = 16, .phys = 16, .le = 1, .signd = 0,
+ .silence = {},
+ },
/* FIXME: the following three formats are not defined properly yet */
[SNDRV_PCM_FORMAT_MPEG] = {
.le = -1, .signd = -1,
@@ -213,7 +221,7 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
* snd_pcm_format_signed - Check the PCM format is signed linear
* @format: the format to check
*
- * Returns 1 if the given PCM format is signed linear, 0 if unsigned
+ * Return: 1 if the given PCM format is signed linear, 0 if unsigned
* linear, and a negative error code for non-linear formats.
*/
int snd_pcm_format_signed(snd_pcm_format_t format)
@@ -232,7 +240,7 @@ EXPORT_SYMBOL(snd_pcm_format_signed);
* snd_pcm_format_unsigned - Check the PCM format is unsigned linear
* @format: the format to check
*
- * Returns 1 if the given PCM format is unsigned linear, 0 if signed
+ * Return: 1 if the given PCM format is unsigned linear, 0 if signed
* linear, and a negative error code for non-linear formats.
*/
int snd_pcm_format_unsigned(snd_pcm_format_t format)
@@ -251,7 +259,7 @@ EXPORT_SYMBOL(snd_pcm_format_unsigned);
* snd_pcm_format_linear - Check the PCM format is linear
* @format: the format to check
*
- * Returns 1 if the given PCM format is linear, 0 if not.
+ * Return: 1 if the given PCM format is linear, 0 if not.
*/
int snd_pcm_format_linear(snd_pcm_format_t format)
{
@@ -264,7 +272,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear);
* snd_pcm_format_little_endian - Check the PCM format is little-endian
* @format: the format to check
*
- * Returns 1 if the given PCM format is little-endian, 0 if
+ * Return: 1 if the given PCM format is little-endian, 0 if
* big-endian, or a negative error code if endian not specified.
*/
int snd_pcm_format_little_endian(snd_pcm_format_t format)
@@ -283,7 +291,7 @@ EXPORT_SYMBOL(snd_pcm_format_little_endian);
* snd_pcm_format_big_endian - Check the PCM format is big-endian
* @format: the format to check
*
- * Returns 1 if the given PCM format is big-endian, 0 if
+ * Return: 1 if the given PCM format is big-endian, 0 if
* little-endian, or a negative error code if endian not specified.
*/
int snd_pcm_format_big_endian(snd_pcm_format_t format)
@@ -302,7 +310,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian);
* snd_pcm_format_width - return the bit-width of the format
* @format: the format to check
*
- * Returns the bit-width of the format, or a negative error code
+ * Return: The bit-width of the format, or a negative error code
* if unknown format.
*/
int snd_pcm_format_width(snd_pcm_format_t format)
@@ -321,7 +329,7 @@ EXPORT_SYMBOL(snd_pcm_format_width);
* snd_pcm_format_physical_width - return the physical bit-width of the format
* @format: the format to check
*
- * Returns the physical bit-width of the format, or a negative error code
+ * Return: The physical bit-width of the format, or a negative error code
* if unknown format.
*/
int snd_pcm_format_physical_width(snd_pcm_format_t format)
@@ -341,7 +349,7 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width);
* @format: the format to check
* @samples: sampling rate
*
- * Returns the byte size of the given samples for the format, or a
+ * Return: The byte size of the given samples for the format, or a
* negative error code if unknown format.
*/
ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples)
@@ -358,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_size);
* snd_pcm_format_silence_64 - return the silent data in 8 bytes array
* @format: the format to check
*
- * Returns the format pattern to fill or NULL if error.
+ * Return: The format pattern to fill or %NULL if error.
*/
const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format)
{
@@ -379,7 +387,7 @@ EXPORT_SYMBOL(snd_pcm_format_silence_64);
*
* Sets the silence data on the buffer for the given samples.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int samples)
{
@@ -449,7 +457,7 @@ EXPORT_SYMBOL(snd_pcm_format_set_silence);
* Determines the rate_min and rate_max fields from the rates bits of
* the given runtime->hw.
*
- * Returns zero if successful.
+ * Return: Zero if successful.
*/
int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
{
@@ -475,7 +483,7 @@ EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
* snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
* @rate: the sample rate to convert
*
- * Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or
+ * Return: The SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or
* SNDRV_PCM_RATE_KNOT for an unknown rate.
*/
unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate)
@@ -493,8 +501,8 @@ EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);
* snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate
* @rate_bit: the rate bit to convert
*
- * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag
- * or 0 for an unknown rate bit
+ * Return: The sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag
+ * or 0 for an unknown rate bit.
*/
unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit)
{
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index eb560fa32321..23e3c46cd0a4 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -898,6 +898,8 @@ static struct action_ops snd_pcm_action_start = {
/**
* snd_pcm_start - start all linked streams
* @substream: the PCM substream instance
+ *
+ * Return: Zero if successful, or a negative error code.
*/
int snd_pcm_start(struct snd_pcm_substream *substream)
{
@@ -951,6 +953,8 @@ static struct action_ops snd_pcm_action_stop = {
* @state: PCM state after stopping the stream
*
* The state of each stream is then changed to the given state unconditionally.
+ *
+ * Return: Zero if succesful, or a negative error code.
*/
int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state)
{
@@ -965,6 +969,8 @@ EXPORT_SYMBOL(snd_pcm_stop);
*
* After stopping, the state is changed to SETUP.
* Unlike snd_pcm_stop(), this affects only the given stream.
+ *
+ * Return: Zero if succesful, or a negative error code.
*/
int snd_pcm_drain_done(struct snd_pcm_substream *substream)
{
@@ -1098,6 +1104,9 @@ static struct action_ops snd_pcm_action_suspend = {
* @substream: the PCM substream
*
* After this call, all streams are changed to SUSPENDED state.
+ *
+ * Return: Zero if successful (or @substream is %NULL), or a negative error
+ * code.
*/
int snd_pcm_suspend(struct snd_pcm_substream *substream)
{
@@ -1120,6 +1129,8 @@ EXPORT_SYMBOL(snd_pcm_suspend);
* @pcm: the PCM instance
*
* After this call, all streams are changed to SUSPENDED state.
+ *
+ * Return: Zero if successful (or @pcm is %NULL), or a negative error code.
*/
int snd_pcm_suspend_all(struct snd_pcm *pcm)
{
@@ -1343,6 +1354,8 @@ static struct action_ops snd_pcm_action_prepare = {
* snd_pcm_prepare - prepare the PCM substream to be triggerable
* @substream: the PCM substream instance
* @file: file to refer f_flags
+ *
+ * Return: Zero if successful, or a negative error code.
*/
static int snd_pcm_prepare(struct snd_pcm_substream *substream,
struct file *file)
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 1bb95aeea084..7b596b5751db 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -863,7 +863,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card,
*
* Reads the data from the internal buffer.
*
- * Returns the size of read data, or a negative error code on failure.
+ * Return: The size of read data, or a negative error code on failure.
*/
int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
const unsigned char *buffer, int count)
@@ -1024,8 +1024,8 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun
/**
* snd_rawmidi_transmit_empty - check whether the output buffer is empty
* @substream: the rawmidi substream
- *
- * Returns 1 if the internal output buffer is empty, 0 if not.
+ *
+ * Return: 1 if the internal output buffer is empty, 0 if not.
*/
int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream)
{
@@ -1055,7 +1055,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream)
* and call snd_rawmidi_transmit_ack() after the transmission is
* finished.
*
- * Returns the size of copied data, or a negative error code on failure.
+ * Return: The size of copied data, or a negative error code on failure.
*/
int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
unsigned char *buffer, int count)
@@ -1107,7 +1107,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
* the given size and updates the condition.
* Call after the transmission is finished.
*
- * Returns the advanced size if successful, or a negative error code on failure.
+ * Return: The advanced size if successful, or a negative error code on failure.
*/
int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count)
{
@@ -1140,7 +1140,7 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count)
*
* Copies data from the buffer to the device and advances the pointer.
*
- * Returns the copied size if successful, or a negative error code on failure.
+ * Return: The copied size if successful, or a negative error code on failure.
*/
int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
unsigned char *buffer, int count)
@@ -1438,7 +1438,7 @@ static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi,
* Creates a new rawmidi instance.
* Use snd_rawmidi_set_ops() to set the operators to the new instance.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_rawmidi_new(struct snd_card *card, char *id, int device,
int output_count, int input_count,
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 70ccdab74153..f002bd911dae 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -102,6 +102,9 @@ static void snd_request_other(int minor)
* This function increments the reference counter of the card instance
* if an associated instance with the given minor number and type is found.
* The caller must call snd_card_unref() appropriately later.
+ *
+ * Return: The user data pointer if the specified device is found. %NULL
+ * otherwise.
*/
void *snd_lookup_minor_data(unsigned int minor, int type)
{
@@ -261,7 +264,7 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev)
* Registers an ALSA device file for the given card.
* The operators have to be set in reg parameter.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
const struct file_operations *f_ops,
@@ -339,7 +342,7 @@ static int find_snd_minor(int type, struct snd_card *card, int dev)
* Unregisters the device file already registered via
* snd_register_device().
*
- * Returns zero if sucecessful, or a negative error code on failure
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_unregister_device(int type, struct snd_card *card, int dev)
{
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 0097f3619faa..02f90b4f8b86 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -365,8 +365,7 @@ static void master_free(struct snd_kcontrol *kcontrol)
* @name: name string of the control element to create
* @tlv: optional TLV int array for dB information
*
- * Creates a virtual matster control with the given name string.
- * Returns the created control element, or NULL for errors (ENOMEM).
+ * Creates a virtual master control with the given name string.
*
* After creating a vmaster element, you can add the slave controls
* via snd_ctl_add_slave() or snd_ctl_add_slave_uncached().
@@ -375,6 +374,8 @@ static void master_free(struct snd_kcontrol *kcontrol)
* for dB scale of the master control. It should be a single element
* with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or
* #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB.
+ *
+ * Return: The created control element, or %NULL for errors (ENOMEM).
*/
struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
const unsigned int *tlv)
@@ -426,6 +427,8 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master);
*
* Adds the given hook to the vmaster control element so that it's called
* at each time when the value is changed.
+ *
+ * Return: Zero.
*/
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
void (*hook)(void *private_data, int),
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index 7d02c322ed93..8545da99b183 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -87,7 +87,7 @@ config SND_ALOOP
configured number of substreams (see the pcm_substreams module
parameter).
- The looback device allow time sychronization with an external
+ The loopback device allows time sychronization with an external
timing source using the time shift universal control (+-20%
of system time).
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 64d534710b51..6f78de9c6fb6 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -325,7 +325,7 @@ static void params_change(struct snd_pcm_substream *substream)
struct loopback_pcm *dpcm = runtime->private_data;
struct loopback_cable *cable = dpcm->cable;
- cable->hw.formats = (1ULL << runtime->format);
+ cable->hw.formats = pcm_format_to_bits(runtime->format);
cable->hw.rate_min = runtime->rate;
cable->hw.rate_max = runtime->rate;
cable->hw.channels_min = runtime->channels;
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 4608c2ca43f8..e3a90d043f03 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -129,6 +129,8 @@ static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu)
* @dev_id: mpu401 instance
*
* Processes the interrupt for MPU401-UART i/o.
+ *
+ * Return: %IRQ_HANDLED if the interrupt was handled. %IRQ_NONE otherwise.
*/
irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id)
{
@@ -148,6 +150,8 @@ EXPORT_SYMBOL(snd_mpu401_uart_interrupt);
* @dev_id: mpu401 instance
*
* Processes the interrupt for MPU401-UART output.
+ *
+ * Return: %IRQ_HANDLED if the interrupt was handled. %IRQ_NONE otherwise.
*/
irqreturn_t snd_mpu401_uart_interrupt_tx(int irq, void *dev_id)
{
@@ -519,7 +523,7 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
* not the mpu401 instance itself. To access to the mpu401 instance,
* cast from rawmidi->private_data (with struct snd_mpu401 magic-cast).
*
- * Returns zero if successful, or a negative error code.
+ * Return: Zero if successful, or a negative error code.
*/
int snd_mpu401_uart_new(struct snd_card *card, int device,
unsigned short hardware,
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index bcc3e8e07122..a59c88818f48 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap)
}
}
dmap->raw_buf = start_addr;
- dmap->raw_buf_phys = virt_to_bus(start_addr);
+ dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL);
for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
SetPageReserved(page);
@@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap)
for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
ClearPageReserved(page);
+ dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL);
free_pages((unsigned long) dmap->raw_buf, sz);
dmap->raw_buf = NULL;
}
diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c
index 7d42c5418d1b..851a1da46be1 100644
--- a/sound/oss/sb_common.c
+++ b/sound/oss/sb_common.c
@@ -626,13 +626,12 @@ int sb_dsp_detect(struct address_info *hw_config, int pci, int pciio, struct sb_
*/
- detected_devc = kmalloc(sizeof(sb_devc), GFP_KERNEL);
+ detected_devc = kmemdup(devc, sizeof(sb_devc), GFP_KERNEL);
if (detected_devc == NULL)
{
printk(KERN_ERR "sb: Can't allocate memory for device information\n");
return 0;
}
- memcpy(detected_devc, devc, sizeof(sb_devc));
MDB(printk(KERN_INFO "SB %d.%02d detected OK (%x)\n", devc->major, devc->minor, hw_config->io_base));
return 1;
}
diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c
index 8e514a676a0d..5433c6f5eca2 100644
--- a/sound/oss/uart401.c
+++ b/sound/oss/uart401.c
@@ -352,23 +352,26 @@ int probe_uart401(struct address_info *hw_config, struct module *owner)
goto cleanup_irq;
}
conf_printf(name, hw_config);
- midi_devs[devc->my_dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL);
+ midi_devs[devc->my_dev] = kmemdup(&uart401_operations,
+ sizeof(struct midi_operations),
+ GFP_KERNEL);
if (!midi_devs[devc->my_dev]) {
printk(KERN_ERR "uart401: Failed to allocate memory\n");
goto cleanup_unload_mididev;
}
- memcpy(midi_devs[devc->my_dev], &uart401_operations, sizeof(struct midi_operations));
if (owner)
midi_devs[devc->my_dev]->owner = owner;
midi_devs[devc->my_dev]->devc = devc;
- midi_devs[devc->my_dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL);
+ midi_devs[devc->my_dev]->converter = kmemdup(&std_midi_synth,
+ sizeof(struct synth_operations),
+ GFP_KERNEL);
+
if (!midi_devs[devc->my_dev]->converter) {
printk(KERN_WARNING "uart401: Failed to allocate memory\n");
goto cleanup_midi_devs;
}
- memcpy(midi_devs[devc->my_dev]->converter, &std_midi_synth, sizeof(struct synth_operations));
strcpy(midi_devs[devc->my_dev]->info.name, name);
midi_devs[devc->my_dev]->converter->id = "UART401";
midi_devs[devc->my_dev]->converter->midi_dev = devc->my_dev;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 8b0f99688303..d37c683cfd7a 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -299,7 +299,7 @@ EXPORT_SYMBOL(snd_ac97_write);
* Reads a value from the given register. This will invoke the read
* callback directly after the register check.
*
- * Returns the read value.
+ * Return: The read value.
*/
unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
@@ -352,7 +352,7 @@ EXPORT_SYMBOL(snd_ac97_write_cache);
* Compares the value with the register cache and updates the value
* only when the value is changed.
*
- * Returns 1 if the value is changed, 0 if no change, or a negative
+ * Return: 1 if the value is changed, 0 if no change, or a negative
* code on failure.
*/
int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short value)
@@ -384,7 +384,7 @@ EXPORT_SYMBOL(snd_ac97_update);
* Updates the masked-bits on the given register only when the value
* is changed.
*
- * Returns 1 if the bits are changed, 0 if no change, or a negative
+ * Return: 1 if the bits are changed, 0 if no change, or a negative
* code on failure.
*/
int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value)
@@ -1836,7 +1836,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m
* snd_ac97_get_short_name - retrieve codec name
* @ac97: the codec instance
*
- * Returns the short identifying name of the codec.
+ * Return: The short identifying name of the codec.
*/
const char *snd_ac97_get_short_name(struct snd_ac97 *ac97)
{
@@ -1910,7 +1910,7 @@ static int ac97_reset_wait(struct snd_ac97 *ac97, int timeout, int with_modem)
* The AC97 bus instance is registered as a low-level device, so you don't
* have to release it manually.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops,
void *private_data, struct snd_ac97_bus **rbus)
@@ -2006,7 +2006,7 @@ static void do_update_power(struct work_struct *work)
* The ac97 instance is registered as a low-level device, so you don't
* have to release it manually.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, struct snd_ac97 **rac97)
{
@@ -2373,6 +2373,8 @@ static struct ac97_power_reg power_regs[PWIDX_SIZE] = {
* @powerup: non-zero when power up the part
*
* Update the AC97 powerdown register bits of the given part.
+ *
+ * Return: Zero.
*/
int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup)
{
@@ -2885,7 +2887,7 @@ static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr)
* headphone (true line-out) control as "Master".
* The quirk-list must be terminated with a zero-filled entry.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override)
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
index f1488fc176d5..eab0fc9ff2e0 100644
--- a/sound/pci/ac97/ac97_pcm.c
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -253,7 +253,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate)
* AC97_SPDIF is accepted as a pseudo register to modify the SPDIF
* status bits.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate)
{
@@ -440,6 +440,8 @@ static unsigned int get_rates(struct ac97_pcm *pcm, unsigned int cidx, unsigned
* It assigns available AC97 slots for given PCMs. If none or only
* some slots are available, pcm->xxx.slots and pcm->xxx.rslots[] members
* are reduced and might be zero.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_pcm_assign(struct snd_ac97_bus *bus,
unsigned short pcms_count,
@@ -562,6 +564,8 @@ EXPORT_SYMBOL(snd_ac97_pcm_assign);
* @slots: a subset of allocated slots (snd_ac97_pcm_assign) for this pcm
*
* It locks the specified slots and sets the given rate to AC97 registers.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
enum ac97_pcm_cfg cfg, unsigned short slots)
@@ -644,6 +648,8 @@ EXPORT_SYMBOL(snd_ac97_pcm_open);
* @pcm: the ac97 pcm instance
*
* It frees the locked AC97 slots.
+ *
+ * Return: Zero.
*/
int snd_ac97_pcm_close(struct ac97_pcm *pcm)
{
@@ -718,6 +724,8 @@ static int double_rate_hw_constraint_channels(struct snd_pcm_hw_params *params,
*
* Installs the hardware constraint rules to prevent using double rates and
* more than two channels at the same time.
+ *
+ * Return: Zero if successful, or a negative error code on failure.
*/
int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime)
{
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index e760af9d1fb6..53754f5edeb1 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -451,10 +451,10 @@ static int snd_ali_reset_5451(struct snd_ali *codec)
if (pci_dev) {
pci_read_config_dword(pci_dev, 0x7c, &dwVal);
pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000);
- udelay(5000);
+ mdelay(5);
pci_read_config_dword(pci_dev, 0x7c, &dwVal);
pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff);
- udelay(5000);
+ mdelay(5);
}
pci_dev = codec->pci;
@@ -463,14 +463,14 @@ static int snd_ali_reset_5451(struct snd_ali *codec)
udelay(500);
pci_read_config_dword(pci_dev, 0x44, &dwVal);
pci_write_config_dword(pci_dev, 0x44, dwVal & 0xfffbffff);
- udelay(5000);
+ mdelay(5);
wCount = 200;
while(wCount--) {
wReg = snd_ali_codec_peek(codec, 0, AC97_POWERDOWN);
if ((wReg & 0x000f) == 0x000f)
return 0;
- udelay(5000);
+ mdelay(5);
}
/* non-fatal if you have a non PM capable codec */
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 0aabfedeecba..fbc17203613c 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -966,7 +966,7 @@ static u64 snd_card_asihpi_playback_formats(struct snd_card_asihpi *asihpi,
if (!err)
err = hpi_outstream_query_format(h_stream, &hpi_format);
if (!err && (hpi_to_alsa_formats[format] != -1))
- formats |= (1ULL << hpi_to_alsa_formats[format]);
+ formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]);
}
return formats;
}
@@ -1141,8 +1141,8 @@ static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi,
format, sample_rate, 128000, 0);
if (!err)
err = hpi_instream_query_format(h_stream, &hpi_format);
- if (!err)
- formats |= (1ULL << hpi_to_alsa_formats[format]);
+ if (!err && (hpi_to_alsa_formats[format] != -1))
+ formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]);
}
return formats;
}
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index e6b016693240..bdd888ec9a84 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -657,14 +657,14 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu)
return 0;
}
-static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu)
+static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu,
+ const struct firmware *fw_entry)
{
int n, i;
int reg;
int value;
unsigned int write_post;
unsigned long flags;
- const struct firmware *fw_entry = emu->firmware;
if (!fw_entry)
return -EIO;
@@ -725,9 +725,34 @@ static int emu1010_firmware_thread(void *data)
/* Return to Audio Dock programming mode */
snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK);
- err = snd_emu1010_load_firmware(emu);
- if (err != 0)
- continue;
+
+ if (!emu->dock_fw) {
+ const char *filename = NULL;
+ switch (emu->card_capabilities->emu_model) {
+ case EMU_MODEL_EMU1010:
+ filename = DOCK_FILENAME;
+ break;
+ case EMU_MODEL_EMU1010B:
+ filename = MICRO_DOCK_FILENAME;
+ break;
+ case EMU_MODEL_EMU1616:
+ filename = MICRO_DOCK_FILENAME;
+ break;
+ }
+ if (filename) {
+ err = request_firmware(&emu->dock_fw,
+ filename,
+ &emu->pci->dev);
+ if (err)
+ continue;
+ }
+ }
+
+ if (emu->dock_fw) {
+ err = snd_emu1010_load_firmware(emu, emu->dock_fw);
+ if (err)
+ continue;
+ }
snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0);
snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg);
@@ -862,7 +887,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
filename, emu->firmware->size);
}
- err = snd_emu1010_load_firmware(emu);
+ err = snd_emu1010_load_firmware(emu, emu->firmware);
if (err != 0) {
snd_printk(KERN_INFO "emu1010: Loading Firmware failed\n");
return err;
@@ -1253,6 +1278,8 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
kthread_stop(emu->emu1010.firmware_thread);
if (emu->firmware)
release_firmware(emu->firmware);
+ if (emu->dock_fw)
+ release_firmware(emu->dock_fw);
if (emu->irq >= 0)
free_irq(emu->irq, emu);
/* remove reserved page */
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index a3ea76a4c9d2..7c11d46b84d3 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -119,6 +119,32 @@ static bool check_pincap_validity(struct hda_codec *codec, hda_nid_t pin,
}
}
+static bool can_be_headset_mic(struct hda_codec *codec,
+ struct auto_pin_cfg_item *item,
+ int seq_number)
+{
+ int attr;
+ unsigned int def_conf;
+ if (item->type != AUTO_PIN_MIC)
+ return false;
+
+ if (item->is_headset_mic || item->is_headphone_mic)
+ return false; /* Already assigned */
+
+ def_conf = snd_hda_codec_get_pincfg(codec, item->pin);
+ attr = snd_hda_get_input_pin_attr(def_conf);
+ if (attr <= INPUT_PIN_ATTR_DOCK)
+ return false;
+
+ if (seq_number >= 0) {
+ int seq = get_defcfg_sequence(def_conf);
+ if (seq != seq_number)
+ return false;
+ }
+
+ return true;
+}
+
/*
* Parse all pin widgets and store the useful pin nids to cfg
*
@@ -260,6 +286,38 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
}
}
+ /* Find a pin that could be a headset or headphone mic */
+ if (cond_flags & HDA_PINCFG_HEADSET_MIC || cond_flags & HDA_PINCFG_HEADPHONE_MIC) {
+ bool hsmic = !!(cond_flags & HDA_PINCFG_HEADSET_MIC);
+ bool hpmic = !!(cond_flags & HDA_PINCFG_HEADPHONE_MIC);
+ for (i = 0; (hsmic || hpmic) && (i < cfg->num_inputs); i++)
+ if (hsmic && can_be_headset_mic(codec, &cfg->inputs[i], 0xc)) {
+ cfg->inputs[i].is_headset_mic = 1;
+ hsmic = false;
+ } else if (hpmic && can_be_headset_mic(codec, &cfg->inputs[i], 0xd)) {
+ cfg->inputs[i].is_headphone_mic = 1;
+ hpmic = false;
+ }
+
+ /* If we didn't find our sequence number mark, fall back to any sequence number */
+ for (i = 0; (hsmic || hpmic) && (i < cfg->num_inputs); i++) {
+ if (!can_be_headset_mic(codec, &cfg->inputs[i], -1))
+ continue;
+ if (hsmic) {
+ cfg->inputs[i].is_headset_mic = 1;
+ hsmic = false;
+ } else if (hpmic) {
+ cfg->inputs[i].is_headphone_mic = 1;
+ hpmic = false;
+ }
+ }
+
+ if (hsmic)
+ snd_printdd("Told to look for a headset mic, but didn't find any.\n");
+ if (hpmic)
+ snd_printdd("Told to look for a headphone mic, but didn't find any.\n");
+ }
+
/* FIX-UP:
* If no line-out is defined but multiple HPs are found,
* some of them might be the real line-outs.
@@ -388,6 +446,7 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
*/
static const char *hda_get_input_pin_label(struct hda_codec *codec,
+ const struct auto_pin_cfg_item *item,
hda_nid_t pin, bool check_location)
{
unsigned int def_conf;
@@ -400,6 +459,10 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec,
switch (get_defcfg_device(def_conf)) {
case AC_JACK_MIC_IN:
+ if (item && item->is_headset_mic)
+ return "Headset Mic";
+ if (item && item->is_headphone_mic)
+ return "Headphone Mic";
if (!check_location)
return "Mic";
attr = snd_hda_get_input_pin_attr(def_conf);
@@ -480,7 +543,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec,
has_multiple_pins = 1;
if (has_multiple_pins && type == AUTO_PIN_MIC)
has_multiple_pins &= check_mic_location_need(codec, cfg, input);
- return hda_get_input_pin_label(codec, cfg->inputs[input].pin,
+ return hda_get_input_pin_label(codec, &cfg->inputs[input],
+ cfg->inputs[input].pin,
has_multiple_pins);
}
EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
@@ -649,7 +713,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
}
}
if (!name)
- name = hda_get_input_pin_label(codec, nid, true);
+ name = hda_get_input_pin_label(codec, NULL, nid, true);
break;
}
if (!name)
diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h
index f74807138b49..e941f604f5e5 100644
--- a/sound/pci/hda/hda_auto_parser.h
+++ b/sound/pci/hda/hda_auto_parser.h
@@ -36,6 +36,8 @@ enum {
struct auto_pin_cfg_item {
hda_nid_t pin;
int type;
+ unsigned int is_headset_mic:1;
+ unsigned int is_headphone_mic:1; /* Mic-only in headphone jack */
};
struct auto_pin_cfg;
@@ -78,8 +80,10 @@ struct auto_pin_cfg {
};
/* bit-flags for snd_hda_parse_pin_def_config() behavior */
-#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */
-#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */
+#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */
+#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */
+#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Try to find headset mic; mark seq number as 0xc to trigger */
+#define HDA_PINCFG_HEADPHONE_MIC (1 << 3) /* Try to find headphone mic; mark seq number as 0xd to trigger */
int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
struct auto_pin_cfg *cfg,
@@ -90,4 +94,25 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
+static inline int auto_cfg_hp_outs(const struct auto_pin_cfg *cfg)
+{
+ return (cfg->line_out_type == AUTO_PIN_HP_OUT) ?
+ cfg->line_outs : cfg->hp_outs;
+}
+static inline const hda_nid_t *auto_cfg_hp_pins(const struct auto_pin_cfg *cfg)
+{
+ return (cfg->line_out_type == AUTO_PIN_HP_OUT) ?
+ cfg->line_out_pins : cfg->hp_pins;
+}
+static inline int auto_cfg_speaker_outs(const struct auto_pin_cfg *cfg)
+{
+ return (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) ?
+ cfg->line_outs : cfg->speaker_outs;
+}
+static inline const hda_nid_t *auto_cfg_speaker_pins(const struct auto_pin_cfg *cfg)
+{
+ return (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) ?
+ cfg->line_out_pins : cfg->speaker_pins;
+}
+
#endif /* __SOUND_HDA_AUTO_PARSER_H */
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0849aac449f2..63c99090a4ec 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -39,13 +39,23 @@ static void snd_hda_generate_beep(struct work_struct *work)
struct hda_beep *beep =
container_of(work, struct hda_beep, beep_work);
struct hda_codec *codec = beep->codec;
+ int tone;
if (!beep->enabled)
return;
+ tone = beep->tone;
+ if (tone && !beep->playing) {
+ snd_hda_power_up(codec);
+ beep->playing = 1;
+ }
/* generate tone */
snd_hda_codec_write(codec, beep->nid, 0,
- AC_VERB_SET_BEEP_CONTROL, beep->tone);
+ AC_VERB_SET_BEEP_CONTROL, tone);
+ if (!tone && beep->playing) {
+ beep->playing = 0;
+ snd_hda_power_down(codec);
+ }
}
/* (non-standard) Linear beep tone calculation for IDT/STAC codecs
@@ -115,14 +125,23 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
return 0;
}
+static void turn_off_beep(struct hda_beep *beep)
+{
+ cancel_work_sync(&beep->beep_work);
+ if (beep->playing) {
+ /* turn off beep */
+ snd_hda_codec_write(beep->codec, beep->nid, 0,
+ AC_VERB_SET_BEEP_CONTROL, 0);
+ beep->playing = 0;
+ snd_hda_power_down(beep->codec);
+ }
+}
+
static void snd_hda_do_detach(struct hda_beep *beep)
{
input_unregister_device(beep->dev);
beep->dev = NULL;
- cancel_work_sync(&beep->beep_work);
- /* turn off beep for sure */
- snd_hda_codec_write(beep->codec, beep->nid, 0,
- AC_VERB_SET_BEEP_CONTROL, 0);
+ turn_off_beep(beep);
}
static int snd_hda_do_attach(struct hda_beep *beep)
@@ -170,12 +189,8 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
enable = !!enable;
if (beep->enabled != enable) {
beep->enabled = enable;
- if (!enable) {
- cancel_work_sync(&beep->beep_work);
- /* turn off beep */
- snd_hda_codec_write(beep->codec, beep->nid, 0,
- AC_VERB_SET_BEEP_CONTROL, 0);
- }
+ if (!enable)
+ turn_off_beep(beep);
return 1;
}
return 0;
@@ -198,7 +213,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
snprintf(beep->phys, sizeof(beep->phys),
"card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
/* enable linear scale */
- snd_hda_codec_write(codec, nid, 0,
+ snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_DIGI_CONVERT_2, 0x01);
beep->nid = nid;
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 4dc6933bc655..cb88464676b6 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -36,6 +36,7 @@ struct hda_beep {
hda_nid_t nid;
unsigned int enabled:1;
unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */
+ unsigned int playing:1;
struct work_struct beep_work; /* scheduled task for beep event */
struct mutex mutex;
};
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4aba7646dd9c..6f9b64700f6e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1065,8 +1065,14 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
{
struct hda_pincfg *pin;
+ /* the check below may be invalid when pins are added by a fixup
+ * dynamically (e.g. via snd_hda_codec_update_widgets()), so disabled
+ * for now
+ */
+ /*
if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
return -EINVAL;
+ */
pin = look_up_pincfg(codec, list, nid);
if (!pin) {
@@ -1300,8 +1306,6 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec,
static unsigned int hda_set_power_state(struct hda_codec *codec,
unsigned int power_state);
-static unsigned int default_power_filter(struct hda_codec *codec, hda_nid_t nid,
- unsigned int power_state);
/**
* snd_hda_codec_new - create a HDA codec
@@ -1422,7 +1426,6 @@ int snd_hda_codec_new(struct hda_bus *bus,
#endif
codec->epss = snd_hda_codec_get_supported_ps(codec, fg,
AC_PWRST_EPSS);
- codec->power_filter = default_power_filter;
/* power-up all before initialization */
hda_set_power_state(codec, AC_PWRST_D0);
@@ -2787,6 +2790,11 @@ void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook)
{
if (!hook->hook || !hook->codec)
return;
+ /* don't call vmaster hook in the destructor since it might have
+ * been already destroyed
+ */
+ if (hook->codec->bus->shutdown)
+ return;
switch (hook->mute_mode) {
case HDA_VMUTE_FOLLOW_MASTER:
snd_ctl_sync_vmaster_hook(hook->sw_kctl);
@@ -3770,8 +3778,9 @@ static unsigned int hda_sync_power_state(struct hda_codec *codec,
}
/* don't power down the widget if it controls eapd and EAPD_BTLENABLE is set */
-static unsigned int default_power_filter(struct hda_codec *codec, hda_nid_t nid,
- unsigned int power_state)
+unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
+ hda_nid_t nid,
+ unsigned int power_state)
{
if (power_state == AC_PWRST_D3 &&
get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_PIN &&
@@ -3783,6 +3792,7 @@ static unsigned int default_power_filter(struct hda_codec *codec, hda_nid_t nid,
}
return power_state;
}
+EXPORT_SYMBOL_HDA(snd_hda_codec_eapd_power_filter);
/*
* set power state of the codec, and return the power state
@@ -3827,8 +3837,8 @@ static void sync_power_up_states(struct hda_codec *codec)
hda_nid_t nid = codec->start_nid;
int i;
- /* don't care if no or standard filter is used */
- if (!codec->power_filter || codec->power_filter == default_power_filter)
+ /* don't care if no filter is used */
+ if (!codec->power_filter)
return;
for (i = 0; i < codec->num_nodes; i++, nid++) {
@@ -5546,14 +5556,12 @@ void *snd_array_new(struct snd_array *array)
if (array->used >= array->alloced) {
int num = array->alloced + array->alloc_align;
int size = (num + 1) * array->elem_size;
- int oldsize = array->alloced * array->elem_size;
void *nlist;
if (snd_BUG_ON(num >= 4096))
return NULL;
- nlist = krealloc(array->list, size, GFP_KERNEL);
+ nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO);
if (!nlist)
return NULL;
- memset(nlist + oldsize, 0, size - oldsize);
array->list = nlist;
array->alloced = num;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 23ca1722aff1..c93f9021f452 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -757,6 +757,9 @@ struct hda_pcm_ops {
struct snd_pcm_substream *substream);
int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
struct snd_pcm_substream *substream);
+ unsigned int (*get_delay)(struct hda_pcm_stream *info,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream);
};
/* PCM information for each substream */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 2dbe767be16b..ac079f93c535 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -34,6 +34,7 @@
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
+#include "hda_beep.h"
#include "hda_generic.h"
@@ -150,15 +151,25 @@ static void parse_user_hints(struct hda_codec *codec)
val = snd_hda_get_bool_hint(codec, "add_stereo_mix_input");
if (val >= 0)
spec->add_stereo_mix_input = !!val;
+ /* the following two are just for compatibility */
val = snd_hda_get_bool_hint(codec, "add_out_jack_modes");
if (val >= 0)
- spec->add_out_jack_modes = !!val;
+ spec->add_jack_modes = !!val;
val = snd_hda_get_bool_hint(codec, "add_in_jack_modes");
if (val >= 0)
- spec->add_in_jack_modes = !!val;
+ spec->add_jack_modes = !!val;
+ val = snd_hda_get_bool_hint(codec, "add_jack_modes");
+ if (val >= 0)
+ spec->add_jack_modes = !!val;
val = snd_hda_get_bool_hint(codec, "power_down_unused");
if (val >= 0)
spec->power_down_unused = !!val;
+ val = snd_hda_get_bool_hint(codec, "add_hp_mic");
+ if (val >= 0)
+ spec->hp_mic = !!val;
+ val = snd_hda_get_bool_hint(codec, "hp_mic_detect");
+ if (val >= 0)
+ spec->suppress_hp_mic_detect = !val;
if (!snd_hda_get_int_hint(codec, "mixer_nid", &val))
spec->mixer_nid = val;
@@ -996,7 +1007,7 @@ enum {
/* Primary DAC shared with main surrounds */
BAD_SHARED_SURROUND = 0x100,
/* No independent HP possible */
- BAD_NO_INDEP_HP = 0x40,
+ BAD_NO_INDEP_HP = 0x10,
/* Primary DAC shared with main CLFE */
BAD_SHARED_CLFE = 0x10,
/* Primary DAC shared with extra surrounds */
@@ -1051,16 +1062,7 @@ static int assign_out_path_ctls(struct hda_codec *codec, struct nid_path *path)
return badness;
}
-struct badness_table {
- int no_primary_dac; /* no primary DAC */
- int no_dac; /* no secondary DACs */
- int shared_primary; /* primary DAC is shared with main output */
- int shared_surr; /* secondary DAC shared with main or primary */
- int shared_clfe; /* third DAC shared with main or primary */
- int shared_surr_main; /* secondary DAC sahred with main/DAC0 */
-};
-
-static struct badness_table main_out_badness = {
+const struct badness_table hda_main_out_badness = {
.no_primary_dac = BAD_NO_PRIMARY_DAC,
.no_dac = BAD_NO_DAC,
.shared_primary = BAD_NO_PRIMARY_DAC,
@@ -1068,8 +1070,9 @@ static struct badness_table main_out_badness = {
.shared_clfe = BAD_SHARED_CLFE,
.shared_surr_main = BAD_SHARED_SURROUND,
};
+EXPORT_SYMBOL_HDA(hda_main_out_badness);
-static struct badness_table extra_out_badness = {
+const struct badness_table hda_extra_out_badness = {
.no_primary_dac = BAD_NO_DAC,
.no_dac = BAD_NO_DAC,
.shared_primary = BAD_NO_EXTRA_DAC,
@@ -1077,6 +1080,7 @@ static struct badness_table extra_out_badness = {
.shared_clfe = BAD_SHARED_EXTRA_SURROUND,
.shared_surr_main = BAD_NO_EXTRA_SURR_DAC,
};
+EXPORT_SYMBOL_HDA(hda_extra_out_badness);
/* get the DAC of the primary output corresponding to the given array index */
static hda_nid_t get_primary_out(struct hda_codec *codec, int idx)
@@ -1367,22 +1371,25 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx)
{
struct hda_gen_spec *spec = codec->spec;
struct nid_path *path;
- hda_nid_t dac, pin;
+ hda_nid_t path_dac, dac, pin;
path = snd_hda_get_path_from_idx(codec, path_idx);
if (!path || !path->depth ||
is_nid_contained(path, spec->mixer_nid))
return 0;
- dac = path->path[0];
+ path_dac = path->path[0];
+ dac = spec->private_dac_nids[0];
pin = path->path[path->depth - 1];
path = snd_hda_add_new_path(codec, dac, pin, spec->mixer_nid);
if (!path) {
- if (dac != spec->multiout.dac_nids[0])
- dac = spec->multiout.dac_nids[0];
+ if (dac != path_dac)
+ dac = path_dac;
else if (spec->multiout.hp_out_nid[0])
dac = spec->multiout.hp_out_nid[0];
else if (spec->multiout.extra_out_nid[0])
dac = spec->multiout.extra_out_nid[0];
+ else
+ dac = 0;
if (dac)
path = snd_hda_add_new_path(codec, dac, pin,
spec->mixer_nid);
@@ -1507,7 +1514,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
badness += try_assign_dacs(codec, cfg->line_outs, cfg->line_out_pins,
spec->private_dac_nids, spec->out_paths,
- &main_out_badness);
+ spec->main_out_badness);
if (fill_mio_first &&
cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
@@ -1522,7 +1529,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
err = try_assign_dacs(codec, cfg->hp_outs, cfg->hp_pins,
spec->multiout.hp_out_nid,
spec->hp_paths,
- &extra_out_badness);
+ spec->extra_out_badness);
if (err < 0)
return err;
badness += err;
@@ -1532,7 +1539,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
cfg->speaker_pins,
spec->multiout.extra_out_nid,
spec->speaker_paths,
- &extra_out_badness);
+ spec->extra_out_badness);
if (err < 0)
return err;
badness += err;
@@ -1926,6 +1933,17 @@ static int create_speaker_out_ctls(struct hda_codec *codec)
* independent HP controls
*/
+/* update HP auto-mute state too */
+static void update_hp_automute_hook(struct hda_codec *codec)
+{
+ struct hda_gen_spec *spec = codec->spec;
+
+ if (spec->hp_automute_hook)
+ spec->hp_automute_hook(codec, NULL);
+ else
+ snd_hda_gen_hp_automute(codec, NULL);
+}
+
static int indep_hp_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -1986,12 +2004,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol,
else
*dacp = spec->alt_dac_nid;
- /* update HP auto-mute state too */
- if (spec->hp_automute_hook)
- spec->hp_automute_hook(codec, NULL);
- else
- snd_hda_gen_hp_automute(codec, NULL);
-
+ update_hp_automute_hook(codec);
ret = 1;
}
unlock:
@@ -2072,6 +2085,14 @@ get_multiio_path(struct hda_codec *codec, int idx)
static void update_automute_all(struct hda_codec *codec);
+/* Default value to be passed as aamix argument for snd_hda_activate_path();
+ * used for output paths
+ */
+static bool aamix_default(struct hda_gen_spec *spec)
+{
+ return !spec->have_aamix_ctl || spec->aamix_mode;
+}
+
static int set_multi_io(struct hda_codec *codec, int idx, bool output)
{
struct hda_gen_spec *spec = codec->spec;
@@ -2087,11 +2108,11 @@ static int set_multi_io(struct hda_codec *codec, int idx, bool output)
if (output) {
set_pin_target(codec, nid, PIN_OUT, true);
- snd_hda_activate_path(codec, path, true, true);
+ snd_hda_activate_path(codec, path, true, aamix_default(spec));
set_pin_eapd(codec, nid, true);
} else {
set_pin_eapd(codec, nid, false);
- snd_hda_activate_path(codec, path, false, true);
+ snd_hda_activate_path(codec, path, false, aamix_default(spec));
set_pin_target(codec, nid, spec->multi_io[idx].ctl_in, true);
path_power_down_sync(codec, path);
}
@@ -2182,8 +2203,8 @@ static void update_aamix_paths(struct hda_codec *codec, bool do_mix,
snd_hda_activate_path(codec, mix_path, true, true);
path_power_down_sync(codec, nomix_path);
} else {
- snd_hda_activate_path(codec, mix_path, false, true);
- snd_hda_activate_path(codec, nomix_path, true, true);
+ snd_hda_activate_path(codec, mix_path, false, false);
+ snd_hda_activate_path(codec, nomix_path, true, false);
path_power_down_sync(codec, mix_path);
}
}
@@ -2240,63 +2261,95 @@ static int create_loopback_mixing_ctl(struct hda_codec *codec)
static void call_update_outputs(struct hda_codec *codec);
/* for shared I/O, change the pin-control accordingly */
-static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic)
+static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force)
{
struct hda_gen_spec *spec = codec->spec;
+ bool as_mic;
unsigned int val;
- hda_nid_t pin = spec->autocfg.inputs[1].pin;
- /* NOTE: this assumes that there are only two inputs, the
- * first is the real internal mic and the second is HP/mic jack.
- */
+ hda_nid_t pin;
- val = snd_hda_get_default_vref(codec, pin);
+ pin = spec->hp_mic_pin;
+ as_mic = spec->cur_mux[adc_mux] == spec->hp_mic_mux_idx;
+
+ if (!force) {
+ val = snd_hda_codec_get_pin_target(codec, pin);
+ if (as_mic) {
+ if (val & PIN_IN)
+ return;
+ } else {
+ if (val & PIN_OUT)
+ return;
+ }
+ }
- /* This pin does not have vref caps - let's enable vref on pin 0x18
- instead, as suggested by Realtek */
+ val = snd_hda_get_default_vref(codec, pin);
+ /* if the HP pin doesn't support VREF and the codec driver gives an
+ * alternative pin, set up the VREF on that pin instead
+ */
if (val == AC_PINCTL_VREF_HIZ && spec->shared_mic_vref_pin) {
const hda_nid_t vref_pin = spec->shared_mic_vref_pin;
unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin);
if (vref_val != AC_PINCTL_VREF_HIZ)
snd_hda_set_pin_ctl_cache(codec, vref_pin,
- PIN_IN | (set_as_mic ? vref_val : 0));
+ PIN_IN | (as_mic ? vref_val : 0));
}
- val = set_as_mic ? val | PIN_IN : PIN_HP;
- set_pin_target(codec, pin, val, true);
-
- spec->automute_speaker = !set_as_mic;
- call_update_outputs(codec);
+ if (!spec->hp_mic_jack_modes) {
+ if (as_mic)
+ val |= PIN_IN;
+ else
+ val = PIN_HP;
+ set_pin_target(codec, pin, val, true);
+ update_hp_automute_hook(codec);
+ }
}
/* create a shared input with the headphone out */
-static int create_shared_input(struct hda_codec *codec)
+static int create_hp_mic(struct hda_codec *codec)
{
struct hda_gen_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
unsigned int defcfg;
hda_nid_t nid;
- /* only one internal input pin? */
- if (cfg->num_inputs != 1)
- return 0;
- defcfg = snd_hda_codec_get_pincfg(codec, cfg->inputs[0].pin);
- if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT)
+ if (!spec->hp_mic) {
+ if (spec->suppress_hp_mic_detect)
+ return 0;
+ /* automatic detection: only if no input or a single internal
+ * input pin is found, try to detect the shared hp/mic
+ */
+ if (cfg->num_inputs > 1)
+ return 0;
+ else if (cfg->num_inputs == 1) {
+ defcfg = snd_hda_codec_get_pincfg(codec, cfg->inputs[0].pin);
+ if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT)
+ return 0;
+ }
+ }
+
+ spec->hp_mic = 0; /* clear once */
+ if (cfg->num_inputs >= AUTO_CFG_MAX_INS)
return 0;
- if (cfg->hp_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
- nid = cfg->hp_pins[0]; /* OK, we have a single HP-out */
- else if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_HP_OUT)
- nid = cfg->line_out_pins[0]; /* OK, we have a single line-out */
- else
- return 0; /* both not available */
+ nid = 0;
+ if (cfg->line_out_type == AUTO_PIN_HP_OUT && cfg->line_outs > 0)
+ nid = cfg->line_out_pins[0];
+ else if (cfg->hp_outs > 0)
+ nid = cfg->hp_pins[0];
+ if (!nid)
+ return 0;
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_IN))
return 0; /* no input */
- cfg->inputs[1].pin = nid;
- cfg->inputs[1].type = AUTO_PIN_MIC;
- cfg->num_inputs = 2;
- spec->shared_mic_hp = 1;
+ cfg->inputs[cfg->num_inputs].pin = nid;
+ cfg->inputs[cfg->num_inputs].type = AUTO_PIN_MIC;
+ cfg->inputs[cfg->num_inputs].is_headphone_mic = 1;
+ cfg->num_inputs++;
+ spec->hp_mic = 1;
+ spec->hp_mic_pin = nid;
+ /* we can't handle auto-mic together with HP-mic */
+ spec->suppress_auto_mic = 1;
snd_printdd("hda-codec: Enable shared I/O jack on NID 0x%x\n", nid);
return 0;
}
@@ -2304,13 +2357,17 @@ static int create_shared_input(struct hda_codec *codec)
/*
* output jack mode
*/
+
+static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin);
+
+static const char * const out_jack_texts[] = {
+ "Line Out", "Headphone Out",
+};
+
static int out_jack_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static const char * const texts[] = {
- "Line Out", "Headphone Out",
- };
- return snd_hda_enum_helper_info(kcontrol, uinfo, 2, texts);
+ return snd_hda_enum_helper_info(kcontrol, uinfo, 2, out_jack_texts);
}
static int out_jack_mode_get(struct snd_kcontrol *kcontrol,
@@ -2372,6 +2429,17 @@ static void get_jack_mode_name(struct hda_codec *codec, hda_nid_t pin,
;
}
+static int get_out_jack_num_items(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ if (spec->add_jack_modes) {
+ unsigned int pincap = snd_hda_query_pin_caps(codec, pin);
+ if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV))
+ return 2;
+ }
+ return 1;
+}
+
static int create_out_jack_modes(struct hda_codec *codec, int num_pins,
hda_nid_t *pins)
{
@@ -2380,8 +2448,13 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins,
for (i = 0; i < num_pins; i++) {
hda_nid_t pin = pins[i];
- unsigned int pincap = snd_hda_query_pin_caps(codec, pin);
- if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV)) {
+ if (pin == spec->hp_mic_pin) {
+ int ret = create_hp_mic_jack_mode(codec, pin);
+ if (ret < 0)
+ return ret;
+ continue;
+ }
+ if (get_out_jack_num_items(codec, pin) > 1) {
struct snd_kcontrol_new *knew;
char name[44];
get_jack_mode_name(codec, pin, name, sizeof(name));
@@ -2502,12 +2575,24 @@ static const struct snd_kcontrol_new in_jack_mode_enum = {
.put = in_jack_mode_put,
};
+static int get_in_jack_num_items(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ int nitems = 0;
+ if (spec->add_jack_modes)
+ nitems = hweight32(get_vref_caps(codec, pin));
+ return nitems ? nitems : 1;
+}
+
static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin)
{
struct hda_gen_spec *spec = codec->spec;
- unsigned int defcfg;
struct snd_kcontrol_new *knew;
char name[44];
+ unsigned int defcfg;
+
+ if (pin == spec->hp_mic_pin)
+ return 0; /* already done in create_out_jack_mode() */
/* no jack mode for fixed pins */
defcfg = snd_hda_codec_get_pincfg(codec, pin);
@@ -2515,7 +2600,7 @@ static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin)
return 0;
/* no multiple vref caps? */
- if (hweight32(get_vref_caps(codec, pin)) <= 1)
+ if (get_in_jack_num_items(codec, pin) <= 1)
return 0;
get_jack_mode_name(codec, pin, name, sizeof(name));
@@ -2526,6 +2611,132 @@ static int create_in_jack_mode(struct hda_codec *codec, hda_nid_t pin)
return 0;
}
+/*
+ * HP/mic shared jack mode
+ */
+static int hp_mic_jack_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ int out_jacks = get_out_jack_num_items(codec, nid);
+ int in_jacks = get_in_jack_num_items(codec, nid);
+ const char *text = NULL;
+ int idx;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = out_jacks + in_jacks;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ idx = uinfo->value.enumerated.item;
+ if (idx < out_jacks) {
+ if (out_jacks > 1)
+ text = out_jack_texts[idx];
+ else
+ text = "Headphone Out";
+ } else {
+ idx -= out_jacks;
+ if (in_jacks > 1) {
+ unsigned int vref_caps = get_vref_caps(codec, nid);
+ text = vref_texts[get_vref_idx(vref_caps, idx)];
+ } else
+ text = "Mic In";
+ }
+
+ strcpy(uinfo->value.enumerated.name, text);
+ return 0;
+}
+
+static int get_cur_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t nid)
+{
+ int out_jacks = get_out_jack_num_items(codec, nid);
+ int in_jacks = get_in_jack_num_items(codec, nid);
+ unsigned int val = snd_hda_codec_get_pin_target(codec, nid);
+ int idx = 0;
+
+ if (val & PIN_OUT) {
+ if (out_jacks > 1 && val == PIN_HP)
+ idx = 1;
+ } else if (val & PIN_IN) {
+ idx = out_jacks;
+ if (in_jacks > 1) {
+ unsigned int vref_caps = get_vref_caps(codec, nid);
+ val &= AC_PINCTL_VREFEN;
+ idx += cvt_from_vref_idx(vref_caps, val);
+ }
+ }
+ return idx;
+}
+
+static int hp_mic_jack_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ ucontrol->value.enumerated.item[0] =
+ get_cur_hp_mic_jack_mode(codec, nid);
+ return 0;
+}
+
+static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ int out_jacks = get_out_jack_num_items(codec, nid);
+ int in_jacks = get_in_jack_num_items(codec, nid);
+ unsigned int val, oldval, idx;
+
+ oldval = get_cur_hp_mic_jack_mode(codec, nid);
+ idx = ucontrol->value.enumerated.item[0];
+ if (oldval == idx)
+ return 0;
+
+ if (idx < out_jacks) {
+ if (out_jacks > 1)
+ val = idx ? PIN_HP : PIN_OUT;
+ else
+ val = PIN_HP;
+ } else {
+ idx -= out_jacks;
+ if (in_jacks > 1) {
+ unsigned int vref_caps = get_vref_caps(codec, nid);
+ val = snd_hda_codec_get_pin_target(codec, nid);
+ val &= ~(AC_PINCTL_VREFEN | PIN_HP);
+ val |= get_vref_idx(vref_caps, idx) | PIN_IN;
+ } else
+ val = snd_hda_get_default_vref(codec, nid);
+ }
+ snd_hda_set_pin_ctl_cache(codec, nid, val);
+ update_hp_automute_hook(codec);
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new hp_mic_jack_mode_enum = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = hp_mic_jack_mode_info,
+ .get = hp_mic_jack_mode_get,
+ .put = hp_mic_jack_mode_put,
+};
+
+static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ struct snd_kcontrol_new *knew;
+
+ if (get_out_jack_num_items(codec, pin) <= 1 &&
+ get_in_jack_num_items(codec, pin) <= 1)
+ return 0; /* no need */
+ knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode",
+ &hp_mic_jack_mode_enum);
+ if (!knew)
+ return -ENOMEM;
+ knew->private_value = pin;
+ spec->hp_mic_jack_modes = 1;
+ return 0;
+}
/*
* Parse input paths
@@ -2648,7 +2859,6 @@ static int check_dyn_adc_switch(struct hda_codec *codec)
unsigned int ok_bits;
int i, n, nums;
- again:
nums = 0;
ok_bits = 0;
for (n = 0; n < spec->num_adc_nids; n++) {
@@ -2663,12 +2873,6 @@ static int check_dyn_adc_switch(struct hda_codec *codec)
}
if (!ok_bits) {
- if (spec->shared_mic_hp) {
- spec->shared_mic_hp = 0;
- imux->num_items = 1;
- goto again;
- }
-
/* check whether ADC-switch is possible */
for (i = 0; i < imux->num_items; i++) {
for (n = 0; n < spec->num_adc_nids; n++) {
@@ -2701,7 +2905,8 @@ static int check_dyn_adc_switch(struct hda_codec *codec)
spec->num_adc_nids = nums;
}
- if (imux->num_items == 1 || spec->shared_mic_hp) {
+ if (imux->num_items == 1 ||
+ (imux->num_items == 2 && spec->hp_mic)) {
snd_printdd("hda-codec: reducing to a single ADC\n");
spec->num_adc_nids = 1; /* reduce to a single ADC */
}
@@ -2738,6 +2943,8 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin,
snd_hda_get_path_idx(codec, path);
if (!imux_added) {
+ if (spec->hp_mic_pin == pin)
+ spec->hp_mic_mux_idx = imux->num_items;
spec->imux_pins[imux->num_items] = pin;
snd_hda_add_imux_item(imux, label, cfg_idx, NULL);
imux_added = true;
@@ -2812,7 +3019,8 @@ static int create_input_ctls(struct hda_codec *codec)
val = PIN_IN;
if (cfg->inputs[i].type == AUTO_PIN_MIC)
val |= snd_hda_get_default_vref(codec, pin);
- set_pin_target(codec, pin, val, false);
+ if (pin != spec->hp_mic_pin)
+ set_pin_target(codec, pin, val, false);
if (mixer) {
if (is_reachable_path(codec, pin, mixer)) {
@@ -2830,7 +3038,7 @@ static int create_input_ctls(struct hda_codec *codec)
if (err < 0)
return err;
- if (spec->add_in_jack_modes) {
+ if (spec->add_jack_modes) {
err = create_in_jack_mode(codec, pin);
if (err < 0)
return err;
@@ -3462,8 +3670,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
spec->cur_mux[adc_idx] = idx;
- if (spec->shared_mic_hp)
- update_shared_mic_hp(codec, spec->cur_mux[adc_idx]);
+ if (spec->hp_mic)
+ update_hp_mic(codec, adc_idx, false);
if (spec->dyn_adc_switch)
dyn_adc_pcm_resetup(codec, idx);
@@ -3511,18 +3719,21 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
for (i = 0; i < num_pins; i++) {
hda_nid_t nid = pins[i];
- unsigned int val;
+ unsigned int val, oldval;
if (!nid)
break;
+ oldval = snd_hda_codec_get_pin_target(codec, nid);
+ if (oldval & PIN_IN)
+ continue; /* no mute for inputs */
/* don't reset VREF value in case it's controlling
* the amp (see alc861_fixup_asus_amp_vref_0f())
*/
if (spec->keep_vref_in_automute)
- val = snd_hda_codec_get_pin_target(codec, nid) & ~PIN_HP;
+ val = oldval & ~PIN_HP;
else
val = 0;
if (!mute)
- val |= snd_hda_codec_get_pin_target(codec, nid);
+ val |= oldval;
/* here we call update_pin_ctl() so that the pinctl is changed
* without changing the pinctl target value;
* the original target value will be still referred at the
@@ -3543,8 +3754,7 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
* in general, HP pins/amps control should be enabled in all cases,
* but currently set only for master_mute, just to be safe
*/
- if (!spec->shared_mic_hp) /* don't change HP-pin when shared with mic */
- do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+ do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins, spec->master_mute);
if (!spec->automute_speaker)
@@ -3649,10 +3859,7 @@ static void update_automute_all(struct hda_codec *codec)
{
struct hda_gen_spec *spec = codec->spec;
- if (spec->hp_automute_hook)
- spec->hp_automute_hook(codec, NULL);
- else
- snd_hda_gen_hp_automute(codec, NULL);
+ update_hp_automute_hook(codec);
if (spec->line_automute_hook)
spec->line_automute_hook(codec, NULL);
else
@@ -3978,6 +4185,11 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
cfg = &spec->autocfg;
}
+ if (!spec->main_out_badness)
+ spec->main_out_badness = &hda_main_out_badness;
+ if (!spec->extra_out_badness)
+ spec->extra_out_badness = &hda_extra_out_badness;
+
fill_all_dac_nids(codec);
if (!cfg->line_outs) {
@@ -4024,7 +4236,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
err = create_loopback_mixing_ctl(codec);
if (err < 0)
return err;
- err = create_shared_input(codec);
+ err = create_hp_mic(codec);
if (err < 0)
return err;
err = create_input_ctls(codec);
@@ -4050,11 +4262,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
if (err < 0)
return err;
- if (!spec->shared_mic_hp) {
- err = check_auto_mic_availability(codec);
- if (err < 0)
- return err;
- }
+ err = check_auto_mic_availability(codec);
+ if (err < 0)
+ return err;
err = create_capture_mixers(codec);
if (err < 0)
@@ -4064,7 +4274,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
if (err < 0)
return err;
- if (spec->add_out_jack_modes) {
+ if (spec->add_jack_modes) {
if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
err = create_out_jack_modes(codec, cfg->line_outs,
cfg->line_out_pins);
@@ -4085,6 +4295,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
if (spec->power_down_unused)
codec->power_filter = snd_hda_gen_path_power_filter;
+ if (!spec->no_analog && spec->beep_nid) {
+ err = snd_hda_attach_beep_device(codec, spec->beep_nid);
+ if (err < 0)
+ return err;
+ }
+
return 1;
}
EXPORT_SYMBOL_HDA(snd_hda_gen_parse_auto_config);
@@ -4161,17 +4377,6 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
free_kctls(spec); /* no longer needed */
- if (spec->shared_mic_hp) {
- int err;
- int nid = spec->autocfg.inputs[1].pin;
- err = snd_hda_jack_add_kctl(codec, nid, "Headphone Mic", 0);
- if (err < 0)
- return err;
- err = snd_hda_jack_detect_enable(codec, nid, 0);
- if (err < 0)
- return err;
- }
-
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -4729,7 +4934,8 @@ static void set_output_and_unmute(struct hda_codec *codec, int path_idx)
return;
pin = path->path[path->depth - 1];
restore_pin_ctl(codec, pin);
- snd_hda_activate_path(codec, path, path->active, true);
+ snd_hda_activate_path(codec, path, path->active,
+ aamix_default(codec->spec));
set_pin_eapd(codec, pin, path->active);
}
@@ -4779,7 +4985,8 @@ static void init_multi_io(struct hda_codec *codec)
if (!spec->multi_io[i].ctl_in)
spec->multi_io[i].ctl_in =
snd_hda_codec_get_pin_target(codec, pin);
- snd_hda_activate_path(codec, path, path->active, true);
+ snd_hda_activate_path(codec, path, path->active,
+ aamix_default(spec));
}
}
@@ -4826,11 +5033,10 @@ static void init_input_src(struct hda_codec *codec)
snd_hda_activate_path(codec, path, active, false);
}
}
+ if (spec->hp_mic)
+ update_hp_mic(codec, c, true);
}
- if (spec->shared_mic_hp)
- update_shared_mic_hp(codec, spec->cur_mux[0]);
-
if (spec->cap_sync_hook)
spec->cap_sync_hook(codec, NULL);
}
@@ -4911,6 +5117,7 @@ EXPORT_SYMBOL_HDA(snd_hda_gen_init);
*/
void snd_hda_gen_free(struct hda_codec *codec)
{
+ snd_hda_detach_beep_device(codec);
snd_hda_gen_spec_free(codec->spec);
kfree(codec->spec);
codec->spec = NULL;
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 009b57be96d3..54e665160379 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -76,6 +76,19 @@ enum {
HDA_GEN_PCM_ACT_CLOSE,
};
+/* DAC assignment badness table */
+struct badness_table {
+ int no_primary_dac; /* no primary DAC */
+ int no_dac; /* no secondary DACs */
+ int shared_primary; /* primary DAC is shared with main output */
+ int shared_surr; /* secondary DAC shared with main or primary */
+ int shared_clfe; /* third DAC shared with main or primary */
+ int shared_surr_main; /* secondary DAC sahred with main/DAC0 */
+};
+
+extern const struct badness_table hda_main_out_badness;
+extern const struct badness_table hda_extra_out_badness;
+
struct hda_gen_spec {
char stream_name_analog[32]; /* analog PCM stream */
const struct hda_pcm_stream *stream_analog_playback;
@@ -145,7 +158,10 @@ struct hda_gen_spec {
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS];
unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS];
+ /* shared hp/mic */
hda_nid_t shared_mic_vref_pin;
+ hda_nid_t hp_mic_pin;
+ int hp_mic_mux_idx;
/* DAC/ADC lists */
int num_all_dacs;
@@ -200,7 +216,8 @@ struct hda_gen_spec {
/* other parse behavior flags */
unsigned int need_dac_fix:1; /* need to limit DACs for multi channels */
- unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
+ unsigned int hp_mic:1; /* Allow HP as a mic-in */
+ unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
@@ -209,8 +226,7 @@ struct hda_gen_spec {
unsigned int indep_hp:1; /* independent HP supported */
unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */
unsigned int add_stereo_mix_input:1; /* add aamix as a capture src */
- unsigned int add_out_jack_modes:1; /* add output jack mode enum ctls */
- unsigned int add_in_jack_modes:1; /* add input jack mode enum ctls */
+ unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */
unsigned int power_down_unused:1; /* power down unused widgets */
/* other internal flags */
@@ -218,10 +234,18 @@ struct hda_gen_spec {
unsigned int dyn_adc_switch:1; /* switch ADCs (for ALC275) */
unsigned int indep_hp_enabled:1; /* independent HP enabled */
unsigned int have_aamix_ctl:1;
+ unsigned int hp_mic_jack_modes:1;
+
+ /* badness tables for output path evaluations */
+ const struct badness_table *main_out_badness;
+ const struct badness_table *extra_out_badness;
/* loopback mixing mode */
bool aamix_mode;
+ /* digital beep */
+ hda_nid_t beep_nid;
+
/* for virtual master */
hda_nid_t vmaster_nid;
unsigned int vmaster_tlv[4];
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index bcd40ee488e3..7b213d589ef6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1889,6 +1889,26 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream,
tc->cycle_last = last;
}
+static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream,
+ u64 nsec)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+ u64 codec_frames, codec_nsecs;
+
+ if (!hinfo->ops.get_delay)
+ return nsec;
+
+ codec_frames = hinfo->ops.get_delay(hinfo, apcm->codec, substream);
+ codec_nsecs = div_u64(codec_frames * 1000000000LL,
+ substream->runtime->rate);
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return nsec + codec_nsecs;
+
+ return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0;
+}
+
static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream,
struct timespec *ts)
{
@@ -1897,6 +1917,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream,
nsec = timecounter_read(&azx_dev->azx_tc);
nsec = div_u64(nsec, 3); /* can be optimized */
+ nsec = azx_adjust_codec_delay(substream, nsec);
*ts = ns_to_timespec(nsec);
@@ -2349,8 +2370,11 @@ static unsigned int azx_get_position(struct azx *chip,
struct azx_dev *azx_dev,
bool with_check)
{
+ struct snd_pcm_substream *substream = azx_dev->substream;
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
unsigned int pos;
- int stream = azx_dev->substream->stream;
+ int stream = substream->stream;
+ struct hda_pcm_stream *hinfo = apcm->hinfo[stream];
int delay = 0;
switch (chip->position_fix[stream]) {
@@ -2381,7 +2405,7 @@ static unsigned int azx_get_position(struct azx *chip,
pos = 0;
/* calculate runtime delay from LPIB */
- if (azx_dev->substream->runtime &&
+ if (substream->runtime &&
chip->position_fix[stream] == POS_FIX_POSBUF &&
(chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) {
unsigned int lpib_pos = azx_sd_readl(azx_dev, SD_LPIB);
@@ -2399,9 +2423,16 @@ static unsigned int azx_get_position(struct azx *chip,
delay = 0;
chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY;
}
- azx_dev->substream->runtime->delay =
- bytes_to_frames(azx_dev->substream->runtime, delay);
+ delay = bytes_to_frames(substream->runtime, delay);
}
+
+ if (substream->runtime) {
+ if (hinfo->ops.get_delay)
+ delay += hinfo->ops.get_delay(hinfo, apcm->codec,
+ substream);
+ substream->runtime->delay = delay;
+ }
+
trace_azx_get_position(chip, azx_dev, pos, delay);
return pos;
}
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 1d035efeff4f..9e0a95288f46 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -394,7 +394,8 @@ static int get_unique_index(struct hda_codec *codec, const char *name, int idx)
}
static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg)
+ const struct auto_pin_cfg *cfg,
+ const char *base_name)
{
unsigned int def_conf, conn;
char name[44];
@@ -410,7 +411,11 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
phantom_jack = (conn != AC_JACK_PORT_COMPLEX) ||
!is_jack_detectable(codec, nid);
- snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
+ if (base_name) {
+ strlcpy(name, base_name, sizeof(name));
+ idx = 0;
+ } else
+ snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
if (phantom_jack)
/* Example final name: "Internal Mic Phantom Jack" */
strncat(name, " Phantom", sizeof(name) - strlen(name) - 1);
@@ -433,39 +438,51 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
const hda_nid_t *p;
int i, err;
+ for (i = 0; i < cfg->num_inputs; i++) {
+ /* If we have headphone mics; make sure they get the right name
+ before grabbed by output pins */
+ if (cfg->inputs[i].is_headphone_mic) {
+ if (auto_cfg_hp_outs(cfg) == 1)
+ err = add_jack_kctl(codec, auto_cfg_hp_pins(cfg)[0],
+ cfg, "Headphone Mic");
+ else
+ err = add_jack_kctl(codec, cfg->inputs[i].pin,
+ cfg, "Headphone Mic");
+ } else
+ err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg,
+ NULL);
+ if (err < 0)
+ return err;
+ }
+
for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, NULL);
if (err < 0)
return err;
}
for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, NULL);
if (err < 0)
return err;
}
for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
- if (err < 0)
- return err;
- }
- for (i = 0; i < cfg->num_inputs; i++) {
- err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
+ err = add_jack_kctl(codec, *p, cfg, NULL);
if (err < 0)
return err;
}
for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, NULL);
if (err < 0)
return err;
}
- err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
+ err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, NULL);
if (err < 0)
return err;
- err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
+ err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, NULL);
if (err < 0)
return err;
return 0;
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 83b7486c8eff..e0bf7534fa1f 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -670,6 +670,10 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid,
return (state != target_state);
}
+unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
+ hda_nid_t nid,
+ unsigned int power_state);
+
/*
* AMP control callbacks
*/
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index df8014b27596..977b0d878dae 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -43,7 +43,6 @@ struct ad198x_spec {
hda_nid_t eapd_nid;
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
- hda_nid_t beep_dev_nid;
#ifdef ENABLE_AD_STATIC_QUIRKS
const struct snd_kcontrol_new *mixers[6];
@@ -609,7 +608,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = {
.build_controls = ad198x_auto_build_controls,
.build_pcms = snd_hda_gen_build_pcms,
.init = snd_hda_gen_init,
- .free = ad198x_free,
+ .free = snd_hda_gen_free,
.unsol_event = snd_hda_jack_unsol_event,
#ifdef CONFIG_PM
.check_power_status = snd_hda_gen_check_power_status,
@@ -638,12 +637,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
- if (spec->beep_dev_nid) {
- err = snd_hda_attach_beep_device(codec, spec->beep_dev_nid);
- if (err < 0)
- return err;
- }
-
codec->patch_ops = ad198x_auto_patch_ops;
return 0;
@@ -1240,7 +1233,7 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
codec->inv_eapd = 1;
spec->gen.mixer_nid = 0x07;
- spec->beep_dev_nid = 0x19;
+ spec->gen.beep_nid = 0x19;
set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
/* AD1986A has a hardware problem that it can't share a stream
@@ -1256,7 +1249,7 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
err = ad198x_parse_auto_config(codec);
if (err < 0) {
- ad198x_free(codec);
+ snd_hda_gen_free(codec);
return err;
}
@@ -1673,7 +1666,7 @@ static int ad1983_parse_auto_config(struct hda_codec *codec)
return err;
spec = codec->spec;
- spec->beep_dev_nid = 0x10;
+ spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
err = ad198x_parse_auto_config(codec);
if (err < 0)
@@ -1684,7 +1677,7 @@ static int ad1983_parse_auto_config(struct hda_codec *codec)
return 0;
error:
- ad198x_free(codec);
+ snd_hda_gen_free(codec);
return err;
}
@@ -2187,7 +2180,7 @@ static int ad1981_parse_auto_config(struct hda_codec *codec)
spec = codec->spec;
spec->gen.mixer_nid = 0x0e;
- spec->beep_dev_nid = 0x10;
+ spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
snd_hda_pick_fixup(codec, NULL, ad1981_fixup_tbl, ad1981_fixups);
@@ -2205,7 +2198,7 @@ static int ad1981_parse_auto_config(struct hda_codec *codec)
return 0;
error:
- ad198x_free(codec);
+ snd_hda_gen_free(codec);
return err;
}
@@ -3236,7 +3229,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
spec->gen.mixer_nid = 0x20;
spec->gen.mixer_merge_nid = 0x21;
- spec->beep_dev_nid = 0x10;
+ spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
err = ad198x_parse_auto_config(codec);
if (err < 0)
@@ -3247,7 +3240,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
return 0;
error:
- ad198x_free(codec);
+ snd_hda_gen_free(codec);
return err;
}
@@ -3653,7 +3646,7 @@ static int ad1884_parse_auto_config(struct hda_codec *codec)
spec = codec->spec;
spec->gen.mixer_nid = 0x20;
- spec->beep_dev_nid = 0x10;
+ spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
snd_hda_pick_fixup(codec, NULL, ad1884_fixup_tbl, ad1884_fixups);
@@ -3671,7 +3664,7 @@ static int ad1884_parse_auto_config(struct hda_codec *codec)
return 0;
error:
- ad198x_free(codec);
+ snd_hda_gen_free(codec);
return err;
}
@@ -5155,7 +5148,7 @@ static int ad1882_parse_auto_config(struct hda_codec *codec)
spec->gen.mixer_nid = 0x20;
spec->gen.mixer_merge_nid = 0x21;
- spec->beep_dev_nid = 0x10;
+ spec->gen.beep_nid = 0x10;
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
err = ad198x_parse_auto_config(codec);
if (err < 0)
@@ -5166,7 +5159,7 @@ static int ad1882_parse_auto_config(struct hda_codec *codec)
return 0;
error:
- ad198x_free(codec);
+ snd_hda_gen_free(codec);
return err;
}
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0792b5725f9c..90ff7a3f72df 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -131,6 +131,13 @@ enum {
/* Effects values size*/
#define EFFECT_VALS_MAX_COUNT 12
+/* Latency introduced by DSP blocks in milliseconds. */
+#define DSP_CAPTURE_INIT_LATENCY 0
+#define DSP_CRYSTAL_VOICE_LATENCY 124
+#define DSP_PLAYBACK_INIT_LATENCY 13
+#define DSP_PLAY_ENHANCEMENT_LATENCY 30
+#define DSP_SPEAKER_OUT_LATENCY 7
+
struct ct_effect {
char name[44];
hda_nid_t nid;
@@ -741,6 +748,9 @@ struct ca0132_spec {
long voicefx_val;
long cur_mic_boost;
+ struct hda_codec *codec;
+ struct delayed_work unsol_hp_work;
+
#ifdef ENABLE_TUNING_CONTROLS
long cur_ctl_vals[TUNING_CTLS_COUNT];
#endif
@@ -2740,6 +2750,31 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
return 0;
}
+static unsigned int ca0132_playback_pcm_delay(struct hda_pcm_stream *info,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int latency = DSP_PLAYBACK_INIT_LATENCY;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (spec->dsp_state != DSP_DOWNLOADED)
+ return 0;
+
+ /* Add latency if playback enhancement and either effect is enabled. */
+ if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) {
+ if ((spec->effects_switch[SURROUND - EFFECT_START_NID]) ||
+ (spec->effects_switch[DIALOG_PLUS - EFFECT_START_NID]))
+ latency += DSP_PLAY_ENHANCEMENT_LATENCY;
+ }
+
+ /* Applying Speaker EQ adds latency as well. */
+ if (spec->cur_out_type == SPEAKER_OUT)
+ latency += DSP_SPEAKER_OUT_LATENCY;
+
+ return (latency * runtime->rate) / 1000;
+}
+
/*
* Digital out
*/
@@ -2808,6 +2843,23 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
return 0;
}
+static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int latency = DSP_CAPTURE_INIT_LATENCY;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (spec->dsp_state != DSP_DOWNLOADED)
+ return 0;
+
+ if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID])
+ latency += DSP_CRYSTAL_VOICE_LATENCY;
+
+ return (latency * runtime->rate) / 1000;
+}
+
/*
* Controls stuffs.
*/
@@ -3227,6 +3279,14 @@ exit:
return err < 0 ? err : 0;
}
+static void ca0132_unsol_hp_delayed(struct work_struct *work)
+{
+ struct ca0132_spec *spec = container_of(
+ to_delayed_work(work), struct ca0132_spec, unsol_hp_work);
+ ca0132_select_out(spec->codec);
+ snd_hda_jack_report_sync(spec->codec);
+}
+
static void ca0132_set_dmic(struct hda_codec *codec, int enable);
static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
@@ -3991,7 +4051,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
.channels_max = 6,
.ops = {
.prepare = ca0132_playback_pcm_prepare,
- .cleanup = ca0132_playback_pcm_cleanup
+ .cleanup = ca0132_playback_pcm_cleanup,
+ .get_delay = ca0132_playback_pcm_delay,
},
};
@@ -4001,7 +4062,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
.channels_max = 2,
.ops = {
.prepare = ca0132_capture_pcm_prepare,
- .cleanup = ca0132_capture_pcm_cleanup
+ .cleanup = ca0132_capture_pcm_cleanup,
+ .get_delay = ca0132_capture_pcm_delay,
},
};
@@ -4399,8 +4461,7 @@ static void ca0132_process_dsp_response(struct hda_codec *codec)
static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd(KERN_INFO "ca0132_unsol_event: 0x%x\n", res);
-
+ struct ca0132_spec *spec = codec->spec;
if (((res >> AC_UNSOL_RES_TAG_SHIFT) & 0x3f) == UNSOL_TAG_DSP) {
ca0132_process_dsp_response(codec);
@@ -4412,8 +4473,13 @@ static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res)
switch (res) {
case UNSOL_TAG_HP:
- ca0132_select_out(codec);
- snd_hda_jack_report_sync(codec);
+ /* Delay enabling the HP amp, to let the mic-detection
+ * state machine run.
+ */
+ cancel_delayed_work_sync(&spec->unsol_hp_work);
+ queue_delayed_work(codec->bus->workq,
+ &spec->unsol_hp_work,
+ msecs_to_jiffies(500));
break;
case UNSOL_TAG_AMIC1:
ca0132_select_mic(codec);
@@ -4588,6 +4654,7 @@ static void ca0132_free(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
+ cancel_delayed_work_sync(&spec->unsol_hp_work);
snd_hda_power_up(codec);
snd_hda_sequence_write(codec, spec->base_exit_verbs);
ca0132_exit_chip(codec);
@@ -4653,6 +4720,7 @@ static int patch_ca0132(struct hda_codec *codec)
if (!spec)
return -ENOMEM;
codec->spec = spec;
+ spec->codec = codec;
spec->num_mixers = 1;
spec->mixers[0] = ca0132_mixer;
@@ -4663,6 +4731,8 @@ static int patch_ca0132(struct hda_codec *codec)
spec->init_verbs[1] = ca0132_init_verbs1;
spec->num_init_verbs = 2;
+ INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed);
+
ca0132_init_chip(codec);
ca0132_config(codec);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 0d9c58f13560..bd8d46cca2b3 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -68,6 +68,7 @@ enum {
enum {
CS421X_CDB4210,
CS421X_SENSE_B,
+ CS421X_STUMPY,
};
/* Vendor-specific processing widget */
@@ -538,6 +539,7 @@ static int patch_cs420x(struct hda_codec *codec)
/* CS4210 board names */
static const struct hda_model_fixup cs421x_models[] = {
{ .id = CS421X_CDB4210, .name = "cdb4210" },
+ { .id = CS421X_STUMPY, .name = "stumpy" },
{}
};
@@ -559,6 +561,17 @@ static const struct hda_pintbl cdb4210_pincfgs[] = {
{} /* terminator */
};
+/* Stumpy ChromeBox */
+static const struct hda_pintbl stumpy_pincfgs[] = {
+ { 0x05, 0x022120f0 },
+ { 0x06, 0x901700f0 },
+ { 0x07, 0x02a120f0 },
+ { 0x08, 0x77a70037 },
+ { 0x09, 0x77a6003e },
+ { 0x0a, 0x434510f0 },
+ {} /* terminator */
+};
+
/* Setup GPIO/SENSE for each board (if used) */
static void cs421x_fixup_sense_b(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
@@ -578,7 +591,11 @@ static const struct hda_fixup cs421x_fixups[] = {
[CS421X_SENSE_B] = {
.type = HDA_FIXUP_FUNC,
.v.func = cs421x_fixup_sense_b,
- }
+ },
+ [CS421X_STUMPY] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = stumpy_pincfgs,
+ },
};
static const struct hda_verb cs421x_coef_init_verbs[] = {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2a89d1eefeb6..84b81c874a4a 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -139,8 +139,12 @@ struct conexant_spec {
#ifdef CONFIG_SND_HDA_INPUT_BEEP
-#define set_beep_amp(spec, nid, idx, dir) \
- ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir))
+static inline void set_beep_amp(struct conexant_spec *spec, hda_nid_t nid,
+ int idx, int dir)
+{
+ spec->gen.beep_nid = nid;
+ spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir);
+}
/* additional beep mixers; the actual parameters are overwritten at build */
static const struct snd_kcontrol_new cxt_beep_mixer[] = {
HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT),
@@ -2942,7 +2946,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
@@ -3191,17 +3194,11 @@ static int cx_auto_build_controls(struct hda_codec *codec)
return 0;
}
-static void cx_auto_free(struct hda_codec *codec)
-{
- snd_hda_detach_beep_device(codec);
- snd_hda_gen_free(codec);
-}
-
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = cx_auto_build_controls,
.build_pcms = snd_hda_gen_build_pcms,
.init = snd_hda_gen_init,
- .free = cx_auto_free,
+ .free = snd_hda_gen_free,
.unsol_event = snd_hda_jack_unsol_event,
#ifdef CONFIG_PM
.check_power_status = snd_hda_gen_check_power_status,
@@ -3310,6 +3307,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
+ SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
@@ -3356,7 +3354,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
switch (codec->vendor_id) {
case 0x14f15045:
codec->single_adc_amp = 1;
- codec->power_filter = NULL; /* Needs speaker amp to D3 to avoid click */
break;
case 0x14f15047:
codec->pin_amp_workaround = 1;
@@ -3396,8 +3393,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
goto error;
codec->patch_ops = cx_auto_patch_ops;
- if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
/* Some laptops with Conexant chips show stalls in S3 resume,
* which falls into the single-cmd mode.
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index de8ac5c07fd0..32930e668854 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -44,16 +44,6 @@ static bool static_hdmi_pcm;
module_param(static_hdmi_pcm, bool, 0644);
MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
-/*
- * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device
- * could support N independent pipes, each of them can be connected to one or
- * more ports (DVI, HDMI or DisplayPort).
- *
- * The HDA correspondence of pipes/ports are converter/pin nodes.
- */
-#define MAX_HDMI_CVTS 8
-#define MAX_HDMI_PINS 8
-
struct hdmi_spec_per_cvt {
hda_nid_t cvt_nid;
int assigned;
@@ -80,16 +70,17 @@ struct hdmi_spec_per_pin {
bool non_pcm;
bool chmap_set; /* channel-map override by ALSA API? */
unsigned char chmap[8]; /* ALSA API channel-map */
+ char pcm_name[8]; /* filled in build_pcm callbacks */
};
struct hdmi_spec {
int num_cvts;
- struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS];
- hda_nid_t cvt_nids[MAX_HDMI_CVTS];
+ struct snd_array cvts; /* struct hdmi_spec_per_cvt */
+ hda_nid_t cvt_nids[4]; /* only for haswell fix */
int num_pins;
- struct hdmi_spec_per_pin pins[MAX_HDMI_PINS];
- struct hda_pcm pcm_rec[MAX_HDMI_PINS];
+ struct snd_array pins; /* struct hdmi_spec_per_pin */
+ struct snd_array pcm_rec; /* struct hda_pcm */
unsigned int channels_max; /* max over all cvts */
struct hdmi_eld temp_eld;
@@ -304,12 +295,19 @@ static struct cea_channel_speaker_allocation channel_allocations[] = {
* HDMI routines
*/
+#define get_pin(spec, idx) \
+ ((struct hdmi_spec_per_pin *)snd_array_elem(&spec->pins, idx))
+#define get_cvt(spec, idx) \
+ ((struct hdmi_spec_per_cvt *)snd_array_elem(&spec->cvts, idx))
+#define get_pcm_rec(spec, idx) \
+ ((struct hda_pcm *)snd_array_elem(&spec->pcm_rec, idx))
+
static int pin_nid_to_pin_index(struct hdmi_spec *spec, hda_nid_t pin_nid)
{
int pin_idx;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++)
- if (spec->pins[pin_idx].pin_nid == pin_nid)
+ if (get_pin(spec, pin_idx)->pin_nid == pin_nid)
return pin_idx;
snd_printk(KERN_WARNING "HDMI: pin nid %d not registered\n", pin_nid);
@@ -322,7 +320,7 @@ static int hinfo_to_pin_index(struct hdmi_spec *spec,
int pin_idx;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++)
- if (&spec->pcm_rec[pin_idx].stream[0] == hinfo)
+ if (get_pcm_rec(spec, pin_idx)->stream == hinfo)
return pin_idx;
snd_printk(KERN_WARNING "HDMI: hinfo %p not registered\n", hinfo);
@@ -334,7 +332,7 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid)
int cvt_idx;
for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++)
- if (spec->cvts[cvt_idx].cvt_nid == cvt_nid)
+ if (get_cvt(spec, cvt_idx)->cvt_nid == cvt_nid)
return cvt_idx;
snd_printk(KERN_WARNING "HDMI: cvt nid %d not registered\n", cvt_nid);
@@ -352,7 +350,7 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
pin_idx = kcontrol->private_value;
- eld = &spec->pins[pin_idx].sink_eld;
+ eld = &get_pin(spec, pin_idx)->sink_eld;
mutex_lock(&eld->lock);
uinfo->count = eld->eld_valid ? eld->eld_size : 0;
@@ -370,7 +368,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
int pin_idx;
pin_idx = kcontrol->private_value;
- eld = &spec->pins[pin_idx].sink_eld;
+ eld = &get_pin(spec, pin_idx)->sink_eld;
mutex_lock(&eld->lock);
if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) {
@@ -410,11 +408,11 @@ static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx,
kctl->private_value = pin_idx;
kctl->id.device = device;
- err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl);
+ err = snd_hda_ctl_add(codec, get_pin(spec, pin_idx)->pin_nid, kctl);
if (err < 0)
return err;
- spec->pins[pin_idx].eld_ctl = kctl;
+ get_pin(spec, pin_idx)->eld_ctl = kctl;
return 0;
}
@@ -875,14 +873,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx,
struct snd_pcm_substream *substream)
{
struct hdmi_spec *spec = codec->spec;
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
int channels = substream->runtime->channels;
struct hdmi_eld *eld;
int ca;
union audio_infoframe ai;
- eld = &spec->pins[pin_idx].sink_eld;
+ eld = &per_pin->sink_eld;
if (!eld->monitor_present)
return;
@@ -977,7 +975,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
if (pin_idx < 0)
return;
- hdmi_present_sense(&spec->pins[pin_idx], 1);
+ hdmi_present_sense(get_pin(spec, pin_idx), 1);
snd_hda_jack_report_sync(codec);
}
@@ -1020,6 +1018,41 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
hdmi_non_intrinsic_event(codec, res);
}
+static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid)
+{
+ int pwr, lamp, ramp;
+
+ pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0);
+ pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT;
+ if (pwr != AC_PWRST_D0) {
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
+ msleep(40);
+ pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0);
+ pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT;
+ snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr);
+ }
+
+ lamp = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT);
+ ramp = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT);
+ if (lamp != ramp) {
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp);
+
+ lamp = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT);
+ ramp = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT);
+ snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp);
+ }
+}
+
/*
* Callbacks
*/
@@ -1034,6 +1067,9 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
int pinctl;
int new_pinctl = 0;
+ if (codec->vendor_id == 0x80862807)
+ haswell_verify_pin_D0(codec, pin_nid);
+
if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) {
pinctl = snd_hda_codec_read(codec, pin_nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
@@ -1083,12 +1119,12 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
pin_idx = hinfo_to_pin_index(spec, hinfo);
if (snd_BUG_ON(pin_idx < 0))
return -EINVAL;
- per_pin = &spec->pins[pin_idx];
+ per_pin = get_pin(spec, pin_idx);
eld = &per_pin->sink_eld;
/* Dynamically assign converter to stream */
for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) {
- per_cvt = &spec->cvts[cvt_idx];
+ per_cvt = get_cvt(spec, cvt_idx);
/* Must not already be assigned */
if (per_cvt->assigned)
@@ -1151,7 +1187,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx)
{
struct hdmi_spec *spec = codec->spec;
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) {
@@ -1275,14 +1311,13 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
return 0;
- if (snd_BUG_ON(spec->num_pins >= MAX_HDMI_PINS))
- return -E2BIG;
-
if (codec->vendor_id == 0x80862807)
intel_haswell_fixup_connect_list(codec, pin_nid);
pin_idx = spec->num_pins;
- per_pin = &spec->pins[pin_idx];
+ per_pin = snd_array_new(&spec->pins);
+ if (!per_pin)
+ return -ENOMEM;
per_pin->pin_nid = pin_nid;
per_pin->non_pcm = false;
@@ -1299,19 +1334,16 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
{
struct hdmi_spec *spec = codec->spec;
- int cvt_idx;
struct hdmi_spec_per_cvt *per_cvt;
unsigned int chans;
int err;
- if (snd_BUG_ON(spec->num_cvts >= MAX_HDMI_CVTS))
- return -E2BIG;
-
chans = get_wcaps(codec, cvt_nid);
chans = get_wcaps_channels(chans);
- cvt_idx = spec->num_cvts;
- per_cvt = &spec->cvts[cvt_idx];
+ per_cvt = snd_array_new(&spec->cvts);
+ if (!per_cvt)
+ return -ENOMEM;
per_cvt->cvt_nid = cvt_nid;
per_cvt->channels_min = 2;
@@ -1328,7 +1360,9 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
if (err < 0)
return err;
- spec->cvt_nids[spec->num_cvts++] = cvt_nid;
+ if (spec->num_cvts < ARRAY_SIZE(spec->cvt_nids))
+ spec->cvt_nids[spec->num_cvts] = cvt_nid;
+ spec->num_cvts++;
return 0;
}
@@ -1384,13 +1418,6 @@ static int hdmi_parse_codec(struct hda_codec *codec)
/*
*/
-static char *get_hdmi_pcm_name(int idx)
-{
- static char names[MAX_HDMI_PINS][8];
- sprintf(&names[idx][0], "HDMI %d", idx);
- return &names[idx][0];
-}
-
static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
{
struct hda_spdif_out *spdif;
@@ -1417,7 +1444,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
hda_nid_t cvt_nid = hinfo->nid;
struct hdmi_spec *spec = codec->spec;
int pin_idx = hinfo_to_pin_index(spec, hinfo);
- hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid;
+ hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid;
bool non_pcm;
non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
@@ -1450,7 +1477,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid);
if (snd_BUG_ON(cvt_idx < 0))
return -EINVAL;
- per_cvt = &spec->cvts[cvt_idx];
+ per_cvt = get_cvt(spec, cvt_idx);
snd_BUG_ON(!per_cvt->assigned);
per_cvt->assigned = 0;
@@ -1459,7 +1486,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
pin_idx = hinfo_to_pin_index(spec, hinfo);
if (snd_BUG_ON(pin_idx < 0))
return -EINVAL;
- per_pin = &spec->pins[pin_idx];
+ per_pin = get_pin(spec, pin_idx);
snd_hda_spdif_ctls_unassign(codec, pin_idx);
per_pin->chmap_set = false;
@@ -1553,7 +1580,7 @@ static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = info->private_data;
struct hdmi_spec *spec = codec->spec;
int pin_idx = kcontrol->private_value;
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
int i;
for (i = 0; i < ARRAY_SIZE(per_pin->chmap); i++)
@@ -1568,7 +1595,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = info->private_data;
struct hdmi_spec *spec = codec->spec;
int pin_idx = kcontrol->private_value;
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
unsigned int ctl_idx;
struct snd_pcm_substream *substream;
unsigned char chmap[8];
@@ -1613,9 +1640,14 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
struct hda_pcm *info;
struct hda_pcm_stream *pstr;
-
- info = &spec->pcm_rec[pin_idx];
- info->name = get_hdmi_pcm_name(pin_idx);
+ struct hdmi_spec_per_pin *per_pin;
+
+ per_pin = get_pin(spec, pin_idx);
+ sprintf(per_pin->pcm_name, "HDMI %d", pin_idx);
+ info = snd_array_new(&spec->pcm_rec);
+ if (!info)
+ return -ENOMEM;
+ info->name = per_pin->pcm_name;
info->pcm_type = HDA_PCM_TYPE_HDMI;
info->own_chmap = true;
@@ -1626,7 +1658,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
}
codec->num_pcms = spec->num_pins;
- codec->pcm_info = spec->pcm_rec;
+ codec->pcm_info = spec->pcm_rec.list;
return 0;
}
@@ -1635,8 +1667,8 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx)
{
char hdmi_str[32] = "HDMI/DP";
struct hdmi_spec *spec = codec->spec;
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
- int pcmdev = spec->pcm_rec[pin_idx].device;
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+ int pcmdev = get_pcm_rec(spec, pin_idx)->device;
if (pcmdev > 0)
sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
@@ -1654,7 +1686,7 @@ static int generic_hdmi_build_controls(struct hda_codec *codec)
int pin_idx;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
err = generic_hdmi_build_jack(codec, pin_idx);
if (err < 0)
@@ -1669,9 +1701,8 @@ static int generic_hdmi_build_controls(struct hda_codec *codec)
snd_hda_spdif_ctls_unassign(codec, pin_idx);
/* add control for ELD Bytes */
- err = hdmi_create_eld_ctl(codec,
- pin_idx,
- spec->pcm_rec[pin_idx].device);
+ err = hdmi_create_eld_ctl(codec, pin_idx,
+ get_pcm_rec(spec, pin_idx)->device);
if (err < 0)
return err;
@@ -1709,7 +1740,7 @@ static int generic_hdmi_init_per_pins(struct hda_codec *codec)
int pin_idx;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
struct hdmi_eld *eld = &per_pin->sink_eld;
per_pin->codec = codec;
@@ -1726,7 +1757,7 @@ static int generic_hdmi_init(struct hda_codec *codec)
int pin_idx;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
hdmi_init_pin(codec, pin_nid);
@@ -1735,13 +1766,27 @@ static int generic_hdmi_init(struct hda_codec *codec)
return 0;
}
+static void hdmi_array_init(struct hdmi_spec *spec, int nums)
+{
+ snd_array_init(&spec->pins, sizeof(struct hdmi_spec_per_pin), nums);
+ snd_array_init(&spec->cvts, sizeof(struct hdmi_spec_per_cvt), nums);
+ snd_array_init(&spec->pcm_rec, sizeof(struct hda_pcm), nums);
+}
+
+static void hdmi_array_free(struct hdmi_spec *spec)
+{
+ snd_array_free(&spec->pins);
+ snd_array_free(&spec->cvts);
+ snd_array_free(&spec->pcm_rec);
+}
+
static void generic_hdmi_free(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
int pin_idx;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
struct hdmi_eld *eld = &per_pin->sink_eld;
cancel_delayed_work(&per_pin->work);
@@ -1749,6 +1794,7 @@ static void generic_hdmi_free(struct hda_codec *codec)
}
flush_workqueue(codec->bus->workq);
+ hdmi_array_free(spec);
kfree(spec);
}
@@ -1775,6 +1821,7 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec,
/* override pins connection list */
snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid);
+ nconns = max(spec->num_cvts, 4);
snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids);
}
@@ -1855,6 +1902,7 @@ static int patch_generic_hdmi(struct hda_codec *codec)
return -ENOMEM;
codec->spec = spec;
+ hdmi_array_init(spec, 4);
snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -1882,24 +1930,30 @@ static int patch_generic_hdmi(struct hda_codec *codec)
static int simple_playback_build_pcms(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
+ struct hda_pcm *info;
unsigned int chans;
struct hda_pcm_stream *pstr;
+ struct hdmi_spec_per_cvt *per_cvt;
- codec->num_pcms = 1;
- codec->pcm_info = info;
-
- chans = get_wcaps(codec, spec->cvts[0].cvt_nid);
+ per_cvt = get_cvt(spec, 0);
+ chans = get_wcaps(codec, per_cvt->cvt_nid);
chans = get_wcaps_channels(chans);
- info->name = get_hdmi_pcm_name(0);
+ info = snd_array_new(&spec->pcm_rec);
+ if (!info)
+ return -ENOMEM;
+ info->name = get_pin(spec, 0)->pcm_name;
+ sprintf(info->name, "HDMI 0");
info->pcm_type = HDA_PCM_TYPE_HDMI;
pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
*pstr = spec->pcm_playback;
- pstr->nid = spec->cvts[0].cvt_nid;
+ pstr->nid = per_cvt->cvt_nid;
if (pstr->channels_max <= 2 && chans && chans <= 16)
pstr->channels_max = chans;
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
return 0;
}
@@ -1919,11 +1973,12 @@ static void simple_hdmi_unsol_event(struct hda_codec *codec,
static int simple_playback_build_controls(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
+ struct hdmi_spec_per_cvt *per_cvt;
int err;
- err = snd_hda_create_spdif_out_ctls(codec,
- spec->cvts[0].cvt_nid,
- spec->cvts[0].cvt_nid);
+ per_cvt = get_cvt(spec, 0);
+ err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid,
+ per_cvt->cvt_nid);
if (err < 0)
return err;
return simple_hdmi_build_jack(codec, 0);
@@ -1932,7 +1987,8 @@ static int simple_playback_build_controls(struct hda_codec *codec)
static int simple_playback_init(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
- hda_nid_t pin = spec->pins[0].pin_nid;
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, 0);
+ hda_nid_t pin = per_pin->pin_nid;
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
@@ -1948,6 +2004,7 @@ static void simple_playback_free(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
+ hdmi_array_free(spec);
kfree(spec);
}
@@ -2111,20 +2168,29 @@ static int patch_simple_hdmi(struct hda_codec *codec,
hda_nid_t cvt_nid, hda_nid_t pin_nid)
{
struct hdmi_spec *spec;
+ struct hdmi_spec_per_cvt *per_cvt;
+ struct hdmi_spec_per_pin *per_pin;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (!spec)
return -ENOMEM;
codec->spec = spec;
+ hdmi_array_init(spec, 1);
spec->multiout.num_dacs = 0; /* no analog */
spec->multiout.max_channels = 2;
spec->multiout.dig_out_nid = cvt_nid;
spec->num_cvts = 1;
spec->num_pins = 1;
- spec->cvts[0].cvt_nid = cvt_nid;
- spec->pins[0].pin_nid = pin_nid;
+ per_pin = snd_array_new(&spec->pins);
+ per_cvt = snd_array_new(&spec->cvts);
+ if (!per_pin || !per_cvt) {
+ simple_playback_free(codec);
+ return -ENOMEM;
+ }
+ per_cvt->cvt_nid = cvt_nid;
+ per_pin->pin_nid = pin_nid;
spec->pcm_playback = simple_pcm_playback;
codec->patch_ops = simple_hdmi_patch_ops;
@@ -2201,9 +2267,11 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
int i;
struct hdmi_spec *spec = codec->spec;
struct hda_spdif_out *spdif;
+ struct hdmi_spec_per_cvt *per_cvt;
mutex_lock(&codec->spdif_mutex);
- spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
+ per_cvt = get_cvt(spec, 0);
+ spdif = snd_hda_spdif_out_of_nid(codec, per_cvt->cvt_nid);
chs = substream->runtime->channels;
@@ -2325,13 +2393,17 @@ static int nvhdmi_7x_8ch_build_pcms(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
int err = simple_playback_build_pcms(codec);
- spec->pcm_rec[0].own_chmap = true;
+ if (!err) {
+ struct hda_pcm *info = get_pcm_rec(spec, 0);
+ info->own_chmap = true;
+ }
return err;
}
static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
+ struct hda_pcm *info;
struct snd_pcm_chmap *chmap;
int err;
@@ -2340,7 +2412,8 @@ static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec)
return err;
/* add channel maps */
- err = snd_pcm_add_chmap_ctls(spec->pcm_rec[0].pcm,
+ info = get_pcm_rec(spec, 0);
+ err = snd_pcm_add_chmap_ctls(info->pcm,
SNDRV_PCM_STREAM_PLAYBACK,
snd_pcm_alt_chmaps, 8, 0, &chmap);
if (err < 0)
@@ -2395,6 +2468,7 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct hdmi_spec *spec = codec->spec;
+ struct hdmi_spec_per_cvt *per_cvt = get_cvt(spec, 0);
int chans = substream->runtime->channels;
int i, err;
@@ -2402,11 +2476,11 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
substream);
if (err < 0)
return err;
- snd_hda_codec_write(codec, spec->cvts[0].cvt_nid, 0,
+ snd_hda_codec_write(codec, per_cvt->cvt_nid, 0,
AC_VERB_SET_CVT_CHAN_COUNT, chans - 1);
/* FIXME: XXX */
for (i = 0; i < chans; i++) {
- snd_hda_codec_write(codec, spec->cvts[0].cvt_nid, 0,
+ snd_hda_codec_write(codec, per_cvt->cvt_nid, 0,
AC_VERB_SET_HDMI_CHAN_SLOT,
(i << 4) | i);
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f15c36bde540..6bf47f7326ad 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -34,7 +34,6 @@
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_auto_parser.h"
-#include "hda_beep.h"
#include "hda_jack.h"
#include "hda_generic.h"
@@ -53,6 +52,20 @@ enum {
ALC_INIT_GPIO3,
};
+enum {
+ ALC_HEADSET_MODE_UNKNOWN,
+ ALC_HEADSET_MODE_UNPLUGGED,
+ ALC_HEADSET_MODE_HEADSET,
+ ALC_HEADSET_MODE_MIC,
+ ALC_HEADSET_MODE_HEADPHONE,
+};
+
+enum {
+ ALC_HEADSET_TYPE_UNKNOWN,
+ ALC_HEADSET_TYPE_CTIA,
+ ALC_HEADSET_TYPE_OMTP,
+};
+
struct alc_customize_define {
unsigned int sku_cfg;
unsigned char port_connectivity;
@@ -86,6 +99,13 @@ struct alc_spec {
int mute_led_polarity;
hda_nid_t mute_led_nid;
+ unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */
+
+ hda_nid_t headset_mic_pin;
+ hda_nid_t headphone_mic_pin;
+ int current_headset_mode;
+ int current_headset_type;
+
/* hooks */
void (*init_hook)(struct hda_codec *codec);
#ifdef CONFIG_PM
@@ -805,17 +825,7 @@ static inline void alc_shutup(struct hda_codec *codec)
snd_hda_shutup_pins(codec);
}
-static void alc_free(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- if (!spec)
- return;
-
- snd_hda_gen_spec_free(&spec->gen);
- snd_hda_detach_beep_device(codec);
- kfree(spec);
-}
+#define alc_free snd_hda_gen_free
#ifdef CONFIG_PM
static void alc_power_eapd(struct hda_codec *codec)
@@ -1401,6 +1411,7 @@ static int patch_alc880(struct hda_codec *codec)
spec = codec->spec;
spec->gen.need_dac_fix = 1;
+ spec->gen.beep_nid = 0x01;
snd_hda_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl,
alc880_fixups);
@@ -1411,12 +1422,8 @@ static int patch_alc880(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
- }
codec->patch_ops = alc_patch_ops;
codec->patch_ops.unsol_event = alc880_unsol_event;
@@ -1455,6 +1462,7 @@ enum {
ALC260_FIXUP_HP_B1900,
ALC260_FIXUP_KN1,
ALC260_FIXUP_FSC_S7020,
+ ALC260_FIXUP_FSC_S7020_JWSE,
};
static void alc260_gpio1_automute(struct hda_codec *codec)
@@ -1516,14 +1524,17 @@ static void alc260_fixup_fsc_s7020(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
struct alc_spec *spec = codec->spec;
-
- switch (action) {
- case HDA_FIXUP_ACT_PRE_PROBE:
- spec->gen.add_out_jack_modes = 1;
- break;
- case HDA_FIXUP_ACT_PROBE:
+ if (action == HDA_FIXUP_ACT_PROBE)
spec->init_amp = ALC_INIT_NONE;
- break;
+}
+
+static void alc260_fixup_fsc_s7020_jwse(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.add_jack_modes = 1;
+ spec->gen.hp_mic = 1;
}
}
@@ -1586,6 +1597,12 @@ static const struct hda_fixup alc260_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc260_fixup_fsc_s7020,
},
+ [ALC260_FIXUP_FSC_S7020_JWSE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc260_fixup_fsc_s7020_jwse,
+ .chained = true,
+ .chain_id = ALC260_FIXUP_FSC_S7020,
+ },
};
static const struct snd_pci_quirk alc260_fixup_tbl[] = {
@@ -1602,6 +1619,14 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = {
{}
};
+static const struct hda_model_fixup alc260_fixup_models[] = {
+ {.id = ALC260_FIXUP_GPIO1, .name = "gpio1"},
+ {.id = ALC260_FIXUP_COEF, .name = "coef"},
+ {.id = ALC260_FIXUP_FSC_S7020, .name = "fujitsu"},
+ {.id = ALC260_FIXUP_FSC_S7020_JWSE, .name = "fujitsu-jwse"},
+ {}
+};
+
/*
*/
static int patch_alc260(struct hda_codec *codec)
@@ -1619,8 +1644,10 @@ static int patch_alc260(struct hda_codec *codec)
* it's almost harmless.
*/
spec->gen.prefer_hp_amp = 1;
+ spec->gen.beep_nid = 0x01;
- snd_hda_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
+ snd_hda_pick_fixup(codec, alc260_fixup_models, alc260_fixup_tbl,
+ alc260_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
/* automatic parse from the BIOS config */
@@ -1628,12 +1655,8 @@ static int patch_alc260(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog)
set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
- }
codec->patch_ops = alc_patch_ops;
spec->shutup = alc_eapd_shutup;
@@ -2132,17 +2155,16 @@ static int patch_alc882(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
+ if (has_cdefine_beep(codec))
+ spec->gen.beep_nid = 0x01;
+
/* automatic parse from the BIOS config */
err = alc882_parse_auto_config(codec);
if (err < 0)
goto error;
- if (!spec->gen.no_analog && has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog && spec->gen.beep_nid)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
- }
codec->patch_ops = alc_patch_ops;
@@ -2295,17 +2317,16 @@ static int patch_alc262(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
+ if (has_cdefine_beep(codec))
+ spec->gen.beep_nid = 0x01;
+
/* automatic parse from the BIOS config */
err = alc262_parse_auto_config(codec);
if (err < 0)
goto error;
- if (!spec->gen.no_analog && has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog && spec->gen.beep_nid)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
- }
codec->patch_ops = alc_patch_ops;
spec->shutup = alc_eapd_shutup;
@@ -2386,16 +2407,7 @@ static const struct snd_pci_quirk alc268_fixup_tbl[] = {
static int alc268_parse_auto_config(struct hda_codec *codec)
{
static const hda_nid_t alc268_ssids[] = { 0x15, 0x1b, 0x14, 0 };
- struct alc_spec *spec = codec->spec;
- int err = alc_parse_auto_config(codec, NULL, alc268_ssids);
- if (err > 0) {
- if (!spec->gen.no_analog &&
- spec->gen.autocfg.speaker_pins[0] != 0x1d) {
- add_mixer(spec, alc268_beep_mixer);
- snd_hda_add_verbs(codec, alc268_beep_init_verbs);
- }
- }
- return err;
+ return alc_parse_auto_config(codec, NULL, alc268_ssids);
}
/*
@@ -2403,7 +2415,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
static int patch_alc268(struct hda_codec *codec)
{
struct alc_spec *spec;
- int i, has_beep, err;
+ int err;
/* ALC268 has no aa-loopback mixer */
err = alc_alloc_spec(codec, 0);
@@ -2411,6 +2423,7 @@ static int patch_alc268(struct hda_codec *codec)
return err;
spec = codec->spec;
+ spec->gen.beep_nid = 0x01;
snd_hda_pick_fixup(codec, alc268_fixup_models, alc268_fixup_tbl, alc268_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -2420,18 +2433,10 @@ static int patch_alc268(struct hda_codec *codec)
if (err < 0)
goto error;
- has_beep = 0;
- for (i = 0; i < spec->num_mixers; i++) {
- if (spec->mixers[i] == alc268_beep_mixer) {
- has_beep = 1;
- break;
- }
- }
-
- if (has_beep) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err > 0 && !spec->gen.no_analog &&
+ spec->gen.autocfg.speaker_pins[0] != 0x1d) {
+ add_mixer(spec, alc268_beep_mixer);
+ snd_hda_add_verbs(codec, alc268_beep_init_verbs);
if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
/* override the amp caps for beep generator */
snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
@@ -2515,6 +2520,7 @@ enum {
ALC269_TYPE_ALC280,
ALC269_TYPE_ALC282,
ALC269_TYPE_ALC284,
+ ALC269_TYPE_ALC286,
};
/*
@@ -2538,6 +2544,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
case ALC269_TYPE_ALC269VB:
case ALC269_TYPE_ALC269VD:
case ALC269_TYPE_ALC282:
+ case ALC269_TYPE_ALC286:
ssids = alc269_ssids;
break;
default:
@@ -2631,7 +2638,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec,
};
unsigned int cfg;
- if (strcmp(codec->chip_name, "ALC271X"))
+ if (strcmp(codec->chip_name, "ALC271X") &&
+ strcmp(codec->chip_name, "ALC269VB"))
return;
cfg = snd_hda_codec_get_pincfg(codec, 0x12);
if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
@@ -2693,6 +2701,34 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec,
spec->gen.automute_hook = alc269_quanta_automute;
}
+static void alc269_x101_hp_automute_hook(struct hda_codec *codec,
+ struct hda_jack_tbl *jack)
+{
+ struct alc_spec *spec = codec->spec;
+ int vref;
+ msleep(200);
+ snd_hda_gen_hp_automute(codec, jack);
+
+ vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+ msleep(100);
+ snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ vref);
+ msleep(500);
+ snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ vref);
+}
+
+static void alc269_fixup_x101_headset_mic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->parse_flags |= HDA_PINCFG_HEADSET_MIC;
+ spec->gen.hp_automute_hook = alc269_x101_hp_automute_hook;
+ }
+}
+
+
/* update mute-LED according to the speaker mute state via mic VREF pin */
static void alc269_fixup_mic_mute_hook(void *private_data, int enabled)
{
@@ -2757,6 +2793,356 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec,
}
}
+/* turn on/off mute LED per vmaster hook */
+static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ struct alc_spec *spec = codec->spec;
+ unsigned int oldval = spec->gpio_led;
+
+ if (enabled)
+ spec->gpio_led &= ~0x08;
+ else
+ spec->gpio_led |= 0x08;
+ if (spec->gpio_led != oldval)
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->gpio_led);
+}
+
+/* turn on/off mic-mute LED per capture hook */
+static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int oldval = spec->gpio_led;
+
+ if (!ucontrol)
+ return;
+
+ if (ucontrol->value.integer.value[0] ||
+ ucontrol->value.integer.value[1])
+ spec->gpio_led &= ~0x10;
+ else
+ spec->gpio_led |= 0x10;
+ if (spec->gpio_led != oldval)
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->gpio_led);
+}
+
+static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ static const struct hda_verb gpio_init[] = {
+ { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 },
+ { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 },
+ {}
+ };
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook;
+ spec->gen.cap_sync_hook = alc269_fixup_hp_gpio_mic_mute_hook;
+ spec->gpio_led = 0;
+ snd_hda_add_verbs(codec, gpio_init);
+ }
+}
+
+static void alc_headset_mode_unplugged(struct hda_codec *codec)
+{
+ int val;
+
+ switch (codec->vendor_id) {
+ case 0x10ec0283:
+ alc_write_coef_idx(codec, 0x1b, 0x0c0b);
+ alc_write_coef_idx(codec, 0x45, 0xc429);
+ val = alc_read_coef_idx(codec, 0x35);
+ alc_write_coef_idx(codec, 0x35, val & 0xbfff);
+ alc_write_coef_idx(codec, 0x06, 0x2104);
+ alc_write_coef_idx(codec, 0x1a, 0x0001);
+ alc_write_coef_idx(codec, 0x26, 0x0004);
+ alc_write_coef_idx(codec, 0x32, 0x42a3);
+ break;
+ case 0x10ec0292:
+ alc_write_coef_idx(codec, 0x76, 0x000e);
+ alc_write_coef_idx(codec, 0x6c, 0x2400);
+ alc_write_coef_idx(codec, 0x18, 0x7308);
+ alc_write_coef_idx(codec, 0x6b, 0xc429);
+ break;
+ case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x15, 0x0d40);
+ alc_write_coef_idx(codec, 0xb7, 0x802b);
+ break;
+ }
+ snd_printdd("Headset jack set to unplugged mode.\n");
+}
+
+
+static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
+ hda_nid_t mic_pin)
+{
+ int val;
+
+ switch (codec->vendor_id) {
+ case 0x10ec0283:
+ alc_write_coef_idx(codec, 0x45, 0xc429);
+ snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
+ val = alc_read_coef_idx(codec, 0x35);
+ alc_write_coef_idx(codec, 0x35, val | 1<<14);
+ alc_write_coef_idx(codec, 0x06, 0x2100);
+ alc_write_coef_idx(codec, 0x1a, 0x0021);
+ alc_write_coef_idx(codec, 0x26, 0x008c);
+ snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
+ break;
+ case 0x10ec0292:
+ snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
+ alc_write_coef_idx(codec, 0x19, 0xa208);
+ alc_write_coef_idx(codec, 0x2e, 0xacf0);
+ break;
+ case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x11, 0x0001);
+ snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
+ alc_write_coef_idx(codec, 0xb7, 0x802b);
+ alc_write_coef_idx(codec, 0xb5, 0x1040);
+ val = alc_read_coef_idx(codec, 0xc3);
+ alc_write_coef_idx(codec, 0xc3, val | 1<<12);
+ snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
+ break;
+ }
+ snd_printdd("Headset jack set to mic-in mode.\n");
+}
+
+static void alc_headset_mode_default(struct hda_codec *codec)
+{
+ switch (codec->vendor_id) {
+ case 0x10ec0283:
+ alc_write_coef_idx(codec, 0x06, 0x2100);
+ alc_write_coef_idx(codec, 0x32, 0x4ea3);
+ break;
+ case 0x10ec0292:
+ alc_write_coef_idx(codec, 0x76, 0x000e);
+ alc_write_coef_idx(codec, 0x6c, 0x2400);
+ alc_write_coef_idx(codec, 0x6b, 0xc429);
+ alc_write_coef_idx(codec, 0x18, 0x7308);
+ break;
+ case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x11, 0x0041);
+ alc_write_coef_idx(codec, 0x15, 0x0d40);
+ alc_write_coef_idx(codec, 0xb7, 0x802b);
+ break;
+ }
+ snd_printdd("Headset jack set to headphone (default) mode.\n");
+}
+
+/* Iphone type */
+static void alc_headset_mode_ctia(struct hda_codec *codec)
+{
+ switch (codec->vendor_id) {
+ case 0x10ec0283:
+ alc_write_coef_idx(codec, 0x45, 0xd429);
+ alc_write_coef_idx(codec, 0x1b, 0x0c2b);
+ alc_write_coef_idx(codec, 0x32, 0x4ea3);
+ break;
+ case 0x10ec0292:
+ alc_write_coef_idx(codec, 0x6b, 0xd429);
+ alc_write_coef_idx(codec, 0x76, 0x0008);
+ alc_write_coef_idx(codec, 0x18, 0x7388);
+ break;
+ case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x15, 0x0d60);
+ alc_write_coef_idx(codec, 0xc3, 0x0000);
+ break;
+ }
+ snd_printdd("Headset jack set to iPhone-style headset mode.\n");
+}
+
+/* Nokia type */
+static void alc_headset_mode_omtp(struct hda_codec *codec)
+{
+ switch (codec->vendor_id) {
+ case 0x10ec0283:
+ alc_write_coef_idx(codec, 0x45, 0xe429);
+ alc_write_coef_idx(codec, 0x1b, 0x0c2b);
+ alc_write_coef_idx(codec, 0x32, 0x4ea3);
+ break;
+ case 0x10ec0292:
+ alc_write_coef_idx(codec, 0x6b, 0xe429);
+ alc_write_coef_idx(codec, 0x76, 0x0008);
+ alc_write_coef_idx(codec, 0x18, 0x7388);
+ break;
+ case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x15, 0x0d50);
+ alc_write_coef_idx(codec, 0xc3, 0x0000);
+ break;
+ }
+ snd_printdd("Headset jack set to Nokia-style headset mode.\n");
+}
+
+static void alc_determine_headset_type(struct hda_codec *codec)
+{
+ int val;
+ bool is_ctia = false;
+ struct alc_spec *spec = codec->spec;
+
+ switch (codec->vendor_id) {
+ case 0x10ec0283:
+ alc_write_coef_idx(codec, 0x45, 0xd029);
+ msleep(300);
+ val = alc_read_coef_idx(codec, 0x46);
+ is_ctia = (val & 0x0070) == 0x0070;
+ break;
+ case 0x10ec0292:
+ alc_write_coef_idx(codec, 0x6b, 0xd429);
+ msleep(300);
+ val = alc_read_coef_idx(codec, 0x6c);
+ is_ctia = (val & 0x001c) == 0x001c;
+ break;
+ case 0x10ec0668:
+ alc_write_coef_idx(codec, 0x11, 0x0001);
+ alc_write_coef_idx(codec, 0xb7, 0x802b);
+ alc_write_coef_idx(codec, 0x15, 0x0d60);
+ alc_write_coef_idx(codec, 0xc3, 0x0c00);
+ msleep(300);
+ val = alc_read_coef_idx(codec, 0xbe);
+ is_ctia = (val & 0x1c02) == 0x1c02;
+ break;
+ }
+
+ snd_printdd("Headset jack detected iPhone-style headset: %s\n",
+ is_ctia ? "yes" : "no");
+ spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP;
+}
+
+static void alc_update_headset_mode(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+ hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+
+ int new_headset_mode;
+
+ if (!snd_hda_jack_detect(codec, hp_pin))
+ new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED;
+ else if (mux_pin == spec->headset_mic_pin)
+ new_headset_mode = ALC_HEADSET_MODE_HEADSET;
+ else if (mux_pin == spec->headphone_mic_pin)
+ new_headset_mode = ALC_HEADSET_MODE_MIC;
+ else
+ new_headset_mode = ALC_HEADSET_MODE_HEADPHONE;
+
+ if (new_headset_mode == spec->current_headset_mode)
+ return;
+
+ switch (new_headset_mode) {
+ case ALC_HEADSET_MODE_UNPLUGGED:
+ alc_headset_mode_unplugged(codec);
+ spec->gen.hp_jack_present = false;
+ break;
+ case ALC_HEADSET_MODE_HEADSET:
+ if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN)
+ alc_determine_headset_type(codec);
+ if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA)
+ alc_headset_mode_ctia(codec);
+ else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP)
+ alc_headset_mode_omtp(codec);
+ spec->gen.hp_jack_present = true;
+ break;
+ case ALC_HEADSET_MODE_MIC:
+ alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin);
+ spec->gen.hp_jack_present = false;
+ break;
+ case ALC_HEADSET_MODE_HEADPHONE:
+ alc_headset_mode_default(codec);
+ spec->gen.hp_jack_present = true;
+ break;
+ }
+ if (new_headset_mode != ALC_HEADSET_MODE_MIC) {
+ snd_hda_set_pin_ctl_cache(codec, hp_pin,
+ AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ if (spec->headphone_mic_pin)
+ snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin,
+ PIN_VREFHIZ);
+ }
+ spec->current_headset_mode = new_headset_mode;
+
+ snd_hda_gen_update_outputs(codec);
+}
+
+static void alc_update_headset_mode_hook(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ alc_update_headset_mode(codec);
+}
+
+static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN;
+ snd_hda_gen_hp_automute(codec, jack);
+}
+
+static void alc_probe_headset_mode(struct hda_codec *codec)
+{
+ int i;
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+ /* Find mic pins */
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].is_headset_mic && !spec->headset_mic_pin)
+ spec->headset_mic_pin = cfg->inputs[i].pin;
+ if (cfg->inputs[i].is_headphone_mic && !spec->headphone_mic_pin)
+ spec->headphone_mic_pin = cfg->inputs[i].pin;
+ }
+
+ spec->gen.cap_sync_hook = alc_update_headset_mode_hook;
+ spec->gen.automute_hook = alc_update_headset_mode;
+ spec->gen.hp_automute_hook = alc_update_headset_jack_cb;
+}
+
+static void alc_fixup_headset_mode(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC;
+ break;
+ case HDA_FIXUP_ACT_PROBE:
+ alc_probe_headset_mode(codec);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ spec->current_headset_mode = 0;
+ alc_update_headset_mode(codec);
+ break;
+ }
+}
+
+static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ struct alc_spec *spec = codec->spec;
+ spec->parse_flags |= HDA_PINCFG_HEADSET_MIC;
+ }
+ else
+ alc_fixup_headset_mode(codec, fix, action);
+}
+
+static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ int val;
+ alc_write_coef_idx(codec, 0xc4, 0x8000);
+ val = alc_read_coef_idx(codec, 0xc2);
+ alc_write_coef_idx(codec, 0xc2, val & 0xfe);
+ snd_hda_set_pin_ctl_cache(codec, 0x18, 0);
+ }
+ alc_fixup_headset_mode(codec, fix, action);
+}
+
static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -2772,6 +3158,38 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
}
}
+static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+ int i;
+
+ /* The mic boosts on level 2 and 3 are too noisy
+ on the internal mic input.
+ Therefore limit the boost to 0 or 1. */
+
+ if (action != HDA_FIXUP_ACT_PROBE)
+ return;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ hda_nid_t nid = cfg->inputs[i].pin;
+ unsigned int defcfg;
+ if (cfg->inputs[i].type != AUTO_PIN_MIC)
+ continue;
+ defcfg = snd_hda_codec_get_pincfg(codec, nid);
+ if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT)
+ continue;
+
+ snd_hda_override_amp_caps(codec, nid, HDA_INPUT,
+ (0x00 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x01 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x2f << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+ }
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -2792,11 +3210,21 @@ enum {
ALC269_FIXUP_HP_MUTE_LED,
ALC269_FIXUP_HP_MUTE_LED_MIC1,
ALC269_FIXUP_HP_MUTE_LED_MIC2,
+ ALC269_FIXUP_HP_GPIO_LED,
ALC269_FIXUP_INV_DMIC,
ALC269_FIXUP_LENOVO_DOCK,
ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT,
+ ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC269_FIXUP_DELL2_MIC_NO_PRESENCE,
+ ALC269_FIXUP_HEADSET_MODE,
+ ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC,
+ ALC269_FIXUP_ASUS_X101_FUNC,
+ ALC269_FIXUP_ASUS_X101_VERB,
+ ALC269_FIXUP_ASUS_X101,
ALC271_FIXUP_AMIC_MIC2,
ALC271_FIXUP_HP_GATE_MIC_JACK,
+ ALC269_FIXUP_ACER_AC700,
+ ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -2931,6 +3359,10 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_hp_mute_led_mic2,
},
+ [ALC269_FIXUP_HP_GPIO_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_hp_gpio_led,
+ },
[ALC269_FIXUP_INV_DMIC] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic_0x12,
@@ -2949,6 +3381,59 @@ static const struct hda_fixup alc269_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
},
+ [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
+ [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x16, 0x21014020 }, /* dock line out */
+ { 0x19, 0x21a19030 }, /* dock mic */
+ { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+ },
+ [ALC269_FIXUP_HEADSET_MODE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_headset_mode,
+ },
+ [ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_headset_mode_no_hp_mic,
+ },
+ [ALC269_FIXUP_ASUS_X101_FUNC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_x101_headset_mic,
+ },
+ [ALC269_FIXUP_ASUS_X101_VERB] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x08},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0310},
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_ASUS_X101_FUNC
+ },
+ [ALC269_FIXUP_ASUS_X101] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x04a1182c }, /* Headset mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_ASUS_X101_VERB
+ },
[ALC271_FIXUP_AMIC_MIC2] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -2965,29 +3450,72 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC271_FIXUP_AMIC_MIC2,
},
+ [ALC269_FIXUP_ACER_AC700] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x99a3092f }, /* int-mic */
+ { 0x14, 0x99130110 }, /* speaker */
+ { 0x18, 0x03a11c20 }, /* mic */
+ { 0x1e, 0x0346101e }, /* SPDIF1 */
+ { 0x21, 0x0321101f }, /* HP out */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC271_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_LIMIT_INT_MIC_BOOST] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+ SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05c5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05c6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05c7, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05c8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05ec, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05ed, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05ee, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05f3, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
+ SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
+ SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+ SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
@@ -3063,6 +3591,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"},
{.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
+ {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
{}
};
@@ -3131,6 +3660,9 @@ static int patch_alc269(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
+ if (has_cdefine_beep(codec))
+ spec->gen.beep_nid = 0x01;
+
switch (codec->vendor_id) {
case 0x10ec0269:
spec->codec_variant = ALC269_TYPE_ALC269VA;
@@ -3172,6 +3704,9 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0292:
spec->codec_variant = ALC269_TYPE_ALC284;
break;
+ case 0x10ec0286:
+ spec->codec_variant = ALC269_TYPE_ALC286;
+ break;
}
/* automatic parse from the BIOS config */
@@ -3179,12 +3714,8 @@ static int patch_alc269(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog && has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog && spec->gen.beep_nid)
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
- }
codec->patch_ops = alc_patch_ops;
#ifdef CONFIG_PM
@@ -3292,6 +3823,7 @@ static int patch_alc861(struct hda_codec *codec)
return err;
spec = codec->spec;
+ spec->gen.beep_nid = 0x23;
snd_hda_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -3301,12 +3833,8 @@ static int patch_alc861(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog) {
- err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog)
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
- }
codec->patch_ops = alc_patch_ops;
#ifdef CONFIG_PM
@@ -3387,6 +3915,7 @@ static int patch_alc861vd(struct hda_codec *codec)
return err;
spec = codec->spec;
+ spec->gen.beep_nid = 0x23;
snd_hda_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -3396,12 +3925,8 @@ static int patch_alc861vd(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog) {
- err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
- }
codec->patch_ops = alc_patch_ops;
@@ -3480,6 +4005,8 @@ enum {
ALC662_FIXUP_NO_JACK_DETECT,
ALC662_FIXUP_ZOTAC_Z68,
ALC662_FIXUP_INV_DMIC,
+ ALC668_FIXUP_DELL_MIC_NO_PRESENCE,
+ ALC668_FIXUP_HEADSET_MODE,
};
static const struct hda_fixup alc662_fixups[] = {
@@ -3640,6 +4167,20 @@ static const struct hda_fixup alc662_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_inv_dmic_0x12,
},
+ [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+ { 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC668_FIXUP_HEADSET_MODE
+ },
+ [ALC668_FIXUP_HEADSET_MODE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_headset_mode_alc668,
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -3648,6 +4189,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+ SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
@@ -3784,10 +4327,14 @@ static int patch_alc662(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
+ if (has_cdefine_beep(codec))
+ spec->gen.beep_nid = 0x01;
+
if ((alc_get_coef0(codec) & (1 << 14)) &&
codec->bus->pci->subsystem_vendor == 0x1025 &&
spec->cdefine.platform_type == 1) {
- if (alc_codec_rename(codec, "ALC272X") < 0)
+ err = alc_codec_rename(codec, "ALC272X");
+ if (err < 0)
goto error;
}
@@ -3796,10 +4343,7 @@ static int patch_alc662(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog && has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (!spec->gen.no_analog && spec->gen.beep_nid) {
switch (codec->vendor_id) {
case 0x10ec0662:
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -3878,6 +4422,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
{ .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
{ .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
+ { .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 },
{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index dafe04ae8c72..1d9d6427e0bf 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -211,7 +211,6 @@ struct sigmatel_spec {
/* beep widgets */
hda_nid_t anabeep_nid;
- hda_nid_t digbeep_nid;
/* SPDIF-out mux */
const char * const *spdif_labels;
@@ -3529,8 +3528,12 @@ static int stac_parse_auto_config(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
int err;
+ int flags = 0;
- err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+ if (spec->headset_jack)
+ flags |= HDA_PINCFG_HEADSET_MIC;
+
+ err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, flags);
if (err < 0)
return err;
@@ -3560,16 +3563,13 @@ static int stac_parse_auto_config(struct hda_codec *codec)
/* setup digital beep controls and input device */
#ifdef CONFIG_SND_HDA_INPUT_BEEP
- if (spec->digbeep_nid > 0) {
- hda_nid_t nid = spec->digbeep_nid;
+ if (spec->gen.beep_nid) {
+ hda_nid_t nid = spec->gen.beep_nid;
unsigned int caps;
err = stac_auto_create_beep_ctls(codec, nid);
if (err < 0)
return err;
- err = snd_hda_attach_beep_device(codec, nid);
- if (err < 0)
- return err;
if (codec->beep) {
/* IDT/STAC codecs have linear beep tone parameter */
codec->beep->linear_tone = spec->linear_tone_beep;
@@ -3657,17 +3657,7 @@ static void stac_shutup(struct hda_codec *codec)
~spec->eapd_mask);
}
-static void stac_free(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
-
- if (!spec)
- return;
-
- snd_hda_gen_spec_free(&spec->gen);
- kfree(spec);
- snd_hda_detach_beep_device(codec);
-}
+#define stac_free snd_hda_gen_free
#ifdef CONFIG_PROC_FS
static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
@@ -3797,6 +3787,7 @@ static int patch_stac9200(struct hda_codec *codec)
spec->gen.own_eapd_ctl = 1;
codec->patch_ops = stac_patch_ops;
+ codec->power_filter = snd_hda_codec_eapd_power_filter;
snd_hda_add_verbs(codec, stac9200_eapd_init);
@@ -3884,7 +3875,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
spec->aloopback_mask = 0x01;
spec->aloopback_shift = 8;
- spec->digbeep_nid = 0x1c;
+ spec->gen.beep_nid = 0x1c; /* digital beep */
/* GPIO0 High = Enable EAPD */
spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
@@ -3968,7 +3959,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
spec->gen.power_down_unused = 1;
spec->gen.mixer_nid = 0x1b;
- spec->digbeep_nid = 0x21;
+ spec->gen.beep_nid = 0x21; /* digital beep */
spec->pwr_nids = stac92hd83xxx_pwr_nids;
spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
spec->default_polarity = -1; /* no default cfg */
@@ -4016,7 +4007,7 @@ static int patch_stac92hd95(struct hda_codec *codec)
spec->gen.own_eapd_ctl = 1;
spec->gen.power_down_unused = 1;
- spec->digbeep_nid = 0x19;
+ spec->gen.beep_nid = 0x19; /* digital beep */
spec->pwr_nids = stac92hd95_pwr_nids;
spec->num_pwrs = ARRAY_SIZE(stac92hd95_pwr_nids);
spec->default_polarity = -1; /* no default cfg */
@@ -4091,7 +4082,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->aloopback_shift = 0;
spec->powerdown_adcs = 1;
- spec->digbeep_nid = 0x26;
+ spec->gen.beep_nid = 0x26; /* digital beep */
spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
spec->pwr_nids = stac92hd71bxx_pwr_nids;
@@ -4173,7 +4164,7 @@ static int patch_stac927x(struct hda_codec *codec)
spec->have_spdif_mux = 1;
spec->spdif_labels = stac927x_spdif_labels;
- spec->digbeep_nid = 0x23;
+ spec->gen.beep_nid = 0x23; /* digital beep */
/* GPIO0 High = Enable EAPD */
spec->eapd_mask = spec->gpio_mask = 0x01;
@@ -4232,7 +4223,7 @@ static int patch_stac9205(struct hda_codec *codec)
spec->gen.own_eapd_ctl = 1;
spec->have_spdif_mux = 1;
- spec->digbeep_nid = 0x23;
+ spec->gen.beep_nid = 0x23; /* digital beep */
snd_hda_add_verbs(codec, stac9205_core_init);
spec->aloopback_ctl = &stac9205_loopback;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index c35338a8771d..e0dadcf2030d 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -626,11 +626,31 @@ static void via_set_jack_unsol_events(struct hda_codec *codec)
}
}
+static const struct badness_table via_main_out_badness = {
+ .no_primary_dac = 0x10000,
+ .no_dac = 0x4000,
+ .shared_primary = 0x10000,
+ .shared_surr = 0x20,
+ .shared_clfe = 0x20,
+ .shared_surr_main = 0x20,
+};
+static const struct badness_table via_extra_out_badness = {
+ .no_primary_dac = 0x4000,
+ .no_dac = 0x4000,
+ .shared_primary = 0x12,
+ .shared_surr = 0x20,
+ .shared_clfe = 0x20,
+ .shared_surr_main = 0x10,
+};
+
static int via_parse_auto_config(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
int err;
+ spec->gen.main_out_badness = &via_main_out_badness;
+ spec->gen.extra_out_badness = &via_extra_out_badness;
+
err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
if (err < 0)
return err;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 223c3d9cc69e..9ea05e956474 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -969,6 +969,7 @@ static int snd_hdspm_create_pcm(struct snd_card *card,
struct hdspm *hdspm);
static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm);
+static inline int hdspm_get_pll_freq(struct hdspm *hdspm);
static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm);
static int hdspm_autosync_ref(struct hdspm *hdspm);
static int snd_hdspm_set_defaults(struct hdspm *hdspm);
@@ -1075,6 +1076,20 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm)
return ret;
}
+/* round arbitary sample rates to commonly known rates */
+static int hdspm_round_frequency(int rate)
+{
+ if (rate < 38050)
+ return 32000;
+ if (rate < 46008)
+ return 44100;
+ else
+ return 48000;
+}
+
+static int hdspm_tco_sync_check(struct hdspm *hdspm);
+static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
+
/* check for external sample rate */
static int hdspm_external_sample_rate(struct hdspm *hdspm)
{
@@ -1216,22 +1231,45 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
break;
}
- /* QS and DS rates normally can not be detected
- * automatically by the card. Only exception is MADI
- * in 96k frame mode.
- *
- * So if we read SS values (32 .. 48k), check for
- * user-provided DS/QS bits in the control register
- * and multiply the base frequency accordingly.
- */
- if (rate <= 48000) {
- if (hdspm->control_register & HDSPM_QuadSpeed)
- rate *= 4;
- else if (hdspm->control_register &
- HDSPM_DoubleSpeed)
- rate *= 2;
+ } /* endif HDSPM_madiLock */
+
+ /* check sample rate from TCO or SYNC_IN */
+ {
+ bool is_valid_input = 0;
+ bool has_sync = 0;
+
+ syncref = hdspm_autosync_ref(hdspm);
+ if (HDSPM_AUTOSYNC_FROM_TCO == syncref) {
+ is_valid_input = 1;
+ has_sync = (HDSPM_SYNC_CHECK_SYNC ==
+ hdspm_tco_sync_check(hdspm));
+ } else if (HDSPM_AUTOSYNC_FROM_SYNC_IN == syncref) {
+ is_valid_input = 1;
+ has_sync = (HDSPM_SYNC_CHECK_SYNC ==
+ hdspm_sync_in_sync_check(hdspm));
+ }
+
+ if (is_valid_input && has_sync) {
+ rate = hdspm_round_frequency(
+ hdspm_get_pll_freq(hdspm));
}
}
+
+ /* QS and DS rates normally can not be detected
+ * automatically by the card. Only exception is MADI
+ * in 96k frame mode.
+ *
+ * So if we read SS values (32 .. 48k), check for
+ * user-provided DS/QS bits in the control register
+ * and multiply the base frequency accordingly.
+ */
+ if (rate <= 48000) {
+ if (hdspm->control_register & HDSPM_QuadSpeed)
+ rate *= 4;
+ else if (hdspm->control_register &
+ HDSPM_DoubleSpeed)
+ rate *= 2;
+ }
break;
}
@@ -1979,16 +2017,25 @@ static void hdspm_midi_tasklet(unsigned long arg)
/* get the system sample rate which is set */
+static inline int hdspm_get_pll_freq(struct hdspm *hdspm)
+{
+ unsigned int period, rate;
+
+ period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ);
+ rate = hdspm_calc_dds_value(hdspm, period);
+
+ return rate;
+}
+
/**
* Calculate the real sample rate from the
* current DDS value.
**/
static int hdspm_get_system_sample_rate(struct hdspm *hdspm)
{
- unsigned int period, rate;
+ unsigned int rate;
- period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ);
- rate = hdspm_calc_dds_value(hdspm, period);
+ rate = hdspm_get_pll_freq(hdspm);
if (rate > 207000) {
/* Unreasonable high sample rate as seen on PCI MADI cards. */
@@ -2128,6 +2175,16 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
return (status >> (idx*4)) & 0xF;
}
+#define ENUMERATED_CTL_INFO(info, texts) \
+{ \
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \
+ uinfo->count = 1; \
+ uinfo->value.enumerated.items = ARRAY_SIZE(texts); \
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; \
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \
+}
+
#define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
@@ -2143,14 +2200,7 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
static int snd_hdspm_info_autosync_sample_rate(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 10;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts_freq[uinfo->value.enumerated.item]);
+ ENUMERATED_CTL_INFO(uinfo, texts_freq);
return 0;
}
@@ -2316,15 +2366,7 @@ static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "Master", "AutoSync" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -2888,6 +2930,112 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
return 0;
}
+
+
+#define HDSPM_TCO_VIDEO_INPUT_FORMAT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |\
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_tco_video_input_format, \
+ .get = snd_hdspm_get_tco_video_input_format, \
+}
+
+static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"No video", "NTSC", "PAL"};
+ ENUMERATED_CTL_INFO(uinfo, texts);
+ return 0;
+}
+
+static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 status;
+ int ret = 0;
+
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ status = hdspm_read(hdspm, HDSPM_RD_TCO + 4);
+ switch (status & (HDSPM_TCO1_Video_Input_Format_NTSC |
+ HDSPM_TCO1_Video_Input_Format_PAL)) {
+ case HDSPM_TCO1_Video_Input_Format_NTSC:
+ /* ntsc */
+ ret = 1;
+ break;
+ case HDSPM_TCO1_Video_Input_Format_PAL:
+ /* pal */
+ ret = 2;
+ break;
+ default:
+ /* no video */
+ ret = 0;
+ break;
+ }
+ ucontrol->value.enumerated.item[0] = ret;
+ return 0;
+}
+
+
+
+#define HDSPM_TCO_LTC_FRAMES(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |\
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_tco_ltc_frames, \
+ .get = snd_hdspm_get_tco_ltc_frames, \
+}
+
+static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
+ "30 fps"};
+ ENUMERATED_CTL_INFO(uinfo, texts);
+ return 0;
+}
+
+static int hdspm_tco_ltc_frames(struct hdspm *hdspm)
+{
+ u32 status;
+ int ret = 0;
+
+ status = hdspm_read(hdspm, HDSPM_RD_TCO + 4);
+ if (status & HDSPM_TCO1_LTC_Input_valid) {
+ switch (status & (HDSPM_TCO1_LTC_Format_LSB |
+ HDSPM_TCO1_LTC_Format_MSB)) {
+ case 0:
+ /* 24 fps */
+ ret = 1;
+ break;
+ case HDSPM_TCO1_LTC_Format_LSB:
+ /* 25 fps */
+ ret = 2;
+ break;
+ case HDSPM_TCO1_LTC_Format_MSB:
+ /* 25 fps */
+ ret = 3;
+ break;
+ default:
+ /* 30 fps */
+ ret = 4;
+ break;
+ }
+ }
+
+ return ret;
+}
+
+static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_tco_ltc_frames(hdspm);
+ return 0;
+}
+
#define HDSPM_TOGGLE_SETTING(xname, xindex) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
@@ -2974,17 +3122,7 @@ static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "optical", "coaxial" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3046,17 +3184,7 @@ static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "Single", "Double" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3129,17 +3257,7 @@ static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "Single", "Double", "Quad" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 3;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3215,17 +3333,7 @@ static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "Single", "Double", "Quad" };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 3;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3445,19 +3553,30 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
.get = snd_hdspm_get_sync_check \
}
+#define HDSPM_TCO_LOCK_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .private_value = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_tco_info_lock_check, \
+ .get = snd_hdspm_get_sync_check \
+}
+
+
static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 4;
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
+ ENUMERATED_CTL_INFO(uinfo, texts);
+ return 0;
+}
+
+static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "No Lock", "Lock" };
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3590,6 +3709,14 @@ static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx)
return 0;
}
+static int hdspm_tco_input_check(struct hdspm *hdspm, u32 mask)
+{
+ u32 status;
+ status = hdspm_read(hdspm, HDSPM_RD_TCO + 4);
+
+ return (status & mask) ? 1 : 0;
+}
+
static int hdspm_tco_sync_check(struct hdspm *hdspm)
{
@@ -3697,6 +3824,22 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
}
+ if (hdspm->tco) {
+ switch (kcontrol->private_value) {
+ case 11:
+ /* Check TCO for lock state of its current input */
+ val = hdspm_tco_input_check(hdspm, HDSPM_TCO1_TCO_lock);
+ break;
+ case 12:
+ /* Check TCO for valid time code on LTC input. */
+ val = hdspm_tco_input_check(hdspm,
+ HDSPM_TCO1_LTC_Input_valid);
+ break;
+ default:
+ break;
+ }
+ }
+
if (-1 == val)
val = 3;
@@ -3813,17 +3956,7 @@ static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "44.1 kHz", "48 kHz" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3869,17 +4002,7 @@ static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 5;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3924,17 +4047,7 @@ static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 3;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -3981,17 +4094,7 @@ static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol,
{
static char *texts[] = { "24 fps", "25 fps", "29.97fps",
"29.97 dfps", "30 fps", "30 dfps" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 6;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4037,17 +4140,7 @@ static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "LTC", "Video", "WCK" };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 3;
-
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
- uinfo->value.enumerated.item =
- uinfo->value.enumerated.items - 1;
-
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
-
+ ENUMERATED_CTL_INFO(uinfo, texts);
return 0;
}
@@ -4145,6 +4238,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_madi[] = {
HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 3),
HDSPM_TOGGLE_SETTING("Line Out", HDSPM_LineOut),
HDSPM_TOGGLE_SETTING("TX 64 channels mode", HDSPM_TX_64ch),
+ HDSPM_TOGGLE_SETTING("Disable 96K frames", HDSPM_SMUX),
HDSPM_TOGGLE_SETTING("Clear Track Marker", HDSPM_clr_tms),
HDSPM_TOGGLE_SETTING("Safe Mode", HDSPM_AutoInp),
HDSPM_INPUT_SELECT("Input Select", 0),
@@ -4272,7 +4366,11 @@ static struct snd_kcontrol_new snd_hdspm_controls_tco[] = {
HDSPM_TCO_WCK_CONVERSION("TCO WCK Conversion", 0),
HDSPM_TCO_FRAME_RATE("TCO Frame Rate", 0),
HDSPM_TCO_SYNC_SOURCE("TCO Sync Source", 0),
- HDSPM_TCO_WORD_TERM("TCO Word Term", 0)
+ HDSPM_TCO_WORD_TERM("TCO Word Term", 0),
+ HDSPM_TCO_LOCK_CHECK("TCO Input Check", 11),
+ HDSPM_TCO_LOCK_CHECK("TCO LTC Valid", 12),
+ HDSPM_TCO_LTC_FRAMES("TCO Detected Frame Rate", 0),
+ HDSPM_TCO_VIDEO_INPUT_FORMAT("Video Input Format", 0)
};
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 5da8ca7aee05..9e675c76436c 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -29,6 +29,10 @@ config SND_SOC_AC97_BUS
config SND_SOC_DMAENGINE_PCM
bool
+config SND_SOC_GENERIC_DMAENGINE_PCM
+ bool
+ select SND_SOC_DMAENGINE_PCM
+
# All the supported SoCs
source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 99f32f7c0692..197b6ae54c8d 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -5,6 +5,10 @@ ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
snd-soc-core-objs += soc-dmaengine-pcm.o
endif
+ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
+snd-soc-core-objs += soc-generic-dmaengine-pcm.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/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 30184a4a147a..1d38fd0bc4e2 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -67,9 +67,10 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = {
static void atmel_pcm_dma_irq(u32 ssc_sr,
struct snd_pcm_substream *substream)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct atmel_pcm_dma_params *prtd;
- prtd = snd_dmaengine_pcm_get_data(substream);
+ prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
if (ssc_sr & prtd->mask->ssc_error) {
if (snd_pcm_running(substream))
@@ -104,15 +105,13 @@ static bool filter(struct dma_chan *chan, void *slave)
}
static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd)
{
- struct atmel_pcm_dma_params *prtd;
struct ssc_device *ssc;
struct dma_chan *dma_chan;
struct dma_slave_config slave_config;
int ret;
- prtd = snd_dmaengine_pcm_get_data(substream);
ssc = prtd->ssc;
ret = snd_hwparams_to_dma_slave_config(substream, params,
@@ -130,8 +129,6 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
slave_config.src_maxburst = 1;
}
- slave_config.device_fc = false;
-
dma_chan = snd_dmaengine_pcm_get_chan(substream);
if (dmaengine_slave_config(dma_chan, &slave_config)) {
pr_err("atmel-pcm: failed to configure dma channel\n");
@@ -158,15 +155,13 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
if (ssc->pdev)
sdata = ssc->pdev->dev.platform_data;
- ret = snd_dmaengine_pcm_open(substream, filter, sdata);
+ ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata);
if (ret) {
pr_err("atmel-pcm: dmaengine pcm open failed\n");
return -EINVAL;
}
- snd_dmaengine_pcm_set_data(substream, prtd);
-
- ret = atmel_pcm_configure_dma(substream, params);
+ ret = atmel_pcm_configure_dma(substream, params, prtd);
if (ret) {
pr_err("atmel-pcm: failed to configure dmai\n");
goto err;
@@ -176,15 +171,16 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
err:
- snd_dmaengine_pcm_close(substream);
+ snd_dmaengine_pcm_close_release_chan(substream);
return ret;
}
static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct atmel_pcm_dma_params *prtd;
- prtd = snd_dmaengine_pcm_get_data(substream);
+ prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error);
ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable);
@@ -199,16 +195,9 @@ static int atmel_pcm_open(struct snd_pcm_substream *substream)
return 0;
}
-static int atmel_pcm_close(struct snd_pcm_substream *substream)
-{
- snd_dmaengine_pcm_close(substream);
-
- return 0;
-}
-
static struct snd_pcm_ops atmel_pcm_ops = {
.open = atmel_pcm_open,
- .close = atmel_pcm_close,
+ .close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = atmel_pcm_hw_params,
.prepare = atmel_pcm_dma_prepare,
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e13580d6c476..f3fdfa07fcb9 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -533,6 +533,49 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
break;
case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+ /*
+ * DSP/PCM Mode A format, CODEC supplies BCLK and LRC clocks.
+ *
+ * The SSC transmit clock is obtained from the BCLK signal on
+ * on the TK line, and the SSC receive clock is
+ * generated from the transmit clock.
+ *
+ * Data is transferred on first BCLK after LRC pulse rising
+ * edge.If stereo, the right channel data is contiguous with
+ * the left channel data.
+ */
+ rcmr = SSC_BF(RCMR_PERIOD, 0)
+ | SSC_BF(RCMR_STTDLY, START_DELAY)
+ | SSC_BF(RCMR_START, SSC_START_RISING_RF)
+ | SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+ | SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(RCMR_CKS, SSC_CKS_PIN);
+
+ rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
+ | SSC_BF(RFMR_FSLEN, 0)
+ | SSC_BF(RFMR_DATNB, (channels - 1))
+ | SSC_BIT(RFMR_MSBF)
+ | SSC_BF(RFMR_LOOP, 0)
+ | SSC_BF(RFMR_DATLEN, (bits - 1));
+
+ tcmr = SSC_BF(TCMR_PERIOD, 0)
+ | SSC_BF(TCMR_STTDLY, START_DELAY)
+ | SSC_BF(TCMR_START, SSC_START_RISING_RF)
+ | SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+ | SSC_BF(TCMR_CKO, SSC_CKO_NONE)
+ | SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+
+ tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+ | SSC_BF(TFMR_FSDEN, 0)
+ | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
+ | SSC_BF(TFMR_FSLEN, 0)
+ | SSC_BF(TFMR_DATNB, (channels - 1))
+ | SSC_BIT(TFMR_MSBF)
+ | SSC_BF(TFMR_DATDEF, 0)
+ | SSC_BF(TFMR_DATLEN, (bits - 1));
+ break;
+
default:
printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
ssc_p->daifmt);
@@ -707,13 +750,18 @@ static struct snd_soc_dai_driver atmel_ssc_dai = {
.ops = &atmel_ssc_dai_ops,
};
+static const struct snd_soc_component_driver atmel_ssc_component = {
+ .name = "atmel-ssc",
+};
+
static int asoc_ssc_init(struct device *dev)
{
struct platform_device *pdev = to_platform_device(dev);
struct ssc_device *ssc = platform_get_drvdata(pdev);
int ret;
- ret = snd_soc_register_dai(dev, &atmel_ssc_dai);
+ ret = snd_soc_register_component(dev, &atmel_ssc_component,
+ &atmel_ssc_dai, 1);
if (ret) {
dev_err(dev, "Could not register DAI: %d\n", ret);
goto err;
@@ -732,7 +780,7 @@ static int asoc_ssc_init(struct device *dev)
return 0;
err_unregister_dai:
- snd_soc_unregister_dai(dev);
+ snd_soc_unregister_component(dev);
err:
return ret;
}
@@ -747,7 +795,7 @@ static void asoc_ssc_exit(struct device *dev)
else
atmel_pcm_pdc_platform_unregister(dev);
- snd_soc_unregister_dai(dev);
+ snd_soc_unregister_component(dev);
}
/**
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
index ea7d9d157022..44b8dcecf571 100644
--- a/sound/soc/au1x/ac97c.c
+++ b/sound/soc/au1x/ac97c.c
@@ -223,6 +223,10 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = {
.ops = &alchemy_ac97c_ops,
};
+static const struct snd_soc_component_driver au1xac97c_component = {
+ .name = "au1xac97c",
+};
+
static int au1xac97c_drvprobe(struct platform_device *pdev)
{
int ret;
@@ -268,7 +272,8 @@ static int au1xac97c_drvprobe(struct platform_device *pdev)
platform_set_drvdata(pdev, ctx);
- ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
+ ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component,
+ &au1xac97c_dai_driver, 1);
if (ret)
return ret;
@@ -280,7 +285,7 @@ static int au1xac97c_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
WR(ctx, AC97_ENABLE, EN_D); /* clock off, disable */
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
index 072448afc219..b3f37f6edbcb 100644
--- a/sound/soc/au1x/i2sc.c
+++ b/sound/soc/au1x/i2sc.c
@@ -225,6 +225,10 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = {
.ops = &au1xi2s_dai_ops,
};
+static const struct snd_soc_component_driver au1xi2s_component = {
+ .name = "au1xi2s",
+};
+
static int au1xi2s_drvprobe(struct platform_device *pdev)
{
struct resource *iores, *dmares;
@@ -260,14 +264,15 @@ static int au1xi2s_drvprobe(struct platform_device *pdev)
platform_set_drvdata(pdev, ctx);
- return snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+ return snd_soc_register_component(&pdev->dev, &au1xi2s_component,
+ &au1xi2s_dai_driver, 1);
}
static int au1xi2s_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
WR(ctx, I2S_ENABLE, EN_D); /* clock off, disable */
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 6ba07e365967..8f1862aa7333 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -361,6 +361,10 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = {
.ops = &au1xpsc_ac97_dai_ops,
};
+static const struct snd_soc_component_driver au1xpsc_ac97_component = {
+ .name = "au1xpsc-ac97",
+};
+
static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
{
int ret;
@@ -419,7 +423,8 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
platform_set_drvdata(pdev, wd);
- ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
+ ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component,
+ &wd->dai_drv, 1);
if (ret)
return ret;
@@ -431,7 +436,7 @@ static int au1xpsc_ac97_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
/* disable PSC completely */
au_writel(0, AC97_CFG(wd));
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 360b4e50d7c8..fe923a7bdc39 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -288,6 +288,10 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = {
.ops = &au1xpsc_i2s_dai_ops,
};
+static const struct snd_soc_component_driver au1xpsc_i2s_component = {
+ .name = "au1xpsc-i2s",
+};
+
static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
{
struct resource *iores, *dmares;
@@ -350,14 +354,15 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
platform_set_drvdata(pdev, wd);
- return snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
+ return snd_soc_register_component(&pdev->dev, &au1xpsc_i2s_component,
+ &wd->dai_drv, 1);
}
static int au1xpsc_i2s_drvremove(struct platform_device *pdev)
{
struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
au_writel(0, I2S_CFG(wd));
au_sync();
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 8e41bcb020eb..490217325975 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -282,6 +282,10 @@ static struct snd_soc_dai_driver bfin_ac97_dai = {
.formats = SNDRV_PCM_FMTBIT_S16_LE, },
};
+static const struct snd_soc_component_driver bfin_ac97_component = {
+ .name = "bfin-ac97",
+};
+
static int asoc_bfin_ac97_probe(struct platform_device *pdev)
{
struct sport_device *sport_handle;
@@ -331,7 +335,8 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev)
goto sport_config_err;
}
- ret = snd_soc_register_dai(&pdev->dev, &bfin_ac97_dai);
+ ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component,
+ &bfin_ac97_dai, 1);
if (ret) {
pr_err("Failed to register DAI: %d\n", ret);
goto sport_config_err;
@@ -356,7 +361,7 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev)
{
struct sport_device *sport_handle = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
sport_done(sport_handle);
#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 168d88bccb41..dd0c2a4f83a3 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -245,6 +245,10 @@ static struct snd_soc_dai_driver bf5xx_i2s_dai = {
.ops = &bf5xx_i2s_dai_ops,
};
+static const struct snd_soc_component_driver bf5xx_i2s_component = {
+ .name = "bf5xx-i2s",
+};
+
static int bf5xx_i2s_probe(struct platform_device *pdev)
{
struct sport_device *sport_handle;
@@ -257,7 +261,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev)
return -ENODEV;
/* register with the ASoC layers */
- ret = snd_soc_register_dai(&pdev->dev, &bf5xx_i2s_dai);
+ ret = snd_soc_register_component(&pdev->dev, &bf5xx_i2s_component,
+ &bf5xx_i2s_dai, 1);
if (ret) {
pr_err("Failed to register DAI: %d\n", ret);
sport_done(sport_handle);
@@ -273,7 +278,7 @@ static int bf5xx_i2s_remove(struct platform_device *pdev)
pr_debug("%s enter\n", __func__);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
sport_done(sport_handle);
return 0;
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index c1e516ec53ad..69e9a3e935bd 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -249,6 +249,10 @@ static struct snd_soc_dai_driver bf5xx_tdm_dai = {
.ops = &bf5xx_tdm_dai_ops,
};
+static const struct snd_soc_component_driver bf5xx_tdm_component = {
+ .name = "bf5xx-tdm",
+};
+
static int bfin_tdm_probe(struct platform_device *pdev)
{
struct sport_device *sport_handle;
@@ -282,7 +286,8 @@ static int bfin_tdm_probe(struct platform_device *pdev)
goto sport_config_err;
}
- ret = snd_soc_register_dai(&pdev->dev, &bf5xx_tdm_dai);
+ ret = snd_soc_register_component(&pdev->dev, &bf5xx_tdm_component,
+ &bf5xx_tdm_dai, 1);
if (ret) {
pr_err("Failed to register DAI: %d\n", ret);
goto sport_config_err;
@@ -299,7 +304,7 @@ static int bfin_tdm_remove(struct platform_device *pdev)
{
struct sport_device *sport_handle = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
sport_done(sport_handle);
return 0;
diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
index 8f337972f438..c02405cc007d 100644
--- a/sound/soc/blackfin/bf6xx-i2s.c
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -186,6 +186,10 @@ static struct snd_soc_dai_driver bfin_i2s_dai = {
.ops = &bfin_i2s_dai_ops,
};
+static const struct snd_soc_component_driver bfin_i2s_component = {
+ .name = "bfin-i2s",
+};
+
static int bfin_i2s_probe(struct platform_device *pdev)
{
struct sport_device *sport;
@@ -197,7 +201,8 @@ static int bfin_i2s_probe(struct platform_device *pdev)
return -ENODEV;
/* register with the ASoC layers */
- ret = snd_soc_register_dai(dev, &bfin_i2s_dai);
+ ret = snd_soc_register_component(dev, &bfin_i2s_component,
+ &bfin_i2s_dai, 1);
if (ret) {
dev_err(dev, "Failed to register DAI: %d\n", ret);
sport_delete(sport);
@@ -212,7 +217,7 @@ static int bfin_i2s_remove(struct platform_device *pdev)
{
struct sport_device *sport = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
sport_delete(sport);
return 0;
diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c
index 5db68cf7b281..c43fb214558a 100644
--- a/sound/soc/cirrus/edb93xx.c
+++ b/sound/soc/cirrus/edb93xx.c
@@ -27,7 +27,6 @@
#include <sound/soc.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
-#include "ep93xx-pcm.h"
static int edb93xx_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index 1738d28fb04f..7798fbd5e81d 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -23,7 +23,6 @@
#include <sound/soc.h>
#include <linux/platform_data/dma-ep93xx.h>
-#include "ep93xx-pcm.h"
/*
* Per channel (1-4) registers.
@@ -101,14 +100,16 @@ struct ep93xx_ac97_info {
/* currently ALSA only supports a single AC97 device */
static struct ep93xx_ac97_info *ep93xx_ac97_info;
-static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_out = {
+static struct ep93xx_dma_data ep93xx_ac97_pcm_out = {
.name = "ac97-pcm-out",
.dma_port = EP93XX_DMA_AAC1,
+ .direction = DMA_MEM_TO_DEV,
};
-static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_in = {
+static struct ep93xx_dma_data ep93xx_ac97_pcm_in = {
.name = "ac97-pcm-in",
.dma_port = EP93XX_DMA_AAC1,
+ .direction = DMA_DEV_TO_MEM,
};
static inline unsigned ep93xx_ac97_read_reg(struct ep93xx_ac97_info *info,
@@ -316,7 +317,7 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
static int ep93xx_ac97_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct ep93xx_pcm_dma_params *dma_data;
+ struct ep93xx_dma_data *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dma_data = &ep93xx_ac97_pcm_out;
@@ -353,6 +354,10 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = {
.ops = &ep93xx_ac97_dai_ops,
};
+static const struct snd_soc_component_driver ep93xx_ac97_component = {
+ .name = "ep93xx-ac97",
+};
+
static int ep93xx_ac97_probe(struct platform_device *pdev)
{
struct ep93xx_ac97_info *info;
@@ -390,7 +395,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev)
ep93xx_ac97_info = info;
platform_set_drvdata(pdev, info);
- ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai);
+ ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component,
+ &ep93xx_ac97_dai, 1);
if (ret)
goto fail;
@@ -407,7 +413,7 @@ static int ep93xx_ac97_remove(struct platform_device *pdev)
{
struct ep93xx_ac97_info *info = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
/* disable the AC97 controller */
ep93xx_ac97_write_reg(info, AC97GCR, 0);
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index 323ed69b7975..5c1102e9e159 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -30,8 +30,6 @@
#include <mach/ep93xx-regs.h>
#include <linux/platform_data/dma-ep93xx.h>
-#include "ep93xx-pcm.h"
-
#define EP93XX_I2S_TXCLKCFG 0x00
#define EP93XX_I2S_RXCLKCFG 0x04
#define EP93XX_I2S_GLCTRL 0x0C
@@ -62,18 +60,20 @@ struct ep93xx_i2s_info {
struct clk *mclk;
struct clk *sclk;
struct clk *lrclk;
- struct ep93xx_pcm_dma_params *dma_params;
+ struct ep93xx_dma_data *dma_data;
void __iomem *regs;
};
-struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = {
+struct ep93xx_dma_data ep93xx_i2s_dma_data[] = {
[SNDRV_PCM_STREAM_PLAYBACK] = {
.name = "i2s-pcm-out",
- .dma_port = EP93XX_DMA_I2S1,
+ .port = EP93XX_DMA_I2S1,
+ .direction = DMA_MEM_TO_DEV,
},
[SNDRV_PCM_STREAM_CAPTURE] = {
.name = "i2s-pcm-in",
- .dma_port = EP93XX_DMA_I2S1,
+ .port = EP93XX_DMA_I2S1,
+ .direction = DMA_DEV_TO_MEM,
},
};
@@ -147,7 +147,7 @@ static int ep93xx_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
snd_soc_dai_set_dma_data(cpu_dai, substream,
- &info->dma_params[substream->stream]);
+ &info->dma_data[substream->stream]);
return 0;
}
@@ -366,6 +366,10 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = {
.ops = &ep93xx_i2s_dai_ops,
};
+static const struct snd_soc_component_driver ep93xx_i2s_component = {
+ .name = "ep93xx-i2s",
+};
+
static int ep93xx_i2s_probe(struct platform_device *pdev)
{
struct ep93xx_i2s_info *info;
@@ -403,9 +407,10 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
}
dev_set_drvdata(&pdev->dev, info);
- info->dma_params = ep93xx_i2s_dma_params;
+ info->dma_data = ep93xx_i2s_dma_data;
- err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai);
+ err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component,
+ &ep93xx_i2s_dai, 1);
if (err)
goto fail_put_lrclk;
@@ -426,7 +431,7 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
{
struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
clk_put(info->sclk);
diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
index 72eb7a49e16a..488032690378 100644
--- a/sound/soc/cirrus/ep93xx-pcm.c
+++ b/sound/soc/cirrus/ep93xx-pcm.c
@@ -29,8 +29,6 @@
#include <mach/hardware.h>
#include <mach/ep93xx-regs.h>
-#include "ep93xx-pcm.h"
-
static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
@@ -68,40 +66,12 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param)
static int ep93xx_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct ep93xx_pcm_dma_params *dma_params;
- struct ep93xx_dma_data *dma_data;
- int ret;
snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware);
- dma_data = kmalloc(sizeof(*dma_data), GFP_KERNEL);
- if (!dma_data)
- return -ENOMEM;
-
- dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream);
- dma_data->port = dma_params->dma_port;
- dma_data->name = dma_params->name;
- dma_data->direction = snd_pcm_substream_to_dma_direction(substream);
-
- ret = snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, dma_data);
- if (ret) {
- kfree(dma_data);
- return ret;
- }
-
- snd_dmaengine_pcm_set_data(substream, dma_data);
-
- return 0;
-}
-
-static int ep93xx_pcm_close(struct snd_pcm_substream *substream)
-{
- struct dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
-
- snd_dmaengine_pcm_close(substream);
- kfree(dma_data);
- return 0;
+ return snd_dmaengine_pcm_open_request_chan(substream,
+ ep93xx_pcm_dma_filter,
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream));
}
static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -131,7 +101,7 @@ static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream,
static struct snd_pcm_ops ep93xx_pcm_ops = {
.open = ep93xx_pcm_open,
- .close = ep93xx_pcm_close,
+ .close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = ep93xx_pcm_hw_params,
.hw_free = ep93xx_pcm_hw_free,
diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h
deleted file mode 100644
index 111e1121ecb8..000000000000
--- a/sound/soc/cirrus/ep93xx-pcm.h
+++ /dev/null
@@ -1,20 +0,0 @@
-/*
- * sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface
- *
- * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org>
- * Copyright (C) 2006 Applied Data Systems
- *
- * 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 _EP93XX_SND_SOC_PCM_H
-#define _EP93XX_SND_SOC_PCM_H
-
-struct ep93xx_pcm_dma_params {
- char *name;
- int dma_port;
-};
-
-#endif /* _EP93XX_SND_SOC_PCM_H */
diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c
index a397bb0d8179..4d094d00c34a 100644
--- a/sound/soc/cirrus/simone.c
+++ b/sound/soc/cirrus/simone.c
@@ -21,8 +21,6 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
-#include "ep93xx-pcm.h"
-
static struct snd_soc_dai_link simone_dai = {
.name = "AC97",
.stream_name = "AC97 HiFi",
diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c
index 9d77fe28dfcc..69041074f2c1 100644
--- a/sound/soc/cirrus/snappercl15.c
+++ b/sound/soc/cirrus/snappercl15.c
@@ -21,7 +21,6 @@
#include <mach/hardware.h>
#include "../codecs/tlv320aic23.h"
-#include "ep93xx-pcm.h"
#define CODEC_CLOCK 5644800
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 45b72561c615..2f45f00e31b0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
+ select SND_SOC_AK5386
select SND_SOC_ALC5623 if I2C
select SND_SOC_ALC5632 if I2C
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
@@ -63,6 +64,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_STA32X if I2C
select SND_SOC_STA529 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
+ select SND_SOC_TAS5086 if I2C
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC32X4 if I2C
@@ -203,6 +205,9 @@ config SND_SOC_AK4642
config SND_SOC_AK4671
tristate
+config SND_SOC_AK5386
+ tristate
+
config SND_SOC_ALC5623
tristate
config SND_SOC_ALC5632
@@ -320,11 +325,14 @@ config SND_SOC_STA529
config SND_SOC_STAC9766
tristate
+config SND_SOC_TAS5086
+ tristate
+
config SND_SOC_TLV320AIC23
tristate
config SND_SOC_TLV320AIC26
- tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
+ tristate
depends on SPI
config SND_SOC_TLV320AIC32X4
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6a3b3c3b8b41..b9e41c9a1f4c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
+snd-soc-ak5386-objs := ak5386.o
snd-soc-arizona-objs := arizona.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs42l51-objs := cs42l51.o
@@ -55,6 +56,7 @@ snd-soc-ssm2602-objs := ssm2602.o
snd-soc-sta32x-objs := sta32x.o
snd-soc-sta529-objs := sta529.o
snd-soc-stac9766-objs := stac9766.o
+snd-soc-tas5086-objs := tas5086.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -137,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o
obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o
obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o
@@ -177,6 +180,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
+obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 068b3ae56a17..1aa10ddf3a61 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -133,6 +133,8 @@ struct adau1373 {
#define ADAU1373_DAI_FORMAT_DSP 0x3
#define ADAU1373_BCLKDIV_SOURCE BIT(5)
+#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2)
+#define ADAU1373_BCLKDIV_BCLK_MASK 0x03
#define ADAU1373_BCLKDIV_32 0x03
#define ADAU1373_BCLKDIV_64 0x02
#define ADAU1373_BCLKDIV_128 0x01
@@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream,
adau1373_dai->enable_src = (div != 0);
snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id),
- ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64);
+ ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK,
+ (div << 2) | ADAU1373_BCLKDIV_64);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index 6f6c335a5baa..c7cfdf957e4d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
int val = 0;
int ret;
@@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
return -EINVAL;
- ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
- AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
- val);
+ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+ AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
+ val);
if (ret < 0)
return ret;
@@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
- int val = 0;
+ struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+ int ret, val = 0;
/* set the IEC958 bits: consumer mode, no copyright bit */
val |= IEC958_AES0_CON_NOT_COPYRIGHT;
- snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val);
+ regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val);
val = 0;
@@ -132,11 +134,33 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val);
+ ret = regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val);
+ if (ret < 0)
+ return ret;
+
+ /* enable transmitter */
+ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+ AK4104_TX_TXE, AK4104_TX_TXE);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int ak4104_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+
+ /* disable transmitter */
+ return regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+ AK4104_TX_TXE, 0);
}
static const struct snd_soc_dai_ops ak4101_dai_ops = {
.hw_params = ak4104_hw_params,
+ .hw_free = ak4104_hw_free,
.set_fmt = ak4104_set_dai_fmt,
};
@@ -160,20 +184,17 @@ static int ak4104_probe(struct snd_soc_codec *codec)
int ret;
codec->control_data = ak4104->regmap;
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
- if (ret != 0)
- return ret;
/* set power-up and non-reset bits */
- ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
- AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
- AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+ AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
+ AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
if (ret < 0)
return ret;
/* enable transmitter */
- ret = snd_soc_update_bits(codec, AK4104_REG_TX,
- AK4104_TX_TXE, AK4104_TX_TXE);
+ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+ AK4104_TX_TXE, AK4104_TX_TXE);
if (ret < 0)
return ret;
@@ -182,8 +203,10 @@ static int ak4104_probe(struct snd_soc_codec *codec)
static int ak4104_remove(struct snd_soc_codec *codec)
{
- snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
- AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
+ struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+
+ regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+ AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
return 0;
}
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
new file mode 100644
index 000000000000..1f303983ae02
--- /dev/null
+++ b/sound/soc/codecs/ak5386.c
@@ -0,0 +1,152 @@
+/*
+ * ALSA SoC driver for
+ * Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC
+ *
+ * (c) 2013 Daniel Mack <zonque@gmail.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/slab.h>
+#include <linux/of.h>
+#include <linux/of_gpio.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+struct ak5386_priv {
+ int reset_gpio;
+};
+
+static struct snd_soc_codec_driver soc_codec_ak5386;
+
+static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ format &= SND_SOC_DAIFMT_FORMAT_MASK;
+ if (format != SND_SOC_DAIFMT_LEFT_J &&
+ format != SND_SOC_DAIFMT_I2S) {
+ dev_err(codec->dev, "Invalid DAI format\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ak5386_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 ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ /*
+ * From the datasheet:
+ *
+ * All external clocks (MCLK, SCLK and LRCK) must be present unless
+ * PDN pin = ā€œLā€. If these clocks are not provided, the AK5386 may
+ * draw excess current due to its use of internal dynamically
+ * refreshed logic. If the external clocks are not present, place
+ * the AK5386 in power-down mode (PDN pin = ā€œLā€).
+ */
+
+ if (gpio_is_valid(priv->reset_gpio))
+ gpio_set_value(priv->reset_gpio, 1);
+
+ return 0;
+}
+
+static int ak5386_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (gpio_is_valid(priv->reset_gpio))
+ gpio_set_value(priv->reset_gpio, 0);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ak5386_dai_ops = {
+ .set_fmt = ak5386_set_dai_fmt,
+ .hw_params = ak5386_hw_params,
+ .hw_free = ak5386_hw_free,
+};
+
+static struct snd_soc_dai_driver ak5386_dai = {
+ .name = "ak5386-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE,
+ },
+ .ops = &ak5386_dai_ops,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id ak5386_dt_ids[] = {
+ { .compatible = "asahi-kasei,ak5386", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, ak5386_dt_ids);
+#endif
+
+static int ak5386_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct ak5386_priv *priv;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->reset_gpio = -EINVAL;
+ dev_set_drvdata(dev, priv);
+
+ if (of_match_device(of_match_ptr(ak5386_dt_ids), dev))
+ priv->reset_gpio = of_get_named_gpio(dev->of_node,
+ "reset-gpio", 0);
+
+ if (gpio_is_valid(priv->reset_gpio))
+ if (devm_gpio_request_one(dev, priv->reset_gpio,
+ GPIOF_OUT_INIT_LOW,
+ "AK5386 Reset"))
+ priv->reset_gpio = -EINVAL;
+
+ return snd_soc_register_codec(dev, &soc_codec_ak5386,
+ &ak5386_dai, 1);
+}
+
+static int ak5386_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver ak5386_driver = {
+ .probe = ak5386_probe,
+ .remove = ak5386_remove,
+ .driver = {
+ .name = "ak5386",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(ak5386_dt_ids),
+ },
+};
+
+module_platform_driver(ak5386_driver);
+
+MODULE_DESCRIPTION("ASoC driver for AK5386 ADC");
+MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e7d34711412c..389f23253831 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -10,6 +10,7 @@
* published by the Free Software Foundation.
*/
+#include <linux/delay.h>
#include <linux/gcd.h>
#include <linux/module.h>
#include <linux/pm_runtime.h>
@@ -65,6 +66,163 @@
#define arizona_aif_dbg(_dai, fmt, ...) \
dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ bool manual_ena = false;
+ int val;
+
+ switch (arizona->type) {
+ case WM5102:
+ switch (arizona->rev) {
+ case 0:
+ break;
+ default:
+ manual_ena = true;
+ break;
+ }
+ default:
+ break;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (!priv->spk_ena && manual_ena) {
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ priv->spk_ena_pending = true;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ val = snd_soc_read(codec, ARIZONA_INTERRUPT_RAW_STATUS_3);
+ if (val & ARIZONA_SPK_SHUTDOWN_STS) {
+ dev_crit(arizona->dev,
+ "Speaker not enabled due to temperature\n");
+ return -EBUSY;
+ }
+
+ snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1,
+ 1 << w->shift, 1 << w->shift);
+
+ if (priv->spk_ena_pending) {
+ msleep(75);
+ snd_soc_write(codec, 0x4f5, 0xda);
+ priv->spk_ena_pending = false;
+ priv->spk_ena++;
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ if (manual_ena) {
+ priv->spk_ena--;
+ if (!priv->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ }
+
+ snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1,
+ 1 << w->shift, 0);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ if (manual_ena) {
+ if (!priv->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x0da);
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static irqreturn_t arizona_thermal_warn(int irq, void *data)
+{
+ struct arizona *arizona = data;
+ unsigned int val;
+ int ret;
+
+ ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3,
+ &val);
+ if (ret != 0) {
+ dev_err(arizona->dev, "Failed to read thermal status: %d\n",
+ ret);
+ } else if (val & ARIZONA_SPK_SHUTDOWN_WARN_STS) {
+ dev_crit(arizona->dev, "Thermal warning\n");
+ }
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t arizona_thermal_shutdown(int irq, void *data)
+{
+ struct arizona *arizona = data;
+ unsigned int val;
+ int ret;
+
+ ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3,
+ &val);
+ if (ret != 0) {
+ dev_err(arizona->dev, "Failed to read thermal status: %d\n",
+ ret);
+ } else if (val & ARIZONA_SPK_SHUTDOWN_STS) {
+ dev_crit(arizona->dev, "Thermal shutdown\n");
+ ret = regmap_update_bits(arizona->regmap,
+ ARIZONA_OUTPUT_ENABLES_1,
+ ARIZONA_OUT4L_ENA |
+ ARIZONA_OUT4R_ENA, 0);
+ if (ret != 0)
+ dev_crit(arizona->dev,
+ "Failed to disable speaker outputs: %d\n",
+ ret);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static const struct snd_soc_dapm_widget arizona_spkl =
+ SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM,
+ ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU);
+
+static const struct snd_soc_dapm_widget arizona_spkr =
+ SND_SOC_DAPM_PGA_E("OUT4R", SND_SOC_NOPM,
+ ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU);
+
+int arizona_init_spk(struct snd_soc_codec *codec)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1);
+ if (ret != 0)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1);
+ if (ret != 0)
+ return ret;
+
+ ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN,
+ "Thermal warning", arizona_thermal_warn,
+ arizona);
+ if (ret != 0)
+ dev_err(arizona->dev,
+ "Failed to get thermal warning IRQ: %d\n",
+ ret);
+
+ ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN,
+ "Thermal shutdown", arizona_thermal_shutdown,
+ arizona);
+ if (ret != 0)
+ dev_err(arizona->dev,
+ "Failed to get thermal shutdown IRQ: %d\n",
+ ret);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_spk);
+
const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"None",
"Tone Generator 1",
@@ -274,6 +432,33 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values);
const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
+const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
+ "SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate",
+};
+EXPORT_SYMBOL_GPL(arizona_rate_text);
+
+const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = {
+ 0, 1, 2, 8,
+};
+EXPORT_SYMBOL_GPL(arizona_rate_val);
+
+
+const struct soc_enum arizona_isrc_fsl[] = {
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2,
+ ARIZONA_ISRC1_FSL_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_2,
+ ARIZONA_ISRC2_FSL_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_2,
+ ARIZONA_ISRC3_FSL_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+};
+EXPORT_SYMBOL_GPL(arizona_isrc_fsl);
+
static const char *arizona_vol_ramp_text[] = {
"0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB",
"15ms/6dB", "30ms/6dB",
@@ -332,9 +517,27 @@ const struct soc_enum arizona_ng_hold =
4, arizona_ng_hold_text);
EXPORT_SYMBOL_GPL(arizona_ng_hold);
+static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+ int i;
+
+ if (ena)
+ val = ARIZONA_IN_VU;
+ else
+ val = 0;
+
+ for (i = 0; i < priv->num_inputs; i++)
+ snd_soc_update_bits(codec,
+ ARIZONA_ADC_DIGITAL_VOLUME_1L + (i * 4),
+ ARIZONA_IN_VU, val);
+}
+
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
unsigned int reg;
if (w->shift % 2)
@@ -343,13 +546,29 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
reg = ARIZONA_ADC_DIGITAL_VOLUME_1R + ((w->shift / 2) * 8);
switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ priv->in_pending++;
+ break;
case SND_SOC_DAPM_POST_PMU:
snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0);
+
+ /* If this is the last input pending then allow VU */
+ priv->in_pending--;
+ if (priv->in_pending == 0) {
+ msleep(1);
+ arizona_in_set_vu(w->codec, 1);
+ }
break;
case SND_SOC_DAPM_PRE_PMD:
- snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE,
- ARIZONA_IN1L_MUTE);
+ snd_soc_update_bits(w->codec, reg,
+ ARIZONA_IN1L_MUTE | ARIZONA_IN_VU,
+ ARIZONA_IN1L_MUTE | ARIZONA_IN_VU);
break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* Disable volume updates if no inputs are enabled */
+ reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES);
+ if (reg == 0)
+ arizona_in_set_vu(w->codec, 0);
}
return 0;
@@ -360,6 +579,24 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ switch (w->shift) {
+ case ARIZONA_OUT1L_ENA_SHIFT:
+ case ARIZONA_OUT1R_ENA_SHIFT:
+ case ARIZONA_OUT2L_ENA_SHIFT:
+ case ARIZONA_OUT2R_ENA_SHIFT:
+ case ARIZONA_OUT3L_ENA_SHIFT:
+ case ARIZONA_OUT3R_ENA_SHIFT:
+ msleep(17);
+ break;
+
+ default:
+ break;
+ }
+ break;
+ }
+
return 0;
}
EXPORT_SYMBOL_GPL(arizona_out_ev);
@@ -502,27 +739,27 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
break;
case 11289600:
case 12288000:
- val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ val |= ARIZONA_CLK_12MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
case 22579200:
case 24576000:
- val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ val |= ARIZONA_CLK_24MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
case 45158400:
case 49152000:
- val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ val |= ARIZONA_CLK_49MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
case 67737600:
case 73728000:
- val |= 4 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ val |= ARIZONA_CLK_73MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
case 90316800:
case 98304000:
- val |= 5 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ val |= ARIZONA_CLK_98MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
case 135475200:
case 147456000:
- val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT;
+ val |= ARIZONA_CLK_147MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
break;
case 0:
dev_dbg(arizona->dev, "%s cleared\n", name);
@@ -816,7 +1053,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
struct arizona *arizona = priv->arizona;
int base = dai->driver->base;
const int *rates;
- int i, ret;
+ int i, ret, val;
int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1];
int bclk, lrclk, wl, frame, bclk_target;
@@ -832,6 +1069,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
bclk_target *= chan_limit;
}
+ /* Force stereo for I2S mode */
+ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
+ if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) {
+ arizona_aif_dbg(dai, "Forcing stereo mode\n");
+ bclk_target *= 2;
+ }
+
for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
if (rates[i] >= bclk_target &&
rates[i] % params_rate(params) == 0) {
@@ -988,6 +1232,16 @@ static struct {
{ 1000000, 13500000, 0, 1 },
};
+static struct {
+ unsigned int min;
+ unsigned int max;
+ u16 gain;
+} fll_gains[] = {
+ { 0, 256000, 0 },
+ { 256000, 1000000, 2 },
+ { 1000000, 13500000, 4 },
+};
+
struct arizona_fll_cfg {
int n;
int theta;
@@ -995,6 +1249,7 @@ struct arizona_fll_cfg {
int refdiv;
int outdiv;
int fratio;
+ int gain;
};
static int arizona_calc_fll(struct arizona_fll *fll,
@@ -1054,6 +1309,18 @@ static int arizona_calc_fll(struct arizona_fll *fll,
return -EINVAL;
}
+ for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
+ if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
+ cfg->gain = fll_gains[i].gain;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_gains)) {
+ arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
+ Fref);
+ return -EINVAL;
+ }
+
cfg->n = target / (ratio * Fref);
if (target % (ratio * Fref)) {
@@ -1081,13 +1348,15 @@ static int arizona_calc_fll(struct arizona_fll *fll,
cfg->n, cfg->theta, cfg->lambda);
arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
+ arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain);
return 0;
}
static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
- struct arizona_fll_cfg *cfg, int source)
+ struct arizona_fll_cfg *cfg, int source,
+ bool sync)
{
regmap_update_bits(arizona->regmap, base + 3,
ARIZONA_FLL1_THETA_MASK, cfg->theta);
@@ -1102,87 +1371,84 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
+ if (sync)
+ regmap_update_bits(arizona->regmap, base + 0x7,
+ ARIZONA_FLL1_GAIN_MASK,
+ cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ else
+ regmap_update_bits(arizona->regmap, base + 0x9,
+ ARIZONA_FLL1_GAIN_MASK,
+ cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+
regmap_update_bits(arizona->regmap, base + 2,
ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
ARIZONA_FLL1_CTRL_UPD | cfg->n);
}
-int arizona_set_fll(struct arizona_fll *fll, int source,
- unsigned int Fref, unsigned int Fout)
+static bool arizona_is_enabled_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
- struct arizona_fll_cfg cfg, sync;
- unsigned int reg, val;
- int syncsrc;
- bool ena;
+ unsigned int reg;
int ret;
- if (fll->fref == Fref && fll->fout == Fout)
- return 0;
-
ret = regmap_read(arizona->regmap, fll->base + 1, &reg);
if (ret != 0) {
arizona_fll_err(fll, "Failed to read current state: %d\n",
ret);
return ret;
}
- ena = reg & ARIZONA_FLL1_ENA;
- if (Fout) {
- /* Do we have a 32kHz reference? */
- regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
- switch (val & ARIZONA_CLK_32K_SRC_MASK) {
- case ARIZONA_CLK_SRC_MCLK1:
- case ARIZONA_CLK_SRC_MCLK2:
- syncsrc = val & ARIZONA_CLK_32K_SRC_MASK;
- break;
- default:
- syncsrc = -1;
- }
+ return reg & ARIZONA_FLL1_ENA;
+}
- if (source == syncsrc)
- syncsrc = -1;
+static void arizona_enable_fll(struct arizona_fll *fll,
+ struct arizona_fll_cfg *ref,
+ struct arizona_fll_cfg *sync)
+{
+ struct arizona *arizona = fll->arizona;
+ int ret;
- if (syncsrc >= 0) {
- ret = arizona_calc_fll(fll, &sync, Fref, Fout);
- if (ret != 0)
- return ret;
+ /*
+ * If we have both REFCLK and SYNCCLK then enable both,
+ * otherwise apply the SYNCCLK settings to REFCLK.
+ */
+ if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) {
+ regmap_update_bits(arizona->regmap, fll->base + 5,
+ ARIZONA_FLL1_OUTDIV_MASK,
+ ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+ arizona_apply_fll(arizona, fll->base, ref, fll->ref_src,
+ false);
+ if (fll->sync_src >= 0)
+ arizona_apply_fll(arizona, fll->base + 0x10, sync,
+ fll->sync_src, true);
+ } else if (fll->sync_src >= 0) {
+ regmap_update_bits(arizona->regmap, fll->base + 5,
+ ARIZONA_FLL1_OUTDIV_MASK,
+ sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+ arizona_apply_fll(arizona, fll->base, sync,
+ fll->sync_src, false);
- ret = arizona_calc_fll(fll, &cfg, 32768, Fout);
- if (ret != 0)
- return ret;
- } else {
- ret = arizona_calc_fll(fll, &cfg, Fref, Fout);
- if (ret != 0)
- return ret;
- }
- } else {
- regmap_update_bits(arizona->regmap, fll->base + 1,
- ARIZONA_FLL1_ENA, 0);
regmap_update_bits(arizona->regmap, fll->base + 0x11,
ARIZONA_FLL1_SYNC_ENA, 0);
-
- if (ena)
- pm_runtime_put_autosuspend(arizona->dev);
-
- fll->fref = Fref;
- fll->fout = Fout;
-
- return 0;
- }
-
- regmap_update_bits(arizona->regmap, fll->base + 5,
- ARIZONA_FLL1_OUTDIV_MASK,
- cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
-
- if (syncsrc >= 0) {
- arizona_apply_fll(arizona, fll->base, &cfg, syncsrc);
- arizona_apply_fll(arizona, fll->base + 0x10, &sync, source);
} else {
- arizona_apply_fll(arizona, fll->base, &cfg, source);
+ arizona_fll_err(fll, "No clocks provided\n");
+ return;
}
- if (!ena)
+ /*
+ * Increase the bandwidth if we're not using a low frequency
+ * sync source.
+ */
+ if (fll->sync_src >= 0 && fll->sync_freq > 100000)
+ regmap_update_bits(arizona->regmap, fll->base + 0x17,
+ ARIZONA_FLL1_SYNC_BW, 0);
+ else
+ regmap_update_bits(arizona->regmap, fll->base + 0x17,
+ ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW);
+
+ if (!arizona_is_enabled_fll(fll))
pm_runtime_get(arizona->dev);
/* Clear any pending completions */
@@ -1190,7 +1456,8 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
regmap_update_bits(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
- if (syncsrc >= 0)
+ if (fll->ref_src >= 0 && fll->sync_src >= 0 &&
+ fll->ref_src != fll->sync_src)
regmap_update_bits(arizona->regmap, fll->base + 0x11,
ARIZONA_FLL1_SYNC_ENA,
ARIZONA_FLL1_SYNC_ENA);
@@ -1199,10 +1466,88 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
msecs_to_jiffies(250));
if (ret == 0)
arizona_fll_warn(fll, "Timed out waiting for lock\n");
+}
+
+static void arizona_disable_fll(struct arizona_fll *fll)
+{
+ struct arizona *arizona = fll->arizona;
+ bool change;
+
+ regmap_update_bits_check(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_ENA, 0, &change);
+ regmap_update_bits(arizona->regmap, fll->base + 0x11,
+ ARIZONA_FLL1_SYNC_ENA, 0);
+
+ if (change)
+ pm_runtime_put_autosuspend(arizona->dev);
+}
+
+int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct arizona_fll_cfg ref, sync;
+ int ret;
- fll->fref = Fref;
+ if (fll->ref_src == source && fll->ref_freq == Fref)
+ return 0;
+
+ if (fll->fout && Fref > 0) {
+ ret = arizona_calc_fll(fll, &ref, Fref, fll->fout);
+ if (ret != 0)
+ return ret;
+
+ if (fll->sync_src >= 0) {
+ ret = arizona_calc_fll(fll, &sync, fll->sync_freq,
+ fll->fout);
+ if (ret != 0)
+ return ret;
+ }
+ }
+
+ fll->ref_src = source;
+ fll->ref_freq = Fref;
+
+ if (fll->fout && Fref > 0) {
+ arizona_enable_fll(fll, &ref, &sync);
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_set_fll_refclk);
+
+int arizona_set_fll(struct arizona_fll *fll, int source,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct arizona_fll_cfg ref, sync;
+ int ret;
+
+ if (fll->sync_src == source &&
+ fll->sync_freq == Fref && fll->fout == Fout)
+ return 0;
+
+ if (Fout) {
+ if (fll->ref_src >= 0) {
+ ret = arizona_calc_fll(fll, &ref, fll->ref_freq,
+ Fout);
+ if (ret != 0)
+ return ret;
+ }
+
+ ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+ if (ret != 0)
+ return ret;
+ }
+
+ fll->sync_src = source;
+ fll->sync_freq = Fref;
fll->fout = Fout;
+ if (Fout) {
+ arizona_enable_fll(fll, &ref, &sync);
+ } else {
+ arizona_disable_fll(fll);
+ }
+
return 0;
}
EXPORT_SYMBOL_GPL(arizona_set_fll);
@@ -1211,12 +1556,26 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
int ok_irq, struct arizona_fll *fll)
{
int ret;
+ unsigned int val;
init_completion(&fll->ok);
fll->id = id;
fll->base = base;
fll->arizona = arizona;
+ fll->sync_src = ARIZONA_FLL_SRC_NONE;
+
+ /* Configure default refclk to 32kHz if we have one */
+ regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
+ switch (val & ARIZONA_CLK_32K_SRC_MASK) {
+ case ARIZONA_CLK_SRC_MCLK1:
+ case ARIZONA_CLK_SRC_MCLK2:
+ fll->ref_src = val & ARIZONA_CLK_32K_SRC_MASK;
+ break;
+ default:
+ fll->ref_src = ARIZONA_FLL_SRC_NONE;
+ }
+ fll->ref_freq = 32768;
snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id);
snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 13dd2916b721..af39f1006427 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -32,6 +32,7 @@
#define ARIZONA_CLK_SRC_AIF2BCLK 0x9
#define ARIZONA_CLK_SRC_AIF3BCLK 0xa
+#define ARIZONA_FLL_SRC_NONE -1
#define ARIZONA_FLL_SRC_MCLK1 0
#define ARIZONA_FLL_SRC_MCLK2 1
#define ARIZONA_FLL_SRC_SLIMCLK 3
@@ -48,6 +49,14 @@
#define ARIZONA_MIXER_VOL_SHIFT 1
#define ARIZONA_MIXER_VOL_WIDTH 7
+#define ARIZONA_CLK_6MHZ 0
+#define ARIZONA_CLK_12MHZ 1
+#define ARIZONA_CLK_24MHZ 2
+#define ARIZONA_CLK_49MHZ 3
+#define ARIZONA_CLK_73MHZ 4
+#define ARIZONA_CLK_98MHZ 5
+#define ARIZONA_CLK_147MHZ 6
+
#define ARIZONA_MAX_DAI 4
#define ARIZONA_MAX_ADSP 4
@@ -64,6 +73,12 @@ struct arizona_priv {
int sysclk;
int asyncclk;
struct arizona_dai_priv dai[ARIZONA_MAX_DAI];
+
+ int num_inputs;
+ unsigned int in_pending;
+
+ unsigned int spk_ena:2;
+ unsigned int spk_ena_pending:1;
};
#define ARIZONA_NUM_MIXER_INPUTS 99
@@ -165,6 +180,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
ARIZONA_MIXER_ROUTES(name, name "L"), \
ARIZONA_MIXER_ROUTES(name, name "R")
+#define ARIZONA_RATE_ENUM_SIZE 4
+extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
+extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
+
+extern const struct soc_enum arizona_isrc_fsl[];
+
extern const struct soc_enum arizona_in_vi_ramp;
extern const struct soc_enum arizona_in_vd_ramp;
@@ -201,8 +222,12 @@ struct arizona_fll {
unsigned int base;
unsigned int vco_mult;
struct completion ok;
- unsigned int fref;
+
unsigned int fout;
+ int sync_src;
+ unsigned int sync_freq;
+ int ref_src;
+ unsigned int ref_freq;
char lock_name[ARIZONA_FLL_NAME_LEN];
char clock_ok_name[ARIZONA_FLL_NAME_LEN];
@@ -210,9 +235,13 @@ struct arizona_fll {
extern int arizona_init_fll(struct arizona *arizona, int id, int base,
int lock_irq, int ok_irq, struct arizona_fll *fll);
+extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
+ unsigned int Fref, unsigned int Fout);
extern int arizona_set_fll(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout);
+extern int arizona_init_spk(struct snd_soc_codec *codec);
+
extern int arizona_init_dai(struct arizona_priv *priv, int dai);
int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 2415a4118dbd..03036b326732 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -39,17 +39,15 @@
/*
* CS4271 registers
- * High byte represents SPI chip address (0x10) + write command (0)
- * Low byte - codec register address
*/
-#define CS4271_MODE1 0x2001 /* Mode Control 1 */
-#define CS4271_DACCTL 0x2002 /* DAC Control */
-#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */
-#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */
-#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */
-#define CS4271_ADCCTL 0x2006 /* ADC Control */
-#define CS4271_MODE2 0x2007 /* Mode Control 2 */
-#define CS4271_CHIPID 0x2008 /* Chip ID */
+#define CS4271_MODE1 0x01 /* Mode Control 1 */
+#define CS4271_DACCTL 0x02 /* DAC Control */
+#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */
+#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */
+#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */
+#define CS4271_ADCCTL 0x06 /* ADC Control */
+#define CS4271_MODE2 0x07 /* Mode Control 2 */
+#define CS4271_CHIPID 0x08 /* Chip ID */
#define CS4271_FIRSTREG CS4271_MODE1
#define CS4271_LASTREG CS4271_MODE2
@@ -144,23 +142,27 @@
* Array do not include Chip ID, as codec driver does not use
* registers read operations at all
*/
-static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = {
- 0,
- 0,
- CS4271_DACCTL_AMUTE,
- CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR,
- 0,
- 0,
- 0,
- 0,
+static const struct reg_default cs4271_reg_defaults[] = {
+ { CS4271_MODE1, 0, },
+ { CS4271_DACCTL, CS4271_DACCTL_AMUTE, },
+ { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, },
+ { CS4271_VOLA, 0, },
+ { CS4271_VOLB, 0, },
+ { CS4271_ADCCTL, 0, },
+ { CS4271_MODE2, 0, },
};
+static bool cs4271_volatile_reg(struct device *dev, unsigned int reg)
+{
+ return reg == CS4271_CHIPID;
+}
+
struct cs4271_private {
/* SND_SOC_I2C or SND_SOC_SPI */
- enum snd_soc_control_type bus_type;
unsigned int mclk;
bool master;
bool deemph;
+ struct regmap *regmap;
/* Current sample rate for de-emphasis control */
int rate;
/* GPIO driving Reset pin, if any */
@@ -210,14 +212,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai,
switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_LEFT_J:
val |= CS4271_MODE1_DAC_DIF_LJ;
- ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+ ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL,
CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ);
if (ret < 0)
return ret;
break;
case SND_SOC_DAIFMT_I2S:
val |= CS4271_MODE1_DAC_DIF_I2S;
- ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+ ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL,
CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S);
if (ret < 0)
return ret;
@@ -227,7 +229,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- ret = snd_soc_update_bits(codec, CS4271_MODE1,
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1,
CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val);
if (ret < 0)
return ret;
@@ -252,7 +254,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec)
val <<= 4;
}
- ret = snd_soc_update_bits(codec, CS4271_DACCTL,
+ ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL,
CS4271_DACCTL_DEM_MASK, val);
if (ret < 0)
return ret;
@@ -341,14 +343,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
!dai->capture_active) ||
(substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
!dai->playback_active)) {
- ret = snd_soc_update_bits(codec, CS4271_MODE2,
- CS4271_MODE2_PDN,
- CS4271_MODE2_PDN);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_PDN,
+ CS4271_MODE2_PDN);
if (ret < 0)
return ret;
- ret = snd_soc_update_bits(codec, CS4271_MODE2,
- CS4271_MODE2_PDN, 0);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_PDN, 0);
if (ret < 0)
return ret;
}
@@ -378,7 +380,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
val |= cs4271_clk_tab[i].ratio_mask;
- ret = snd_soc_update_bits(codec, CS4271_MODE1,
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1,
CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);
if (ret < 0)
return ret;
@@ -386,22 +388,29 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
return cs4271_set_deemph(codec);
}
-static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
+static int cs4271_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
{
struct snd_soc_codec *codec = dai->codec;
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
int ret;
int val_a = 0;
int val_b = 0;
+ if (stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute) {
val_a = CS4271_VOLA_MUTE;
val_b = CS4271_VOLB_MUTE;
}
- ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA,
+ CS4271_VOLA_MUTE, val_a);
if (ret < 0)
return ret;
- ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b);
+
+ ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB,
+ CS4271_VOLB_MUTE, val_b);
if (ret < 0)
return ret;
@@ -436,7 +445,7 @@ static const struct snd_soc_dai_ops cs4271_dai_ops = {
.hw_params = cs4271_hw_params,
.set_sysclk = cs4271_set_dai_sysclk,
.set_fmt = cs4271_set_dai_fmt,
- .digital_mute = cs4271_digital_mute,
+ .mute_stream = cs4271_mute_stream,
};
static struct snd_soc_dai_driver cs4271_dai = {
@@ -463,25 +472,33 @@ static struct snd_soc_dai_driver cs4271_dai = {
static int cs4271_soc_suspend(struct snd_soc_codec *codec)
{
int ret;
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
/* Set power-down bit */
- ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN,
- CS4271_MODE2_PDN);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_PDN, CS4271_MODE2_PDN);
if (ret < 0)
return ret;
+
return 0;
}
static int cs4271_soc_resume(struct snd_soc_codec *codec)
{
int ret;
+ struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
/* Restore codec state */
- ret = snd_soc_cache_sync(codec);
+ ret = regcache_sync(cs4271->regmap);
if (ret < 0)
return ret;
+
/* then disable the power-down bit */
- ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_PDN, 0);
if (ret < 0)
return ret;
+
return 0;
}
#else
@@ -542,40 +559,22 @@ static int cs4271_probe(struct snd_soc_codec *codec)
cs4271->gpio_nreset = gpio_nreset;
- /*
- * In case of I2C, chip address specified in board data.
- * So cache IO operations use 8 bit codec register address.
- * In case of SPI, chip address and register address
- * passed together as 16 bit value.
- * Anyway, register address is masked with 0xFF inside
- * soc-cache code.
- */
- if (cs4271->bus_type == SND_SOC_SPI)
- ret = snd_soc_codec_set_cache_io(codec, 16, 8,
- cs4271->bus_type);
- else
- ret = snd_soc_codec_set_cache_io(codec, 8, 8,
- cs4271->bus_type);
- if (ret) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_update_bits(codec, CS4271_MODE2,
- CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
- CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
if (ret < 0)
return ret;
- ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+ ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_PDN, 0);
if (ret < 0)
return ret;
/* Power-up sequence requires 85 uS */
udelay(85);
if (amutec_eq_bmutec)
- snd_soc_update_bits(codec, CS4271_MODE2,
- CS4271_MODE2_MUTECAEQUB,
- CS4271_MODE2_MUTECAEQUB);
+ regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+ CS4271_MODE2_MUTECAEQUB,
+ CS4271_MODE2_MUTECAEQUB);
return snd_soc_add_codec_controls(codec, cs4271_snd_controls,
ARRAY_SIZE(cs4271_snd_controls));
@@ -597,13 +596,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
.remove = cs4271_remove,
.suspend = cs4271_soc_suspend,
.resume = cs4271_soc_resume,
- .reg_cache_default = cs4271_dflt_reg,
- .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg),
- .reg_word_size = sizeof(cs4271_dflt_reg[0]),
- .compress_type = SND_SOC_FLAT_COMPRESSION,
};
#if defined(CONFIG_SPI_MASTER)
+
+static const struct regmap_config cs4271_spi_regmap = {
+ .reg_bits = 16,
+ .val_bits = 8,
+ .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),
+ .cache_type = REGCACHE_RBTREE,
+
+ .volatile_reg = cs4271_volatile_reg,
+};
+
static int cs4271_spi_probe(struct spi_device *spi)
{
struct cs4271_private *cs4271;
@@ -613,7 +623,9 @@ static int cs4271_spi_probe(struct spi_device *spi)
return -ENOMEM;
spi_set_drvdata(spi, cs4271);
- cs4271->bus_type = SND_SOC_SPI;
+ 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);
@@ -643,6 +655,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
+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)
{
@@ -653,7 +677,9 @@ static int cs4271_i2c_probe(struct i2c_client *client,
return -ENOMEM;
i2c_set_clientdata(client, cs4271);
- cs4271->bus_type = SND_SOC_I2C;
+ cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap);
+ if (IS_ERR(cs4271->regmap))
+ return PTR_ERR(cs4271->regmap);
return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271,
&cs4271_dai, 1);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 6361dab48bd1..3b20c86cdb01 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1180,7 +1180,11 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
priv->config[id].mmcc &= 0xC0;
priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
priv->config[id].spc &= 0xFC;
- priv->config[id].spc |= MCK_SCLK_MCLK;
+ /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */
+ if (priv->mclk >= 6400000)
+ priv->config[id].spc |= MCK_SCLK_64FS;
+ else
+ priv->config[id].spc |= MCK_SCLK_MCLK;
} else {
/* CS42L73 Slave */
priv->config[id].spc &= 0xFC;
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index a4c16fd70f77..3eeada57e87d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = {
2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
};
+static const unsigned int max98088_hp_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0),
+ 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0),
+ 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0),
+ 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0),
+ 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0),
+};
+
+static const unsigned int max98088_spk_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0),
+ 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0),
+ 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0),
+ 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0),
+ 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0),
+};
+
static const struct snd_kcontrol_new max98088_snd_controls[] = {
- SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L,
- M98088_REG_3A_LVL_HP_R, 0, 31, 0),
- SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L,
- M98088_REG_3E_LVL_SPK_R, 0, 31, 0),
- SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L,
- M98088_REG_3C_LVL_REC_R, 0, 31, 0),
+ SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L,
+ M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv),
+ SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L,
+ M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv),
+ SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L,
+ M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv),
SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L,
M98088_REG_3A_LVL_HP_R, 7, 1, 1),
@@ -2006,7 +2024,7 @@ static int max98088_probe(struct snd_soc_codec *codec)
ret);
goto err_access;
}
- dev_info(codec->dev, "revision %c\n", ret + 'A');
+ dev_info(codec->dev, "revision %c\n", ret - 0x40 + 'A');
snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index fc176044994d..ce0d36412c97 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -23,8 +23,6 @@
#include <sound/max98090.h>
#include "max98090.h"
-#include <linux/version.h>
-
#define DEBUG
#define EXTMIC_METHOD
#define EXTMIC_METHOD_TEST
@@ -509,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol,
return 0;
}
-static const char * max98090_perf_pwr_text[] =
+static const char *max98090_perf_pwr_text[] =
{ "High Performance", "Low Power" };
-static const char * max98090_pwr_perf_text[] =
+static const char *max98090_pwr_perf_text[] =
{ "Low Power", "High Performance" };
static const struct soc_enum max98090_vcmbandgap_enum =
SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT,
ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
-static const char * max98090_osr128_text[] = { "64*fs", "128*fs" };
+static const char *max98090_osr128_text[] = { "64*fs", "128*fs" };
static const struct soc_enum max98090_osr128_enum =
SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT,
@@ -535,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum =
M98090_FLT_DMIC34MODE_SHIFT,
ARRAY_SIZE(max98090_mode_text), max98090_mode_text);
-static const char * max98090_drcatk_text[] =
+static const char *max98090_drcatk_text[] =
{ "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" };
static const struct soc_enum max98090_drcatk_enum =
SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT,
ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text);
-static const char * max98090_drcrls_text[] =
+static const char *max98090_drcrls_text[] =
{ "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" };
static const struct soc_enum max98090_drcrls_enum =
SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT,
ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text);
-static const char * max98090_alccmp_text[] =
+static const char *max98090_alccmp_text[] =
{ "1:1", "1:1.5", "1:2", "1:4", "1:INF" };
static const struct soc_enum max98090_alccmp_enum =
SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT,
ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text);
-static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" };
+static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" };
static const struct soc_enum max98090_drcexp_enum =
SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT,
@@ -859,7 +857,7 @@ static const struct soc_enum mic2_mux_enum =
static const struct snd_kcontrol_new max98090_mic2_mux =
SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum);
-static const char * max98090_micpre_text[] = { "Off", "On" };
+static const char *max98090_micpre_text[] = { "Off", "On" };
static const struct soc_enum max98090_pa1en_enum =
SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT,
@@ -1703,9 +1701,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai,
* seen for the case of TDM mode. The remaining cases have
* normal logic.
*/
- if (max98090->tdm_slots > 1) {
+ if (max98090->tdm_slots > 1)
regval ^= M98090_BCI_MASK;
- }
snd_soc_write(codec,
M98090_REG_INTERFACE_FORMAT, regval);
@@ -2059,17 +2056,14 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
if (!active)
return IRQ_NONE;
- if (active & M98090_CLD_MASK) {
+ if (active & M98090_CLD_MASK)
dev_err(codec->dev, "M98090_CLD_MASK\n");
- }
- if (active & M98090_SLD_MASK) {
+ if (active & M98090_SLD_MASK)
dev_dbg(codec->dev, "M98090_SLD_MASK\n");
- }
- if (active & M98090_ULK_MASK) {
+ if (active & M98090_ULK_MASK)
dev_err(codec->dev, "M98090_ULK_MASK\n");
- }
if (active & M98090_JDET_MASK) {
dev_dbg(codec->dev, "M98090_JDET_MASK\n");
@@ -2080,13 +2074,11 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
msecs_to_jiffies(100));
}
- if (active & M98090_DRCACT_MASK) {
+ if (active & M98090_DRCACT_MASK)
dev_dbg(codec->dev, "M98090_DRCACT_MASK\n");
- }
- if (active & M98090_DRCCLP_MASK) {
+ if (active & M98090_DRCCLP_MASK)
dev_err(codec->dev, "M98090_DRCCLP_MASK\n");
- }
return IRQ_HANDLED;
}
@@ -2324,7 +2316,7 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
max98090->pdata = i2c->dev.platform_data;
max98090->irq = i2c->irq;
- max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap);
+ max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap);
if (IS_ERR(max98090->regmap)) {
ret = PTR_ERR(max98090->regmap);
dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
@@ -2334,18 +2326,13 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_max98090, max98090_dai,
ARRAY_SIZE(max98090_dai));
- if (ret < 0)
- regmap_exit(max98090->regmap);
-
err_enable:
return ret;
}
static int max98090_i2c_remove(struct i2c_client *client)
{
- struct max98090_priv *max98090 = dev_get_drvdata(&client->dev);
snd_soc_unregister_codec(&client->dev);
- regmap_exit(max98090->regmap);
return 0;
}
@@ -2369,7 +2356,7 @@ static int max98090_runtime_suspend(struct device *dev)
return 0;
}
-static struct dev_pm_ops max98090_pm = {
+static const struct dev_pm_ops max98090_pm = {
SET_RUNTIME_PM_OPS(max98090_runtime_suspend,
max98090_runtime_resume, NULL)
};
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index 566ea3256e2d..721587c9cd84 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -1,3 +1,22 @@
+/*
+ * sound/soc/codecs/si476x.c -- Codec driver for SI476X chips
+ *
+ * Copyright (C) 2012 Innovative Converged Devices(ICD)
+ * Copyright (C) 2013 Andrey Smirnov
+ *
+ * Author: Andrey Smirnov <andrew.smirnov@gmail.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/slab.h>
#include <sound/pcm.h>
@@ -45,13 +64,23 @@ static unsigned int si476x_codec_read(struct snd_soc_codec *codec,
unsigned int reg)
{
int err;
+ unsigned int val;
struct si476x_core *core = codec->control_data;
si476x_core_lock(core);
- err = si476x_core_cmd_get_property(core, reg);
+ if (!si476x_core_is_powered_up(core))
+ regcache_cache_only(core->regmap, true);
+
+ err = regmap_read(core->regmap, reg, &val);
+
+ if (!si476x_core_is_powered_up(core))
+ regcache_cache_only(core->regmap, false);
si476x_core_unlock(core);
- return err;
+ if (err < 0)
+ return err;
+
+ return val;
}
static int si476x_codec_write(struct snd_soc_codec *codec,
@@ -61,7 +90,13 @@ static int si476x_codec_write(struct snd_soc_codec *codec,
struct si476x_core *core = codec->control_data;
si476x_core_lock(core);
- err = si476x_core_cmd_set_property(core, reg, val);
+ if (!si476x_core_is_powered_up(core))
+ regcache_cache_only(core->regmap, true);
+
+ err = regmap_write(core->regmap, reg, val);
+
+ if (!si476x_core_is_powered_up(core))
+ regcache_cache_only(core->regmap, false);
si476x_core_unlock(core);
return err;
@@ -140,7 +175,7 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
dev_err(codec_dai->codec->dev, "Failed to set output format\n");
return err;
}
-
+
return 0;
}
@@ -182,7 +217,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT,
SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK,
- (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) |
+ (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) |
(width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT));
if (err < 0) {
dev_err(dai->codec->dev, "Failed to set output width\n");
@@ -251,6 +286,6 @@ static struct platform_driver si476x_platform_driver = {
};
module_platform_driver(si476x_platform_driver);
-MODULE_AUTHOR("Andrey Smirnov <andrey.smirnov@convergeddevices.net>");
+MODULE_AUTHOR("Andrey Smirnov <andrew.smirnov@gmail.com>");
MODULE_DESCRIPTION("ASoC Si4761/64 codec driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
new file mode 100644
index 000000000000..d447c4aa1d5e
--- /dev/null
+++ b/sound/soc/codecs/tas5086.c
@@ -0,0 +1,591 @@
+/*
+ * TAS5086 ASoC codec driver
+ *
+ * Copyright (c) 2013 Daniel Mack <zonque@gmail.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; 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.
+ *
+ * TODO:
+ * - implement DAPM and input muxing
+ * - implement modulation limit
+ * - implement non-default PWM start
+ *
+ * Note that this chip has a very unusual register layout, specifically
+ * because the registers are of unequal size, and multi-byte registers
+ * require bulk writes to take effect. Regmap does not support that kind
+ * of devices.
+ *
+ * Currently, the driver does not touch any of the registers >= 0x20, so
+ * it doesn't matter because the entire map can be accessed as 8-bit
+ * array. In case more features will be added in the future
+ * that require access to higher registers, the entire regmap H/W I/O
+ * routines have to be open-coded.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <linux/of_device.h>
+#include <linux/of_gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/tas5086.h>
+
+#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE)
+
+#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+ SNDRV_PCM_RATE_192000)
+
+/*
+ * TAS5086 registers
+ */
+#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */
+#define TAS5086_CLOCK_RATE(val) (val << 5)
+#define TAS5086_CLOCK_RATE_MASK (0x7 << 5)
+#define TAS5086_CLOCK_RATIO(val) (val << 2)
+#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2)
+#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1)
+#define TAS5086_CLOCK_VALID (1 << 0)
+
+#define TAS5086_DEEMPH_MASK 0x03
+#define TAS5086_SOFT_MUTE_ALL 0x3f
+
+#define TAS5086_DEV_ID 0x01 /* Device ID register */
+#define TAS5086_ERROR_STATUS 0x02 /* Error status register */
+#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */
+#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */
+#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */
+#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */
+#define TAS5086_MASTER_VOL 0x07 /* Master volume */
+#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */
+#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */
+#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */
+#define TAS5086_PWM_START 0x18 /* PWM start register */
+#define TAS5086_SURROUND 0x19 /* Surround register */
+#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */
+#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */
+#define TAS5086_BKNDERR 0x1c
+
+/*
+ * Default TAS5086 power-up configuration
+ */
+static const struct reg_default tas5086_reg_defaults[] = {
+ { 0x00, 0x6c },
+ { 0x01, 0x03 },
+ { 0x02, 0x00 },
+ { 0x03, 0xa0 },
+ { 0x04, 0x05 },
+ { 0x05, 0x60 },
+ { 0x06, 0x00 },
+ { 0x07, 0xff },
+ { 0x08, 0x30 },
+ { 0x09, 0x30 },
+ { 0x0a, 0x30 },
+ { 0x0b, 0x30 },
+ { 0x0c, 0x30 },
+ { 0x0d, 0x30 },
+ { 0x0e, 0xb1 },
+ { 0x0f, 0x00 },
+ { 0x10, 0x02 },
+ { 0x11, 0x00 },
+ { 0x12, 0x00 },
+ { 0x13, 0x00 },
+ { 0x14, 0x00 },
+ { 0x15, 0x00 },
+ { 0x16, 0x00 },
+ { 0x17, 0x00 },
+ { 0x18, 0x3f },
+ { 0x19, 0x00 },
+ { 0x1a, 0x18 },
+ { 0x1b, 0x82 },
+ { 0x1c, 0x05 },
+};
+
+static bool tas5086_accessible_reg(struct device *dev, unsigned int reg)
+{
+ return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17));
+}
+
+static bool tas5086_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case TAS5086_DEV_ID:
+ case TAS5086_ERROR_STATUS:
+ return true;
+ }
+
+ return false;
+}
+
+static bool tas5086_writeable_reg(struct device *dev, unsigned int reg)
+{
+ return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID);
+}
+
+struct tas5086_private {
+ struct regmap *regmap;
+ unsigned int mclk, sclk;
+ unsigned int format;
+ bool deemph;
+ /* Current sample rate for de-emphasis control */
+ int rate;
+ /* GPIO driving Reset pin, if any */
+ int gpio_nreset;
+};
+
+static int tas5086_deemph[] = { 0, 32000, 44100, 48000 };
+
+static int tas5086_set_deemph(struct snd_soc_codec *codec)
+{
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+ int i, val = 0;
+
+ if (priv->deemph)
+ for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++)
+ if (tas5086_deemph[i] == priv->rate)
+ val = i;
+
+ return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1,
+ TAS5086_DEEMPH_MASK, val);
+}
+
+static int tas5086_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = priv->deemph;
+
+ return 0;
+}
+
+static int tas5086_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ priv->deemph = ucontrol->value.enumerated.item[0];
+
+ return tas5086_set_deemph(codec);
+}
+
+
+static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case TAS5086_CLK_IDX_MCLK:
+ priv->mclk = freq;
+ break;
+ case TAS5086_CLK_IDX_SCLK:
+ priv->sclk = freq;
+ break;
+ }
+
+ return 0;
+}
+
+static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ /* The TAS5086 can only be slave to all clocks */
+ if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+ dev_err(codec->dev, "Invalid clocking mode\n");
+ return -EINVAL;
+ }
+
+ /* we need to refer to the data format from hw_params() */
+ priv->format = format;
+
+ return 0;
+}
+
+static const int tas5086_sample_rates[] = {
+ 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+
+static const int tas5086_ratios[] = {
+ 64, 128, 192, 256, 384, 512
+};
+
+static int index_in_array(const int *array, int len, int needle)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ if (array[i] == needle)
+ return i;
+
+ return -ENOENT;
+}
+
+static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+ int val;
+ int ret;
+
+ priv->rate = params_rate(params);
+
+ /* Look up the sample rate and refer to the offset in the list */
+ val = index_in_array(tas5086_sample_rates,
+ ARRAY_SIZE(tas5086_sample_rates), priv->rate);
+
+ if (val < 0) {
+ dev_err(codec->dev, "Invalid sample rate\n");
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+ TAS5086_CLOCK_RATE_MASK,
+ TAS5086_CLOCK_RATE(val));
+ if (ret < 0)
+ return ret;
+
+ /* MCLK / Fs ratio */
+ val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios),
+ priv->mclk / priv->rate);
+ if (val < 0) {
+ dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n");
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+ TAS5086_CLOCK_RATIO_MASK,
+ TAS5086_CLOCK_RATIO(val));
+ if (ret < 0)
+ return ret;
+
+
+ ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+ TAS5086_CLOCK_SCLK_RATIO_48,
+ (priv->sclk == 48 * priv->rate) ?
+ TAS5086_CLOCK_SCLK_RATIO_48 : 0);
+ if (ret < 0)
+ return ret;
+
+ /*
+ * The chip has a very unituitive register mapping and muxes information
+ * about data format and sample depth into the same register, but not on
+ * a logical bit-boundary. Hence, we have to refer to the format passed
+ * in the set_dai_fmt() callback and set up everything from here.
+ *
+ * First, determine the 'base' value, using the format ...
+ */
+ switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ val = 0x00;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ val = 0x03;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ val = 0x06;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid DAI format\n");
+ return -EINVAL;
+ }
+
+ /* ... then add the offset for the sample bit depth. */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val += 0;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val += 1;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ val += 2;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid bit width\n");
+ return -EINVAL;
+ };
+
+ ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val);
+ if (ret < 0)
+ return ret;
+
+ /* clock is considered valid now */
+ ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+ TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID);
+ if (ret < 0)
+ return ret;
+
+ return tas5086_set_deemph(codec);
+}
+
+static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int val = 0;
+
+ if (mute)
+ val = TAS5086_SOFT_MUTE_ALL;
+
+ return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val);
+}
+
+/* TAS5086 controls */
+static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1);
+
+static const struct snd_kcontrol_new tas5086_controls[] = {
+ SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL,
+ 0, 0xff, 1, tas5086_dac_tlv),
+ SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume",
+ TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1),
+ 0, 0xff, 1, tas5086_dac_tlv),
+ SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume",
+ TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3),
+ 0, 0xff, 1, tas5086_dac_tlv),
+ SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume",
+ TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5),
+ 0, 0xff, 1, tas5086_dac_tlv),
+ SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0,
+ tas5086_get_deemph, tas5086_put_deemph),
+};
+
+static const struct snd_soc_dai_ops tas5086_dai_ops = {
+ .hw_params = tas5086_hw_params,
+ .set_sysclk = tas5086_set_dai_sysclk,
+ .set_fmt = tas5086_set_dai_fmt,
+ .mute_stream = tas5086_mute_stream,
+};
+
+static struct snd_soc_dai_driver tas5086_dai = {
+ .name = "tas5086-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 6,
+ .rates = TAS5086_PCM_RATES,
+ .formats = TAS5086_PCM_FORMATS,
+ },
+ .ops = &tas5086_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int tas5086_soc_resume(struct snd_soc_codec *codec)
+{
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ /* Restore codec state */
+ return regcache_sync(priv->regmap);
+}
+#else
+#define tas5086_soc_resume NULL
+#endif /* CONFIG_PM */
+
+#ifdef CONFIG_OF
+static const struct of_device_id tas5086_dt_ids[] = {
+ { .compatible = "ti,tas5086", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, tas5086_dt_ids);
+#endif
+
+/* charge period values in microseconds */
+static const int tas5086_charge_period[] = {
+ 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200,
+ 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000,
+ 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000,
+};
+
+static int tas5086_probe(struct snd_soc_codec *codec)
+{
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+ int charge_period = 1300000; /* hardware default is 1300 ms */
+ int i, ret;
+
+ if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) {
+ struct device_node *of_node = codec->dev->of_node;
+ of_property_read_u32(of_node, "ti,charge-period", &charge_period);
+ }
+
+ /* lookup and set split-capacitor charge period */
+ if (charge_period == 0) {
+ regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0);
+ } else {
+ i = index_in_array(tas5086_charge_period,
+ ARRAY_SIZE(tas5086_charge_period),
+ charge_period);
+ if (i >= 0)
+ regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE,
+ i + 0x08);
+ else
+ dev_warn(codec->dev,
+ "Invalid split-cap charge period of %d ns.\n",
+ charge_period);
+ }
+
+ /* enable factory trim */
+ ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00);
+ if (ret < 0)
+ return ret;
+
+ /* start all channels */
+ ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20);
+ if (ret < 0)
+ return ret;
+
+ /* set master volume to 0 dB */
+ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30);
+ if (ret < 0)
+ return ret;
+
+ /* mute all channels for now */
+ ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE,
+ TAS5086_SOFT_MUTE_ALL);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int tas5086_remove(struct snd_soc_codec *codec)
+{
+ struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (gpio_is_valid(priv->gpio_nreset))
+ /* Set codec to the reset state */
+ gpio_set_value(priv->gpio_nreset, 0);
+
+ return 0;
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_tas5086 = {
+ .probe = tas5086_probe,
+ .remove = tas5086_remove,
+ .resume = tas5086_soc_resume,
+ .controls = tas5086_controls,
+ .num_controls = ARRAY_SIZE(tas5086_controls),
+};
+
+static const struct i2c_device_id tas5086_i2c_id[] = {
+ { "tas5086", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id);
+
+static const struct regmap_config tas5086_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = ARRAY_SIZE(tas5086_reg_defaults),
+ .reg_defaults = tas5086_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+ .volatile_reg = tas5086_volatile_reg,
+ .writeable_reg = tas5086_writeable_reg,
+ .readable_reg = tas5086_accessible_reg,
+};
+
+static int tas5086_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct tas5086_private *priv;
+ struct device *dev = &i2c->dev;
+ int gpio_nreset = -EINVAL;
+ int i, ret;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap);
+ if (IS_ERR(priv->regmap)) {
+ ret = PTR_ERR(priv->regmap);
+ dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret);
+ return ret;
+ }
+
+ i2c_set_clientdata(i2c, priv);
+
+ if (of_match_device(of_match_ptr(tas5086_dt_ids), dev)) {
+ struct device_node *of_node = dev->of_node;
+ gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0);
+ }
+
+ if (gpio_is_valid(gpio_nreset))
+ if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset"))
+ gpio_nreset = -EINVAL;
+
+ if (gpio_is_valid(gpio_nreset)) {
+ /* Reset codec - minimum assertion time is 400ns */
+ gpio_direction_output(gpio_nreset, 0);
+ udelay(1);
+ gpio_set_value(gpio_nreset, 1);
+
+ /* Codec needs ~15ms to wake up */
+ msleep(15);
+ }
+
+ priv->gpio_nreset = gpio_nreset;
+
+ /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */
+ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i);
+ if (ret < 0)
+ return ret;
+
+ if (i != 0x3) {
+ dev_err(dev,
+ "Failed to identify TAS5086 codec (got %02x)\n", i);
+ return -ENODEV;
+ }
+
+ return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086,
+ &tas5086_dai, 1);
+}
+
+static int tas5086_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+ return 0;
+}
+
+static struct i2c_driver tas5086_i2c_driver = {
+ .driver = {
+ .name = "tas5086",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(tas5086_dt_ids),
+ },
+ .id_table = tas5086_i2c_id,
+ .probe = tas5086_i2c_probe,
+ .remove = tas5086_i2c_remove,
+};
+
+module_i2c_driver(tas5086_i2c_driver);
+
+MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>");
+MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index ad2fee4bb4cd..8df2b6e1a1a6 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -342,7 +342,7 @@ static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len)
data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i]));
}
-static int wm0010_firmware_load(char *name, struct snd_soc_codec *codec)
+static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
{
struct spi_device *spi = to_spi_device(codec->dev);
struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
@@ -361,8 +361,8 @@ static int wm0010_firmware_load(char *name, struct snd_soc_codec *codec)
ret = request_firmware(&fw, name, codec->dev);
if (ret != 0) {
- dev_err(codec->dev, "Failed to request application: %d\n",
- ret);
+ dev_err(codec->dev, "Failed to request application(%s): %d\n",
+ name, ret);
return ret;
}
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index f2ac38b61a1b..7fefd766b582 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -761,6 +761,8 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
case WM2000_REG_SYS_CTL2:
case WM2000_REG_ANC_STAT:
case WM2000_REG_IF_CTL:
+ case WM2000_REG_ANA_MIC_CTL:
+ case WM2000_REG_SPK_CTL:
return true;
default:
return false;
@@ -771,7 +773,7 @@ static const struct regmap_config wm2000_regmap = {
.reg_bits = 16,
.val_bits = 8,
- .max_register = WM2000_REG_IF_CTL,
+ .max_register = WM2000_REG_SPK_CTL,
.readable_reg = wm2000_readable_reg,
};
diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h
index fb812cd9e77d..3870c0e1d246 100644
--- a/sound/soc/codecs/wm2000.h
+++ b/sound/soc/codecs/wm2000.h
@@ -30,6 +30,8 @@
#define WM2000_REG_SYS_CTL2 0xf004
#define WM2000_REG_ANC_STAT 0xf005
#define WM2000_REG_IF_CTL 0xf006
+#define WM2000_REG_ANA_MIC_CTL 0xf028
+#define WM2000_REG_SPK_CTL 0xf034
/* SPEECH_CLARITY */
#define WM2000_SPEECH_CLARITY 0x01
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index ddc98f02ecbd..57ba315d0c84 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1565,7 +1565,7 @@ static int wm2200_probe(struct snd_soc_codec *codec)
return ret;
}
- ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2);
+ ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2);
if (ret != 0)
return ret;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 15bc31f1abb1..e895d3939eef 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,9 +36,6 @@
struct wm5102_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
-
- unsigned int spk_ena:2;
- unsigned int spk_ena_pending:1;
};
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -615,6 +612,26 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
return 0;
}
+static const char *wm5102_osr_text[] = {
+ "Low power", "Normal", "High performance",
+};
+
+static const unsigned int wm5102_osr_val[] = {
+ 0x0, 0x3, 0x5,
+};
+
+static const struct soc_enum wm5102_hpout_osr[] = {
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
+ ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+ wm5102_osr_text, wm5102_osr_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L,
+ ARIZONA_OUT2_OSR_SHIFT, 0x7, 3,
+ wm5102_osr_text, wm5102_osr_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
+ ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+ wm5102_osr_text, wm5102_osr_val),
+};
+
#define WM5102_NG_SRC(name, base) \
SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \
SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \
@@ -745,6 +762,9 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
+SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
+
ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -761,6 +781,8 @@ ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+ ARIZONA_OUT4_OSR_SHIFT, 1, 0),
SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
ARIZONA_OUT5_OSR_SHIFT, 1, 0),
@@ -790,6 +812,10 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
0xbf, 0, digital_tlv),
+SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]),
+SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]),
+SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]),
+
SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp),
SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp),
@@ -828,47 +854,6 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
};
-static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol,
- int event)
-{
- struct snd_soc_codec *codec = w->codec;
- struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
-
- if (arizona->rev < 1)
- return 0;
-
- switch (event) {
- case SND_SOC_DAPM_PRE_PMU:
- if (!wm5102->spk_ena) {
- snd_soc_write(codec, 0x4f5, 0x25a);
- wm5102->spk_ena_pending = true;
- }
- break;
- case SND_SOC_DAPM_POST_PMU:
- if (wm5102->spk_ena_pending) {
- msleep(75);
- snd_soc_write(codec, 0x4f5, 0xda);
- wm5102->spk_ena_pending = false;
- wm5102->spk_ena++;
- }
- break;
- case SND_SOC_DAPM_PRE_PMD:
- wm5102->spk_ena--;
- if (!wm5102->spk_ena)
- snd_soc_write(codec, 0x4f5, 0x25a);
- break;
- case SND_SOC_DAPM_POST_PMD:
- if (!wm5102->spk_ena)
- snd_soc_write(codec, 0x4f5, 0x0da);
- break;
- }
-
- return 0;
-}
-
-
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -984,22 +969,28 @@ SND_SOC_DAPM_INPUT("IN3R"),
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
@@ -1146,12 +1137,6 @@ SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
-SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
-SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
@@ -1494,6 +1479,12 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout);
case WM5102_FLL2:
return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout);
+ case WM5102_FLL1_REFCLK:
+ return arizona_set_fll_refclk(&wm5102->fll[0], source, Fref,
+ Fout);
+ case WM5102_FLL2_REFCLK:
+ return arizona_set_fll_refclk(&wm5102->fll[1], source, Fref,
+ Fout);
default:
return -EINVAL;
}
@@ -1581,10 +1572,12 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec)
if (ret != 0)
return ret;
- ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 1);
+ ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2);
if (ret != 0)
return ret;
+ arizona_init_spk(codec);
+
snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
priv->core.arizona->dapm = &codec->dapm;
@@ -1604,13 +1597,6 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec)
#define WM5102_DIG_VU 0x0200
static unsigned int wm5102_digital_vu[] = {
- ARIZONA_ADC_DIGITAL_VOLUME_1L,
- ARIZONA_ADC_DIGITAL_VOLUME_1R,
- ARIZONA_ADC_DIGITAL_VOLUME_2L,
- ARIZONA_ADC_DIGITAL_VOLUME_2R,
- ARIZONA_ADC_DIGITAL_VOLUME_3L,
- ARIZONA_ADC_DIGITAL_VOLUME_3R,
-
ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R,
ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -1653,6 +1639,7 @@ static int wm5102_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, wm5102);
wm5102->core.arizona = arizona;
+ wm5102->core.num_inputs = 6;
wm5102->core.adsp[0].part = "wm5102";
wm5102->core.adsp[0].num = 1;
@@ -1677,6 +1664,12 @@ static int wm5102_probe(struct platform_device *pdev)
ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
&wm5102->fll[1]);
+ /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */
+ regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2,
+ ARIZONA_SAMPLE_RATE_2_MASK, 0x11);
+ regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3,
+ ARIZONA_SAMPLE_RATE_3_MASK, 0x12);
+
for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++)
arizona_init_dai(&wm5102->core, i);
diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h
index d30477f3070c..adb38040f661 100644
--- a/sound/soc/codecs/wm5102.h
+++ b/sound/soc/codecs/wm5102.h
@@ -15,7 +15,9 @@
#include "arizona.h"
-#define WM5102_FLL1 1
-#define WM5102_FLL2 2
+#define WM5102_FLL1 1
+#define WM5102_FLL2 2
+#define WM5102_FLL1_REFCLK 3
+#define WM5102_FLL2_REFCLK 4
#endif
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 7841b42a819c..731884e04776 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -416,28 +416,36 @@ SND_SOC_DAPM_INPUT("IN4R"),
SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT,
0, NULL, 0, arizona_in_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
@@ -569,12 +577,6 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
SND_SOC_DAPM_PGA_E("OUT3R", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
-SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
-SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
@@ -880,6 +882,12 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout);
case WM5110_FLL2:
return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout);
+ case WM5110_FLL1_REFCLK:
+ return arizona_set_fll_refclk(&wm5110->fll[0], source, Fref,
+ Fout);
+ case WM5110_FLL2_REFCLK:
+ return arizona_set_fll_refclk(&wm5110->fll[1], source, Fref,
+ Fout);
default:
return -EINVAL;
}
@@ -987,15 +995,6 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec)
#define WM5110_DIG_VU 0x0200
static unsigned int wm5110_digital_vu[] = {
- ARIZONA_ADC_DIGITAL_VOLUME_1L,
- ARIZONA_ADC_DIGITAL_VOLUME_1R,
- ARIZONA_ADC_DIGITAL_VOLUME_2L,
- ARIZONA_ADC_DIGITAL_VOLUME_2R,
- ARIZONA_ADC_DIGITAL_VOLUME_3L,
- ARIZONA_ADC_DIGITAL_VOLUME_3R,
- ARIZONA_ADC_DIGITAL_VOLUME_4L,
- ARIZONA_ADC_DIGITAL_VOLUME_4R,
-
ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R,
ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -1040,6 +1039,7 @@ static int wm5110_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, wm5110);
wm5110->core.arizona = arizona;
+ wm5110->core.num_inputs = 8;
for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++)
wm5110->fll[i].vco_mult = 3;
diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h
index 75e9351ccab0..e6c0cd4235c5 100644
--- a/sound/soc/codecs/wm5110.h
+++ b/sound/soc/codecs/wm5110.h
@@ -15,7 +15,9 @@
#include "arizona.h"
-#define WM5110_FLL1 1
-#define WM5110_FLL2 2
+#define WM5110_FLL1 1
+#define WM5110_FLL2 2
+#define WM5110_FLL1_REFCLK 3
+#define WM5110_FLL2_REFCLK 4
#endif
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index f8a31ad0b203..9d88437cdcd1 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -478,6 +478,8 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
/* ALSA can only do steps of .01dB */
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0);
+
static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
@@ -698,6 +700,8 @@ SOC_ENUM("DAC Mute Mode", mute_mode),
SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0),
SOC_ENUM("DAC Companding Mode", dac_companding),
SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0),
+SOC_SINGLE_TLV("DAC Boost Volume", WM8903_AUDIO_INTERFACE_0, 9, 3, 0,
+ dac_boost_tlv),
SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
wm8903_get_deemph, wm8903_put_deemph),
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index a64b93425ae3..0a4ffdd1d2a7 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -204,6 +204,7 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
static const struct snd_kcontrol_new wm8960_snd_controls[] = {
SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
@@ -213,6 +214,15 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
7, 1, 0),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
+ WM8960_INBMIX1, 4, 7, 0, boost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
+ WM8960_INBMIX1, 1, 7, 0, boost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
+ WM8960_INBMIX2, 4, 7, 0, boost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
+ WM8960_INBMIX2, 1, 7, 0, boost_tlv),
+
SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
0, 255, 0, dac_tlv),
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index c9bd445c4976..14094f558e03 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2209,7 +2209,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
vmid_reference(codec);
break;
case WM8958:
- if (wm8994->revision < 1)
+ if (control->revision < 1)
vmid_reference(codec);
break;
default:
@@ -2244,7 +2244,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
vmid_dereference(codec);
break;
case WM8958:
- if (wm8994->revision < 1)
+ if (control->revision < 1)
vmid_dereference(codec);
break;
default:
@@ -2268,10 +2268,26 @@ out:
*/
if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) {
dev_dbg(codec->dev, "Configuring AIFs for 128fs\n");
+
+ wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE)
+ & WM8994_AIF1CLK_RATE_MASK;
+ wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE)
+ & WM8994_AIF1CLK_RATE_MASK;
+
snd_soc_update_bits(codec, WM8994_AIF1_RATE,
WM8994_AIF1CLK_RATE_MASK, 0x1);
snd_soc_update_bits(codec, WM8994_AIF2_RATE,
WM8994_AIF2CLK_RATE_MASK, 0x1);
+ } else if (wm8994->aifdiv[0]) {
+ snd_soc_update_bits(codec, WM8994_AIF1_RATE,
+ WM8994_AIF1CLK_RATE_MASK,
+ wm8994->aifdiv[0]);
+ snd_soc_update_bits(codec, WM8994_AIF2_RATE,
+ WM8994_AIF2CLK_RATE_MASK,
+ wm8994->aifdiv[1]);
+
+ wm8994->aifdiv[0] = 0;
+ wm8994->aifdiv[1] = 0;
}
return 0;
@@ -2368,10 +2384,26 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
*/
if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) {
dev_dbg(codec->dev, "Configuring AIFs for 128fs\n");
+
+ wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE)
+ & WM8994_AIF1CLK_RATE_MASK;
+ wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE)
+ & WM8994_AIF1CLK_RATE_MASK;
+
snd_soc_update_bits(codec, WM8994_AIF1_RATE,
WM8994_AIF1CLK_RATE_MASK, 0x1);
snd_soc_update_bits(codec, WM8994_AIF2_RATE,
WM8994_AIF2CLK_RATE_MASK, 0x1);
+ } else if (wm8994->aifdiv[0]) {
+ snd_soc_update_bits(codec, WM8994_AIF1_RATE,
+ WM8994_AIF1CLK_RATE_MASK,
+ wm8994->aifdiv[0]);
+ snd_soc_update_bits(codec, WM8994_AIF2_RATE,
+ WM8994_AIF2CLK_RATE_MASK,
+ wm8994->aifdiv[1]);
+
+ wm8994->aifdiv[0] = 0;
+ wm8994->aifdiv[1] = 0;
}
return 0;
@@ -2411,7 +2443,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
switch (control->type) {
case WM8958:
- if (wm8994->revision == 0) {
+ if (control->revision == 0) {
/* Optimise performance for rev A */
snd_soc_update_bits(codec,
WM8958_CHARGE_PUMP_2,
@@ -2656,6 +2688,8 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct wm8994 *control = wm8994->wm8994;
+ struct wm8994_pdata *pdata = &control->pdata;
int aif1_reg;
int aif2_reg;
int bclk_reg;
@@ -2723,7 +2757,14 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
}
wm8994->channels[id] = params_channels(params);
- switch (params_channels(params)) {
+ if (pdata->max_channels_clocked[id] &&
+ wm8994->channels[id] > pdata->max_channels_clocked[id]) {
+ dev_dbg(dai->dev, "Constraining channels to %d from %d\n",
+ pdata->max_channels_clocked[id], wm8994->channels[id]);
+ wm8994->channels[id] = pdata->max_channels_clocked[id];
+ }
+
+ switch (wm8994->channels[id]) {
case 1:
case 2:
bclk_rate *= 2;
@@ -2745,7 +2786,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n",
dai->id, wm8994->aifclk[id], bclk_rate);
- if (params_channels(params) == 1 &&
+ if (wm8994->channels[id] == 1 &&
(snd_soc_read(codec, aif1_reg) & 0x18) == 0x18)
aif2 |= WM8994_AIF1_MONO;
@@ -3053,7 +3094,7 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec)
int i, ret;
unsigned int val, mask;
- if (wm8994->revision < 4) {
+ if (control->revision < 4) {
/* force a HW read */
ret = regmap_read(control->regmap,
WM8994_POWER_MANAGEMENT_5, &val);
@@ -3870,7 +3911,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
codec->dapm.idle_bias_off = 1;
/* Set revision-specific configuration */
- wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION);
switch (control->type) {
case WM8994:
/* Single ended line outputs should have VMID on. */
@@ -3878,7 +3918,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
!control->pdata.lineout2_diff)
codec->dapm.idle_bias_off = 0;
- switch (wm8994->revision) {
+ switch (control->revision) {
case 2:
case 3:
wm8994->hubs.dcs_codes_l = -5;
@@ -3897,7 +3937,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.hp_startup_mode = 1;
- switch (wm8994->revision) {
+ switch (control->revision) {
case 0:
break;
default:
@@ -4000,7 +4040,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
switch (control->type) {
case WM1811:
- if (control->cust_id > 1 || wm8994->revision > 1) {
+ if (control->cust_id > 1 || control->revision > 1) {
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_GPIO(6),
wm1811_jackdet_irq, "JACKDET",
@@ -4114,7 +4154,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
case WM8994:
snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
ARRAY_SIZE(wm8994_specific_dapm_widgets));
- if (wm8994->revision < 4) {
+ if (control->revision < 4) {
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
ARRAY_SIZE(wm8994_lateclk_revd_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
@@ -4135,7 +4175,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
ARRAY_SIZE(wm8958_snd_controls));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
- if (wm8994->revision < 1) {
+ if (control->revision < 1) {
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
ARRAY_SIZE(wm8994_lateclk_revd_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
@@ -4174,7 +4214,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
- if (wm8994->revision < 4) {
+ if (control->revision < 4) {
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
ARRAY_SIZE(wm8994_revd_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
@@ -4185,7 +4225,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
}
break;
case WM8958:
- if (wm8994->revision < 1) {
+ if (control->revision < 1) {
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 45f192702024..55ddf4d57d9b 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -79,6 +79,7 @@ struct wm8994_priv {
int sysclk_rate[2];
int mclk[2];
int aifclk[2];
+ int aifdiv[2];
int channels[2];
struct wm8994_fll_config fll[2], fll_suspend[2];
struct completion fll_locked[2];
@@ -146,8 +147,6 @@ struct wm8994_priv {
wm1811_mic_id_cb mic_id_cb;
void *mic_id_cb_data;
- int revision;
-
unsigned int aif1clk_enable:1;
unsigned int aif2clk_enable:1;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 9af1bddc4c62..3470b649c0b2 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -31,6 +31,7 @@
#include <linux/mfd/arizona/registers.h>
+#include "arizona.h"
#include "wm_adsp.h"
#define adsp_crit(_dsp, fmt, ...) \
@@ -193,17 +194,25 @@ static void wm_adsp_buf_free(struct list_head *list)
#define WM_ADSP_NUM_FW 4
+#define WM_ADSP_FW_MBC_VSS 0
+#define WM_ADSP_FW_TX 1
+#define WM_ADSP_FW_TX_SPK 2
+#define WM_ADSP_FW_RX_ANC 3
+
static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = {
- "MBC/VSS", "Tx", "Tx Speaker", "Rx ANC"
+ [WM_ADSP_FW_MBC_VSS] = "MBC/VSS",
+ [WM_ADSP_FW_TX] = "Tx",
+ [WM_ADSP_FW_TX_SPK] = "Tx Speaker",
+ [WM_ADSP_FW_RX_ANC] = "Rx ANC",
};
static struct {
const char *file;
} wm_adsp_fw[WM_ADSP_NUM_FW] = {
- { .file = "mbc-vss" },
- { .file = "tx" },
- { .file = "tx-spk" },
- { .file = "rx-anc" },
+ [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" },
+ [WM_ADSP_FW_TX] = { .file = "tx" },
+ [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" },
+ [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" },
};
static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol,
@@ -246,17 +255,52 @@ static const struct soc_enum wm_adsp_fw_enum[] = {
SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text),
};
-const struct snd_kcontrol_new wm_adsp_fw_controls[] = {
+const struct snd_kcontrol_new wm_adsp1_fw_controls[] = {
+ SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0],
+ wm_adsp_fw_get, wm_adsp_fw_put),
+ SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1],
+ wm_adsp_fw_get, wm_adsp_fw_put),
+ SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2],
+ wm_adsp_fw_get, wm_adsp_fw_put),
+};
+EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls);
+
+#if IS_ENABLED(CONFIG_SND_SOC_ARIZONA)
+static const struct soc_enum wm_adsp2_rate_enum[] = {
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1,
+ ARIZONA_DSP1_RATE_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1,
+ ARIZONA_DSP1_RATE_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1,
+ ARIZONA_DSP1_RATE_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+ SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1,
+ ARIZONA_DSP1_RATE_SHIFT, 0xf,
+ ARIZONA_RATE_ENUM_SIZE,
+ arizona_rate_text, arizona_rate_val),
+};
+
+const struct snd_kcontrol_new wm_adsp2_fw_controls[] = {
SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0],
wm_adsp_fw_get, wm_adsp_fw_put),
+ SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]),
SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1],
wm_adsp_fw_get, wm_adsp_fw_put),
+ SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]),
SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2],
wm_adsp_fw_get, wm_adsp_fw_put),
+ SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]),
SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3],
wm_adsp_fw_get, wm_adsp_fw_put),
+ SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]),
};
-EXPORT_SYMBOL_GPL(wm_adsp_fw_controls);
+EXPORT_SYMBOL_GPL(wm_adsp2_fw_controls);
+#endif
static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp,
int type)
@@ -549,13 +593,30 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
buf_size = sizeof(adsp1_id);
algs = be32_to_cpu(adsp1_id.algs);
+ dsp->fw_id = be32_to_cpu(adsp1_id.fw.id);
adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n",
- be32_to_cpu(adsp1_id.fw.id),
+ dsp->fw_id,
(be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16,
(be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8,
be32_to_cpu(adsp1_id.fw.ver) & 0xff,
algs);
+ region = kzalloc(sizeof(*region), GFP_KERNEL);
+ if (!region)
+ return -ENOMEM;
+ region->type = WMFW_ADSP1_ZM;
+ region->alg = be32_to_cpu(adsp1_id.fw.id);
+ region->base = be32_to_cpu(adsp1_id.zm);
+ list_add_tail(&region->list, &dsp->alg_regions);
+
+ region = kzalloc(sizeof(*region), GFP_KERNEL);
+ if (!region)
+ return -ENOMEM;
+ region->type = WMFW_ADSP1_DM;
+ region->alg = be32_to_cpu(adsp1_id.fw.id);
+ region->base = be32_to_cpu(adsp1_id.dm);
+ list_add_tail(&region->list, &dsp->alg_regions);
+
pos = sizeof(adsp1_id) / 2;
term = pos + ((sizeof(*adsp1_alg) * algs) / 2);
break;
@@ -573,13 +634,38 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
buf_size = sizeof(adsp2_id);
algs = be32_to_cpu(adsp2_id.algs);
+ dsp->fw_id = be32_to_cpu(adsp2_id.fw.id);
adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n",
- be32_to_cpu(adsp2_id.fw.id),
+ dsp->fw_id,
(be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16,
(be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8,
be32_to_cpu(adsp2_id.fw.ver) & 0xff,
algs);
+ region = kzalloc(sizeof(*region), GFP_KERNEL);
+ if (!region)
+ return -ENOMEM;
+ region->type = WMFW_ADSP2_XM;
+ region->alg = be32_to_cpu(adsp2_id.fw.id);
+ region->base = be32_to_cpu(adsp2_id.xm);
+ list_add_tail(&region->list, &dsp->alg_regions);
+
+ region = kzalloc(sizeof(*region), GFP_KERNEL);
+ if (!region)
+ return -ENOMEM;
+ region->type = WMFW_ADSP2_YM;
+ region->alg = be32_to_cpu(adsp2_id.fw.id);
+ region->base = be32_to_cpu(adsp2_id.ym);
+ list_add_tail(&region->list, &dsp->alg_regions);
+
+ region = kzalloc(sizeof(*region), GFP_KERNEL);
+ if (!region)
+ return -ENOMEM;
+ region->type = WMFW_ADSP2_ZM;
+ region->alg = be32_to_cpu(adsp2_id.fw.id);
+ region->base = be32_to_cpu(adsp2_id.zm);
+ list_add_tail(&region->list, &dsp->alg_regions);
+
pos = sizeof(adsp2_id) / 2;
term = pos + ((sizeof(*adsp2_alg) * algs) / 2);
break;
@@ -781,8 +867,24 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
case (WMFW_INFO_TEXT << 8):
break;
case (WMFW_ABSOLUTE << 8):
- region_name = "register";
- reg = offset;
+ /*
+ * Old files may use this for global
+ * coefficients.
+ */
+ if (le32_to_cpu(blk->id) == dsp->fw_id &&
+ offset == 0) {
+ region_name = "global coefficients";
+ mem = wm_adsp_find_region(dsp, type);
+ if (!mem) {
+ adsp_err(dsp, "No ZM\n");
+ break;
+ }
+ reg = wm_adsp_region_to_reg(mem, 0);
+
+ } else {
+ region_name = "register";
+ reg = offset;
+ }
break;
case WMFW_ADSP1_DM:
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index cb8871a3ec00..fea514627526 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -46,6 +46,8 @@ struct wm_adsp {
struct list_head alg_regions;
+ int fw_id;
+
const struct wm_adsp_region *mem;
int num_mems;
@@ -65,7 +67,8 @@ struct wm_adsp {
.shift = num, .event = wm_adsp2_event, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD }
-extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
+extern const struct snd_kcontrol_new wm_adsp1_fw_controls[];
+extern const struct snd_kcontrol_new wm_adsp2_fw_controls[];
int wm_adsp1_init(struct wm_adsp *adsp);
int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 867ae97ddcec..f5d81b948759 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -199,11 +199,12 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg)
list_add_tail(&cache->list, &hubs->dcs_cache);
}
-static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
+static int wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
u16 *reg_l, u16 *reg_r)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
u16 dcs_reg, reg;
+ int ret = 0;
switch (hubs->dcs_readback_mode) {
case 2:
@@ -236,8 +237,9 @@ static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
break;
default:
WARN(1, "Unknown DCS readback method\n");
- return;
+ ret = -1;
}
+ return ret;
}
/*
@@ -286,7 +288,8 @@ static void enable_dc_servo(struct snd_soc_codec *codec)
WM8993_DCS_TRIG_STARTUP_1);
}
- wm_hubs_read_dc_servo(codec, &reg_l, &reg_r);
+ if (wm_hubs_read_dc_servo(codec, &reg_l, &reg_r) < 0)
+ return;
dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 821831207180..ebe82947bab3 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -645,6 +645,10 @@ static struct snd_soc_dai_driver davinci_i2s_dai = {
};
+static const struct snd_soc_component_driver davinci_i2s_component = {
+ .name = "davinci-i2s",
+};
+
static int davinci_i2s_probe(struct platform_device *pdev)
{
struct snd_platform_data *pdata = pdev->dev.platform_data;
@@ -727,20 +731,21 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev_set_drvdata(&pdev->dev, dev);
- ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai);
+ ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component,
+ &davinci_i2s_dai, 1);
if (ret != 0)
goto err_release_clk;
ret = davinci_soc_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
- goto err_unregister_dai;
+ goto err_unregister_component;
}
return 0;
-err_unregister_dai:
- snd_soc_unregister_dai(&pdev->dev);
+err_unregister_component:
+ snd_soc_unregister_component(&pdev->dev);
err_release_clk:
clk_disable(dev->clk);
clk_put(dev->clk);
@@ -751,7 +756,7 @@ static int davinci_i2s_remove(struct platform_device *pdev)
{
struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
davinci_soc_platform_unregister(&pdev->dev);
clk_disable(dev->clk);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9321e5c9d8c1..8b85049daab0 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -235,6 +235,8 @@
#define DISMOD (val)(val<<2)
#define TXSTATE BIT(4)
#define RXSTATE BIT(5)
+#define SRMOD_MASK 3
+#define SRMOD_INACTIVE 0
/*
* DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
@@ -634,35 +636,43 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
* callback, take it into account here. That allows us to for example
* send 32 bits per channel to the codec, while only 16 of them carry
* audio payload.
- * The clock ratio is given for a full period of data (both left and
- * right channels), so it has to be divided by 2.
+ * The clock ratio is given for a full period of data (for I2S format
+ * both left and right channels), so it has to be divided by number of
+ * tdm-slots (for I2S - divided by 2).
*/
if (dev->bclk_lrclk_ratio)
- word_length = dev->bclk_lrclk_ratio / 2;
+ word_length = dev->bclk_lrclk_ratio / dev->tdm_slots;
/* mapping of the XSSZ bit-field as described in the datasheet */
fmt = (word_length >> 1) - 1;
- mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
- RXSSZ(fmt), RXSSZ(0x0F));
- mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
- TXSSZ(fmt), TXSSZ(0x0F));
- mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate),
- TXROT(7));
- mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate),
- RXROT(7));
+ if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) {
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
+ RXSSZ(fmt), RXSSZ(0x0F));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+ TXSSZ(fmt), TXSSZ(0x0F));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+ TXROT(rotate), TXROT(7));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
+ RXROT(rotate), RXROT(7));
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
+ mask);
+ }
+
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
- mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask);
return 0;
}
-static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
+static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream,
+ int channels)
{
int i;
u8 tx_ser = 0;
u8 rx_ser = 0;
-
+ u8 ser;
+ u8 slots = dev->tdm_slots;
+ u8 max_active_serializers = (channels + slots - 1) / slots;
/* Default configuration */
mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
@@ -682,17 +692,33 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
for (i = 0; i < dev->num_serializer; i++) {
mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
dev->serial_dir[i]);
- if (dev->serial_dir[i] == TX_MODE) {
+ if (dev->serial_dir[i] == TX_MODE &&
+ tx_ser < max_active_serializers) {
mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
AXR(i));
tx_ser++;
- } else if (dev->serial_dir[i] == RX_MODE) {
+ } else if (dev->serial_dir[i] == RX_MODE &&
+ rx_ser < max_active_serializers) {
mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
AXR(i));
rx_ser++;
+ } else {
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
+ SRMOD_INACTIVE, SRMOD_MASK);
}
}
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ser = tx_ser;
+ else
+ ser = rx_ser;
+
+ if (ser < max_active_serializers) {
+ dev_warn(dev->dev, "stream has more channels (%d) than are "
+ "enabled in mcasp (%d)\n", channels, ser * slots);
+ return -EINVAL;
+ }
+
if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (dev->txnumevt * tx_ser > 64)
dev->txnumevt = 1;
@@ -729,6 +755,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
}
}
+
+ return 0;
}
static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
@@ -772,12 +800,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
/* S/PDIF */
static void davinci_hw_dit_param(struct davinci_audio_dev *dev)
{
- /* Set the PDIR for Serialiser as output */
- mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX);
-
- /* TXMASK for 24 bits */
- mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF);
-
/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
and LSB first */
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
@@ -812,12 +834,21 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
&dev->dma_params[substream->stream];
int word_length;
u8 fifo_level;
+ u8 slots = dev->tdm_slots;
+ u8 active_serializers;
+ int channels;
+ struct snd_interval *pcm_channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ channels = pcm_channels->min;
+
+ active_serializers = (channels + slots - 1) / slots;
- davinci_hw_common_param(dev, substream->stream);
+ if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL)
+ return -EINVAL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- fifo_level = dev->txnumevt;
+ fifo_level = dev->txnumevt * active_serializers;
else
- fifo_level = dev->rxnumevt;
+ fifo_level = dev->rxnumevt * active_serializers;
if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
davinci_hw_dit_param(dev);
@@ -936,13 +967,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.name = "davinci-mcasp.0",
.playback = {
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
.capture = {
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
@@ -962,6 +993,10 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
};
+static const struct snd_soc_component_driver davinci_mcasp_component = {
+ .name = "davinci-mcasp",
+};
+
static const struct of_device_id mcasp_dt_ids[] = {
{
.compatible = "ti,dm646x-mcasp-audio",
@@ -1015,8 +1050,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
pdata->op_mode = val;
ret = of_property_read_u32(np, "tdm-slots", &val);
- if (ret >= 0)
+ if (ret >= 0) {
+ if (val < 2 || val > 32) {
+ dev_err(&pdev->dev,
+ "tdm-slots must be in rage [2-32]\n");
+ ret = -EINVAL;
+ goto nodata;
+ }
+
pdata->tdm_slots = val;
+ }
ret = of_property_read_u32(np, "num-serializer", &val);
if (ret >= 0)
@@ -1170,7 +1213,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data->channel = res->start;
dev_set_drvdata(&pdev->dev, dev);
- ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]);
+ ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component,
+ &davinci_mcasp_dai[pdata->op_mode], 1);
if (ret != 0)
goto err_release_clk;
@@ -1178,13 +1222,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
ret = davinci_soc_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
- goto err_unregister_dai;
+ goto err_unregister_component;
}
return 0;
-err_unregister_dai:
- snd_soc_unregister_dai(&pdev->dev);
+err_unregister_component:
+ snd_soc_unregister_component(&pdev->dev);
err_release_clk:
pm_runtime_put_sync(&pdev->dev);
pm_runtime_disable(&pdev->dev);
@@ -1194,7 +1238,7 @@ err_release_clk:
static int davinci_mcasp_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
davinci_soc_platform_unregister(&pdev->dev);
pm_runtime_put_sync(&pdev->dev);
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 0edd3b5a37fd..a9ac0c11da71 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -38,7 +38,7 @@ struct davinci_audio_dev {
u8 num_serializer;
u8 *serial_dir;
u8 version;
- u8 bclk_lrclk_ratio;
+ u16 bclk_lrclk_ratio;
/* McASP FIFO related */
u8 txnumevt;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index afab81f844ae..b2f27c2e5fdc 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -200,7 +200,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
src = dma_pos;
dst = prtd->params->dma_addr;
src_bidx = data_type;
- dst_bidx = 0;
+ dst_bidx = 4;
src_cidx = data_type * fifo_level;
dst_cidx = 0;
} else {
@@ -223,9 +223,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0,
ASYNC);
else
- edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level,
- count, fifo_level,
- ABSYNC);
+ edma_set_transfer_params(prtd->asp_link[0], acnt,
+ fifo_level,
+ count, fifo_level,
+ ABSYNC);
}
static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 07bde2e6f84e..30587c0cdbd2 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -204,6 +204,10 @@ static struct snd_soc_dai_driver davinci_vcif_dai = {
};
+static const struct snd_soc_component_driver davinci_vcif_component = {
+ .name = "davinci-vcif",
+};
+
static int davinci_vcif_probe(struct platform_device *pdev)
{
struct davinci_vc *davinci_vc = pdev->dev.platform_data;
@@ -234,7 +238,8 @@ static int davinci_vcif_probe(struct platform_device *pdev)
dev_set_drvdata(&pdev->dev, davinci_vcif_dev);
- ret = snd_soc_register_dai(&pdev->dev, &davinci_vcif_dai);
+ ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component,
+ &davinci_vcif_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "could not register dai\n");
return ret;
@@ -243,7 +248,7 @@ static int davinci_vcif_probe(struct platform_device *pdev)
ret = davinci_soc_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return ret;
}
@@ -252,7 +257,7 @@ static int davinci_vcif_probe(struct platform_device *pdev)
static int davinci_vcif_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
davinci_soc_platform_unregister(&pdev->dev);
return 0;
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index deb30d59965e..593a3ea12d4c 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -297,6 +297,10 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = {
.trigger = dw_i2s_trigger,
};
+static const struct snd_soc_component_driver dw_i2s_component = {
+ .name = "dw-i2s",
+};
+
#ifdef CONFIG_PM
static int dw_i2s_suspend(struct snd_soc_dai *dai)
@@ -413,7 +417,8 @@ static int dw_i2s_probe(struct platform_device *pdev)
dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
- ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai);
+ ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component,
+ dw_i2s_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "not able to register dai\n");
goto err_set_drvdata;
@@ -434,7 +439,7 @@ static int dw_i2s_remove(struct platform_device *pdev)
{
struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
dev_set_drvdata(&pdev->dev, NULL);
clk_put(dev->clk);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 3b98159d9645..3843a18d4e56 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -118,7 +118,7 @@ config SND_SOC_IMX_PCM_FIQ
config SND_SOC_IMX_PCM_DMA
bool
- select SND_SOC_DMAENGINE_PCM
+ select SND_SOC_GENERIC_DMAENGINE_PCM
select SND_SOC_IMX_PCM
config SND_SOC_IMX_AUDMUX
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 7decbd9b2340..0f0bed6def9e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -27,6 +27,7 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include "fsl_ssi.h"
#include "imx-pcm.h"
@@ -122,8 +123,10 @@ struct fsl_ssi_private {
bool ssi_on_imx;
struct clk *clk;
struct platform_device *imx_pcm_pdev;
- struct imx_pcm_dma_params dma_params_tx;
- struct imx_pcm_dma_params dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct imx_dma_data filter_data_tx;
+ struct imx_dma_data filter_data_rx;
struct {
unsigned int rfrc;
@@ -422,12 +425,6 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
ssi_private->second_stream = substream;
}
- if (ssi_private->ssi_on_imx)
- snd_soc_dai_set_dma_data(dai, substream,
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
- &ssi_private->dma_params_tx :
- &ssi_private->dma_params_rx);
-
return 0;
}
@@ -549,6 +546,18 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
}
}
+static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai);
+
+ if (ssi_private->ssi_on_imx) {
+ dai->playback_dma_data = &ssi_private->dma_params_tx;
+ dai->capture_dma_data = &ssi_private->dma_params_rx;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
.startup = fsl_ssi_startup,
.hw_params = fsl_ssi_hw_params,
@@ -558,6 +567,7 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
/* Template for the CPU dai driver structure */
static struct snd_soc_dai_driver fsl_ssi_dai_template = {
+ .probe = fsl_ssi_dai_probe,
.playback = {
/* The SSI does not support monaural audio. */
.channels_min = 2,
@@ -574,6 +584,10 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
.ops = &fsl_ssi_dai_ops,
};
+static const struct snd_soc_component_driver fsl_ssi_component = {
+ .name = "fsl-ssi",
+};
+
/* Show the statistics of a flag only if its interrupt is enabled. The
* compiler will optimze this code to a no-op if the interrupt is not
* enabled.
@@ -649,6 +663,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
const uint32_t *iprop;
struct resource res;
char name[64];
+ bool shared;
/* SSIs that are not connected on the board should have a
* status = "disabled"
@@ -737,14 +752,18 @@ static int fsl_ssi_probe(struct platform_device *pdev)
* We have burstsize be "fifo_depth - 2" to match the SSI
* watermark setting in fsl_ssi_startup().
*/
- ssi_private->dma_params_tx.burstsize =
+ ssi_private->dma_params_tx.maxburst =
ssi_private->fifo_depth - 2;
- ssi_private->dma_params_rx.burstsize =
+ ssi_private->dma_params_rx.maxburst =
ssi_private->fifo_depth - 2;
- ssi_private->dma_params_tx.dma_addr =
+ ssi_private->dma_params_tx.addr =
ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0);
- ssi_private->dma_params_rx.dma_addr =
+ ssi_private->dma_params_rx.addr =
ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0);
+ ssi_private->dma_params_tx.filter_data =
+ &ssi_private->filter_data_tx;
+ ssi_private->dma_params_rx.filter_data =
+ &ssi_private->filter_data_rx;
/*
* TODO: This is a temporary solution and should be changed
* to use generic DMA binding later when the helplers get in.
@@ -755,14 +774,14 @@ static int fsl_ssi_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "could not get dma events\n");
goto error_clk;
}
- ssi_private->dma_params_tx.dma = dma_events[0];
- ssi_private->dma_params_rx.dma = dma_events[1];
- ssi_private->dma_params_tx.shared_peripheral =
- of_device_is_compatible(of_get_parent(np),
- "fsl,spba-bus");
- ssi_private->dma_params_rx.shared_peripheral =
- ssi_private->dma_params_tx.shared_peripheral;
+ shared = of_device_is_compatible(of_get_parent(np),
+ "fsl,spba-bus");
+
+ imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx,
+ dma_events[0], shared);
+ imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx,
+ dma_events[1], shared);
}
/* Initialize the the device_attribute structure */
@@ -782,7 +801,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
/* Register with ASoC */
dev_set_drvdata(&pdev->dev, ssi_private);
- ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv);
+ ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
+ &ssi_private->cpu_dai_drv, 1);
if (ret) {
dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
goto error_dev;
@@ -835,7 +855,7 @@ done:
error_dai:
if (ssi_private->ssi_on_imx)
platform_device_unregister(ssi_private->imx_pcm_pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
error_dev:
dev_set_drvdata(&pdev->dev, NULL);
@@ -873,7 +893,7 @@ static int fsl_ssi_remove(struct platform_device *pdev)
clk_disable_unprepare(ssi_private->clk);
clk_put(ssi_private->clk);
}
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
device_remove_file(&pdev->dev, &ssi_private->dev_attr);
free_irq(ssi_private->irq, ssi_private);
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index 217300029b5b..e6b9a69e2a68 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -196,5 +196,13 @@ struct ccsr_ssi {
#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT)
#define CCSR_SSI_SOR_SYNRST 0x00000001
+#define CCSR_SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5)
+#define CCSR_SSI_SACNT_WR 0x00000010
+#define CCSR_SSI_SACNT_RD 0x00000008
+#define CCSR_SSI_SACNT_RDWR_MASK 0x00000018
+#define CCSR_SSI_SACNT_TIF 0x00000004
+#define CCSR_SSI_SACNT_FV 0x00000002
+#define CCSR_SSI_SACNT_AC97EN 0x00000001
+
#endif
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 3f333e5b4673..47f046a8fdab 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -262,7 +262,7 @@ static int imx_audmux_probe(struct platform_device *pdev)
return PTR_ERR(pinctrl);
}
- audmux_clk = clk_get(&pdev->dev, "audmux");
+ audmux_clk = devm_clk_get(&pdev->dev, "audmux");
if (IS_ERR(audmux_clk)) {
dev_dbg(&pdev->dev, "cannot get clock: %ld\n",
PTR_ERR(audmux_clk));
@@ -282,7 +282,6 @@ static int imx_audmux_remove(struct platform_device *pdev)
{
if (audmux_type == IMX31_AUDMUX)
audmux_debugfs_remove();
- clk_put(audmux_clk);
return 0;
}
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index 500f8ce55d78..c246fb514930 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -11,74 +11,30 @@
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
-#include <linux/clk.h>
-#include <linux/delay.h>
-#include <linux/device.h>
-#include <linux/dma-mapping.h>
-#include <linux/init.h>
-#include <linux/interrupt.h>
-#include <linux/module.h>
#include <linux/platform_device.h>
-#include <linux/slab.h>
#include <linux/dmaengine.h>
#include <linux/types.h>
#include <sound/core.h>
-#include <sound/initval.h>
#include <sound/pcm.h>
-#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/dmaengine_pcm.h>
-#include <linux/platform_data/dma-imx.h>
-
#include "imx-pcm.h"
static bool filter(struct dma_chan *chan, void *param)
{
+ struct snd_dmaengine_dai_dma_data *dma_data = param;
+
if (!imx_dma_is_general_purpose(chan))
return false;
- chan->private = param;
+ chan->private = dma_data->filter_data;
return true;
}
-static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
- struct imx_pcm_dma_params *dma_params;
- struct dma_slave_config slave_config;
- int ret;
-
- dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config);
- if (ret)
- return ret;
-
- slave_config.device_fc = false;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- slave_config.dst_addr = dma_params->dma_addr;
- slave_config.dst_maxburst = dma_params->burstsize;
- } else {
- slave_config.src_addr = dma_params->dma_addr;
- slave_config.src_maxburst = dma_params->burstsize;
- }
-
- ret = dmaengine_slave_config(chan, &slave_config);
- if (ret)
- return ret;
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
- return 0;
-}
-
-static struct snd_pcm_hardware snd_imx_hardware = {
+static const struct snd_pcm_hardware imx_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP |
@@ -97,64 +53,22 @@ static struct snd_pcm_hardware snd_imx_hardware = {
.fifo_size = 0,
};
-static int snd_imx_open(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct imx_pcm_dma_params *dma_params;
- struct imx_dma_data *dma_data;
- int ret;
-
- snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
-
- dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL);
- if (!dma_data)
- return -ENOMEM;
-
- dma_data->peripheral_type = dma_params->shared_peripheral ?
- IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI;
- dma_data->priority = DMA_PRIO_HIGH;
- dma_data->dma_request = dma_params->dma;
-
- ret = snd_dmaengine_pcm_open(substream, filter, dma_data);
- if (ret) {
- kfree(dma_data);
- return ret;
- }
-
- snd_dmaengine_pcm_set_data(substream, dma_data);
-
- return 0;
-}
+static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = {
+ .pcm_hardware = &imx_pcm_hardware,
+ .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+ .compat_filter_fn = filter,
+ .prealloc_buffer_size = IMX_SSI_DMABUF_SIZE,
+};
-static int snd_imx_close(struct snd_pcm_substream *substream)
+int imx_pcm_dma_init(struct platform_device *pdev)
{
- struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
-
- snd_dmaengine_pcm_close(substream);
- kfree(dma_data);
-
- return 0;
+ return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config,
+ SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
+ SND_DMAENGINE_PCM_FLAG_NO_DT |
+ SND_DMAENGINE_PCM_FLAG_COMPAT);
}
-static struct snd_pcm_ops imx_pcm_ops = {
- .open = snd_imx_open,
- .close = snd_imx_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_imx_pcm_hw_params,
- .trigger = snd_dmaengine_pcm_trigger,
- .pointer = snd_dmaengine_pcm_pointer_no_residue,
- .mmap = snd_imx_pcm_mmap,
-};
-
-static struct snd_soc_platform_driver imx_soc_platform_mx2 = {
- .ops = &imx_pcm_ops,
- .pcm_new = imx_pcm_new,
- .pcm_free = imx_pcm_free,
-};
-
-int imx_pcm_dma_init(struct platform_device *pdev)
+void imx_pcm_dma_exit(struct platform_device *pdev)
{
- return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2);
+ snd_dmaengine_pcm_unregister(&pdev->dev);
}
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 920f945cb2f4..670b96b0ce2f 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -34,7 +34,7 @@
#include "imx-ssi.h"
struct imx_pcm_runtime_data {
- int period;
+ unsigned int period;
int periods;
unsigned long offset;
unsigned long last_offset;
@@ -299,8 +299,8 @@ int imx_pcm_fiq_init(struct platform_device *pdev)
imx_ssi_fiq_base = (unsigned long)ssi->base;
- ssi->dma_params_tx.burstsize = 4;
- ssi->dma_params_rx.burstsize = 6;
+ ssi->dma_params_tx.maxburst = 4;
+ ssi->dma_params_rx.maxburst = 6;
ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq);
if (ret)
diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c
index 0d0625bfcb65..c49896442d8e 100644
--- a/sound/soc/fsl/imx-pcm.c
+++ b/sound/soc/fsl/imx-pcm.c
@@ -114,7 +114,11 @@ static int imx_pcm_probe(struct platform_device *pdev)
static int imx_pcm_remove(struct platform_device *pdev)
{
- snd_soc_unregister_platform(&pdev->dev);
+ if (strcmp(pdev->id_entry->name, "imx-fiq-pcm-audio") == 0)
+ snd_soc_unregister_platform(&pdev->dev);
+ else
+ imx_pcm_dma_exit(pdev);
+
return 0;
}
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 5ae13a13a353..b7fa0d75c687 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -13,17 +13,24 @@
#ifndef _IMX_PCM_H
#define _IMX_PCM_H
+#include <linux/platform_data/dma-imx.h>
+
/*
* Do not change this as the FIQ handler depends on this size
*/
#define IMX_SSI_DMABUF_SIZE (64 * 1024)
-struct imx_pcm_dma_params {
- int dma;
- unsigned long dma_addr;
- int burstsize;
- bool shared_peripheral; /* The peripheral is on SPBA bus */
-};
+static inline void
+imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data,
+ int dma, bool shared)
+{
+ dma_data->dma_request = dma;
+ dma_data->priority = DMA_PRIO_HIGH;
+ if (shared)
+ dma_data->peripheral_type = IMX_DMATYPE_SSI_SP;
+ else
+ dma_data->peripheral_type = IMX_DMATYPE_SSI;
+}
int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma);
@@ -32,11 +39,16 @@ void imx_pcm_free(struct snd_pcm *pcm);
#ifdef CONFIG_SND_SOC_IMX_PCM_DMA
int imx_pcm_dma_init(struct platform_device *pdev);
+void imx_pcm_dma_exit(struct platform_device *pdev);
#else
static inline int imx_pcm_dma_init(struct platform_device *pdev)
{
return -ENODEV;
}
+
+static inline void imx_pcm_dma_exit(struct platform_device *pdev)
+{
+}
#endif
#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 424347e9b2d7..9584e78858df 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -148,7 +148,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
data->dai.stream_name = "HiFi";
data->dai.codec_dai_name = "sgtl5000";
data->dai.codec_of_node = codec_np;
- data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev);
+ data->dai.cpu_of_node = ssi_np;
data->dai.platform_name = "imx-pcm-audio";
data->dai.init = &imx_sgtl5000_dai_init;
data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 810c7eeb7b03..902fab02b851 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -232,23 +232,6 @@ static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
return 0;
}
-static int imx_ssi_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai);
- struct imx_pcm_dma_params *dma_data;
-
- /* Tx/Rx config */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dma_data = &ssi->dma_params_tx;
- else
- dma_data = &ssi->dma_params_rx;
-
- snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
-
- return 0;
-}
-
/*
* Should only be called when port is inactive (i.e. SSIEN = 0),
* although can be called multiple times by upper layers.
@@ -353,7 +336,6 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
}
static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
- .startup = imx_ssi_startup,
.hw_params = imx_ssi_hw_params,
.set_fmt = imx_ssi_set_dai_fmt,
.set_clkdiv = imx_ssi_set_dai_clkdiv,
@@ -369,10 +351,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
snd_soc_dai_set_drvdata(dai, ssi);
- val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
- SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
+ val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.maxburst) |
+ SSI_SFCSR_RFWM0(ssi->dma_params_rx.maxburst);
writel(val, ssi->base + SSI_SFCSR);
+ /* Tx/Rx config */
+ dai->playback_dma_data = &ssi->dma_params_tx;
+ dai->capture_dma_data = &ssi->dma_params_rx;
+
return 0;
}
@@ -400,7 +386,7 @@ static struct snd_soc_dai_driver imx_ac97_dai = {
.stream_name = "AC97 Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_48000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
@@ -413,6 +399,10 @@ static struct snd_soc_dai_driver imx_ac97_dai = {
.ops = &imx_ssi_pcm_dai_ops,
};
+static const struct snd_soc_component_driver imx_component = {
+ .name = DRV_NAME,
+};
+
static void setup_channel_to_ac97(struct imx_ssi *imx_ssi)
{
void __iomem *base = imx_ssi->base;
@@ -575,23 +565,31 @@ static int imx_ssi_probe(struct platform_device *pdev)
writel(0x0, ssi->base + SSI_SIER);
- ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0;
- ssi->dma_params_tx.dma_addr = res->start + SSI_STX0;
+ ssi->dma_params_rx.addr = res->start + SSI_SRX0;
+ ssi->dma_params_tx.addr = res->start + SSI_STX0;
+
+ ssi->dma_params_tx.maxburst = 6;
+ ssi->dma_params_rx.maxburst = 4;
- ssi->dma_params_tx.burstsize = 6;
- ssi->dma_params_rx.burstsize = 4;
+ ssi->dma_params_tx.filter_data = &ssi->filter_data_tx;
+ ssi->dma_params_rx.filter_data = &ssi->filter_data_rx;
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
- if (res)
- ssi->dma_params_tx.dma = res->start;
+ if (res) {
+ imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start,
+ false);
+ }
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0");
- if (res)
- ssi->dma_params_rx.dma = res->start;
+ if (res) {
+ imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start,
+ false);
+ }
platform_set_drvdata(pdev, ssi);
- ret = snd_soc_register_dai(&pdev->dev, dai);
+ ret = snd_soc_register_component(&pdev->dev, &imx_component,
+ dai, 1);
if (ret) {
dev_err(&pdev->dev, "register DAI failed\n");
goto failed_register;
@@ -632,7 +630,7 @@ failed_pdev_alloc:
failed_pdev_fiq_add:
platform_device_put(ssi->soc_platform_pdev_fiq);
failed_pdev_fiq_alloc:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
failed_register:
release_mem_region(res->start, resource_size(res));
failed_get_resource:
@@ -650,7 +648,7 @@ static int imx_ssi_remove(struct platform_device *pdev)
platform_device_unregister(ssi->soc_platform_pdev);
platform_device_unregister(ssi->soc_platform_pdev_fiq);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
if (ssi->flags & IMX_SSI_USE_AC97)
ac97_ssi = NULL;
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index dc114bdedce5..bb6b3dbb13fd 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -187,6 +187,7 @@
#include <linux/dmaengine.h>
#include <linux/platform_data/dma-imx.h>
+#include <sound/dmaengine_pcm.h>
#include "imx-pcm.h"
struct imx_ssi {
@@ -204,8 +205,10 @@ struct imx_ssi {
void (*ac97_reset) (struct snd_ac97 *ac97);
void (*ac97_warm_reset)(struct snd_ac97 *ac97);
- struct imx_pcm_dma_params dma_params_rx;
- struct imx_pcm_dma_params dma_params_tx;
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct imx_dma_data filter_data_tx;
+ struct imx_dma_data filter_data_rx;
int enabled;
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index a4aec0488dd3..4141b35ef0bb 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -270,6 +270,9 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = {
.ops = &psc_ac97_digital_ops,
} };
+static const struct snd_soc_component_driver psc_ac97_component = {
+ .name = DRV_NAME,
+};
/* ---------------------------------------------------------------------
@@ -287,7 +290,8 @@ static int psc_ac97_of_probe(struct platform_device *op)
if (rc != 0)
return rc;
- rc = snd_soc_register_dais(&op->dev, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+ rc = snd_soc_register_component(&op->dev, &psc_ac97_component,
+ psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
if (rc != 0) {
dev_err(&op->dev, "Failed to register DAI\n");
return rc;
@@ -313,7 +317,7 @@ static int psc_ac97_of_probe(struct platform_device *op)
static int psc_ac97_of_remove(struct platform_device *op)
{
mpc5200_audio_dma_destroy(op);
- snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_ac97_dai));
+ snd_soc_unregister_component(&op->dev);
return 0;
}
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index b95b966f25a0..f4efaadb80a2 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -148,6 +148,10 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{
.ops = &psc_i2s_dai_ops,
} };
+static const struct snd_soc_component_driver psc_i2s_component = {
+ .name = "mpc5200-i2s",
+};
+
/* ---------------------------------------------------------------------
* OF platform bus binding code:
* - Probe/remove operations
@@ -163,7 +167,8 @@ static int psc_i2s_of_probe(struct platform_device *op)
if (rc != 0)
return rc;
- rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
+ rc = snd_soc_register_component(&op->dev, &psc_i2s_component,
+ psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
if (rc != 0) {
pr_err("Failed to register DAI\n");
return rc;
@@ -208,7 +213,7 @@ static int psc_i2s_of_probe(struct platform_device *op)
static int psc_i2s_of_remove(struct platform_device *op)
{
mpc5200_audio_dma_destroy(op);
- snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_i2s_dai));
+ snd_soc_unregister_component(&op->dev);
return 0;
}
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 6cef491f4823..9a126441c5f3 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -425,6 +425,10 @@ static struct snd_soc_dai_driver jz4740_i2s_dai = {
.resume = jz4740_i2s_resume,
};
+static const struct snd_soc_component_driver jz4740_i2s_component = {
+ .name = "jz4740-i2s",
+};
+
static int jz4740_i2s_dev_probe(struct platform_device *pdev)
{
struct jz4740_i2s *i2s;
@@ -469,7 +473,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev)
}
platform_set_drvdata(pdev, i2s);
- ret = snd_soc_register_dai(&pdev->dev, &jz4740_i2s_dai);
+ ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component,
+ &jz4740_i2s_dai, 1);
if (ret) {
dev_err(&pdev->dev, "Failed to register DAI\n");
@@ -496,7 +501,7 @@ static int jz4740_i2s_dev_remove(struct platform_device *pdev)
{
struct jz4740_i2s *i2s = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
clk_put(i2s->clk_i2s);
clk_put(i2s->clk_aic);
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index c74c89065493..befe68f59285 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -451,6 +451,10 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = {
.ops = &kirkwood_i2s_dai_ops,
};
+static const struct snd_soc_component_driver kirkwood_i2s_component = {
+ .name = DRV_NAME,
+};
+
static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
{
struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data;
@@ -524,10 +528,11 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_128;
}
- err = snd_soc_register_dai(&pdev->dev, soc_dai);
+ err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component,
+ soc_dai, 1);
if (!err)
return 0;
- dev_err(&pdev->dev, "snd_soc_register_dai failed\n");
+ dev_err(&pdev->dev, "snd_soc_register_component failed\n");
if (!IS_ERR(priv->extclk)) {
clk_disable_unprepare(priv->extclk);
@@ -542,7 +547,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
{
struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
if (!IS_ERR(priv->extclk)) {
clk_disable_unprepare(priv->extclk);
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index a263cbed8624..392fc0b8f5b8 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -1,7 +1,7 @@
/*
* sst_platform.c - Intel MID Platform driver
*
- * Copyright (C) 2010-2012 Intel Corp
+ * Copyright (C) 2010-2013 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -165,6 +165,10 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
},
};
+static const struct snd_soc_component_driver sst_component = {
+ .name = "sst",
+};
+
/* helper functions */
static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
int state)
@@ -652,11 +656,21 @@ static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
return stream->compr_ops->get_codec_caps(codec);
}
+static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream,
+ struct snd_compr_metadata *metadata)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->set_metadata(stream->id, metadata);
+}
+
static struct snd_compr_ops sst_platform_compr_ops = {
.open = sst_platform_compr_open,
.free = sst_platform_compr_free,
.set_params = sst_platform_compr_set_params,
+ .set_metadata = sst_platform_compr_set_metadata,
.trigger = sst_platform_compr_trigger,
.pointer = sst_platform_compr_pointer,
.ack = sst_platform_compr_ack,
@@ -683,7 +697,7 @@ static int sst_platform_probe(struct platform_device *pdev)
return ret;
}
- ret = snd_soc_register_dais(&pdev->dev,
+ ret = snd_soc_register_component(&pdev->dev, &sst_component,
sst_platform_dai, ARRAY_SIZE(sst_platform_dai));
if (ret) {
pr_err("registering cpu dais failed\n");
@@ -695,7 +709,7 @@ static int sst_platform_probe(struct platform_device *pdev)
static int sst_platform_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai));
+ snd_soc_unregister_component(&pdev->dev);
snd_soc_unregister_platform(&pdev->dev);
pr_debug("sst_platform_remove success\n");
return 0;
diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h
index d61c5d514ffa..cacc9066ec52 100644
--- a/sound/soc/mid-x86/sst_platform.h
+++ b/sound/soc/mid-x86/sst_platform.h
@@ -124,6 +124,8 @@ struct compress_sst_ops {
int (*close) (unsigned int str_id);
int (*get_caps) (struct snd_compr_caps *caps);
int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
+ int (*set_metadata) (unsigned int str_id,
+ struct snd_compr_metadata *mdata);
};
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index b6fa77678d97..78d321cbe8b4 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -1,7 +1,7 @@
menuconfig SND_MXS_SOC
tristate "SoC Audio for Freescale MXS CPUs"
depends on ARCH_MXS
- select SND_SOC_DMAENGINE_PCM
+ select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M if you want to add support for codecs attached to
the MXS SAIF interface.
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 564b5b60319d..b41fffc056fb 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -18,38 +18,24 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#include <linux/clk.h>
-#include <linux/delay.h>
#include <linux/device.h>
-#include <linux/dma-mapping.h>
#include <linux/init.h>
-#include <linux/interrupt.h>
#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dmaengine.h>
-#include <linux/fsl/mxs-dma.h>
#include <sound/core.h>
-#include <sound/initval.h>
#include <sound/pcm.h>
-#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/dmaengine_pcm.h>
#include "mxs-pcm.h"
-struct mxs_pcm_dma_data {
- struct mxs_dma_data dma_data;
- struct mxs_pcm_dma_params *dma_params;
-};
-
-static struct snd_pcm_hardware snd_mxs_hardware = {
+static const struct snd_pcm_hardware snd_mxs_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_INTERLEAVED,
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_HALF_DUPLEX,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S20_3LE |
SNDRV_PCM_FMTBIT_S24_LE,
@@ -61,13 +47,11 @@ static struct snd_pcm_hardware snd_mxs_hardware = {
.periods_max = 52,
.buffer_bytes_max = 64 * 1024,
.fifo_size = 32,
-
};
static bool filter(struct dma_chan *chan, void *param)
{
- struct mxs_pcm_dma_data *pcm_dma_data = param;
- struct mxs_pcm_dma_params *dma_params = pcm_dma_data->dma_params;
+ struct mxs_pcm_dma_params *dma_params = param;
if (!mxs_dma_is_apbx(chan))
return false;
@@ -75,160 +59,30 @@ static bool filter(struct dma_chan *chan, void *param)
if (chan->chan_id != dma_params->chan_num)
return false;
- chan->private = &pcm_dma_data->dma_data;
+ chan->private = &dma_params->dma_data;
return true;
}
-static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
- return 0;
-}
-
-static int snd_mxs_open(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct mxs_pcm_dma_data *pcm_dma_data;
- int ret;
-
- pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL);
- if (pcm_dma_data == NULL)
- return -ENOMEM;
-
- pcm_dma_data->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- pcm_dma_data->dma_data.chan_irq = pcm_dma_data->dma_params->chan_irq;
-
- ret = snd_dmaengine_pcm_open(substream, filter, pcm_dma_data);
- if (ret) {
- kfree(pcm_dma_data);
- return ret;
- }
-
- snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware);
-
- snd_dmaengine_pcm_set_data(substream, pcm_dma_data);
-
- return 0;
-}
-
-static int snd_mxs_close(struct snd_pcm_substream *substream)
-{
- struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream);
-
- snd_dmaengine_pcm_close(substream);
- kfree(pcm_dma_data);
-
- return 0;
-}
-
-static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- return dma_mmap_writecombine(substream->pcm->card->dev, vma,
- runtime->dma_area,
- runtime->dma_addr,
- runtime->dma_bytes);
-}
-
-static struct snd_pcm_ops mxs_pcm_ops = {
- .open = snd_mxs_open,
- .close = snd_mxs_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_mxs_pcm_hw_params,
- .trigger = snd_dmaengine_pcm_trigger,
- .pointer = snd_dmaengine_pcm_pointer_no_residue,
- .mmap = snd_mxs_pcm_mmap,
-};
-
-static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *buf = &substream->dma_buffer;
- size_t size = snd_mxs_hardware.buffer_bytes_max;
-
- buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
- buf->private_data = NULL;
- buf->area = dma_alloc_writecombine(pcm->card->dev, size,
- &buf->addr, GFP_KERNEL);
- if (!buf->area)
- return -ENOMEM;
- buf->bytes = size;
-
- return 0;
-}
-
-static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32);
-static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_card *card = rtd->card->snd_card;
- struct snd_pcm *pcm = rtd->pcm;
- int ret = 0;
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &mxs_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
-
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = mxs_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = mxs_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
-
-out:
- return ret;
-}
-
-static void mxs_pcm_free(struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (!substream)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
-
- dma_free_writecombine(pcm->card->dev, buf->bytes,
- buf->area, buf->addr);
- buf->area = NULL;
- }
-}
-
-static struct snd_soc_platform_driver mxs_soc_platform = {
- .ops = &mxs_pcm_ops,
- .pcm_new = mxs_pcm_new,
- .pcm_free = mxs_pcm_free,
+static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = {
+ .pcm_hardware = &snd_mxs_hardware,
+ .compat_filter_fn = filter,
+ .prealloc_buffer_size = 64 * 1024,
};
int mxs_pcm_platform_register(struct device *dev)
{
- return snd_soc_register_platform(dev, &mxs_soc_platform);
+ return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config,
+ SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
+ SND_DMAENGINE_PCM_FLAG_NO_DT |
+ SND_DMAENGINE_PCM_FLAG_COMPAT |
+ SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX);
}
EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
void mxs_pcm_platform_unregister(struct device *dev)
{
- snd_soc_unregister_platform(dev);
+ snd_dmaengine_pcm_unregister(dev);
}
EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister);
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
index 35ba2ca42384..3aa918f9ed3e 100644
--- a/sound/soc/mxs/mxs-pcm.h
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -19,8 +19,10 @@
#ifndef _MXS_PCM_H
#define _MXS_PCM_H
+#include <linux/fsl/mxs-dma.h>
+
struct mxs_pcm_dma_params {
- int chan_irq;
+ struct mxs_dma_data dma_data;
int chan_num;
};
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 41a6136e3535..d31dc52fa862 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -370,7 +370,6 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
- snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param);
/* clear error status to 0 for each re-open */
saif->fifo_underrun = 0;
@@ -606,6 +605,8 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai)
struct mxs_saif *saif = dev_get_drvdata(dai->dev);
snd_soc_dai_set_drvdata(dai, saif);
+ dai->playback_dma_data = &saif->dma_param;
+ dai->capture_dma_data = &saif->dma_param;
return 0;
}
@@ -628,6 +629,10 @@ static struct snd_soc_dai_driver mxs_saif_dai = {
.ops = &mxs_saif_dai_ops,
};
+static const struct snd_soc_component_driver mxs_saif_component = {
+ .name = "mxs-saif",
+};
+
static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
{
struct mxs_saif *saif = dev_id;
@@ -754,9 +759,9 @@ static int mxs_saif_probe(struct platform_device *pdev)
return ret;
}
- saif->dma_param.chan_irq = platform_get_irq(pdev, 1);
- if (saif->dma_param.chan_irq < 0) {
- ret = saif->dma_param.chan_irq;
+ saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1);
+ if (saif->dma_param.dma_data.chan_irq < 0) {
+ ret = saif->dma_param.dma_data.chan_irq;
dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
ret);
return ret;
@@ -764,7 +769,8 @@ static int mxs_saif_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, saif);
- ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai);
+ ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component,
+ &mxs_saif_dai, 1);
if (ret) {
dev_err(&pdev->dev, "register DAI failed\n");
return ret;
@@ -779,7 +785,7 @@ static int mxs_saif_probe(struct platform_device *pdev)
return 0;
failed_pdev_alloc:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return ret;
}
@@ -787,7 +793,7 @@ failed_pdev_alloc:
static int mxs_saif_remove(struct platform_device *pdev)
{
mxs_pcm_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 0418467a4848..fe3285ceaf5b 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -314,6 +314,10 @@ static struct snd_soc_dai_driver nuc900_ac97_dai = {
.ops = &nuc900_ac97_dai_ops,
};
+static const struct snd_soc_component_driver nuc900_ac97_component = {
+ .name = "nuc900-ac97",
+};
+
static int nuc900_ac97_drvprobe(struct platform_device *pdev)
{
struct nuc900_audio *nuc900_audio;
@@ -361,7 +365,8 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev)
nuc900_ac97_data = nuc900_audio;
- ret = snd_soc_register_dai(&pdev->dev, &nuc900_ac97_dai);
+ ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
+ &nuc900_ac97_dai, 1);
if (ret)
goto out3;
@@ -384,7 +389,7 @@ out0:
static int nuc900_ac97_drvremove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
clk_put(nuc900_ac97_data->clk);
iounmap(nuc900_ac97_data->mmio);
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index c1900b2a6f28..994dcf345975 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -28,7 +28,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#include "../codecs/tlv320aic23.h"
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2600447fa74f..629446482a91 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -36,7 +36,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#include "../codecs/cx20442.h"
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 285c8368cb47..eb68c7db1cf3 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -1018,9 +1018,10 @@ int omap_mcbsp_init(struct platform_device *pdev)
return -ENODEV;
}
/* RX DMA request number, and port address configuration */
- mcbsp->dma_data[1].name = "Audio Capture";
- mcbsp->dma_data[1].dma_req = res->start;
- mcbsp->dma_data[1].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 1);
+ mcbsp->dma_req[1] = res->start;
+ mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1];
+ mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1);
+ mcbsp->dma_data[1].maxburst = 4;
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx");
if (!res) {
@@ -1028,9 +1029,10 @@ int omap_mcbsp_init(struct platform_device *pdev)
return -ENODEV;
}
/* TX DMA request number, and port address configuration */
- mcbsp->dma_data[0].name = "Audio Playback";
- mcbsp->dma_data[0].dma_req = res->start;
- mcbsp->dma_data[0].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 0);
+ mcbsp->dma_req[0] = res->start;
+ mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0];
+ mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0);
+ mcbsp->dma_data[0].maxburst = 4;
mcbsp->fclk = clk_get(&pdev->dev, "fck");
if (IS_ERR(mcbsp->fclk)) {
diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h
index f93e0b0af303..96d1b086bcf8 100644
--- a/sound/soc/omap/mcbsp.h
+++ b/sound/soc/omap/mcbsp.h
@@ -24,14 +24,14 @@
#ifndef __ASOC_MCBSP_H
#define __ASOC_MCBSP_H
-#include "omap-pcm.h"
-
#ifdef CONFIG_ARCH_OMAP1
#define mcbsp_omap1() 1
#else
#define mcbsp_omap1() 0
#endif
+#include <sound/dmaengine_pcm.h>
+
/* McBSP register numbers. Register address offset = num * reg_step */
enum {
/* Common registers */
@@ -312,7 +312,8 @@ struct omap_mcbsp {
struct omap_mcbsp_platform_data *pdata;
struct omap_mcbsp_st_data *st_data;
struct omap_mcbsp_reg_cfg cfg_regs;
- struct omap_pcm_dma_data dma_data[2];
+ struct snd_dmaengine_dai_dma_data dma_data[2];
+ unsigned int dma_req[2];
int dma_op_mode;
u16 max_tx_thres;
u16 max_rx_thres;
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index ee7cd53aa3ee..5e8d640d314f 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -34,7 +34,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define N810_HEADSET_AMP_GPIO 10
#define N810_SPEAKER_AMP_GPIO 101
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index e7d93fa412a9..70cd5c7b2e14 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -34,7 +34,6 @@
#include "omap-dmic.h"
#include "omap-mcpdm.h"
-#include "omap-pcm.h"
#include "../codecs/twl6040.h"
struct abe_twl6040 {
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index ba49ccd9eed9..2ad0370146fd 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -39,8 +39,8 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
-#include "omap-pcm.h"
#include "omap-dmic.h"
struct omap_dmic {
@@ -55,13 +55,9 @@ struct omap_dmic {
u32 ch_enabled;
bool active;
struct mutex mutex;
-};
-/*
- * Stream DMA parameters
- */
-static struct omap_pcm_dma_data omap_dmic_dai_dma_params = {
- .name = "DMIC capture",
+ struct snd_dmaengine_dai_dma_data dma_data;
+ unsigned int dma_req;
};
static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val)
@@ -118,7 +114,7 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
mutex_unlock(&dmic->mutex);
- snd_soc_dai_set_dma_data(dai, substream, &omap_dmic_dai_dma_params);
+ snd_soc_dai_set_dma_data(dai, substream, &dmic->dma_data);
return ret;
}
@@ -203,7 +199,7 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
int channels;
dmic->clk_div = omap_dmic_select_divider(dmic, params_rate(params));
@@ -230,7 +226,7 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
/* packet size is threshold * channels */
dma_data = snd_soc_dai_get_dma_data(dai, substream);
- dma_data->packet_size = dmic->threshold * channels;
+ dma_data->maxburst = dmic->threshold * channels;
return 0;
}
@@ -448,6 +444,10 @@ static struct snd_soc_dai_driver omap_dmic_dai = {
.ops = &omap_dmic_dai_ops,
};
+static const struct snd_soc_component_driver omap_dmic_component = {
+ .name = "omap-dmic",
+};
+
static int asoc_dmic_probe(struct platform_device *pdev)
{
struct omap_dmic *dmic;
@@ -476,7 +476,7 @@ static int asoc_dmic_probe(struct platform_device *pdev)
ret = -ENODEV;
goto err_put_clk;
}
- omap_dmic_dai_dma_params.port_addr = res->start + OMAP_DMIC_DATA_REG;
+ dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG;
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!res) {
@@ -484,7 +484,9 @@ static int asoc_dmic_probe(struct platform_device *pdev)
ret = -ENODEV;
goto err_put_clk;
}
- omap_dmic_dai_dma_params.dma_req = res->start;
+
+ dmic->dma_req = res->start;
+ dmic->dma_data.filter_data = &dmic->dma_req;
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
if (!res) {
@@ -493,21 +495,12 @@ static int asoc_dmic_probe(struct platform_device *pdev)
goto err_put_clk;
}
- if (!devm_request_mem_region(&pdev->dev, res->start,
- resource_size(res), pdev->name)) {
- dev_err(dmic->dev, "memory region already claimed\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
-
- dmic->io_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!dmic->io_base) {
- ret = -ENOMEM;
- goto err_put_clk;
- }
+ dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(dmic->io_base))
+ return PTR_ERR(dmic->io_base);
- ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai);
+ ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component,
+ &omap_dmic_dai, 1);
if (ret)
goto err_put_clk;
@@ -522,7 +515,7 @@ static int asoc_dmic_remove(struct platform_device *pdev)
{
struct omap_dmic *dmic = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
clk_put(dmic->fclk);
return 0;
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index 32fa840c493e..ced3b88b44d4 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -32,15 +32,16 @@
#include <sound/soc.h>
#include <sound/asound.h>
#include <sound/asoundef.h>
+#include <sound/dmaengine_pcm.h>
#include <video/omapdss.h>
-#include "omap-pcm.h"
#include "omap-hdmi.h"
#define DRV_NAME "omap-hdmi-audio-dai"
struct hdmi_priv {
- struct omap_pcm_dma_data dma_params;
+ struct snd_dmaengine_dai_dma_data dma_data;
+ unsigned int dma_req;
struct omap_dss_audio dss_audio;
struct snd_aes_iec958 iec;
struct snd_cea_861_aud_if cea;
@@ -68,7 +69,7 @@ static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
return -ENODEV;
}
- snd_soc_dai_set_dma_data(dai, substream, &priv->dma_params);
+ snd_soc_dai_set_dma_data(dai, substream, &priv->dma_data);
return 0;
}
@@ -88,25 +89,20 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
struct snd_aes_iec958 *iec = &priv->iec;
struct snd_cea_861_aud_if *cea = &priv->cea;
- struct omap_pcm_dma_data *dma_data;
int err = 0;
- dma_data = snd_soc_dai_get_dma_data(dai, substream);
-
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- dma_data->packet_size = 16;
+ priv->dma_data.maxburst = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- dma_data->packet_size = 32;
+ priv->dma_data.maxburst = 32;
break;
default:
dev_err(dai->dev, "format not supported!\n");
return -EINVAL;
}
- dma_data->data_type = 32;
-
/*
* fill the IEC-60958 channel status word
*/
@@ -264,6 +260,10 @@ static struct snd_soc_dai_driver omap_hdmi_dai = {
.ops = &omap_hdmi_dai_ops,
};
+static const struct snd_soc_component_driver omap_hdmi_component = {
+ .name = DRV_NAME,
+};
+
static int omap_hdmi_probe(struct platform_device *pdev)
{
int ret;
@@ -283,8 +283,7 @@ static int omap_hdmi_probe(struct platform_device *pdev)
return -ENODEV;
}
- hdmi_data->dma_params.port_addr = hdmi_rsrc->start
- + OMAP_HDMI_AUDIO_DMA_PORT;
+ hdmi_data->dma_data.addr = hdmi_rsrc->start + OMAP_HDMI_AUDIO_DMA_PORT;
hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!hdmi_rsrc) {
@@ -292,8 +291,9 @@ static int omap_hdmi_probe(struct platform_device *pdev)
return -ENODEV;
}
- hdmi_data->dma_params.dma_req = hdmi_rsrc->start;
- hdmi_data->dma_params.name = "HDMI playback";
+ hdmi_data->dma_req = hdmi_rsrc->start;
+ hdmi_data->dma_data.filter_data = &hdmi_data->dma_req;
+ hdmi_data->dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
/*
* TODO: We assume that there is only one DSS HDMI device. Future
@@ -321,7 +321,8 @@ static int omap_hdmi_probe(struct platform_device *pdev)
}
dev_set_drvdata(&pdev->dev, hdmi_data);
- ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai);
+ ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component,
+ &omap_hdmi_dai, 1);
return ret;
}
@@ -330,7 +331,7 @@ static int omap_hdmi_remove(struct platform_device *pdev)
{
struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
if (hdmi_data == NULL) {
dev_err(&pdev->dev, "cannot obtain HDMi data\n");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8d2defd6fdbe..eadbfb6b5000 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -33,11 +33,11 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "mcbsp.h"
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000)
@@ -62,24 +62,22 @@ enum {
* Stream DMA parameters. DMA request line and port address are set runtime
* since they are different between OMAP1 and later OMAPs
*/
-static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
+ unsigned int packet_size)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_pcm_dma_data *dma_data;
int words;
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
/*
* Configure McBSP threshold based on either:
* packet_size, when the sDMA is in packet mode, or based on the
* period size in THRESHOLD mode, otherwise use McBSP threshold = 1
* for mono streams.
*/
- if (dma_data->packet_size)
- words = dma_data->packet_size;
+ if (packet_size)
+ words = packet_size;
else
words = 1;
@@ -226,7 +224,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
{
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
int wlen, channels, wpf;
int pkt_size = 0;
unsigned int format, div, framesize, master;
@@ -245,7 +243,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
if (mcbsp->pdata->buffer_size) {
- dma_data->set_threshold = omap_mcbsp_set_threshold;
if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
int period_words, max_thrsh;
int divider = 0;
@@ -276,9 +273,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
/* Use packet mode for non mono streams */
pkt_size = channels;
}
+ omap_mcbsp_set_threshold(substream, pkt_size);
}
- dma_data->packet_size = pkt_size;
+ dma_data->maxburst = pkt_size;
if (mcbsp->configured) {
/* McBSP already configured by another stream */
@@ -586,6 +584,10 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = {
.ops = &mcbsp_dai_ops,
};
+static const struct snd_soc_component_driver omap_mcbsp_component = {
+ .name = "omap-mcbsp",
+};
+
static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -793,7 +795,8 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
ret = omap_mcbsp_init(pdev);
if (!ret)
- return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai);
+ return snd_soc_register_component(&pdev->dev, &omap_mcbsp_component,
+ &omap_mcbsp_dai, 1);
return ret;
}
@@ -802,7 +805,7 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
{
struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
if (mcbsp->pdata->ops && mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(mcbsp->id);
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 5ca11bdac21e..eb05c7ed6d05 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -39,11 +39,14 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include "omap-mcpdm.h"
-#include "omap-pcm.h"
-#define OMAP44XX_MCPDM_L3_BASE 0x49032000
+struct mcpdm_link_config {
+ u32 link_mask; /* channel mask for the direction */
+ u32 threshold; /* FIFO threshold */
+};
struct omap_mcpdm {
struct device *dev;
@@ -53,29 +56,22 @@ struct omap_mcpdm {
struct mutex mutex;
- /* channel data */
- u32 dn_channels;
- u32 up_channels;
-
- /* McPDM FIFO thresholds */
- u32 dn_threshold;
- u32 up_threshold;
+ /* Playback/Capture configuration */
+ struct mcpdm_link_config config[2];
/* McPDM dn offsets for rx1, and 2 channels */
u32 dn_rx_offset;
+
+ /* McPDM needs to be restarted due to runtime reconfiguration */
+ bool restart;
+
+ struct snd_dmaengine_dai_dma_data dma_data[2];
+ unsigned int dma_req[2];
};
/*
* Stream DMA parameters
*/
-static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = {
- {
- .name = "Audio playback",
- },
- {
- .name = "Audio capture",
- },
-};
static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val)
{
@@ -130,11 +126,12 @@ static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {}
static void omap_mcpdm_start(struct omap_mcpdm *mcpdm)
{
u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+ u32 link_mask = mcpdm->config[0].link_mask | mcpdm->config[1].link_mask;
ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
- ctrl |= mcpdm->dn_channels | mcpdm->up_channels;
+ ctrl |= link_mask;
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
@@ -148,11 +145,12 @@ static void omap_mcpdm_start(struct omap_mcpdm *mcpdm)
static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm)
{
u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+ u32 link_mask = MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK;
ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
- ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels);
+ ctrl &= ~(link_mask);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
@@ -188,8 +186,10 @@ static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm)
omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset);
}
- omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold);
- omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold);
+ omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN,
+ mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold);
+ omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP,
+ mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold);
omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET,
MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE);
@@ -267,7 +267,7 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
mutex_unlock(&mcpdm->mutex);
snd_soc_dai_set_dma_data(dai, substream,
- &omap_mcpdm_dai_dma_params[substream->stream]);
+ &mcpdm->dma_data[substream->stream]);
return 0;
}
@@ -283,6 +283,8 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
if (omap_mcpdm_active(mcpdm)) {
omap_mcpdm_stop(mcpdm);
omap_mcpdm_close_streams(mcpdm);
+ mcpdm->config[0].link_mask = 0;
+ mcpdm->config[1].link_mask = 0;
}
}
@@ -295,7 +297,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
int stream = substream->stream;
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
+ u32 threshold;
int channels;
int link_mask = 0;
@@ -325,16 +328,32 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
dma_data = snd_soc_dai_get_dma_data(dai, substream);
+ threshold = mcpdm->config[stream].threshold;
/* Configure McPDM channels, and DMA packet size */
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- mcpdm->dn_channels = link_mask << 3;
- dma_data->packet_size =
- (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels;
+ link_mask <<= 3;
+
+ /* If capture is not running assume a stereo stream to come */
+ if (!mcpdm->config[!stream].link_mask)
+ mcpdm->config[!stream].link_mask = 0x3;
+
+ dma_data->maxburst =
+ (MCPDM_DN_THRES_MAX - threshold) * channels;
} else {
- mcpdm->up_channels = link_mask << 0;
- dma_data->packet_size = mcpdm->up_threshold * channels;
+ /* If playback is not running assume a stereo stream to come */
+ if (!mcpdm->config[!stream].link_mask)
+ mcpdm->config[!stream].link_mask = (0x3 << 3);
+
+ dma_data->maxburst = threshold * channels;
}
+ /* Check if we need to restart McPDM with this stream */
+ if (mcpdm->config[stream].link_mask &&
+ mcpdm->config[stream].link_mask != link_mask)
+ mcpdm->restart = true;
+
+ mcpdm->config[stream].link_mask = link_mask;
+
return 0;
}
@@ -346,6 +365,11 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
if (!omap_mcpdm_active(mcpdm)) {
omap_mcpdm_start(mcpdm);
omap_mcpdm_reg_dump(mcpdm);
+ } else if (mcpdm->restart) {
+ omap_mcpdm_stop(mcpdm);
+ omap_mcpdm_start(mcpdm);
+ mcpdm->restart = false;
+ omap_mcpdm_reg_dump(mcpdm);
}
return 0;
@@ -369,7 +393,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
pm_runtime_get_sync(mcpdm->dev);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00);
- ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+ ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler,
0, "McPDM", (void *)mcpdm);
pm_runtime_put_sync(mcpdm->dev);
@@ -380,8 +404,9 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
}
/* Configure McPDM threshold values */
- mcpdm->dn_threshold = 2;
- mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3;
+ mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold = 2;
+ mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold =
+ MCPDM_UP_THRES_MAX - 3;
return ret;
}
@@ -389,7 +414,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
- free_irq(mcpdm->irq, (void *)mcpdm);
pm_runtime_disable(mcpdm->dev);
return 0;
@@ -420,6 +444,10 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = {
.ops = &omap_mcpdm_dai_ops,
};
+static const struct snd_soc_component_driver omap_mcpdm_component = {
+ .name = "omap-mcpdm",
+};
+
void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
u8 rx1, u8 rx2)
{
@@ -446,33 +474,30 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
if (res == NULL)
return -ENOMEM;
- omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA;
- omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA;
+ mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA;
+ mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA;
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link");
if (!res)
return -ENODEV;
- omap_mcpdm_dai_dma_params[0].dma_req = res->start;
+ mcpdm->dma_req[0] = res->start;
+ mcpdm->dma_data[0].filter_data = &mcpdm->dma_req[0];
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link");
if (!res)
return -ENODEV;
- omap_mcpdm_dai_dma_params[1].dma_req = res->start;
+ mcpdm->dma_req[1] = res->start;
+ mcpdm->dma_data[1].filter_data = &mcpdm->dma_req[1];
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
if (res == NULL)
return -ENOMEM;
- if (!devm_request_mem_region(&pdev->dev, res->start,
- resource_size(res), "McPDM"))
- return -EBUSY;
-
- mcpdm->io_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!mcpdm->io_base)
- return -ENOMEM;
+ mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(mcpdm->io_base))
+ return PTR_ERR(mcpdm->io_base);
mcpdm->irq = platform_get_irq(pdev, 0);
if (mcpdm->irq < 0)
@@ -480,12 +505,13 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
mcpdm->dev = &pdev->dev;
- return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai);
+ return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component,
+ &omap_mcpdm_dai, 1);
}
static int asoc_mcpdm_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index c722c2ef9665..c28e042f2208 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -32,8 +32,6 @@
#include <sound/dmaengine_pcm.h>
#include <sound/soc.h>
-#include "omap-pcm.h"
-
#ifdef CONFIG_ARCH_OMAP1
#define pcm_omap1510() cpu_is_omap1510()
#else
@@ -56,25 +54,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
.buffer_bytes_max = 128 * 1024,
};
-static int omap_pcm_get_dma_buswidth(int num_bits)
-{
- int buswidth;
-
- switch (num_bits) {
- case 16:
- buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
- break;
- case 32:
- buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
- break;
- default:
- buswidth = -EINVAL;
- break;
- }
- return buswidth;
-}
-
-
/* this may get called several times by oss emulation */
static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -105,20 +84,9 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
if (err)
return err;
- /* Override the *_dma addr_width if requested by the DAI driver */
- if (dma_data->data_type) {
- int buswidth = omap_pcm_get_dma_buswidth(dma_data->data_type);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- config.dst_addr_width = buswidth;
- else
- config.src_addr_width = buswidth;
- }
-
- config.src_addr = dma_data->port_addr;
- config.dst_addr = dma_data->port_addr;
- config.src_maxburst = dma_data->packet_size;
- config.dst_maxburst = dma_data->packet_size;
+ snd_dmaengine_pcm_set_config_from_dai_data(substream,
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+ &config);
return dmaengine_slave_config(chan, &config);
}
@@ -129,37 +97,6 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
return 0;
}
-static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct omap_pcm_dma_data *dma_data;
- int ret = 0;
-
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- /* Configure McBSP internal buffer usage */
- if (dma_data->set_threshold)
- dma_data->set_threshold(substream);
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- break;
- default:
- ret = -EINVAL;
- }
-
- if (ret == 0)
- ret = snd_dmaengine_pcm_trigger(substream, cmd);
-
- return ret;
-}
-
static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
{
snd_pcm_uframes_t offset;
@@ -175,20 +112,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
static int omap_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- return snd_dmaengine_pcm_open(substream, omap_dma_filter_fn,
- &dma_data->dma_req);
-}
-
-static int omap_pcm_close(struct snd_pcm_substream *substream)
-{
- snd_dmaengine_pcm_close(substream);
- return 0;
+ return snd_dmaengine_pcm_open_request_chan(substream,
+ omap_dma_filter_fn,
+ dma_data->filter_data);
}
static int omap_pcm_mmap(struct snd_pcm_substream *substream,
@@ -204,11 +136,11 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream,
static struct snd_pcm_ops omap_pcm_ops = {
.open = omap_pcm_open,
- .close = omap_pcm_close,
+ .close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = omap_pcm_hw_params,
.hw_free = omap_pcm_hw_free,
- .trigger = omap_pcm_trigger,
+ .trigger = snd_dmaengine_pcm_trigger,
.pointer = omap_pcm_pointer,
.mmap = omap_pcm_mmap,
};
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
deleted file mode 100644
index cabe74c4068b..000000000000
--- a/sound/soc/omap/omap-pcm.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * omap-pcm.h
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.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.
- *
- * 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
- *
- */
-
-#ifndef __OMAP_PCM_H__
-#define __OMAP_PCM_H__
-
-struct snd_pcm_substream;
-
-struct omap_pcm_dma_data {
- char *name; /* stream identifier */
- int dma_req; /* DMA request line */
- unsigned long port_addr; /* transmit/receive register */
- void (*set_threshold)(struct snd_pcm_substream *substream);
- int data_type; /* 8, 16, 32 (bits) or 0 to let omap-pcm
- * to decide the sDMA data type */
- int packet_size; /* packet size only in PACKET mode */
-};
-
-#endif
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index fd98509d0f49..2a9324f794d8 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -43,7 +43,6 @@
#include <sound/jack.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
struct omap_twl4030 {
int jack_detect; /* board can detect jack events */
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 805512f2555a..cf604a2faa18 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -34,7 +34,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define OMAP3_PANDORA_DAC_POWER_GPIO 118
#define OMAP3_PANDORA_AMP_POWER_GPIO 14
@@ -80,12 +79,18 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
+ int ret;
+
/*
* The PCM1773 DAC datasheet requires 1ms delay between switching
* VCC power on/off and /PD pin high/low
*/
if (SND_SOC_DAPM_EVENT_ON(event)) {
- regulator_enable(omap3pandora_dac_reg);
+ ret = regulator_enable(omap3pandora_dac_reg);
+ if (ret) {
+ dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret);
+ return ret;
+ }
mdelay(1);
gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
} else {
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 06ef8d67ed1c..d03e57da7708 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -33,7 +33,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#include "../codecs/tlv320aic23.h"
#define CODEC_CLOCK 12000000
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 3cd525748975..249cd230ad8f 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -37,7 +37,6 @@
#include <asm/mach-types.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define RX51_TVOUT_SEL_GPIO 40
#define RX51_JACK_DETECT_GPIO 177
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 190eb0bccf5f..349930015264 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -118,9 +118,8 @@ static int mmp_pcm_open(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct platform_device *pdev = to_platform_device(rtd->platform->dev);
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct mmp_dma_data *dma_data;
+ struct mmp_dma_data dma_data;
struct resource *r;
- int ret;
r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
if (!r)
@@ -128,33 +127,12 @@ static int mmp_pcm_open(struct snd_pcm_substream *substream)
snd_soc_set_runtime_hwparams(substream,
&mmp_pcm_hardware[substream->stream]);
- dma_data = devm_kzalloc(&pdev->dev,
- sizeof(struct mmp_dma_data), GFP_KERNEL);
- if (dma_data == NULL)
- return -ENOMEM;
- dma_data->dma_res = r;
- dma_data->ssp_id = cpu_dai->id;
+ dma_data.dma_res = r;
+ dma_data.ssp_id = cpu_dai->id;
- ret = snd_dmaengine_pcm_open(substream, filter, dma_data);
- if (ret) {
- devm_kfree(&pdev->dev, dma_data);
- return ret;
- }
-
- snd_dmaengine_pcm_set_data(substream, dma_data);
- return 0;
-}
-
-static int mmp_pcm_close(struct snd_pcm_substream *substream)
-{
- struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct platform_device *pdev = to_platform_device(rtd->platform->dev);
-
- snd_dmaengine_pcm_close(substream);
- devm_kfree(&pdev->dev, dma_data);
- return 0;
+ return snd_dmaengine_pcm_open_request_chan(substream, filter,
+ &dma_data);
}
static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
@@ -171,7 +149,7 @@ static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
struct snd_pcm_ops mmp_pcm_ops = {
.open = mmp_pcm_open,
- .close = mmp_pcm_close,
+ .close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = mmp_pcm_hw_params,
.trigger = snd_dmaengine_pcm_trigger,
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index 9140c4abafbc..a64779980177 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -405,6 +405,10 @@ struct snd_soc_dai_driver mmp_sspa_dai = {
.ops = &mmp_sspa_dai_ops,
};
+static const struct snd_soc_component_driver mmp_sspa_component = {
+ .name = "mmp-sspa",
+};
+
static int asoc_mmp_sspa_probe(struct platform_device *pdev)
{
struct sspa_priv *priv;
@@ -450,7 +454,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev)
priv->dai_fmt = (unsigned int) -1;
platform_set_drvdata(pdev, priv);
- return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai);
+ return snd_soc_register_component(&pdev->dev, &mmp_sspa_component,
+ &mmp_sspa_dai, 1);
}
static int asoc_mmp_sspa_remove(struct platform_device *pdev)
@@ -460,7 +465,7 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev)
clk_disable(priv->audio_clk);
clk_put(priv->audio_clk);
clk_put(priv->sysclk);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d3eb0c2eec77..6f4dd7543e82 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -794,14 +794,19 @@ static struct snd_soc_dai_driver pxa_ssp_dai = {
.ops = &pxa_ssp_dai_ops,
};
+static const struct snd_soc_component_driver pxa_ssp_component = {
+ .name = "pxa-ssp",
+};
+
static int asoc_ssp_probe(struct platform_device *pdev)
{
- return snd_soc_register_dai(&pdev->dev, &pxa_ssp_dai);
+ return snd_soc_register_component(&pdev->dev, &pxa_ssp_component,
+ &pxa_ssp_dai, 1);
}
static int asoc_ssp_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 4b0a009bd683..57ea8e6c5488 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -47,6 +47,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
.warm_reset = pxa2xx_ac97_warm_reset,
.reset = pxa2xx_ac97_cold_reset,
};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
.name = "AC97 PCM Stereo out",
@@ -232,7 +233,9 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
},
};
-EXPORT_SYMBOL_GPL(soc_ac97_ops);
+static const struct snd_soc_component_driver pxa_ac97_component = {
+ .name = "pxa-ac97",
+};
static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
{
@@ -245,13 +248,13 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
* driver to do interesting things with the clocking to get us up
* and running.
*/
- return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver,
- ARRAY_SIZE(pxa_ac97_dai_driver));
+ return snd_soc_register_component(&pdev->dev, &pxa_ac97_component,
+ pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver));
}
static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver));
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b1a06f67564..f7ca71664112 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -360,14 +360,19 @@ static struct snd_soc_dai_driver pxa_i2s_dai = {
.symmetric_rates = 1,
};
+static const struct snd_soc_component_driver pxa_i2s_component = {
+ .name = "pxa-i2s",
+};
+
static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
{
- return snd_soc_register_dai(&pdev->dev, &pxa_i2s_dai);
+ return snd_soc_register_component(&pdev->dev, &pxa_i2s_component,
+ &pxa_i2s_dai, 1);
}
static int pxa2xx_i2s_drv_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index fee4d477a49c..73bb99f0109a 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -436,6 +436,10 @@ static struct snd_soc_dai_driver s6000_i2s_dai = {
.ops = &s6000_i2s_dai_ops,
};
+static const struct snd_soc_component_driver s6000_i2s_component = {
+ .name = "s6000-i2s",
+};
+
static int s6000_i2s_probe(struct platform_device *pdev)
{
struct s6000_i2s_dev *dev;
@@ -543,7 +547,8 @@ static int s6000_i2s_probe(struct platform_device *pdev)
S6_I2S_INT_UNDERRUN |
S6_I2S_INT_OVERRUN);
- ret = snd_soc_register_dai(&pdev->dev, &s6000_i2s_dai);
+ ret = snd_soc_register_component(&pdev->dev, &s6000_i2s_component,
+ &s6000_i2s_dai, 1);
if (ret)
goto err_release_dev;
@@ -572,7 +577,7 @@ static void s6000_i2s_remove(struct platform_device *pdev)
struct resource *region;
void __iomem *mmio = dev->scbbase;
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
s6000_i2s_stop_channel(dev, 0);
s6000_i2s_stop_channel(dev, 1);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 90e7e6653233..475fb0d8b3c6 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -35,11 +35,10 @@ config SND_SAMSUNG_I2S
tristate
config SND_SOC_SAMSUNG_NEO1973_WM8753
- tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)"
- depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02)
+ tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)"
+ depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02
select SND_S3C24XX_I2S
select SND_SOC_WM8753
- select SND_SOC_LM4857 if MACH_NEO1973_GTA01
select SND_SOC_DFBMCS320
help
Say Y here to enable audio support for the Openmoko Neo1973
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 0df3c5644cfa..cb88ead98917 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -20,7 +20,7 @@
#include <sound/soc.h>
#include <mach/dma.h>
-#include <plat/regs-ac97.h>
+#include "regs-ac97.h"
#include <linux/platform_data/asoc-s3c.h>
#include "dma.h"
@@ -370,6 +370,10 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = {
},
};
+static const struct snd_soc_component_driver s3c_ac97_component = {
+ .name = "s3c-ac97",
+};
+
static int s3c_ac97_probe(struct platform_device *pdev)
{
struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res;
@@ -457,8 +461,8 @@ static int s3c_ac97_probe(struct platform_device *pdev)
goto err4;
}
- ret = snd_soc_register_dais(&pdev->dev, s3c_ac97_dai,
- ARRAY_SIZE(s3c_ac97_dai));
+ ret = snd_soc_register_component(&pdev->dev, &s3c_ac97_component,
+ s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai));
if (ret)
goto err5;
@@ -470,7 +474,7 @@ static int s3c_ac97_probe(struct platform_device *pdev)
return 0;
err6:
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(s3c_ac97_dai));
+ snd_soc_unregister_component(&pdev->dev);
err5:
free_irq(irq_res->start, NULL);
err4:
@@ -490,7 +494,7 @@ static int s3c_ac97_remove(struct platform_device *pdev)
struct resource *mem_res, *irq_res;
asoc_dma_platform_unregister(&pdev->dev);
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(s3c_ac97_dai));
+ snd_soc_unregister_component(&pdev->dev);
irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
if (irq_res)
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index d37ede58e0a8..415ad81999c4 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -218,6 +218,10 @@ static struct snd_soc_dai_driver voice_dai = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
};
+static const struct snd_soc_component_driver voice_component = {
+ .name = "goni-voice",
+};
+
static struct snd_soc_ops goni_voice_ops = {
.hw_params = goni_voice_hw_params,
};
@@ -270,7 +274,8 @@ static int __init goni_init(void)
return -ENOMEM;
/* register voice DAI here */
- ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai);
+ ret = snd_soc_register_component(&goni_snd_device->dev, &voice_component,
+ &voice_dai, 1);
if (ret) {
platform_device_put(goni_snd_device);
return ret;
@@ -280,7 +285,7 @@ static int __init goni_init(void)
ret = platform_device_add(goni_snd_device);
if (ret) {
- snd_soc_unregister_dai(&goni_snd_device->dev);
+ snd_soc_unregister_component(&goni_snd_device->dev);
platform_device_put(goni_snd_device);
}
@@ -289,7 +294,7 @@ static int __init goni_init(void)
static void __exit goni_exit(void)
{
- snd_soc_unregister_dai(&goni_snd_device->dev);
+ snd_soc_unregister_component(&goni_snd_device->dev);
platform_device_unregister(goni_snd_device);
}
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 15a3817aa5c8..fa91376e323d 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -20,7 +20,7 @@
#include <sound/soc.h>
#include <sound/jack.h>
-#include <plat/regs-iis.h>
+#include "regs-iis.h"
#include <asm/mach-types.h>
#include "s3c24xx-i2s.h"
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 6bbeb0bf1a73..82ebb1a51479 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -963,6 +963,10 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = {
.delay = i2s_delay,
};
+static const struct snd_soc_component_driver samsung_i2s_component = {
+ .name = "samsung-i2s",
+};
+
#define SAMSUNG_I2S_RATES SNDRV_PCM_RATE_8000_96000
#define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \
@@ -1114,8 +1118,9 @@ static int samsung_i2s_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Unable to get drvdata\n");
return -EFAULT;
}
- snd_soc_register_dai(&sec_dai->pdev->dev,
- &sec_dai->i2s_dai_drv);
+ snd_soc_register_component(&sec_dai->pdev->dev,
+ &samsung_i2s_component,
+ &sec_dai->i2s_dai_drv, 1);
asoc_dma_platform_register(&pdev->dev);
return 0;
}
@@ -1244,7 +1249,8 @@ static int samsung_i2s_probe(struct platform_device *pdev)
}
}
- snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv);
+ snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component,
+ &pri_dai->i2s_dai_drv, 1);
pm_runtime_enable(&pdev->dev);
@@ -1283,7 +1289,7 @@ static int samsung_i2s_remove(struct platform_device *pdev)
i2s->sec_dai = NULL;
asoc_dma_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
@@ -1298,7 +1304,7 @@ static struct platform_device_id samsung_i2s_driver_ids[] = {
},
{},
};
-MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids);
+MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids);
#ifdef CONFIG_OF
static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = {
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index a07950b0c8ce..6e5fed30aa27 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -68,6 +68,8 @@ static struct idma_info {
dma_addr_t lp_tx_addr;
} idma;
+static int idma_irq;
+
static void idma_getpos(dma_addr_t *src)
{
*src = idma.lp_tx_addr +
@@ -305,7 +307,7 @@ static int idma_open(struct snd_pcm_substream *substream)
if (prtd == NULL)
return -ENOMEM;
- ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd);
+ ret = request_irq(idma_irq, iis_irq, 0, "i2s", prtd);
if (ret < 0) {
pr_err("fail to claim i2s irq , ret = %d\n", ret);
kfree(prtd);
@@ -324,7 +326,7 @@ static int idma_close(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct idma_ctrl *prtd = runtime->private_data;
- free_irq(IRQ_I2S0, prtd);
+ free_irq(idma_irq, prtd);
if (!prtd)
pr_err("idma_close called with prtd == NULL\n");
@@ -409,6 +411,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr)
idma.regs = regs;
idma.lp_tx_addr = addr;
}
+EXPORT_SYMBOL_GPL(idma_reg_addr_init);
static struct snd_soc_platform_driver asoc_idma_platform = {
.ops = &idma_ops,
@@ -418,6 +421,10 @@ static struct snd_soc_platform_driver asoc_idma_platform = {
static int asoc_idma_platform_probe(struct platform_device *pdev)
{
+ idma_irq = platform_get_irq(pdev, 0);
+ if (idma_irq < 0)
+ return idma_irq;
+
return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform);
}
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index a301d8cfaa34..e591c386917a 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -21,8 +21,7 @@
#include <sound/soc.h>
#include <asm/mach-types.h>
-#include <plat/regs-iis.h>
-#include <mach/gta02.h>
+#include "regs-iis.h"
#include "../codecs/wm8753.h"
#include "s3c24xx-i2s.h"
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 13bab79ad93d..1566afe9ef52 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -490,6 +490,10 @@ static struct snd_soc_dai_driver s3c_pcm_dai[] = {
},
};
+static const struct snd_soc_component_driver s3c_pcm_component = {
+ .name = "s3c-pcm",
+};
+
static int s3c_pcm_dev_probe(struct platform_device *pdev)
{
struct s3c_pcm_info *pcm;
@@ -583,7 +587,8 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
pm_runtime_enable(&pdev->dev);
- ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]);
+ ret = snd_soc_register_component(&pdev->dev, &s3c_pcm_component,
+ &s3c_pcm_dai[pdev->id], 1);
if (ret != 0) {
dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret);
goto err5;
@@ -598,7 +603,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
return 0;
err6:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
err5:
clk_disable_unprepare(pcm->pclk);
clk_put(pcm->pclk);
@@ -619,7 +624,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev)
struct resource *mem_res;
asoc_dma_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
pm_runtime_disable(&pdev->dev);
diff --git a/sound/soc/samsung/regs-ac97.h b/sound/soc/samsung/regs-ac97.h
new file mode 100644
index 000000000000..c3878f7acb83
--- /dev/null
+++ b/sound/soc/samsung/regs-ac97.h
@@ -0,0 +1,67 @@
+/* arch/arm/mach-s3c2410/include/mach/regs-ac97.h
+ *
+ * Copyright (c) 2006 Simtec Electronics <linux@simtec.co.uk>
+ * http://www.simtec.co.uk/products/SWLINUX/
+ *
+ * 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.
+ *
+ * S3C2440 AC97 Controller
+*/
+
+#ifndef __ASM_ARCH_REGS_AC97_H
+#define __ASM_ARCH_REGS_AC97_H __FILE__
+
+#define S3C_AC97_GLBCTRL (0x00)
+
+#define S3C_AC97_GLBCTRL_CODECREADYIE (1<<22)
+#define S3C_AC97_GLBCTRL_PCMOUTURIE (1<<21)
+#define S3C_AC97_GLBCTRL_PCMINORIE (1<<20)
+#define S3C_AC97_GLBCTRL_MICINORIE (1<<19)
+#define S3C_AC97_GLBCTRL_PCMOUTTIE (1<<18)
+#define S3C_AC97_GLBCTRL_PCMINTIE (1<<17)
+#define S3C_AC97_GLBCTRL_MICINTIE (1<<16)
+#define S3C_AC97_GLBCTRL_PCMOUTTM_OFF (0<<12)
+#define S3C_AC97_GLBCTRL_PCMOUTTM_PIO (1<<12)
+#define S3C_AC97_GLBCTRL_PCMOUTTM_DMA (2<<12)
+#define S3C_AC97_GLBCTRL_PCMOUTTM_MASK (3<<12)
+#define S3C_AC97_GLBCTRL_PCMINTM_OFF (0<<10)
+#define S3C_AC97_GLBCTRL_PCMINTM_PIO (1<<10)
+#define S3C_AC97_GLBCTRL_PCMINTM_DMA (2<<10)
+#define S3C_AC97_GLBCTRL_PCMINTM_MASK (3<<10)
+#define S3C_AC97_GLBCTRL_MICINTM_OFF (0<<8)
+#define S3C_AC97_GLBCTRL_MICINTM_PIO (1<<8)
+#define S3C_AC97_GLBCTRL_MICINTM_DMA (2<<8)
+#define S3C_AC97_GLBCTRL_MICINTM_MASK (3<<8)
+#define S3C_AC97_GLBCTRL_TRANSFERDATAENABLE (1<<3)
+#define S3C_AC97_GLBCTRL_ACLINKON (1<<2)
+#define S3C_AC97_GLBCTRL_WARMRESET (1<<1)
+#define S3C_AC97_GLBCTRL_COLDRESET (1<<0)
+
+#define S3C_AC97_GLBSTAT (0x04)
+
+#define S3C_AC97_GLBSTAT_CODECREADY (1<<22)
+#define S3C_AC97_GLBSTAT_PCMOUTUR (1<<21)
+#define S3C_AC97_GLBSTAT_PCMINORI (1<<20)
+#define S3C_AC97_GLBSTAT_MICINORI (1<<19)
+#define S3C_AC97_GLBSTAT_PCMOUTTI (1<<18)
+#define S3C_AC97_GLBSTAT_PCMINTI (1<<17)
+#define S3C_AC97_GLBSTAT_MICINTI (1<<16)
+#define S3C_AC97_GLBSTAT_MAINSTATE_IDLE (0<<0)
+#define S3C_AC97_GLBSTAT_MAINSTATE_INIT (1<<0)
+#define S3C_AC97_GLBSTAT_MAINSTATE_READY (2<<0)
+#define S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE (3<<0)
+#define S3C_AC97_GLBSTAT_MAINSTATE_LP (4<<0)
+#define S3C_AC97_GLBSTAT_MAINSTATE_WARM (5<<0)
+
+#define S3C_AC97_CODEC_CMD (0x08)
+
+#define S3C_AC97_CODEC_CMD_READ (1<<23)
+
+#define S3C_AC97_STAT (0x0c)
+#define S3C_AC97_PCM_ADDR (0x10)
+#define S3C_AC97_PCM_DATA (0x18)
+#define S3C_AC97_MIC_DATA (0x1C)
+
+#endif /* __ASM_ARCH_REGS_AC97_H */
diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h
new file mode 100644
index 000000000000..a18d35e7a735
--- /dev/null
+++ b/sound/soc/samsung/regs-iis.h
@@ -0,0 +1,70 @@
+/* arch/arm/plat-samsung/include/plat/regs-iis.h
+ *
+ * Copyright (c) 2003 Simtec Electronics <linux@simtec.co.uk>
+ * http://www.simtec.co.uk/products/SWLINUX/
+ *
+ * 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.
+ *
+ * S3C2410 IIS register definition
+*/
+
+#ifndef __ASM_ARCH_REGS_IIS_H
+#define __ASM_ARCH_REGS_IIS_H
+
+#define S3C2410_IISCON (0x00)
+
+#define S3C2410_IISCON_LRINDEX (1 << 8)
+#define S3C2410_IISCON_TXFIFORDY (1 << 7)
+#define S3C2410_IISCON_RXFIFORDY (1 << 6)
+#define S3C2410_IISCON_TXDMAEN (1 << 5)
+#define S3C2410_IISCON_RXDMAEN (1 << 4)
+#define S3C2410_IISCON_TXIDLE (1 << 3)
+#define S3C2410_IISCON_RXIDLE (1 << 2)
+#define S3C2410_IISCON_PSCEN (1 << 1)
+#define S3C2410_IISCON_IISEN (1 << 0)
+
+#define S3C2410_IISMOD (0x04)
+
+#define S3C2440_IISMOD_MPLL (1 << 9)
+#define S3C2410_IISMOD_SLAVE (1 << 8)
+#define S3C2410_IISMOD_NOXFER (0 << 6)
+#define S3C2410_IISMOD_RXMODE (1 << 6)
+#define S3C2410_IISMOD_TXMODE (2 << 6)
+#define S3C2410_IISMOD_TXRXMODE (3 << 6)
+#define S3C2410_IISMOD_LR_LLOW (0 << 5)
+#define S3C2410_IISMOD_LR_RLOW (1 << 5)
+#define S3C2410_IISMOD_IIS (0 << 4)
+#define S3C2410_IISMOD_MSB (1 << 4)
+#define S3C2410_IISMOD_8BIT (0 << 3)
+#define S3C2410_IISMOD_16BIT (1 << 3)
+#define S3C2410_IISMOD_BITMASK (1 << 3)
+#define S3C2410_IISMOD_256FS (0 << 2)
+#define S3C2410_IISMOD_384FS (1 << 2)
+#define S3C2410_IISMOD_16FS (0 << 0)
+#define S3C2410_IISMOD_32FS (1 << 0)
+#define S3C2410_IISMOD_48FS (2 << 0)
+#define S3C2410_IISMOD_FS_MASK (3 << 0)
+
+#define S3C2410_IISPSR (0x08)
+
+#define S3C2410_IISPSR_INTMASK (31 << 5)
+#define S3C2410_IISPSR_INTSHIFT (5)
+#define S3C2410_IISPSR_EXTMASK (31 << 0)
+#define S3C2410_IISPSR_EXTSHFIT (0)
+
+#define S3C2410_IISFCON (0x0c)
+
+#define S3C2410_IISFCON_TXDMA (1 << 15)
+#define S3C2410_IISFCON_RXDMA (1 << 14)
+#define S3C2410_IISFCON_TXENABLE (1 << 13)
+#define S3C2410_IISFCON_RXENABLE (1 << 12)
+#define S3C2410_IISFCON_TXMASK (0x3f << 6)
+#define S3C2410_IISFCON_TXSHIFT (6)
+#define S3C2410_IISFCON_RXMASK (0x3f)
+#define S3C2410_IISFCON_RXSHIFT (0)
+
+#define S3C2410_IISFIFO (0x10)
+
+#endif /* __ASM_ARCH_REGS_IIS_H */
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index a5826ea9cad6..704460a37005 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -24,7 +24,7 @@
#include <sound/soc.h>
#include <sound/jack.h>
-#include <plat/regs-iis.h>
+#include "regs-iis.h"
#include <asm/mach-types.h>
#include "s3c24xx-i2s.h"
diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 7a73380b3560..20e98d1dded2 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -731,8 +731,9 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
#define s3c2412_i2s_resume NULL
#endif
-int s3c_i2sv2_register_dai(struct device *dev, int id,
- struct snd_soc_dai_driver *drv)
+int s3c_i2sv2_register_component(struct device *dev, int id,
+ struct snd_soc_component_driver *cmp_drv,
+ struct snd_soc_dai_driver *dai_drv)
{
struct snd_soc_dai_ops *ops = drv->ops;
@@ -750,8 +751,8 @@ int s3c_i2sv2_register_dai(struct device *dev, int id,
drv->suspend = s3c2412_i2s_suspend;
drv->resume = s3c2412_i2s_resume;
- return snd_soc_register_dai(dev, drv);
+ return snd_soc_register_component(dev, cmp_drv, dai_drv, 1);
}
-EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component);
MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h
index f8297d9bb8a3..90abab364b49 100644
--- a/sound/soc/samsung/s3c-i2s-v2.h
+++ b/sound/soc/samsung/s3c-i2s-v2.h
@@ -92,7 +92,7 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai,
unsigned long base);
/**
- * s3c_i2sv2_register_dai - register dai with soc core
+ * s3c_i2sv2_register_component - register component and dai with soc core
* @dev: DAI device
* @id: DAI ID
* @drv: The driver structure to register
@@ -100,7 +100,8 @@ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai,
* Fill in any missing fields and then register the given dai with the
* soc core.
*/
-extern int s3c_i2sv2_register_dai(struct device *dev, int id,
- struct snd_soc_dai_driver *drv);
+extern int s3c_i2sv2_register_component(struct device *dev, int id,
+ struct snd_soc_component_driver *cmp_drv,
+ struct snd_soc_dai_driver *dai_drv);
#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 221337716393..47e23864ea72 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -160,11 +160,17 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
.ops = &s3c2412_i2s_dai_ops,
};
+static const struct snd_soc_component_driver s3c2412_i2s_component = {
+ .name = "s3c2412-i2s",
+};
+
static int s3c2412_iis_dev_probe(struct platform_device *pdev)
{
int ret = 0;
- ret = s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai);
+ ret = s3c_i2sv2_register_component(&pdev->dev, -1,
+ &s3c2412_i2s_component,
+ &s3c2412_i2s_dai);
if (ret) {
pr_err("failed to register the dai\n");
return ret;
@@ -178,14 +184,14 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev)
return 0;
err:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return ret;
}
static int s3c2412_iis_dev_remove(struct platform_device *pdev)
{
asoc_dma_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 13f6dd1ceb00..8b3414551a62 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -24,7 +24,7 @@
#include <sound/pcm_params.h>
#include <mach/dma.h>
-#include <plat/regs-iis.h>
+#include "regs-iis.h"
#include "dma.h"
#include "s3c24xx-i2s.h"
@@ -465,11 +465,16 @@ static struct snd_soc_dai_driver s3c24xx_i2s_dai = {
.ops = &s3c24xx_i2s_dai_ops,
};
+static const struct snd_soc_component_driver s3c24xx_i2s_component = {
+ .name = "s3c24xx-i2s",
+};
+
static int s3c24xx_iis_dev_probe(struct platform_device *pdev)
{
int ret = 0;
- ret = snd_soc_register_dai(&pdev->dev, &s3c24xx_i2s_dai);
+ ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component,
+ &s3c24xx_i2s_dai, 1);
if (ret) {
pr_err("failed to register the dai\n");
return ret;
@@ -483,14 +488,14 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev)
return 0;
err:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return ret;
}
static int s3c24xx_iis_dev_remove(struct platform_device *pdev)
{
asoc_dma_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 333e1b7f06c7..1b7b52b0af97 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -18,7 +18,7 @@
#include <sound/soc.h>
#include <sound/s3c24xx_uda134x.h>
-#include <plat/regs-iis.h>
+#include "regs-iis.h"
#include "s3c24xx-i2s.h"
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 5008e5bd6ed8..2e5ebb2f1982 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -357,6 +357,10 @@ static struct snd_soc_dai_driver samsung_spdif_dai = {
.resume = spdif_resume,
};
+static const struct snd_soc_component_driver samsung_spdif_component = {
+ .name = "samsung-spdif",
+};
+
static int spdif_probe(struct platform_device *pdev)
{
struct s3c_audio_pdata *spdif_pdata;
@@ -424,7 +428,8 @@ static int spdif_probe(struct platform_device *pdev)
dev_set_drvdata(&pdev->dev, spdif);
- ret = snd_soc_register_dai(&pdev->dev, &samsung_spdif_dai);
+ ret = snd_soc_register_component(&pdev->dev, &samsung_spdif_component,
+ &samsung_spdif_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "fail to register dai\n");
goto err4;
@@ -445,7 +450,7 @@ static int spdif_probe(struct platform_device *pdev)
return 0;
err5:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
err4:
iounmap(spdif->regs);
err3:
@@ -466,7 +471,7 @@ static int spdif_remove(struct platform_device *pdev)
struct resource *mem_res;
asoc_dma_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
iounmap(spdif->regs);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index c724026a246f..f830c41f97dd 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -296,7 +296,6 @@ struct fsi_core {
struct fsi_master {
void __iomem *base;
- int irq;
struct fsi_priv fsia;
struct fsi_priv fsib;
const struct fsi_core *core;
@@ -1886,6 +1885,10 @@ static struct snd_soc_platform_driver fsi_soc_platform = {
.pcm_free = fsi_pcm_free,
};
+static const struct snd_soc_component_driver fsi_soc_component = {
+ .name = "fsi",
+};
+
/*
* platform function
*/
@@ -2002,7 +2005,6 @@ static int fsi_probe(struct platform_device *pdev)
}
/* master setting */
- master->irq = irq;
master->core = core;
spin_lock_init(&master->lock);
@@ -2046,10 +2048,10 @@ static int fsi_probe(struct platform_device *pdev)
goto exit_fsib;
}
- ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
- ARRAY_SIZE(fsi_soc_dai));
+ ret = snd_soc_register_component(&pdev->dev, &fsi_soc_component,
+ fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
if (ret < 0) {
- dev_err(&pdev->dev, "cannot snd dai register\n");
+ dev_err(&pdev->dev, "cannot snd component register\n");
goto exit_snd_soc;
}
@@ -2074,7 +2076,7 @@ static int fsi_remove(struct platform_device *pdev)
pm_runtime_disable(&pdev->dev);
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+ snd_soc_unregister_component(&pdev->dev);
snd_soc_unregister_platform(&pdev->dev);
fsi_stream_remove(&master->fsia);
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index 4cc2d64ef476..af19f77b7bf0 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -310,15 +310,19 @@ static struct snd_soc_dai_driver sh4_hac_dai[] = {
#endif
};
+static const struct snd_soc_component_driver sh4_hac_component = {
+ .name = "sh4-hac",
+};
+
static int hac_soc_platform_probe(struct platform_device *pdev)
{
- return snd_soc_register_dais(&pdev->dev, sh4_hac_dai,
- ARRAY_SIZE(sh4_hac_dai));
+ return snd_soc_register_component(&pdev->dev, &sh4_hac_component,
+ sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
}
static int hac_soc_platform_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_hac_dai));
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index 8526e1edaf45..5014a884afee 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -153,7 +153,7 @@ static int migor_dai_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link migor_dai = {
.name = "wm8978",
.stream_name = "WM8978",
- .cpu_dai_name = "siu-i2s-dai",
+ .cpu_dai_name = "siu-pcm-audio",
.codec_dai_name = "wm8978-hifi",
.platform_name = "siu-pcm-audio",
.codec_name = "wm8978.0-001a",
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index 34facdc9e4ac..9dc24ffa892a 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -726,6 +726,10 @@ static struct snd_soc_dai_driver siu_i2s_dai = {
.ops = &siu_dai_ops,
};
+static const struct snd_soc_component_driver siu_i2s_component = {
+ .name = "siu-i2s",
+};
+
static int siu_probe(struct platform_device *pdev)
{
const struct firmware *fw_entry;
@@ -783,7 +787,8 @@ static int siu_probe(struct platform_device *pdev)
dev_set_drvdata(&pdev->dev, info);
/* register using ARRAY version so we can keep dai name */
- ret = snd_soc_register_dais(&pdev->dev, &siu_i2s_dai, 1);
+ ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component,
+ &siu_i2s_dai, 1);
if (ret < 0)
goto edaiinit;
@@ -796,7 +801,7 @@ static int siu_probe(struct platform_device *pdev)
return ret;
esocregp:
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
edaiinit:
iounmap(info->reg);
emapreg:
@@ -823,7 +828,7 @@ static int siu_remove(struct platform_device *pdev)
pm_runtime_disable(&pdev->dev);
snd_soc_unregister_platform(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
iounmap(info->reg);
iounmap(info->yram);
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index c8e73a703934..e889405ebd38 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -379,15 +379,19 @@ static struct snd_soc_dai_driver sh4_ssi_dai[] = {
#endif
};
+static const struct snd_soc_component_driver sh4_ssi_component = {
+ .name = "sh4-ssi",
+};
+
static int sh4_soc_dai_probe(struct platform_device *pdev)
{
- return snd_soc_register_dais(&pdev->dev, sh4_ssi_dai,
- ARRAY_SIZE(sh4_ssi_dai));
+ return snd_soc_register_component(&pdev->dev, &sh4_ssi_component,
+ sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
}
static int sh4_soc_dai_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index ed0bfb0ddb96..3853f7eb3f28 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -315,7 +315,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream,
}
static int soc_compr_copy(struct snd_compr_stream *cstream,
- const char __user *buf, size_t count)
+ char __user *buf, size_t count)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_platform *platform = rtd->platform;
@@ -330,11 +330,38 @@ static int soc_compr_copy(struct snd_compr_stream *cstream,
return ret;
}
+static int sst_compr_set_metadata(struct snd_compr_stream *cstream,
+ struct snd_compr_metadata *metadata)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->set_metadata)
+ ret = platform->driver->compr_ops->set_metadata(cstream, metadata);
+
+ return ret;
+}
+
+static int sst_compr_get_metadata(struct snd_compr_stream *cstream,
+ struct snd_compr_metadata *metadata)
+{
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ int ret = 0;
+
+ if (platform->driver->compr_ops && platform->driver->compr_ops->get_metadata)
+ ret = platform->driver->compr_ops->get_metadata(cstream, metadata);
+
+ return ret;
+}
/* ASoC Compress operations */
static struct snd_compr_ops soc_compr_ops = {
.open = soc_compr_open,
.free = soc_compr_free,
.set_params = soc_compr_set_params,
+ .set_metadata = sst_compr_set_metadata,
+ .get_metadata = sst_compr_get_metadata,
.get_params = soc_compr_get_params,
.trigger = soc_compr_trigger,
.pointer = soc_compr_pointer,
@@ -357,7 +384,14 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
/* check client and interface hw capabilities */
snprintf(new_name, sizeof(new_name), "%s %s-%d",
rtd->dai_link->stream_name, codec_dai->name, num);
- direction = SND_COMPRESS_PLAYBACK;
+
+ if (codec_dai->driver->playback.channels_min)
+ direction = SND_COMPRESS_PLAYBACK;
+ else if (codec_dai->driver->capture.channels_min)
+ direction = SND_COMPRESS_CAPTURE;
+ else
+ return -EINVAL;
+
compr = kzalloc(sizeof(*compr), GFP_KERNEL);
if (compr == NULL) {
snd_printk(KERN_ERR "Cannot allocate compr\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ff4b45a5d796..d56bbea6e75e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -58,6 +58,7 @@ static DEFINE_MUTEX(client_mutex);
static LIST_HEAD(dai_list);
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
+static LIST_HEAD(component_list);
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -3740,7 +3741,7 @@ static inline char *fmt_multiple_name(struct device *dev,
*
* @dai: DAI to register
*/
-int snd_soc_register_dai(struct device *dev,
+static int snd_soc_register_dai(struct device *dev,
struct snd_soc_dai_driver *dai_drv)
{
struct snd_soc_codec *codec;
@@ -3787,14 +3788,13 @@ int snd_soc_register_dai(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_GPL(snd_soc_register_dai);
/**
* snd_soc_unregister_dai - Unregister a DAI from the ASoC core
*
* @dai: DAI to unregister
*/
-void snd_soc_unregister_dai(struct device *dev)
+static void snd_soc_unregister_dai(struct device *dev)
{
struct snd_soc_dai *dai;
@@ -3813,7 +3813,6 @@ found:
kfree(dai->name);
kfree(dai);
}
-EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
/**
* snd_soc_register_dais - Register multiple DAIs with the ASoC core
@@ -3821,7 +3820,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
* @dai: Array of DAIs to register
* @count: Number of DAIs
*/
-int snd_soc_register_dais(struct device *dev,
+static int snd_soc_register_dais(struct device *dev,
struct snd_soc_dai_driver *dai_drv, size_t count)
{
struct snd_soc_codec *codec;
@@ -3885,7 +3884,6 @@ err:
return ret;
}
-EXPORT_SYMBOL_GPL(snd_soc_register_dais);
/**
* snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
@@ -3893,31 +3891,23 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dais);
* @dai: Array of DAIs to unregister
* @count: Number of DAIs
*/
-void snd_soc_unregister_dais(struct device *dev, size_t count)
+static void snd_soc_unregister_dais(struct device *dev, size_t count)
{
int i;
for (i = 0; i < count; i++)
snd_soc_unregister_dai(dev);
}
-EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
/**
- * snd_soc_register_platform - Register a platform with the ASoC core
- *
- * @platform: platform to register
+ * snd_soc_add_platform - Add a platform to the ASoC core
+ * @dev: The parent device for the platform
+ * @platform: The platform to add
+ * @platform_driver: The driver for the platform
*/
-int snd_soc_register_platform(struct device *dev,
- struct snd_soc_platform_driver *platform_drv)
+int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
+ const struct snd_soc_platform_driver *platform_drv)
{
- struct snd_soc_platform *platform;
-
- dev_dbg(dev, "ASoC: platform register %s\n", dev_name(dev));
-
- platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL);
- if (platform == NULL)
- return -ENOMEM;
-
/* create platform component name */
platform->name = fmt_single_name(dev, &platform->id);
if (platform->name == NULL) {
@@ -3940,30 +3930,76 @@ int snd_soc_register_platform(struct device *dev,
return 0;
}
-EXPORT_SYMBOL_GPL(snd_soc_register_platform);
+EXPORT_SYMBOL_GPL(snd_soc_add_platform);
/**
- * snd_soc_unregister_platform - Unregister a platform from the ASoC core
+ * snd_soc_register_platform - Register a platform with the ASoC core
*
- * @platform: platform to unregister
+ * @platform: platform to register
*/
-void snd_soc_unregister_platform(struct device *dev)
+int snd_soc_register_platform(struct device *dev,
+ const struct snd_soc_platform_driver *platform_drv)
{
struct snd_soc_platform *platform;
+ int ret;
- list_for_each_entry(platform, &platform_list, list) {
- if (dev == platform->dev)
- goto found;
- }
- return;
+ dev_dbg(dev, "ASoC: platform register %s\n", dev_name(dev));
-found:
+ platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL);
+ if (platform == NULL)
+ return -ENOMEM;
+
+ ret = snd_soc_add_platform(dev, platform, platform_drv);
+ if (ret)
+ kfree(platform);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_platform);
+
+/**
+ * snd_soc_remove_platform - Remove a platform from the ASoC core
+ * @platform: the platform to remove
+ */
+void snd_soc_remove_platform(struct snd_soc_platform *platform)
+{
mutex_lock(&client_mutex);
list_del(&platform->list);
mutex_unlock(&client_mutex);
- dev_dbg(dev, "ASoC: Unregistered platform '%s'\n", platform->name);
+ dev_dbg(platform->dev, "ASoC: Unregistered platform '%s'\n",
+ platform->name);
kfree(platform->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_remove_platform);
+
+struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev)
+{
+ struct snd_soc_platform *platform;
+
+ list_for_each_entry(platform, &platform_list, list) {
+ if (dev == platform->dev)
+ return platform;
+ }
+
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_lookup_platform);
+
+/**
+ * snd_soc_unregister_platform - Unregister a platform from the ASoC core
+ *
+ * @platform: platform to unregister
+ */
+void snd_soc_unregister_platform(struct device *dev)
+{
+ struct snd_soc_platform *platform;
+
+ platform = snd_soc_lookup_platform(dev);
+ if (!platform)
+ return;
+
+ snd_soc_remove_platform(platform);
kfree(platform);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
@@ -4024,8 +4060,8 @@ int snd_soc_register_codec(struct device *dev,
/* create CODEC component name */
codec->name = fmt_single_name(dev, &codec->id);
if (codec->name == NULL) {
- kfree(codec);
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto fail_codec;
}
if (codec_drv->compress_type)
@@ -4064,7 +4100,7 @@ int snd_soc_register_codec(struct device *dev,
reg_size, GFP_KERNEL);
if (!codec->reg_def_copy) {
ret = -ENOMEM;
- goto fail;
+ goto fail_codec_name;
}
}
}
@@ -4088,18 +4124,22 @@ int snd_soc_register_codec(struct device *dev,
mutex_unlock(&client_mutex);
/* register any DAIs */
- if (num_dai) {
- ret = snd_soc_register_dais(dev, dai_drv, num_dai);
- if (ret < 0)
- dev_err(codec->dev, "ASoC: Failed to regster"
- " DAIs: %d\n", ret);
+ ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+ if (ret < 0) {
+ dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+ goto fail_codec_name;
}
dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name);
return 0;
-fail:
+fail_codec_name:
+ mutex_lock(&client_mutex);
+ list_del(&codec->list);
+ mutex_unlock(&client_mutex);
+
kfree(codec->name);
+fail_codec:
kfree(codec);
return ret;
}
@@ -4113,7 +4153,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec);
void snd_soc_unregister_codec(struct device *dev)
{
struct snd_soc_codec *codec;
- int i;
list_for_each_entry(codec, &codec_list, list) {
if (dev == codec->dev)
@@ -4122,9 +4161,7 @@ void snd_soc_unregister_codec(struct device *dev)
return;
found:
- if (codec->num_dai)
- for (i = 0; i < codec->num_dai; i++)
- snd_soc_unregister_dai(dev);
+ snd_soc_unregister_dais(dev, codec->num_dai);
mutex_lock(&client_mutex);
list_del(&codec->list);
@@ -4139,6 +4176,92 @@ found:
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
+
+/**
+ * snd_soc_register_component - Register a component with the ASoC core
+ *
+ */
+int snd_soc_register_component(struct device *dev,
+ const struct snd_soc_component_driver *cmpnt_drv,
+ struct snd_soc_dai_driver *dai_drv,
+ int num_dai)
+{
+ struct snd_soc_component *cmpnt;
+ int ret;
+
+ dev_dbg(dev, "component register %s\n", dev_name(dev));
+
+ cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL);
+ if (!cmpnt) {
+ dev_err(dev, "ASoC: Failed to allocate memory\n");
+ return -ENOMEM;
+ }
+
+ cmpnt->name = fmt_single_name(dev, &cmpnt->id);
+ if (!cmpnt->name) {
+ dev_err(dev, "ASoC: Failed to simplifying name\n");
+ return -ENOMEM;
+ }
+
+ cmpnt->dev = dev;
+ cmpnt->driver = cmpnt_drv;
+ cmpnt->num_dai = num_dai;
+
+ /*
+ * snd_soc_register_dai() uses fmt_single_name(), and
+ * snd_soc_register_dais() uses fmt_multiple_name()
+ * for dai->name which is used for name based matching
+ */
+ if (1 == num_dai)
+ ret = snd_soc_register_dai(dev, dai_drv);
+ else
+ ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+ if (ret < 0) {
+ dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+ goto error_component_name;
+ }
+
+ mutex_lock(&client_mutex);
+ list_add(&cmpnt->list, &component_list);
+ mutex_unlock(&client_mutex);
+
+ dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name);
+
+ return ret;
+
+error_component_name:
+ kfree(cmpnt->name);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_component);
+
+/**
+ * snd_soc_unregister_component - Unregister a component from the ASoC core
+ *
+ */
+void snd_soc_unregister_component(struct device *dev)
+{
+ struct snd_soc_component *cmpnt;
+
+ list_for_each_entry(cmpnt, &component_list, list) {
+ if (dev == cmpnt->dev)
+ goto found;
+ }
+ return;
+
+found:
+ snd_soc_unregister_dais(dev, cmpnt->num_dai);
+
+ mutex_lock(&client_mutex);
+ list_del(&cmpnt->list);
+ mutex_unlock(&client_mutex);
+
+ dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name);
+ kfree(cmpnt->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
+
/* Retrieve a card's name from device tree */
int snd_soc_of_parse_card_name(struct snd_soc_card *card,
const char *propname)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d6d9ba2e6916..21779a6a781a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -504,17 +504,27 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
return 0;
}
-/* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
+/*
+ * Determine if a kcontrol is shared. If it is, look it up. If it isn't,
+ * create it. Either way, add the widget into the control's widget list
+ */
+static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
+ int kci, struct snd_soc_dapm_path *path)
{
struct snd_soc_dapm_context *dapm = w->dapm;
- int i, ret = 0;
- size_t name_len, prefix_len;
- struct snd_soc_dapm_path *path;
struct snd_card *card = dapm->card->snd_card;
const char *prefix;
+ size_t prefix_len;
+ int shared;
+ struct snd_kcontrol *kcontrol;
struct snd_soc_dapm_widget_list *wlist;
+ int wlistentries;
size_t wlistsize;
+ bool wname_in_long_name, kcname_in_long_name;
+ size_t name_len;
+ char *long_name;
+ const char *name;
+ int ret;
if (dapm->codec)
prefix = dapm->codec->name_prefix;
@@ -526,103 +536,141 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
else
prefix_len = 0;
- /* add kcontrol */
- for (i = 0; i < w->num_kcontrols; i++) {
+ shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[kci],
+ &kcontrol);
- /* match name */
- list_for_each_entry(path, &w->sources, list_sink) {
+ if (kcontrol) {
+ wlist = kcontrol->private_data;
+ wlistentries = wlist->num_widgets + 1;
+ } else {
+ wlist = NULL;
+ wlistentries = 1;
+ }
- /* mixer/mux paths name must match control name */
- if (path->name != (char *)w->kcontrol_news[i].name)
- continue;
+ wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
+ wlistentries * sizeof(struct snd_soc_dapm_widget *);
+ wlist = krealloc(wlist, wlistsize, GFP_KERNEL);
+ if (wlist == NULL) {
+ dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n",
+ w->name);
+ return -ENOMEM;
+ }
+ wlist->num_widgets = wlistentries;
+ wlist->widgets[wlistentries - 1] = w;
- if (w->kcontrols[i]) {
- path->kcontrol = w->kcontrols[i];
- continue;
+ if (!kcontrol) {
+ if (shared) {
+ wname_in_long_name = false;
+ kcname_in_long_name = true;
+ } else {
+ switch (w->id) {
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer:
+ wname_in_long_name = true;
+ kcname_in_long_name = true;
+ break;
+ case snd_soc_dapm_mixer_named_ctl:
+ wname_in_long_name = false;
+ kcname_in_long_name = true;
+ break;
+ case snd_soc_dapm_mux:
+ case snd_soc_dapm_virt_mux:
+ case snd_soc_dapm_value_mux:
+ wname_in_long_name = true;
+ kcname_in_long_name = false;
+ break;
+ default:
+ kfree(wlist);
+ return -EINVAL;
}
+ }
+
+ if (wname_in_long_name && kcname_in_long_name) {
+ name_len = strlen(w->name) - prefix_len + 1 +
+ strlen(w->kcontrol_news[kci].name) + 1;
- wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
- sizeof(struct snd_soc_dapm_widget *),
- wlist = kzalloc(wlistsize, GFP_KERNEL);
- if (wlist == NULL) {
- dev_err(dapm->dev,
- "ASoC: can't allocate widget list for %s\n",
- w->name);
+ long_name = kmalloc(name_len, GFP_KERNEL);
+ if (long_name == NULL) {
+ kfree(wlist);
return -ENOMEM;
}
- wlist->num_widgets = 1;
- wlist->widgets[0] = w;
-
- /* add dapm control with long name.
- * for dapm_mixer this is the concatenation of the
- * mixer and kcontrol name.
- * for dapm_mixer_named_ctl this is simply the
- * kcontrol name.
+
+ /*
+ * The control will get a prefix from the control
+ * creation process but we're also using the same
+ * prefix for widgets so cut the prefix off the
+ * front of the widget name.
*/
- name_len = strlen(w->kcontrol_news[i].name) + 1;
- if (w->id != snd_soc_dapm_mixer_named_ctl)
- name_len += 1 + strlen(w->name);
+ snprintf(long_name, name_len, "%s %s",
+ w->name + prefix_len,
+ w->kcontrol_news[kci].name);
+ long_name[name_len - 1] = '\0';
+
+ name = long_name;
+ } else if (wname_in_long_name) {
+ long_name = NULL;
+ name = w->name + prefix_len;
+ } else {
+ long_name = NULL;
+ name = w->kcontrol_news[kci].name;
+ }
- path->long_name = kmalloc(name_len, GFP_KERNEL);
+ kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name,
+ prefix);
+ ret = snd_ctl_add(card, kcontrol);
+ if (ret < 0) {
+ dev_err(dapm->dev,
+ "ASoC: failed to add widget %s dapm kcontrol %s: %d\n",
+ w->name, name, ret);
+ kfree(wlist);
+ kfree(long_name);
+ return ret;
+ }
- if (path->long_name == NULL) {
- kfree(wlist);
- return -ENOMEM;
- }
+ path->long_name = long_name;
+ }
- switch (w->id) {
- default:
- /* The control will get a prefix from
- * the control creation process but
- * we're also using the same prefix
- * for widgets so cut the prefix off
- * the front of the widget name.
- */
- snprintf((char *)path->long_name, name_len,
- "%s %s", w->name + prefix_len,
- w->kcontrol_news[i].name);
- break;
- case snd_soc_dapm_mixer_named_ctl:
- snprintf((char *)path->long_name, name_len,
- "%s", w->kcontrol_news[i].name);
- break;
- }
+ kcontrol->private_data = wlist;
+ w->kcontrols[kci] = kcontrol;
+ path->kcontrol = kcontrol;
- ((char *)path->long_name)[name_len - 1] = '\0';
+ return 0;
+}
- path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i],
- wlist, path->long_name,
- prefix);
- ret = snd_ctl_add(card, path->kcontrol);
- if (ret < 0) {
- dev_err(dapm->dev, "ASoC: failed to add widget"
- " %s dapm kcontrol %s: %d\n",
- w->name, path->long_name, ret);
- kfree(wlist);
- kfree(path->long_name);
- path->long_name = NULL;
- return ret;
+/* create new dapm mixer control */
+static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
+{
+ int i, ret;
+ struct snd_soc_dapm_path *path;
+
+ /* add kcontrol */
+ for (i = 0; i < w->num_kcontrols; i++) {
+ /* match name */
+ list_for_each_entry(path, &w->sources, list_sink) {
+ /* mixer/mux paths name must match control name */
+ if (path->name != (char *)w->kcontrol_news[i].name)
+ continue;
+
+ if (w->kcontrols[i]) {
+ path->kcontrol = w->kcontrols[i];
+ continue;
}
- w->kcontrols[i] = path->kcontrol;
+
+ ret = dapm_create_or_share_mixmux_kcontrol(w, i, path);
+ if (ret < 0)
+ return ret;
}
}
- return ret;
+
+ return 0;
}
/* create new dapm mux control */
static int dapm_new_mux(struct snd_soc_dapm_widget *w)
{
struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_dapm_path *path = NULL;
- struct snd_kcontrol *kcontrol;
- struct snd_card *card = dapm->card->snd_card;
- const char *prefix;
- size_t prefix_len;
+ struct snd_soc_dapm_path *path;
int ret;
- struct snd_soc_dapm_widget_list *wlist;
- int shared, wlistentries;
- size_t wlistsize;
- const char *name;
if (w->num_kcontrols != 1) {
dev_err(dapm->dev,
@@ -631,65 +679,19 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
return -EINVAL;
}
- shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0],
- &kcontrol);
- if (kcontrol) {
- wlist = kcontrol->private_data;
- wlistentries = wlist->num_widgets + 1;
- } else {
- wlist = NULL;
- wlistentries = 1;
- }
- wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
- wlistentries * sizeof(struct snd_soc_dapm_widget *),
- wlist = krealloc(wlist, wlistsize, GFP_KERNEL);
- if (wlist == NULL) {
- dev_err(dapm->dev,
- "ASoC: can't allocate widget list for %s\n", w->name);
- return -ENOMEM;
- }
- wlist->num_widgets = wlistentries;
- wlist->widgets[wlistentries - 1] = w;
-
- if (!kcontrol) {
- if (dapm->codec)
- prefix = dapm->codec->name_prefix;
- else
- prefix = NULL;
-
- if (shared) {
- name = w->kcontrol_news[0].name;
- prefix_len = 0;
- } else {
- name = w->name;
- if (prefix)
- prefix_len = strlen(prefix) + 1;
- else
- prefix_len = 0;
- }
-
- /*
- * The control will get a prefix from the control creation
- * process but we're also using the same prefix for widgets so
- * cut the prefix off the front of the widget name.
- */
- kcontrol = snd_soc_cnew(&w->kcontrol_news[0], wlist,
- name + prefix_len, prefix);
- ret = snd_ctl_add(card, kcontrol);
- if (ret < 0) {
- dev_err(dapm->dev, "ASoC: failed to add kcontrol %s: %d\n",
- w->name, ret);
- kfree(wlist);
- return ret;
- }
+ path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
+ list_sink);
+ if (!path) {
+ dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name);
+ return -EINVAL;
}
- kcontrol->private_data = wlist;
-
- w->kcontrols[0] = kcontrol;
+ ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path);
+ if (ret < 0)
+ return ret;
list_for_each_entry(path, &w->sources, list_sink)
- path->kcontrol = kcontrol;
+ path->kcontrol = w->kcontrols[0];
return 0;
}
@@ -705,14 +707,33 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w)
}
/* reset 'walked' bit for each dapm path */
-static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm)
+static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm,
+ struct list_head *sink)
{
struct snd_soc_dapm_path *p;
- list_for_each_entry(p, &dapm->card->paths, list)
- p->walked = 0;
+ list_for_each_entry(p, sink, list_source) {
+ if (p->walked) {
+ p->walked = 0;
+ dapm_clear_walk_output(dapm, &p->sink->sinks);
+ }
+ }
}
+static void dapm_clear_walk_input(struct snd_soc_dapm_context *dapm,
+ struct list_head *source)
+{
+ struct snd_soc_dapm_path *p;
+
+ list_for_each_entry(p, source, list_sink) {
+ if (p->walked) {
+ p->walked = 0;
+ dapm_clear_walk_input(dapm, &p->source->sources);
+ }
+ }
+}
+
+
/* We implement power down on suspend by checking the power state of
* the ALSA card - when we are suspending the ALSA state for the card
* is set to D3.
@@ -995,13 +1016,17 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
dapm_reset(card);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
paths = is_connected_output_ep(dai->playback_widget, list);
- else
+ dapm_clear_walk_output(&card->dapm,
+ &dai->playback_widget->sinks);
+ } else {
paths = is_connected_input_ep(dai->capture_widget, list);
+ dapm_clear_walk_input(&card->dapm,
+ &dai->capture_widget->sources);
+ }
trace_snd_soc_dapm_connected(paths, stream);
- dapm_clear_walk(&card->dapm);
mutex_unlock(&card->dapm_mutex);
return paths;
@@ -1104,9 +1129,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
DAPM_UPDATE_STAT(w, power_checks);
in = is_connected_input_ep(w, NULL);
- dapm_clear_walk(w->dapm);
+ dapm_clear_walk_input(w->dapm, &w->sources);
out = is_connected_output_ep(w, NULL);
- dapm_clear_walk(w->dapm);
+ dapm_clear_walk_output(w->dapm, &w->sinks);
return out != 0 && in != 0;
}
@@ -1129,7 +1154,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
if (w->active) {
in = is_connected_input_ep(w, NULL);
- dapm_clear_walk(w->dapm);
+ dapm_clear_walk_input(w->dapm, &w->sources);
return in != 0;
} else {
return dapm_generic_check_power(w);
@@ -1145,7 +1170,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w)
if (w->active) {
out = is_connected_output_ep(w, NULL);
- dapm_clear_walk(w->dapm);
+ dapm_clear_walk_output(w->dapm, &w->sinks);
return out != 0;
} else {
return dapm_generic_check_power(w);
@@ -1177,8 +1202,6 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
return 1;
}
- dapm_clear_walk(w->dapm);
-
return 0;
}
@@ -1759,9 +1782,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
return -ENOMEM;
in = is_connected_input_ep(w, NULL);
- dapm_clear_walk(w->dapm);
+ dapm_clear_walk_input(w->dapm, &w->sources);
out = is_connected_output_ep(w, NULL);
- dapm_clear_walk(w->dapm);
+ dapm_clear_walk_output(w->dapm, &w->sinks);
ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
w->name, w->power ? "On" : "Off",
@@ -3137,7 +3160,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
break;
}
- dapm->n_widgets++;
w->dapm = dapm;
w->codec = dapm->codec;
w->platform = dapm->platform;
diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c
index 111b7d921e89..aa924d9b7986 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/soc/soc-dmaengine-pcm.c
@@ -33,8 +33,6 @@ struct dmaengine_pcm_runtime_data {
dma_cookie_t cookie;
unsigned int pos;
-
- void *data;
};
static inline struct dmaengine_pcm_runtime_data *substream_to_prtd(
@@ -43,33 +41,6 @@ static inline struct dmaengine_pcm_runtime_data *substream_to_prtd(
return substream->runtime->private_data;
}
-/**
- * snd_dmaengine_pcm_set_data - Set dmaengine substream private data
- * @substream: PCM substream
- * @data: Data to set
- */
-void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data)
-{
- struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
-
- prtd->data = data;
-}
-EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_data);
-
-/**
- * snd_dmaengine_pcm_get_data - Get dmaeinge substream private data
- * @substream: PCM substream
- *
- * Returns the data previously set with snd_dmaengine_pcm_set_data
- */
-void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream)
-{
- struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
-
- return prtd->data;
-}
-EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_data);
-
struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
{
struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
@@ -118,10 +89,49 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
slave_config->src_addr_width = buswidth;
}
+ slave_config->device_fc = false;
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config);
+/**
+ * snd_dmaengine_pcm_set_config_from_dai_data() - Initializes a dma slave config
+ * using DAI DMA data.
+ * @substream: PCM substream
+ * @dma_data: DAI DMA data
+ * @slave_config: DMA slave configuration
+ *
+ * Initializes the {dst,src}_addr, {dst,src}_maxburst, {dst,src}_addr_width and
+ * slave_id fields of the DMA slave config from the same fields of the DAI DMA
+ * data struct. The src and dst fields will be initialized depending on the
+ * direction of the substream. If the substream is a playback stream the dst
+ * fields will be initialized, if it is a capture stream the src fields will be
+ * initialized. The {dst,src}_addr_width field will only be initialized if the
+ * addr_width field of the DAI DMA data struct is not equal to
+ * DMA_SLAVE_BUSWIDTH_UNDEFINED.
+ */
+void snd_dmaengine_pcm_set_config_from_dai_data(
+ const struct snd_pcm_substream *substream,
+ const struct snd_dmaengine_dai_dma_data *dma_data,
+ struct dma_slave_config *slave_config)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ slave_config->dst_addr = dma_data->addr;
+ slave_config->dst_maxburst = dma_data->maxburst;
+ if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
+ slave_config->dst_addr_width = dma_data->addr_width;
+ } else {
+ slave_config->src_addr = dma_data->addr;
+ slave_config->src_maxburst = dma_data->maxburst;
+ if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
+ slave_config->src_addr_width = dma_data->addr_width;
+ }
+
+ slave_config->slave_id = dma_data->slave_id;
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data);
+
static void dmaengine_pcm_dma_complete(void *arg)
{
struct snd_pcm_substream *substream = arg;
@@ -244,44 +254,48 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
-static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd,
- dma_filter_fn filter_fn, void *filter_data)
+/**
+ * snd_dmaengine_pcm_request_channel - Request channel for the dmaengine PCM
+ * @filter_fn: Filter function used to request the DMA channel
+ * @filter_data: Data passed to the DMA filter function
+ *
+ * Returns NULL or the requested DMA channel.
+ *
+ * This function request a DMA channel for usage with dmaengine PCM.
+ */
+struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn,
+ void *filter_data)
{
dma_cap_mask_t mask;
dma_cap_zero(mask);
dma_cap_set(DMA_SLAVE, mask);
dma_cap_set(DMA_CYCLIC, mask);
- prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data);
- if (!prtd->dma_chan)
- return -ENXIO;
-
- return 0;
+ return dma_request_channel(mask, filter_fn, filter_data);
}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel);
/**
* snd_dmaengine_pcm_open - Open a dmaengine based PCM substream
* @substream: PCM substream
- * @filter_fn: Filter function used to request the DMA channel
- * @filter_data: Data passed to the DMA filter function
+ * @chan: DMA channel to use for data transfers
*
* Returns 0 on success, a negative error code otherwise.
*
- * This function will request a DMA channel using the passed filter function and
- * data. The function should usually be called from the pcm open callback.
- *
- * Note that this function will use private_data field of the substream's
- * runtime. So it is not availabe to your pcm driver implementation. If you need
- * to keep additional data attached to a substream use
- * snd_dmaengine_pcm_{set,get}_data.
+ * The function should usually be called from the pcm open callback. Note that
+ * this function will use private_data field of the substream's runtime. So it
+ * is not availabe to your pcm driver implementation.
*/
int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
- dma_filter_fn filter_fn, void *filter_data)
+ struct dma_chan *chan)
{
struct dmaengine_pcm_runtime_data *prtd;
int ret;
+ if (!chan)
+ return -ENXIO;
+
ret = snd_pcm_hw_constraint_integer(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
@@ -291,11 +305,7 @@ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
if (!prtd)
return -ENOMEM;
- ret = dmaengine_pcm_request_channel(prtd, filter_fn, filter_data);
- if (ret < 0) {
- kfree(prtd);
- return ret;
- }
+ prtd->dma_chan = chan;
substream->runtime->private_data = prtd;
@@ -304,6 +314,27 @@ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open);
/**
+ * snd_dmaengine_pcm_open_request_chan - Open a dmaengine based PCM substream and request channel
+ * @substream: PCM substream
+ * @filter_fn: Filter function used to request the DMA channel
+ * @filter_data: Data passed to the DMA filter function
+ *
+ * Returns 0 on success, a negative error code otherwise.
+ *
+ * This function will request a DMA channel using the passed filter function and
+ * data. The function should usually be called from the pcm open callback. Note
+ * that this function will use private_data field of the substream's runtime. So
+ * it is not availabe to your pcm driver implementation.
+ */
+int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
+ dma_filter_fn filter_fn, void *filter_data)
+{
+ return snd_dmaengine_pcm_open(substream,
+ snd_dmaengine_pcm_request_channel(filter_fn, filter_data));
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan);
+
+/**
* snd_dmaengine_pcm_close - Close a dmaengine based PCM substream
* @substream: PCM substream
*/
@@ -311,11 +342,26 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream)
{
struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
- dma_release_channel(prtd->dma_chan);
kfree(prtd);
return 0;
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close);
+/**
+ * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel
+ * @substream: PCM substream
+ *
+ * Releases the DMA channel associated with the PCM substream.
+ */
+int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream)
+{
+ struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+
+ dma_release_channel(prtd->dma_chan);
+
+ return snd_dmaengine_pcm_close(substream);
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close_release_chan);
+
MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
new file mode 100644
index 000000000000..e29ec3cd84b1
--- /dev/null
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -0,0 +1,300 @@
+/*
+ * Copyright (C) 2013, Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ * 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.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/dmaengine.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <linux/dma-mapping.h>
+#include <linux/of.h>
+
+#include <sound/dmaengine_pcm.h>
+
+struct dmaengine_pcm {
+ struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1];
+ const struct snd_dmaengine_pcm_config *config;
+ struct snd_soc_platform platform;
+ unsigned int flags;
+};
+
+static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p)
+{
+ return container_of(p, struct dmaengine_pcm, platform);
+}
+
+/**
+ * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback
+ * @substream: PCM substream
+ * @params: hw_params
+ * @slave_config: DMA slave config to prepare
+ *
+ * This function can be used as a generic prepare_slave_config callback for
+ * platforms which make use of the snd_dmaengine_dai_dma_data struct for their
+ * DAI DMA data. Internally the function will first call
+ * snd_hwparams_to_dma_slave_config to fill in the slave config based on the
+ * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the
+ * remaining fields based on the DAI DMA data.
+ */
+int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
+ int ret;
+
+ dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+ ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
+ if (ret)
+ return ret;
+
+ snd_dmaengine_pcm_set_config_from_dai_data(substream, dma_data,
+ slave_config);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_prepare_slave_config);
+
+static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct dma_slave_config slave_config;
+ int ret;
+
+ if (pcm->config->prepare_slave_config) {
+ ret = pcm->config->prepare_slave_config(substream, params,
+ &slave_config);
+ if (ret)
+ return ret;
+
+ ret = dmaengine_slave_config(chan, &slave_config);
+ if (ret)
+ return ret;
+ }
+
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+}
+
+static int dmaengine_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+ struct dma_chan *chan = pcm->chan[substream->stream];
+ int ret;
+
+ ret = snd_soc_set_runtime_hwparams(substream,
+ pcm->config->pcm_hardware);
+ if (ret)
+ return ret;
+
+ return snd_dmaengine_pcm_open(substream, chan);
+}
+
+static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm,
+ struct snd_pcm_substream *substream)
+{
+ if (!pcm->chan[substream->stream])
+ return NULL;
+
+ return pcm->chan[substream->stream]->device->dev;
+}
+
+static void dmaengine_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static struct dma_chan *dmaengine_pcm_compat_request_channel(
+ struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_substream *substream)
+{
+ struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+
+ if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0])
+ return pcm->chan[0];
+
+ if (pcm->config->compat_request_channel)
+ return pcm->config->compat_request_channel(rtd, substream);
+
+ return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn,
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream));
+}
+
+static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+ const struct snd_dmaengine_pcm_config *config = pcm->config;
+ struct snd_pcm_substream *substream;
+ unsigned int i;
+ int ret;
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+ substream = rtd->pcm->streams[i].substream;
+ if (!substream)
+ continue;
+
+ if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) {
+ pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd,
+ substream);
+ }
+
+ if (!pcm->chan[i]) {
+ dev_err(rtd->platform->dev,
+ "Missing dma channel for stream: %d\n", i);
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ ret = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ dmaengine_dma_dev(pcm, substream),
+ config->prealloc_buffer_size,
+ config->pcm_hardware->buffer_bytes_max);
+ if (ret)
+ goto err_free;
+ }
+
+ return 0;
+
+err_free:
+ dmaengine_pcm_free(rtd->pcm);
+ return ret;
+}
+
+static const struct snd_pcm_ops dmaengine_pcm_ops = {
+ .open = dmaengine_pcm_open,
+ .close = snd_dmaengine_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = dmaengine_pcm_hw_params,
+ .hw_free = snd_pcm_lib_free_pages,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+};
+
+static const struct snd_soc_platform_driver dmaengine_pcm_platform = {
+ .ops = &dmaengine_pcm_ops,
+ .pcm_new = dmaengine_pcm_new,
+ .pcm_free = dmaengine_pcm_free,
+ .probe_order = SND_SOC_COMP_ORDER_LATE,
+};
+
+static const struct snd_pcm_ops dmaengine_no_residue_pcm_ops = {
+ .open = dmaengine_pcm_open,
+ .close = snd_dmaengine_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = dmaengine_pcm_hw_params,
+ .hw_free = snd_pcm_lib_free_pages,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer_no_residue,
+};
+
+static const struct snd_soc_platform_driver dmaengine_no_residue_pcm_platform = {
+ .ops = &dmaengine_no_residue_pcm_ops,
+ .pcm_new = dmaengine_pcm_new,
+ .pcm_free = dmaengine_pcm_free,
+ .probe_order = SND_SOC_COMP_ORDER_LATE,
+};
+
+static const char * const dmaengine_pcm_dma_channel_names[] = {
+ [SNDRV_PCM_STREAM_PLAYBACK] = "tx",
+ [SNDRV_PCM_STREAM_CAPTURE] = "rx",
+};
+
+static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
+ struct device *dev)
+{
+ unsigned int i;
+
+ if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node)
+ return;
+
+ if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) {
+ pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx");
+ pcm->chan[1] = pcm->chan[0];
+ } else {
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+ pcm->chan[i] = dma_request_slave_channel(dev,
+ dmaengine_pcm_dma_channel_names[i]);
+ }
+ }
+}
+
+/**
+ * snd_dmaengine_pcm_register - Register a dmaengine based PCM device
+ * @dev: The parent device for the PCM device
+ * @config: Platform specific PCM configuration
+ * @flags: Platform specific quirks
+ */
+int snd_dmaengine_pcm_register(struct device *dev,
+ const struct snd_dmaengine_pcm_config *config, unsigned int flags)
+{
+ struct dmaengine_pcm *pcm;
+
+ pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ pcm->config = config;
+ pcm->flags = flags;
+
+ dmaengine_pcm_request_chan_of(pcm, dev);
+
+ if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
+ return snd_soc_add_platform(dev, &pcm->platform,
+ &dmaengine_no_residue_pcm_platform);
+ else
+ return snd_soc_add_platform(dev, &pcm->platform,
+ &dmaengine_pcm_platform);
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register);
+
+/**
+ * snd_dmaengine_pcm_unregister - Removes a dmaengine based PCM device
+ * @dev: Parent device the PCM was register with
+ *
+ * Removes a dmaengine based PCM device previously registered with
+ * snd_dmaengine_pcm_register.
+ */
+void snd_dmaengine_pcm_unregister(struct device *dev)
+{
+ struct snd_soc_platform *platform;
+ struct dmaengine_pcm *pcm;
+ unsigned int i;
+
+ platform = snd_soc_lookup_platform(dev);
+ if (!platform)
+ return;
+
+ pcm = soc_platform_to_pcm(platform);
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+ if (pcm->chan[i]) {
+ dma_release_channel(pcm->chan[i]);
+ if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+ break;
+ }
+ }
+
+ snd_soc_remove_platform(platform);
+ kfree(pcm);
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 29183ef2b93d..8ca9ecc5ac57 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -158,10 +158,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
return -EINVAL;
}
- if (IS_ERR(codec->control_data))
- return PTR_ERR(codec->control_data);
-
- return 0;
+ return PTR_RET(codec->control_data);
}
EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
#else
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index fe4541df498c..4b3be6c3c91e 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -90,8 +90,33 @@ static struct snd_soc_platform_driver dummy_platform = {
};
static struct snd_soc_codec_driver dummy_codec;
+
+#define STUB_RATES SNDRV_PCM_RATE_8000_192000
+#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_U24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | \
+ SNDRV_PCM_FMTBIT_U32_LE | \
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
static struct snd_soc_dai_driver dummy_dai = {
.name = "snd-soc-dummy-dai",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 384,
+ .rates = STUB_RATES,
+ .formats = STUB_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 384,
+ .rates = STUB_RATES,
+ .formats = STUB_FORMATS,
+ },
};
static int snd_soc_dummy_probe(struct platform_device *pdev)
diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index c7c4b20395bb..14d57e89bcba 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -170,6 +170,10 @@ struct snd_soc_dai_driver spdif_in_dai = {
.ops = &spdif_in_dai_ops,
};
+static const struct snd_soc_component_driver spdif_in_component = {
+ .name = "spdif-in",
+};
+
static irqreturn_t spdif_in_irq(int irq, void *arg)
{
struct spdif_in_dev *host = (struct spdif_in_dev *)arg;
@@ -258,7 +262,8 @@ static int spdif_in_probe(struct platform_device *pdev)
return ret;
}
- ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai);
+ ret = snd_soc_register_component(&pdev->dev, &spdif_in_component,
+ &spdif_in_dai, 1);
if (ret != 0) {
clk_put(host->clk);
return ret;
@@ -271,7 +276,7 @@ static int spdif_in_remove(struct platform_device *pdev)
{
struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
dev_set_drvdata(&pdev->dev, NULL);
clk_put(host->clk);
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 5eac4cda2fd7..1e3c3dda3598 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -270,6 +270,10 @@ static struct snd_soc_dai_driver spdif_out_dai = {
.ops = &spdif_out_dai_ops,
};
+static const struct snd_soc_component_driver spdif_out_component = {
+ .name = "spdif-out",
+};
+
static int spdif_out_probe(struct platform_device *pdev)
{
struct spdif_out_dev *host;
@@ -314,7 +318,8 @@ static int spdif_out_probe(struct platform_device *pdev)
dev_set_drvdata(&pdev->dev, host);
- ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai);
+ ret = snd_soc_register_component(&pdev->dev, &spdif_out_component,
+ &spdif_out_dai, 1);
if (ret != 0) {
clk_put(host->clk);
return ret;
@@ -327,7 +332,7 @@ static int spdif_out_remove(struct platform_device *pdev)
{
struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
dev_set_drvdata(&pdev->dev, NULL);
clk_put(host->clk);
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 5e7aebe1e664..2fbd4899d8ef 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -25,7 +25,7 @@
#include <sound/soc.h>
#include <sound/spear_dma.h>
-struct snd_pcm_hardware spear_pcm_hardware = {
+static struct snd_pcm_hardware spear_pcm_hardware = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
@@ -64,21 +64,8 @@ static int spear_pcm_open(struct snd_pcm_substream *substream)
if (ret)
return ret;
- ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data);
- if (ret)
- return ret;
-
- snd_dmaengine_pcm_set_data(substream, dma_data);
-
- return 0;
-}
-
-static int spear_pcm_close(struct snd_pcm_substream *substream)
-{
-
- snd_dmaengine_pcm_close(substream);
-
- return 0;
+ return snd_dmaengine_pcm_open_request_chan(substream, dma_data->filter,
+ dma_data);
}
static int spear_pcm_mmap(struct snd_pcm_substream *substream,
@@ -93,7 +80,7 @@ static int spear_pcm_mmap(struct snd_pcm_substream *substream,
static struct snd_pcm_ops spear_pcm_ops = {
.open = spear_pcm_open,
- .close = spear_pcm_close,
+ .close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = spear_pcm_hw_params,
.hw_free = spear_pcm_hw_free,
@@ -178,7 +165,7 @@ static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-struct snd_soc_platform_driver spear_soc_platform = {
+static struct snd_soc_platform_driver spear_soc_platform = {
.ops = &spear_pcm_ops,
.pcm_new = spear_pcm_new,
.pcm_free = spear_pcm_free,
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index dbc27ce1d4de..b1c9d573da05 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -2,7 +2,7 @@ config SND_SOC_TEGRA
tristate "SoC Audio for the Tegra System-on-Chip"
depends on ARCH_TEGRA && TEGRA20_APB_DMA
select REGMAP_MMIO
- select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA
+ select SND_SOC_GENERIC_DMAENGINE_PCM if TEGRA20_APB_DMA
help
Say Y or M here if you want support for SoC audio on Tegra.
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index 336dcdd3e8a4..2f70ea7f6618 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -35,6 +35,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include "tegra_asoc_utils.h"
#include "tegra20_ac97.h"
@@ -248,6 +249,10 @@ static struct snd_soc_dai_driver tegra20_ac97_dai = {
.ops = &tegra20_ac97_dai_ops,
};
+static const struct snd_soc_component_driver tegra20_ac97_component = {
+ .name = DRV_NAME,
+};
+
static bool tegra20_ac97_wr_rd_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -389,16 +394,17 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
}
ac97->capture_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_RX1;
- ac97->capture_dma_data.wrap = 4;
- ac97->capture_dma_data.width = 32;
- ac97->capture_dma_data.req_sel = of_dma[1];
+ ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ ac97->capture_dma_data.maxburst = 4;
+ ac97->capture_dma_data.slave_id = of_dma[1];
ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1;
- ac97->playback_dma_data.wrap = 4;
- ac97->playback_dma_data.width = 32;
- ac97->playback_dma_data.req_sel = of_dma[1];
+ ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ ac97->capture_dma_data.maxburst = 4;
+ ac97->capture_dma_data.slave_id = of_dma[0];
- ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1);
+ ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component,
+ &tegra20_ac97_dai, 1);
if (ret) {
dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
ret = -ENOMEM;
@@ -408,7 +414,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
ret = tegra_pcm_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
- goto err_unregister_dai;
+ goto err_unregister_component;
}
ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev);
@@ -434,8 +440,8 @@ err_asoc_utils_fini:
tegra_asoc_utils_fini(&ac97->util_data);
err_unregister_pcm:
tegra_pcm_platform_unregister(&pdev->dev);
-err_unregister_dai:
- snd_soc_unregister_dai(&pdev->dev);
+err_unregister_component:
+ snd_soc_unregister_component(&pdev->dev);
err_clk_put:
clk_put(ac97->clk_ac97);
err:
@@ -447,7 +453,7 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev)
struct tegra20_ac97 *ac97 = dev_get_drvdata(&pdev->dev);
tegra_pcm_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
tegra_asoc_utils_fini(&ac97->util_data);
diff --git a/sound/soc/tegra/tegra20_ac97.h b/sound/soc/tegra/tegra20_ac97.h
index dddc6828004e..4acb3aaba29b 100644
--- a/sound/soc/tegra/tegra20_ac97.h
+++ b/sound/soc/tegra/tegra20_ac97.h
@@ -85,8 +85,8 @@
struct tegra20_ac97 {
struct clk *clk_ac97;
- struct tegra_pcm_dma_params capture_dma_data;
- struct tegra_pcm_dma_params playback_dma_data;
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
struct regmap *regmap;
int reset_gpio;
int sync_gpio;
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index caa772de5a18..52af7f6fb37f 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -41,6 +41,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include "tegra20_i2s.h"
@@ -276,6 +277,10 @@ static const struct snd_soc_dai_driver tegra20_i2s_dai_template = {
.symmetric_rates = 1,
};
+static const struct snd_soc_component_driver tegra20_i2s_component = {
+ .name = DRV_NAME,
+};
+
static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -403,14 +408,14 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev)
}
i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2;
- i2s->capture_dma_data.wrap = 4;
- i2s->capture_dma_data.width = 32;
- i2s->capture_dma_data.req_sel = dma_ch;
+ i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ i2s->capture_dma_data.maxburst = 4;
+ i2s->capture_dma_data.slave_id = dma_ch;
i2s->playback_dma_data.addr = mem->start + TEGRA20_I2S_FIFO1;
- i2s->playback_dma_data.wrap = 4;
- i2s->playback_dma_data.width = 32;
- i2s->playback_dma_data.req_sel = dma_ch;
+ i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ i2s->playback_dma_data.maxburst = 4;
+ i2s->playback_dma_data.slave_id = dma_ch;
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
@@ -419,7 +424,8 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev)
goto err_pm_disable;
}
- ret = snd_soc_register_dai(&pdev->dev, &i2s->dai);
+ ret = snd_soc_register_component(&pdev->dev, &tegra20_i2s_component,
+ &i2s->dai, 1);
if (ret) {
dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
ret = -ENOMEM;
@@ -429,13 +435,13 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev)
ret = tegra_pcm_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
- goto err_unregister_dai;
+ goto err_unregister_component;
}
return 0;
-err_unregister_dai:
- snd_soc_unregister_dai(&pdev->dev);
+err_unregister_component:
+ snd_soc_unregister_component(&pdev->dev);
err_suspend:
if (!pm_runtime_status_suspended(&pdev->dev))
tegra20_i2s_runtime_suspend(&pdev->dev);
@@ -456,7 +462,7 @@ static int tegra20_i2s_platform_remove(struct platform_device *pdev)
tegra20_i2s_runtime_suspend(&pdev->dev);
tegra_pcm_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
clk_put(i2s->clk_i2s);
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index 729958713cd4..fa6c29cc12b9 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -155,8 +155,8 @@
struct tegra20_i2s {
struct snd_soc_dai_driver dai;
struct clk *clk_i2s;
- struct tegra_pcm_dma_params capture_dma_data;
- struct tegra_pcm_dma_params playback_dma_data;
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
struct regmap *regmap;
};
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 04771d14d343..5eaa12cdc6eb 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -32,6 +32,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
#include "tegra20_spdif.h"
@@ -182,6 +183,10 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = {
.ops = &tegra20_spdif_dai_ops,
};
+static const struct snd_soc_component_driver tegra20_spdif_component = {
+ .name = DRV_NAME,
+};
+
static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
@@ -318,9 +323,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev)
}
spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT;
- spdif->playback_dma_data.wrap = 4;
- spdif->playback_dma_data.width = 32;
- spdif->playback_dma_data.req_sel = dmareq->start;
+ spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ spdif->capture_dma_data.maxburst = 4;
+ spdif->playback_dma_data.slave_id = dmareq->start;
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
@@ -329,7 +334,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev)
goto err_pm_disable;
}
- ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai);
+ ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component,
+ &tegra20_spdif_dai, 1);
if (ret) {
dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
ret = -ENOMEM;
@@ -339,13 +345,13 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev)
ret = tegra_pcm_platform_register(&pdev->dev);
if (ret) {
dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
- goto err_unregister_dai;
+ goto err_unregister_component;
}
return 0;
-err_unregister_dai:
- snd_soc_unregister_dai(&pdev->dev);
+err_unregister_component:
+ snd_soc_unregister_component(&pdev->dev);
err_suspend:
if (!pm_runtime_status_suspended(&pdev->dev))
tegra20_spdif_runtime_suspend(&pdev->dev);
@@ -366,7 +372,7 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev)
tegra20_spdif_runtime_suspend(&pdev->dev);
tegra_pcm_platform_unregister(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
clk_put(spdif->clk_spdif_out);
diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h
index b48d699fd583..85a9aefcc287 100644
--- a/sound/soc/tegra/tegra20_spdif.h
+++ b/sound/soc/tegra/tegra20_spdif.h
@@ -462,8 +462,8 @@
struct tegra20_spdif {
struct clk *clk_spdif_out;
- struct tegra_pcm_dma_params capture_dma_data;
- struct tegra_pcm_dma_params playback_dma_data;
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
struct regmap *regmap;
};
diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index e5cfb4ac41ba..23e592f453fa 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -95,8 +95,8 @@ static int tegra30_ahub_runtime_resume(struct device *dev)
}
int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif,
- unsigned long *fiforeg,
- unsigned long *reqsel)
+ dma_addr_t *fiforeg,
+ unsigned int *reqsel)
{
int channel;
u32 reg, val;
@@ -178,8 +178,8 @@ int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif)
EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo);
int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif,
- unsigned long *fiforeg,
- unsigned long *reqsel)
+ dma_addr_t *fiforeg,
+ unsigned int *reqsel)
{
int channel;
u32 reg, val;
@@ -287,16 +287,27 @@ int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif)
}
EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source);
-static const char * const configlink_clocks[] = {
- "i2s0",
- "i2s1",
- "i2s2",
- "i2s3",
- "i2s4",
- "dam0",
- "dam1",
- "dam2",
- "spdif_in",
+#define CLK_LIST_MASK_TEGRA30 BIT(0)
+#define CLK_LIST_MASK_TEGRA114 BIT(1)
+
+#define CLK_LIST_MASK_TEGRA30_OR_LATER \
+ (CLK_LIST_MASK_TEGRA30 | CLK_LIST_MASK_TEGRA114)
+
+static const struct {
+ const char *clk_name;
+ u32 clk_list_mask;
+} configlink_clocks[] = {
+ { "i2s0", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "i2s1", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "i2s2", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "i2s3", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "i2s4", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "dam0", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "dam1", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "dam2", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "spdif_in", CLK_LIST_MASK_TEGRA30_OR_LATER },
+ { "amx", CLK_LIST_MASK_TEGRA114 },
+ { "adx", CLK_LIST_MASK_TEGRA114 },
};
#define LAST_REG(name) \
@@ -424,8 +435,24 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = {
.cache_type = REGCACHE_RBTREE,
};
+static struct tegra30_ahub_soc_data soc_data_tegra30 = {
+ .clk_list_mask = CLK_LIST_MASK_TEGRA30,
+};
+
+static struct tegra30_ahub_soc_data soc_data_tegra114 = {
+ .clk_list_mask = CLK_LIST_MASK_TEGRA114,
+};
+
+static const struct of_device_id tegra30_ahub_of_match[] = {
+ { .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 },
+ { .compatible = "nvidia,tegra30-ahub", .data = &soc_data_tegra30 },
+ {},
+};
+
static int tegra30_ahub_probe(struct platform_device *pdev)
{
+ const struct of_device_id *match;
+ const struct tegra30_ahub_soc_data *soc_data;
struct clk *clk;
int i;
struct resource *res0, *res1, *region;
@@ -436,16 +463,24 @@ static int tegra30_ahub_probe(struct platform_device *pdev)
if (ahub)
return -ENODEV;
+ match = of_match_device(tegra30_ahub_of_match, &pdev->dev);
+ if (!match)
+ return -EINVAL;
+ soc_data = match->data;
+
/*
* The AHUB hosts a register bus: the "configlink". For this to
* operate correctly, all devices on this bus must be out of reset.
* Ensure that here.
*/
for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) {
- clk = clk_get(&pdev->dev, configlink_clocks[i]);
+ if (!(configlink_clocks[i].clk_list_mask &
+ soc_data->clk_list_mask))
+ continue;
+ clk = clk_get(&pdev->dev, configlink_clocks[i].clk_name);
if (IS_ERR(clk)) {
dev_err(&pdev->dev, "Can't get clock %s\n",
- configlink_clocks[i]);
+ configlink_clocks[i].clk_name);
ret = PTR_ERR(clk);
goto err;
}
@@ -592,11 +627,6 @@ static int tegra30_ahub_remove(struct platform_device *pdev)
return 0;
}
-static const struct of