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-rw-r--r--sound/atmel/ac97c.c2
-rw-r--r--sound/core/jack.c5
-rw-r--r--sound/core/pcm_lib.c41
-rw-r--r--sound/core/pcm_native.c12
-rw-r--r--sound/core/sound_oss.c6
-rw-r--r--sound/drivers/aloop.c62
-rw-r--r--sound/firewire/amdtp.c49
-rw-r--r--sound/firewire/amdtp.h29
-rw-r--r--sound/firewire/cmp.c2
-rw-r--r--sound/firewire/lib.c28
-rw-r--r--sound/firewire/lib.h1
-rw-r--r--sound/i2c/other/tea575x-tuner.c3
-rw-r--r--sound/isa/als100.c2
-rw-r--r--sound/pci/Kconfig2
-rw-r--r--sound/pci/ad1889.c15
-rw-r--r--sound/pci/ali5451/ali5451.c15
-rw-r--r--sound/pci/als300.c15
-rw-r--r--sound/pci/als4000.c15
-rw-r--r--sound/pci/atiixp.c16
-rw-r--r--sound/pci/atiixp_modem.c16
-rw-r--r--sound/pci/au88x0/au88x0.c17
-rw-r--r--sound/pci/aw2/aw2-alsa.c23
-rw-r--r--sound/pci/azt3328.c23
-rw-r--r--sound/pci/bt87x.c19
-rw-r--r--sound/pci/ca0106/ca0106_main.c17
-rw-r--r--sound/pci/cmipci.c15
-rw-r--r--sound/pci/cs4281.c15
-rw-r--r--sound/pci/cs46xx/cs46xx.c15
-rw-r--r--sound/pci/cs5530.c16
-rw-r--r--sound/pci/cs5535audio/cs5535audio.c15
-rw-r--r--sound/pci/ctxfi/xfi.c13
-rw-r--r--sound/pci/echoaudio/echoaudio.c22
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.c2
-rw-r--r--sound/pci/emu10k1/emu10k1.c15
-rw-r--r--sound/pci/emu10k1/emu10k1x.c17
-rw-r--r--sound/pci/ens1370.c15
-rw-r--r--sound/pci/es1938.c15
-rw-r--r--sound/pci/es1968.c15
-rw-r--r--sound/pci/fm801.c15
-rw-r--r--sound/pci/hda/Makefile2
-rw-r--r--sound/pci/hda/hda_auto_parser.c760
-rw-r--r--sound/pci/hda/hda_auto_parser.h160
-rw-r--r--sound/pci/hda/hda_codec.c1097
-rw-r--r--sound/pci/hda/hda_codec.h18
-rw-r--r--sound/pci/hda/hda_intel.c363
-rw-r--r--sound/pci/hda/hda_jack.c1
-rw-r--r--sound/pci/hda/hda_jack.h2
-rw-r--r--sound/pci/hda/hda_local.h122
-rw-r--r--sound/pci/hda/patch_analog.c14
-rw-r--r--sound/pci/hda/patch_ca0110.c8
-rw-r--r--sound/pci/hda/patch_ca0132.c9
-rw-r--r--sound/pci/hda/patch_cirrus.c30
-rw-r--r--sound/pci/hda/patch_cmedia.c1
-rw-r--r--sound/pci/hda/patch_conexant.c186
-rw-r--r--sound/pci/hda/patch_hdmi.c4
-rw-r--r--sound/pci/hda/patch_realtek.c521
-rw-r--r--sound/pci/hda/patch_sigmatel.c120
-rw-r--r--sound/pci/hda/patch_via.c33
-rw-r--r--sound/pci/ice1712/ice1712.c15
-rw-r--r--sound/pci/ice1712/ice1724.c15
-rw-r--r--sound/pci/intel8x0.c16
-rw-r--r--sound/pci/intel8x0m.c16
-rw-r--r--sound/pci/korg1212/korg1212.c15
-rw-r--r--sound/pci/lola/lola.c15
-rw-r--r--sound/pci/lx6464es/lx6464es.c17
-rw-r--r--sound/pci/maestro3.c15
-rw-r--r--sound/pci/mixart/mixart.c15
-rw-r--r--sound/pci/nm256/nm256.c16
-rw-r--r--sound/pci/oxygen/oxygen.c21
-rw-r--r--sound/pci/oxygen/virtuoso.c13
-rw-r--r--sound/pci/oxygen/xonar_dg.c7
-rw-r--r--sound/pci/pcxhr/pcxhr.c15
-rw-r--r--sound/pci/riptide/riptide.c3
-rw-r--r--sound/pci/rme32.c15
-rw-r--r--sound/pci/rme96.c15
-rw-r--r--sound/pci/rme9652/hdsp.c16
-rw-r--r--sound/pci/rme9652/hdspm.c16
-rw-r--r--sound/pci/rme9652/rme9652.c15
-rw-r--r--sound/pci/sis7019.c13
-rw-r--r--sound/pci/sonicvibes.c15
-rw-r--r--sound/pci/trident/trident.c15
-rw-r--r--sound/pci/via82xx.c15
-rw-r--r--sound/pci/via82xx_modem.c15
-rw-r--r--sound/pci/vx222/vx222.c15
-rw-r--r--sound/pci/ymfpci/ymfpci.c15
-rw-r--r--sound/sh/sh_dac_audio.c4
-rw-r--r--sound/soc/Kconfig5
-rw-r--r--sound/soc/Makefile3
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c37
-rw-r--r--sound/soc/codecs/Kconfig20
-rw-r--r--sound/soc/codecs/Makefile12
-rw-r--r--sound/soc/codecs/ac97.c6
-rw-r--r--sound/soc/codecs/ad1836.c4
-rw-r--r--sound/soc/codecs/ad193x.c4
-rw-r--r--sound/soc/codecs/adau1701.c3
-rw-r--r--sound/soc/codecs/ak4104.c3
-rw-r--r--sound/soc/codecs/ak4535.c3
-rw-r--r--sound/soc/codecs/ak4641.c113
-rw-r--r--sound/soc/codecs/alc5623.c23
-rw-r--r--sound/soc/codecs/alc5632.c31
-rw-r--r--sound/soc/codecs/cs4270.c11
-rw-r--r--sound/soc/codecs/cs4271.c3
-rw-r--r--sound/soc/codecs/cs42l51.c9
-rw-r--r--sound/soc/codecs/cs42l52.c1295
-rw-r--r--sound/soc/codecs/cs42l52.h274
-rw-r--r--sound/soc/codecs/cs42l73.c109
-rw-r--r--sound/soc/codecs/da7210.c379
-rw-r--r--sound/soc/codecs/jz4740.c3
-rw-r--r--sound/soc/codecs/lm49453.c1550
-rw-r--r--sound/soc/codecs/lm49453.h380
-rw-r--r--sound/soc/codecs/max98095.c158
-rw-r--r--sound/soc/codecs/max98095.h22
-rw-r--r--sound/soc/codecs/mc13783.c786
-rw-r--r--sound/soc/codecs/mc13783.h28
-rw-r--r--sound/soc/codecs/ml26124.c681
-rw-r--r--sound/soc/codecs/ml26124.h184
-rw-r--r--sound/soc/codecs/omap-hdmi.c69
-rw-r--r--sound/soc/codecs/rt5631.c110
-rw-r--r--sound/soc/codecs/sgtl5000.c33
-rw-r--r--sound/soc/codecs/ssm2602.c138
-rw-r--r--sound/soc/codecs/sta32x.c3
-rw-r--r--sound/soc/codecs/tlv320aic23.c13
-rw-r--r--sound/soc/codecs/tlv320aic26.c3
-rw-r--r--sound/soc/codecs/tlv320aic3x.c21
-rw-r--r--sound/soc/codecs/tlv320dac33.c35
-rw-r--r--sound/soc/codecs/twl4030.c18
-rw-r--r--sound/soc/codecs/twl6040.c450
-rw-r--r--sound/soc/codecs/uda134x.c6
-rw-r--r--sound/soc/codecs/uda1380.c6
-rw-r--r--sound/soc/codecs/wl1273.c6
-rw-r--r--sound/soc/codecs/wm1250-ev1.c65
-rw-r--r--sound/soc/codecs/wm5100-tables.c125
-rw-r--r--sound/soc/codecs/wm5100.c47
-rw-r--r--sound/soc/codecs/wm5100.h159
-rw-r--r--sound/soc/codecs/wm8350.c187
-rw-r--r--sound/soc/codecs/wm8400.c135
-rw-r--r--sound/soc/codecs/wm8510.c3
-rw-r--r--sound/soc/codecs/wm8523.c3
-rw-r--r--sound/soc/codecs/wm8728.c3
-rw-r--r--sound/soc/codecs/wm8731.c37
-rw-r--r--sound/soc/codecs/wm8737.c3
-rw-r--r--sound/soc/codecs/wm8741.c3
-rw-r--r--sound/soc/codecs/wm8750.c3
-rw-r--r--sound/soc/codecs/wm8753.c6
-rw-r--r--sound/soc/codecs/wm8900.c3
-rw-r--r--sound/soc/codecs/wm8903.c3
-rw-r--r--sound/soc/codecs/wm8940.c3
-rw-r--r--sound/soc/codecs/wm8960.c3
-rw-r--r--sound/soc/codecs/wm8962.c18
-rw-r--r--sound/soc/codecs/wm8971.c3
-rw-r--r--sound/soc/codecs/wm8978.c3
-rw-r--r--sound/soc/codecs/wm8988.c3
-rw-r--r--sound/soc/codecs/wm8990.c3
-rw-r--r--sound/soc/codecs/wm8993.c86
-rw-r--r--sound/soc/codecs/wm8994.c292
-rw-r--r--sound/soc/codecs/wm8994.h3
-rw-r--r--sound/soc/codecs/wm8996.c12
-rw-r--r--sound/soc/codecs/wm9081.c5
-rw-r--r--sound/soc/codecs/wm9705.c6
-rw-r--r--sound/soc/codecs/wm9712.c10
-rw-r--r--sound/soc/codecs/wm_hubs.c220
-rw-r--r--sound/soc/codecs/wm_hubs.h12
-rw-r--r--sound/soc/ep93xx/ep93xx-ac97.c74
-rw-r--r--sound/soc/ep93xx/ep93xx-i2s.c49
-rw-r--r--sound/soc/fsl/Kconfig129
-rw-r--r--sound/soc/fsl/Makefile31
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c (renamed from sound/soc/imx/eukrea-tlv320.c)2
-rw-r--r--sound/soc/fsl/fsl_ssi.c167
-rw-r--r--sound/soc/fsl/fsl_utils.c91
-rw-r--r--sound/soc/fsl/fsl_utils.h26
-rw-r--r--sound/soc/fsl/imx-audmux.c (renamed from sound/soc/imx/imx-audmux.c)0
-rw-r--r--sound/soc/fsl/imx-audmux.h (renamed from sound/soc/imx/imx-audmux.h)0
-rw-r--r--sound/soc/fsl/imx-mc13783.c156
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c (renamed from sound/soc/imx/imx-pcm-dma-mx2.c)3
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c (renamed from sound/soc/imx/imx-pcm-fiq.c)0
-rw-r--r--sound/soc/fsl/imx-pcm.c (renamed from sound/soc/imx/imx-pcm.c)0
-rw-r--r--sound/soc/fsl/imx-pcm.h (renamed from sound/soc/imx/imx-pcm.h)1
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c221
-rw-r--r--sound/soc/fsl/imx-ssi.c (renamed from sound/soc/imx/imx-ssi.c)2
-rw-r--r--sound/soc/fsl/imx-ssi.h (renamed from sound/soc/imx/imx-ssi.h)0
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c166
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c (renamed from sound/soc/imx/mx27vis-aic32x4.c)0
-rw-r--r--sound/soc/fsl/p1022_ds.c158
-rw-r--r--sound/soc/fsl/phycore-ac97.c (renamed from sound/soc/imx/phycore-ac97.c)0
-rw-r--r--sound/soc/fsl/wm1133-ev1.c (renamed from sound/soc/imx/wm1133-ev1.c)0
-rw-r--r--sound/soc/generic/Kconfig4
-rw-r--r--sound/soc/generic/Makefile3
-rw-r--r--sound/soc/generic/simple-card.c114
-rw-r--r--sound/soc/imx/Kconfig79
-rw-r--r--sound/soc/imx/Makefile22
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c4
-rw-r--r--sound/soc/mxs/mxs-pcm.c24
-rw-r--r--sound/soc/mxs/mxs-pcm.h3
-rw-r--r--sound/soc/mxs/mxs-saif.c92
-rw-r--r--sound/soc/mxs/mxs-saif.h1
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c50
-rw-r--r--sound/soc/omap/Kconfig8
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/mcbsp.c115
-rw-r--r--sound/soc/omap/mcbsp.h8
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c68
-rw-r--r--sound/soc/omap/omap-dmic.c8
-rw-r--r--sound/soc/omap/omap-hdmi-card.c87
-rw-r--r--sound/soc/omap/omap-hdmi.c238
-rw-r--r--sound/soc/omap/omap-hdmi.h4
-rw-r--r--sound/soc/omap/omap-mcbsp.c45
-rw-r--r--sound/soc/omap/omap-mcpdm.c8
-rw-r--r--sound/soc/omap/omap4-hdmi-card.c121
-rw-r--r--sound/soc/pxa/pxa-ssp.c28
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c4
-rw-r--r--sound/soc/samsung/littlemill.c102
-rw-r--r--sound/soc/samsung/lowland.c75
-rw-r--r--sound/soc/samsung/speyside.c33
-rw-r--r--sound/soc/sh/Kconfig24
-rw-r--r--sound/soc/sh/Makefile6
-rw-r--r--sound/soc/sh/fsi-ak4642.c108
-rw-r--r--sound/soc/sh/fsi-da7210.c81
-rw-r--r--sound/soc/sh/fsi-hdmi.c118
-rw-r--r--sound/soc/sh/fsi.c224
-rw-r--r--sound/soc/sh/migor.c2
-rw-r--r--sound/soc/soc-core.c690
-rw-r--r--sound/soc/soc-dapm.c562
-rw-r--r--sound/soc/soc-jack.c5
-rw-r--r--sound/soc/soc-pcm.c1718
-rw-r--r--sound/soc/tegra/Kconfig68
-rw-r--r--sound/soc/tegra/Makefile20
-rw-r--r--sound/soc/tegra/tegra20_das.c233
-rw-r--r--sound/soc/tegra/tegra20_das.h134
-rw-r--r--sound/soc/tegra/tegra20_i2s.c494
-rw-r--r--sound/soc/tegra/tegra20_i2s.h164
-rw-r--r--sound/soc/tegra/tegra20_spdif.c404
-rw-r--r--sound/soc/tegra/tegra20_spdif.h471
-rw-r--r--sound/soc/tegra/tegra30_ahub.c631
-rw-r--r--sound/soc/tegra/tegra30_ahub.h483
-rw-r--r--sound/soc/tegra/tegra30_i2s.c536
-rw-r--r--sound/soc/tegra/tegra30_i2s.h242
-rw-r--r--sound/soc/tegra/tegra_alc5632.c48
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.c37
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h9
-rw-r--r--sound/soc/tegra/tegra_das.c261
-rw-r--r--sound/soc/tegra/tegra_das.h135
-rw-r--r--sound/soc/tegra/tegra_i2s.c459
-rw-r--r--sound/soc/tegra/tegra_i2s.h166
-rw-r--r--sound/soc/tegra/tegra_pcm.c28
-rw-r--r--sound/soc/tegra/tegra_pcm.h5
-rw-r--r--sound/soc/tegra/tegra_spdif.c364
-rw-r--r--sound/soc/tegra/tegra_spdif.h473
-rw-r--r--sound/soc/tegra/tegra_wm8753.c224
-rw-r--r--sound/soc/tegra/tegra_wm8903.c29
-rw-r--r--sound/soc/tegra/trimslice.c41
-rw-r--r--sound/soc/ux500/Kconfig14
-rw-r--r--sound/soc/ux500/Makefile4
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c843
-rw-r--r--sound/soc/ux500/ux500_msp_dai.h79
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.c742
-rw-r--r--sound/soc/ux500/ux500_msp_i2s.h553
-rw-r--r--sound/sound_core.c2
-rw-r--r--sound/usb/card.c10
-rw-r--r--sound/usb/card.h86
-rw-r--r--sound/usb/endpoint.c1609
-rw-r--r--sound/usb/endpoint.h32
-rw-r--r--sound/usb/mixer.c50
-rw-r--r--sound/usb/mixer.h3
-rw-r--r--sound/usb/mixer_maps.c13
-rw-r--r--sound/usb/mixer_quirks.c472
-rw-r--r--sound/usb/pcm.c453
-rw-r--r--sound/usb/proc.c38
-rw-r--r--sound/usb/stream.c31
-rw-r--r--sound/usb/usbaudio.h2
269 files changed, 22060 insertions, 8307 deletions
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 115313ef54d..f5ded640b39 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -991,6 +991,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
gpio_direction_output(pdata->reset_pin, 1);
chip->reset_pin = pdata->reset_pin;
}
+ } else {
+ chip->reset_pin = -EINVAL;
}
snd_card_set_dev(card, &pdev->dev);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 471e1e3b0a9..a06b1651fcb 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -155,7 +155,7 @@ EXPORT_SYMBOL(snd_jack_new);
* @jack: The jack to configure
* @parent: The device to set as parent for the jack.
*
- * Set the parent for the jack input device in the device tree. This
+ * Set the parent for the jack devices in the device tree. This
* function is only valid prior to registration of the jack. If no
* parent is configured then the parent device will be the sound card.
*/
@@ -179,6 +179,9 @@ EXPORT_SYMBOL(snd_jack_set_parent);
* mapping is provided but keys are enabled in the jack type then
* BTN_n numeric buttons will be reported.
*
+ * If jacks are not reporting via the input API this call will have no
+ * effect.
+ *
* Note that this is intended to be use by simple devices with small
* numbers of keys that can be reported. It is also possible to
* access the input device directly - devices with complex input
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4d18941178e..8f312fa6c28 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -313,9 +313,22 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base;
snd_pcm_sframes_t hdelta, delta;
unsigned long jdelta;
+ unsigned long curr_jiffies;
+ struct timespec curr_tstamp;
old_hw_ptr = runtime->status->hw_ptr;
+
+ /*
+ * group pointer, time and jiffies reads to allow for more
+ * accurate correlations/corrections.
+ * The values are stored at the end of this routine after
+ * corrections for hw_ptr position
+ */
pos = substream->ops->pointer(substream);
+ curr_jiffies = jiffies;
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp);
+
if (pos == SNDRV_PCM_POS_XRUN) {
xrun(substream);
return -EPIPE;
@@ -343,7 +356,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
delta = runtime->hw_ptr_interrupt + runtime->period_size;
if (delta > new_hw_ptr) {
/* check for double acknowledged interrupts */
- hdelta = jiffies - runtime->hw_ptr_jiffies;
+ hdelta = curr_jiffies - runtime->hw_ptr_jiffies;
if (hdelta > runtime->hw_ptr_buffer_jiffies/2) {
hw_base += runtime->buffer_size;
if (hw_base >= runtime->boundary)
@@ -388,7 +401,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
* Without regular period interrupts, we have to check
* the elapsed time to detect xruns.
*/
- jdelta = jiffies - runtime->hw_ptr_jiffies;
+ jdelta = curr_jiffies - runtime->hw_ptr_jiffies;
if (jdelta < runtime->hw_ptr_buffer_jiffies / 2)
goto no_delta_check;
hdelta = jdelta - delta * HZ / runtime->rate;
@@ -430,7 +443,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
if (hdelta < runtime->delay)
goto no_jiffies_check;
hdelta -= runtime->delay;
- jdelta = jiffies - runtime->hw_ptr_jiffies;
+ jdelta = curr_jiffies - runtime->hw_ptr_jiffies;
if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
delta = jdelta /
(((runtime->period_size * HZ) / runtime->rate)
@@ -492,9 +505,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
}
runtime->hw_ptr_base = hw_base;
runtime->status->hw_ptr = new_hw_ptr;
- runtime->hw_ptr_jiffies = jiffies;
+ runtime->hw_ptr_jiffies = curr_jiffies;
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
- snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
+ runtime->status->tstamp = curr_tstamp;
return snd_pcm_update_state(substream, runtime);
}
@@ -1894,6 +1907,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t xfer = 0;
snd_pcm_uframes_t offset = 0;
+ snd_pcm_uframes_t avail;
int err = 0;
if (size == 0)
@@ -1917,13 +1931,12 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
}
runtime->twake = runtime->control->avail_min ? : 1;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+ avail = snd_pcm_playback_avail(runtime);
while (size > 0) {
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
- snd_pcm_uframes_t avail;
snd_pcm_uframes_t cont;
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- snd_pcm_update_hw_ptr(substream);
- avail = snd_pcm_playback_avail(runtime);
if (!avail) {
if (nonblock) {
err = -EAGAIN;
@@ -1971,6 +1984,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
offset += frames;
size -= frames;
xfer += frames;
+ avail -= frames;
if (runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) {
err = snd_pcm_start(substream);
@@ -2111,6 +2125,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t xfer = 0;
snd_pcm_uframes_t offset = 0;
+ snd_pcm_uframes_t avail;
int err = 0;
if (size == 0)
@@ -2141,13 +2156,12 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
}
runtime->twake = runtime->control->avail_min ? : 1;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+ avail = snd_pcm_capture_avail(runtime);
while (size > 0) {
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
- snd_pcm_uframes_t avail;
snd_pcm_uframes_t cont;
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- snd_pcm_update_hw_ptr(substream);
- avail = snd_pcm_capture_avail(runtime);
if (!avail) {
if (runtime->status->state ==
SNDRV_PCM_STATE_DRAINING) {
@@ -2202,6 +2216,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
offset += frames;
size -= frames;
xfer += frames;
+ avail -= frames;
}
_end_unlock:
runtime->twake = 0;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 3fe99e644eb..53b5ada8f7c 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1360,7 +1360,14 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state)
{
- substream->runtime->trigger_master = substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_DISCONNECTED:
+ case SNDRV_PCM_STATE_SUSPENDED:
+ return -EBADFD;
+ }
+ runtime->trigger_master = substream;
return 0;
}
@@ -1379,6 +1386,9 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
case SNDRV_PCM_STATE_RUNNING:
runtime->status->state = SNDRV_PCM_STATE_DRAINING;
break;
+ case SNDRV_PCM_STATE_XRUN:
+ runtime->status->state = SNDRV_PCM_STATE_SETUP;
+ break;
default:
break;
}
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index c7009204306..e9528333e36 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -35,7 +35,7 @@
#include <linux/sound.h>
#include <linux/mutex.h>
-#define SNDRV_OSS_MINORS 128
+#define SNDRV_OSS_MINORS 256
static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS];
static DEFINE_MUTEX(sound_oss_mutex);
@@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev,
int register1 = -1, register2 = -1;
struct device *carddev = snd_card_get_device_link(card);
- if (card && card->number >= 8)
+ if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
return 0; /* ignore silently */
if (minor < 0)
return minor;
@@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev)
int track2 = -1;
struct snd_minor *mptr;
- if (card && card->number >= 8)
+ if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
return 0;
if (minor < 0)
return minor;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index ad079b63b8b..8b5c36f4d30 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -117,6 +117,7 @@ struct loopback_pcm {
/* timer stuff */
unsigned int irq_pos; /* fractional IRQ position */
unsigned int period_size_frac;
+ unsigned int last_drift;
unsigned long last_jiffies;
struct timer_list timer;
};
@@ -264,6 +265,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd)
return err;
dpcm->last_jiffies = jiffies;
dpcm->pcm_rate_shift = 0;
+ dpcm->last_drift = 0;
spin_lock(&cable->lock);
cable->running |= stream;
cable->pause &= ~stream;
@@ -444,34 +446,30 @@ static void copy_play_buf(struct loopback_pcm *play,
}
}
-#define BYTEPOS_UPDATE_POSONLY 0
-#define BYTEPOS_UPDATE_CLEAR 1
-#define BYTEPOS_UPDATE_COPY 2
-
-static void loopback_bytepos_update(struct loopback_pcm *dpcm,
- unsigned int delta,
- unsigned int cmd)
+static inline unsigned int bytepos_delta(struct loopback_pcm *dpcm,
+ unsigned int jiffies_delta)
{
- unsigned int count;
unsigned long last_pos;
+ unsigned int delta;
last_pos = byte_pos(dpcm, dpcm->irq_pos);
- dpcm->irq_pos += delta * dpcm->pcm_bps;
- count = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
- if (!count)
- return;
- if (cmd == BYTEPOS_UPDATE_CLEAR)
- clear_capture_buf(dpcm, count);
- else if (cmd == BYTEPOS_UPDATE_COPY)
- copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK],
- dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE],
- count);
- dpcm->buf_pos += count;
- dpcm->buf_pos %= dpcm->pcm_buffer_size;
+ dpcm->irq_pos += jiffies_delta * dpcm->pcm_bps;
+ delta = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
+ if (delta >= dpcm->last_drift)
+ delta -= dpcm->last_drift;
+ dpcm->last_drift = 0;
if (dpcm->irq_pos >= dpcm->period_size_frac) {
dpcm->irq_pos %= dpcm->period_size_frac;
dpcm->period_update_pending = 1;
}
+ return delta;
+}
+
+static inline void bytepos_finish(struct loopback_pcm *dpcm,
+ unsigned int delta)
+{
+ dpcm->buf_pos += delta;
+ dpcm->buf_pos %= dpcm->pcm_buffer_size;
}
static unsigned int loopback_pos_update(struct loopback_cable *cable)
@@ -481,7 +479,7 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
struct loopback_pcm *dpcm_capt =
cable->streams[SNDRV_PCM_STREAM_CAPTURE];
unsigned long delta_play = 0, delta_capt = 0;
- unsigned int running;
+ unsigned int running, count1, count2;
unsigned long flags;
spin_lock_irqsave(&cable->lock, flags);
@@ -500,12 +498,13 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
goto unlock;
if (delta_play > delta_capt) {
- loopback_bytepos_update(dpcm_play, delta_play - delta_capt,
- BYTEPOS_UPDATE_POSONLY);
+ count1 = bytepos_delta(dpcm_play, delta_play - delta_capt);
+ bytepos_finish(dpcm_play, count1);
delta_play = delta_capt;
} else if (delta_play < delta_capt) {
- loopback_bytepos_update(dpcm_capt, delta_capt - delta_play,
- BYTEPOS_UPDATE_CLEAR);
+ count1 = bytepos_delta(dpcm_capt, delta_capt - delta_play);
+ clear_capture_buf(dpcm_capt, count1);
+ bytepos_finish(dpcm_capt, count1);
delta_capt = delta_play;
}
@@ -513,8 +512,17 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
goto unlock;
/* note delta_capt == delta_play at this moment */
- loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY);
- loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY);
+ count1 = bytepos_delta(dpcm_play, delta_play);
+ count2 = bytepos_delta(dpcm_capt, delta_capt);
+ if (count1 < count2) {
+ dpcm_capt->last_drift = count2 - count1;
+ count1 = count2;
+ } else if (count1 > count2) {
+ dpcm_play->last_drift = count1 - count2;
+ }
+ copy_play_buf(dpcm_play, dpcm_capt, count1);
+ bytepos_finish(dpcm_play, count1);
+ bytepos_finish(dpcm_capt, count1);
unlock:
spin_unlock_irqrestore(&cable->lock, flags);
return running;
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 87657dd7714..ea995af6d04 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -31,6 +31,8 @@
#define INTERRUPT_INTERVAL 16
#define QUEUE_LENGTH 48
+static void pcm_period_tasklet(unsigned long data);
+
/**
* amdtp_out_stream_init - initialize an AMDTP output stream structure
* @s: the AMDTP output stream to initialize
@@ -47,6 +49,7 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
s->flags = flags;
s->context = ERR_PTR(-1);
mutex_init(&s->mutex);
+ tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s);
s->packet_index = 0;
return 0;
@@ -164,6 +167,21 @@ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
}
EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format);
+/**
+ * amdtp_out_stream_pcm_prepare - prepare PCM device for running
+ * @s: the AMDTP output stream
+ *
+ * This function should be called from the PCM device's .prepare callback.
+ */
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
+{
+ tasklet_kill(&s->period_tasklet);
+ s->pcm_buffer_pointer = 0;
+ s->pcm_period_pointer = 0;
+ s->pointer_flush = true;
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_prepare);
+
static unsigned int calculate_data_blocks(struct amdtp_out_stream *s)
{
unsigned int phase, data_blocks;
@@ -376,11 +394,21 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
s->pcm_period_pointer += data_blocks;
if (s->pcm_period_pointer >= pcm->runtime->period_size) {
s->pcm_period_pointer -= pcm->runtime->period_size;
- snd_pcm_period_elapsed(pcm);
+ s->pointer_flush = false;
+ tasklet_hi_schedule(&s->period_tasklet);
}
}
}
+static void pcm_period_tasklet(unsigned long data)
+{
+ struct amdtp_out_stream *s = (void *)data;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+
+ if (pcm)
+ snd_pcm_period_elapsed(pcm);
+}
+
static void out_packet_callback(struct fw_iso_context *context, u32 cycle,
size_t header_length, void *header, void *data)
{
@@ -506,6 +534,24 @@ err_unlock:
EXPORT_SYMBOL(amdtp_out_stream_start);
/**
+ * amdtp_out_stream_pcm_pointer - get the PCM buffer position
+ * @s: the AMDTP output stream that transports the PCM data
+ *
+ * Returns the current buffer position, in frames.
+ */
+unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
+{
+ /* this optimization is allowed to be racy */
+ if (s->pointer_flush)
+ fw_iso_context_flush_completions(s->context);
+ else
+ s->pointer_flush = true;
+
+ return ACCESS_ONCE(s->pcm_buffer_pointer);
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_pointer);
+
+/**
* amdtp_out_stream_update - update the stream after a bus reset
* @s: the AMDTP output stream
*/
@@ -532,6 +578,7 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s)
return;
}
+ tasklet_kill(&s->period_tasklet);
fw_iso_context_stop(s->context);
fw_iso_context_destroy(s->context);
s->context = ERR_PTR(-1);
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index 537a9cb8358..b680c5ef01d 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -1,6 +1,7 @@
#ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED
#define SOUND_FIREWIRE_AMDTP_H_INCLUDED
+#include <linux/interrupt.h>
#include <linux/mutex.h>
#include <linux/spinlock.h>
#include "packets-buffer.h"
@@ -55,6 +56,7 @@ struct amdtp_out_stream {
struct iso_packets_buffer buffer;
struct snd_pcm_substream *pcm;
+ struct tasklet_struct period_tasklet;
int packet_index;
unsigned int data_block_counter;
@@ -66,6 +68,7 @@ struct amdtp_out_stream {
unsigned int pcm_buffer_pointer;
unsigned int pcm_period_pointer;
+ bool pointer_flush;
};
int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
@@ -81,6 +84,8 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s);
void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
snd_pcm_format_t format);
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s);
+unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s);
void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s);
/**
@@ -123,18 +128,6 @@ static inline bool amdtp_out_streaming_error(struct amdtp_out_stream *s)
}
/**
- * amdtp_out_stream_pcm_prepare - prepare PCM device for running
- * @s: the AMDTP output stream
- *
- * This function should be called from the PCM device's .prepare callback.
- */
-static inline void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
-{
- s->pcm_buffer_pointer = 0;
- s->pcm_period_pointer = 0;
-}
-
-/**
* amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device
* @s: the AMDTP output stream
* @pcm: the PCM device to be started, or %NULL to stop the current device
@@ -149,18 +142,6 @@ static inline void amdtp_out_stream_pcm_trigger(struct amdtp_out_stream *s,
ACCESS_ONCE(s->pcm) = pcm;
}
-/**
- * amdtp_out_stream_pcm_pointer - get the PCM buffer position
- * @s: the AMDTP output stream that transports the PCM data
- *
- * Returns the current buffer position, in frames.
- */
-static inline unsigned long
-amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
-{
- return ACCESS_ONCE(s->pcm_buffer_pointer);
-}
-
static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
{
return sfc & 1;
diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c
index 76294f2ae47..645cb0ba429 100644
--- a/sound/firewire/cmp.c
+++ b/sound/firewire/cmp.c
@@ -84,7 +84,7 @@ static int pcr_modify(struct cmp_connection *c,
return 0;
io_error:
- cmp_error(c, "transaction failed: %s\n", rcode_string(rcode));
+ cmp_error(c, "transaction failed: %s\n", fw_rcode_string(rcode));
return -EIO;
bus_reset:
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index 4750cea2210..14eb4149837 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -14,32 +14,6 @@
#define ERROR_RETRY_DELAY_MS 5
/**
- * rcode_string - convert a firewire result code to a string
- * @rcode: the result
- */
-const char *rcode_string(unsigned int rcode)
-{
- static const char *const names[] = {
- [RCODE_COMPLETE] = "complete",
- [RCODE_CONFLICT_ERROR] = "conflict error",
- [RCODE_DATA_ERROR] = "data error",
- [RCODE_TYPE_ERROR] = "type error",
- [RCODE_ADDRESS_ERROR] = "address error",
- [RCODE_SEND_ERROR] = "send error",
- [RCODE_CANCELLED] = "cancelled",
- [RCODE_BUSY] = "busy",
- [RCODE_GENERATION] = "generation",
- [RCODE_NO_ACK] = "no ack",
- };
-
- if (rcode < ARRAY_SIZE(names) && names[rcode])
- return names[rcode];
- else
- return "unknown";
-}
-EXPORT_SYMBOL(rcode_string);
-
-/**
* snd_fw_transaction - send a request and wait for its completion
* @unit: the driver's unit on the target device
* @tcode: the transaction code
@@ -71,7 +45,7 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode,
if (rcode_is_permanent_error(rcode) || ++tries >= 3) {
dev_err(&unit->device, "transaction failed: %s\n",
- rcode_string(rcode));
+ fw_rcode_string(rcode));
return -EIO;
}
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index 064f3fd9ab0..aef301476ea 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -8,7 +8,6 @@ struct fw_unit;
int snd_fw_transaction(struct fw_unit *unit, int tcode,
u64 offset, void *buffer, size_t length);
-const char *rcode_string(unsigned int rcode);
/* returns true if retrying the transaction would not make sense */
static inline bool rcode_is_permanent_error(int rcode)
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index a63faec5e7f..582aace20ea 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -375,6 +375,9 @@ int snd_tea575x_init(struct snd_tea575x *tea)
tea->vd.v4l2_dev = tea->v4l2_dev;
tea->vd.ctrl_handler = &tea->ctrl_handler;
set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags);
+ /* disable hw_freq_seek if we can't use it */
+ if (tea->cannot_read_data)
+ v4l2_disable_ioctl(&tea->vd, VIDIOC_S_HW_FREQ_SEEK);
v4l2_ctrl_handler_init(&tea->ctrl_handler, 1);
v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1);
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index d1f4351fb6e..2d67c78c9f4 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -7,7 +7,7 @@
Thanks to Pierfrancesco 'qM2' Passerini.
Generalised for soundcards based on DT-0196 and ALS-007 chips
- by Jonathan Woithe <jwoithe@physics.adelaide.edu.au>: June 2002.
+ by Jonathan Woithe <jwoithe@just42.net>: June 2002.
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
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 5ca0939e422..ff3af6e77d6 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -228,7 +228,7 @@ config SND_OXYGEN
Say Y here to include support for sound cards based on the
C-Media CMI8788 (Oxygen HD Audio) chip:
* Asound A-8788
- * Asus Xonar DG
+ * Asus Xonar DG/DGX
* AuzenTech X-Meridian
* AuzenTech X-Meridian 2G
* Bgears b-Enspirer
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 9d91d61902b..e672ff4df2d 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -1062,17 +1062,4 @@ static struct pci_driver ad1889_pci_driver = {
.remove = __devexit_p(snd_ad1889_remove),
};
-static int __init
-alsa_ad1889_init(void)
-{
- return pci_register_driver(&ad1889_pci_driver);
-}
-
-static void __exit
-alsa_ad1889_fini(void)
-{
- pci_unregister_driver(&ad1889_pci_driver);
-}
-
-module_init(alsa_ad1889_init);
-module_exit(alsa_ad1889_fini);
+module_pci_driver(ad1889_pci_driver);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index bdd6164e9c7..9dfc27bf6cc 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2294,7 +2294,7 @@ static void __devexit snd_ali_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ali5451_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ali_ids,
.probe = snd_ali_probe,
@@ -2305,15 +2305,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_ali_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ali_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ali_init)
-module_exit(alsa_card_ali_exit)
+module_pci_driver(ali5451_driver);
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8196e229b2d..59d65388faf 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -852,7 +852,7 @@ static int __devinit snd_als300_probe(struct pci_dev *pci,
return 0;
}
-static struct pci_driver driver = {
+static struct pci_driver als300_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_als300_ids,
.probe = snd_als300_probe,
@@ -863,15 +863,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_als300_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_als300_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_als300_init)
-module_exit(alsa_card_als300_exit)
+module_pci_driver(als300_driver);
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 3269b8011ea..7d7f2598c74 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -1036,7 +1036,7 @@ static int snd_als4000_resume(struct pci_dev *pci)
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver als4000_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_als4000_ids,
.probe = snd_card_als4000_probe,
@@ -1047,15 +1047,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_als4000_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_als4000_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_als4000_init)
-module_exit(alsa_card_als4000_exit)
+module_pci_driver(als4000_driver);
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 590682f115e..156a94f8a12 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1700,7 +1700,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver atiixp_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
@@ -1711,16 +1711,4 @@ static struct pci_driver driver = {
#endif
};
-
-static int __init alsa_card_atiixp_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_atiixp_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_atiixp_init)
-module_exit(alsa_card_atiixp_exit)
+module_pci_driver(atiixp_driver);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 524d35f3123..30a4fd96ce7 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1331,7 +1331,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver atiixp_modem_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
@@ -1342,16 +1342,4 @@ static struct pci_driver driver = {
#endif
};
-
-static int __init alsa_card_atiixp_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_atiixp_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_atiixp_init)
-module_exit(alsa_card_atiixp_exit)
+module_pci_driver(atiixp_modem_driver);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index f13ad536b2d..ffc376f9f4e 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -375,24 +375,11 @@ static void __devexit snd_vortex_remove(struct pci_dev *pci)
}
// pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver vortex_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_vortex_ids,
.probe = snd_vortex_probe,
.remove = __devexit_p(snd_vortex_remove),
};
-// initialization of the module
-static int __init alsa_card_vortex_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_vortex_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_vortex_init)
-module_exit(alsa_card_vortex_exit)
+module_pci_driver(vortex_driver);
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 1c523193146..0f804741825 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -112,8 +112,6 @@ struct aw2 {
/*********************************
* FUNCTION DECLARATIONS
********************************/
-static int __init alsa_card_aw2_init(void);
-static void __exit alsa_card_aw2_exit(void);
static int snd_aw2_dev_free(struct snd_device *device);
static int __devinit snd_aw2_create(struct snd_card *card,
struct pci_dev *pci, struct aw2 **rchip);
@@ -171,13 +169,15 @@ static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = {
MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver aw2_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_aw2_ids,
.probe = snd_aw2_probe,
.remove = __devexit_p(snd_aw2_remove),
};
+module_pci_driver(aw2_driver);
+
/* operators for playback PCM alsa interface */
static struct snd_pcm_ops snd_aw2_playback_ops = {
.open = snd_aw2_pcm_playback_open,
@@ -217,23 +217,6 @@ static struct snd_kcontrol_new aw2_control __devinitdata = {
* FUNCTION IMPLEMENTATIONS
********************************/
-/* initialization of the module */
-static int __init alsa_card_aw2_init(void)
-{
- snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n");
- return pci_register_driver(&driver);
-}
-
-/* clean up the module */
-static void __exit alsa_card_aw2_exit(void)
-{
- snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n");
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_aw2_init);
-module_exit(alsa_card_aw2_exit);
-
/* component-destructor */
static int snd_aw2_dev_free(struct snd_device *device)
{
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 496f14c1a73..f0b4d7493af 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2862,7 +2862,7 @@ snd_azf3328_resume(struct pci_dev *pci)
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver azf3328_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_azf3328_ids,
.probe = snd_azf3328_probe,
@@ -2873,23 +2873,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init
-alsa_card_azf3328_init(void)
-{
- int err;
- snd_azf3328_dbgcallenter();
- err = pci_register_driver(&driver);
- snd_azf3328_dbgcallleave();
- return err;
-}
-
-static void __exit
-alsa_card_azf3328_exit(void)
-{
- snd_azf3328_dbgcallenter();
- pci_unregister_driver(&driver);
- snd_azf3328_dbgcallleave();
-}
-
-module_init(alsa_card_azf3328_init)
-module_exit(alsa_card_azf3328_exit)
+module_pci_driver(azf3328_driver);
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 62d6163fc9d..b6a95eeca09 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -836,8 +836,6 @@ static struct {
{0x7063, 0x2000}, /* pcHDTV HD-2000 TV */
};
-static struct pci_driver driver;
-
/* return the id of the card, or a negative value if it's blacklisted */
static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
{
@@ -964,24 +962,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = {
{ }
};
-static struct pci_driver driver = {
+static struct pci_driver bt87x_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_bt87x_ids,
.probe = snd_bt87x_probe,
.remove = __devexit_p(snd_bt87x_remove),
};
-static int __init alsa_card_bt87x_init(void)
-{
- if (load_all)
- driver.id_table = snd_bt87x_default_ids;
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_bt87x_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_bt87x_init)
-module_exit(alsa_card_bt87x_exit)
+module_pci_driver(bt87x_driver);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 08d6ebfe5a6..e76d68a7081 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1932,7 +1932,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = {
MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
// pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver ca0106_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ca0106_ids,
.probe = snd_ca0106_probe,
@@ -1943,17 +1943,4 @@ static struct pci_driver driver = {
#endif
};
-// initialization of the module
-static int __init alsa_card_ca0106_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_ca0106_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ca0106_init)
-module_exit(alsa_card_ca0106_exit)
+module_pci_driver(ca0106_driver);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 19b06269adc..3815bd4c677 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3398,7 +3398,7 @@ static int snd_cmipci_resume(struct pci_dev *pci)
}
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver cmipci_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cmipci_ids,
.probe = snd_cmipci_probe,
@@ -3409,15 +3409,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_cmipci_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cmipci_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cmipci_init)
-module_exit(alsa_card_cmipci_exit)
+module_pci_driver(cmipci_driver);
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index a9f368f60df..33506ee569b 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -2084,7 +2084,7 @@ static int cs4281_resume(struct pci_dev *pci)
}
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver cs4281_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs4281_ids,
.probe = snd_cs4281_probe,
@@ -2095,15 +2095,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_cs4281_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs4281_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs4281_init)
-module_exit(alsa_card_cs4281_exit)
+module_pci_driver(cs4281_driver);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 819d79d0586..6cc7404e0e8 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -161,7 +161,7 @@ static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver cs46xx_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs46xx_ids,
.probe = snd_card_cs46xx_probe,
@@ -172,15 +172,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_cs46xx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs46xx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs46xx_init)
-module_exit(alsa_card_cs46xx_exit)
+module_pci_driver(cs46xx_driver);
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index c47cabff2bf..f1e4229993a 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -291,23 +291,11 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci,
return 0;
}
-static struct pci_driver driver = {
+static struct pci_driver cs5530_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs5530_ids,
.probe = snd_cs5530_probe,
.remove = __devexit_p(snd_cs5530_remove),
};
-static int __init alsa_card_cs5530_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs5530_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs5530_init)
-module_exit(alsa_card_cs5530_exit)
-
+module_pci_driver(cs5530_driver);
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index a2fb2173e98..2c9697cf0a1 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -394,7 +394,7 @@ static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver cs5535audio_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs5535audio_ids,
.probe = snd_cs5535audio_probe,
@@ -405,18 +405,7 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_cs5535audio_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs5535audio_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs5535audio_init)
-module_exit(alsa_card_cs5535audio_exit)
+module_pci_driver(cs5535audio_driver);
MODULE_AUTHOR("Jaya Kumar");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index 15d95d2bace..75aa2c33841 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -154,15 +154,4 @@ static struct pci_driver ct_driver = {
#endif
};
-static int __init ct_card_init(void)
-{
- return pci_register_driver(&ct_driver);
-}
-
-static void __exit ct_card_exit(void)
-{
- pci_unregister_driver(&ct_driver);
-}
-
-module_init(ct_card_init)
-module_exit(ct_card_exit)
+module_pci_driver(ct_driver);
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 595c11f904b..0f8eda1dafd 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2328,7 +2328,7 @@ static void __devexit snd_echo_remove(struct pci_dev *pci)
******************************************************************************/
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver echo_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_echo_ids,
.probe = snd_echo_probe,
@@ -2339,22 +2339,4 @@ static struct pci_driver driver = {
#endif /* CONFIG_PM */
};
-
-
-/* initialization of the module */
-static int __init alsa_card_echo_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-
-
-/* clean up the module */
-static void __exit alsa_card_echo_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-
-module_init(alsa_card_echo_init)
-module_exit(alsa_card_echo_exit)
+module_pci_driver(echo_driver);
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 64417a73322..d8c670c9d62 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip)
const struct firmware *fw;
int box_type, err;
- if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
+ if (snd_BUG_ON(!chip->comm_page))
return -EPERM;
/* See if the ASIC is present and working - only if the DSP is already loaded */
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 790c65d980c..7fdbbe4d996 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -263,7 +263,7 @@ static int snd_emu10k1_resume(struct pci_dev *pci)
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver emu10k1_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_emu10k1_ids,
.probe = snd_card_emu10k1_probe,
@@ -274,15 +274,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_emu10k1_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_emu10k1_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_emu10k1_init)
-module_exit(alsa_card_emu10k1_exit)
+module_pci_driver(emu10k1_driver);
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 47a651cb6e8..5c8978b2c4d 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1612,24 +1612,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = {
MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids);
// pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver emu10k1x_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_emu10k1x_ids,
.probe = snd_emu10k1x_probe,
.remove = __devexit_p(snd_emu10k1x_remove),
};
-// initialization of the module
-static int __init alsa_card_emu10k1x_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_emu10k1x_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_emu10k1x_init)
-module_exit(alsa_card_emu10k1x_exit)
+module_pci_driver(emu10k1x_driver);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 47a245e8419..3821c81d1c9 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2488,7 +2488,7 @@ static void __devexit snd_audiopci_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ens137x_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_audiopci_ids,
.probe = snd_audiopci_probe,
@@ -2499,15 +2499,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_ens137x_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ens137x_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ens137x_init)
-module_exit(alsa_card_ens137x_exit)
+module_pci_driver(ens137x_driver);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 53eb76b4110..82c8d8c5c52 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1882,7 +1882,7 @@ static void __devexit snd_es1938_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver es1938_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_es1938_ids,
.probe = snd_es1938_probe,
@@ -1893,15 +1893,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_es1938_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_es1938_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_es1938_init)
-module_exit(alsa_card_es1938_exit)
+module_pci_driver(es1938_driver);
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index a8faae1c85e..67f47d89195 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2898,7 +2898,7 @@ static void __devexit snd_es1968_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver es1968_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_es1968_ids,
.probe = snd_es1968_probe,
@@ -2909,15 +2909,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_es1968_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_es1968_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_es1968_init)
-module_exit(alsa_card_es1968_exit)
+module_pci_driver(es1968_driver);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index a416ea8af3e..f6966232275 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1416,7 +1416,7 @@ static int snd_fm801_resume(struct pci_dev *pci)
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver fm801_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_fm801_ids,
.probe = snd_card_fm801_probe,
@@ -1427,15 +1427,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_fm801_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_fm801_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_fm801_init)
-module_exit(alsa_card_fm801_exit)
+module_pci_driver(fm801_driver);
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index ace157cc3d1..bd4149f1aaf 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,6 +1,6 @@
snd-hda-intel-objs := hda_intel.o
-snd-hda-codec-y := hda_codec.o hda_jack.o
+snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o
snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
new file mode 100644
index 00000000000..6e9ef3e2509
--- /dev/null
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -0,0 +1,760 @@
+/*
+ * BIOS auto-parser helper functions for HD-audio
+ *
+ * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#include <linux/slab.h>
+#include <linux/export.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_auto_parser.h"
+
+#define SFX "hda_codec: "
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
+{
+ for (; *list; list++)
+ if (*list == nid)
+ return 1;
+ return 0;
+}
+
+
+/*
+ * Sort an associated group of pins according to their sequence numbers.
+ */
+static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences,
+ int num_pins)
+{
+ int i, j;
+ short seq;
+ hda_nid_t nid;
+
+ for (i = 0; i < num_pins; i++) {
+ for (j = i + 1; j < num_pins; j++) {
+ if (sequences[i] > sequences[j]) {
+ seq = sequences[i];
+ sequences[i] = sequences[j];
+ sequences[j] = seq;
+ nid = pins[i];
+ pins[i] = pins[j];
+ pins[j] = nid;
+ }
+ }
+ }
+}
+
+
+/* add the found input-pin to the cfg->inputs[] table */
+static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid,
+ int type)
+{
+ if (cfg->num_inputs < AUTO_CFG_MAX_INS) {
+ cfg->inputs[cfg->num_inputs].pin = nid;
+ cfg->inputs[cfg->num_inputs].type = type;
+ cfg->num_inputs++;
+ }
+}
+
+/* sort inputs in the order of AUTO_PIN_* type */
+static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
+{
+ int i, j;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ for (j = i + 1; j < cfg->num_inputs; j++) {
+ if (cfg->inputs[i].type > cfg->inputs[j].type) {
+ struct auto_pin_cfg_item tmp;
+ tmp = cfg->inputs[i];
+ cfg->inputs[i] = cfg->inputs[j];
+ cfg->inputs[j] = tmp;
+ }
+ }
+ }
+}
+
+/* Reorder the surround channels
+ * ALSA sequence is front/surr/clfe/side
+ * HDA sequence is:
+ * 4-ch: front/surr => OK as it is
+ * 6-ch: front/clfe/surr
+ * 8-ch: front/clfe/rear/side|fc
+ */
+static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
+{
+ hda_nid_t nid;
+
+ switch (nums) {
+ case 3:
+ case 4:
+ nid = pins[1];
+ pins[1] = pins[2];
+ pins[2] = nid;
+ break;
+ }
+}
+
+/*
+ * Parse all pin widgets and store the useful pin nids to cfg
+ *
+ * The number of line-outs or any primary output is stored in line_outs,
+ * and the corresponding output pins are assigned to line_out_pins[],
+ * in the order of front, rear, CLFE, side, ...
+ *
+ * If more extra outputs (speaker and headphone) are found, the pins are
+ * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack
+ * is detected, one of speaker of HP pins is assigned as the primary
+ * output, i.e. to line_out_pins[0]. So, line_outs is always positive
+ * if any analog output exists.
+ *
+ * The analog input pins are assigned to inputs array.
+ * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
+ * respectively.
+ */
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ const hda_nid_t *ignore_nids,
+ unsigned int cond_flags)
+{
+ hda_nid_t nid, end_nid;
+ short seq, assoc_line_out;
+ short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
+ short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
+ short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
+ int i;
+
+ memset(cfg, 0, sizeof(*cfg));
+
+ memset(sequences_line_out, 0, sizeof(sequences_line_out));
+ memset(sequences_speaker, 0, sizeof(sequences_speaker));
+ memset(sequences_hp, 0, sizeof(sequences_hp));
+ assoc_line_out = 0;
+
+ codec->ignore_misc_bit = true;
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ unsigned int wid_caps = get_wcaps(codec, nid);
+ unsigned int wid_type = get_wcaps_type(wid_caps);
+ unsigned int def_conf;
+ short assoc, loc, conn, dev;
+
+ /* read all default configuration for pin complex */
+ if (wid_type != AC_WID_PIN)
+ continue;
+ /* ignore the given nids (e.g. pc-beep returns error) */
+ if (ignore_nids && is_in_nid_list(nid, ignore_nids))
+ continue;
+
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+ AC_DEFCFG_MISC_NO_PRESENCE))
+ codec->ignore_misc_bit = false;
+ conn = get_defcfg_connect(def_conf);
+ if (conn == AC_JACK_PORT_NONE)
+ continue;
+ loc = get_defcfg_location(def_conf);
+ dev = get_defcfg_device(def_conf);
+
+ /* workaround for buggy BIOS setups */
+ if (dev == AC_JACK_LINE_OUT) {
+ if (conn == AC_JACK_PORT_FIXED)
+ dev = AC_JACK_SPEAKER;
+ }
+
+ switch (dev) {
+ case AC_JACK_LINE_OUT:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+
+ if (!(wid_caps & AC_WCAP_STEREO))
+ if (!cfg->mono_out_pin)
+ cfg->mono_out_pin = nid;
+ if (!assoc)
+ continue;
+ if (!assoc_line_out)
+ assoc_line_out = assoc;
+ else if (assoc_line_out != assoc)
+ continue;
+ if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
+ continue;
+ cfg->line_out_pins[cfg->line_outs] = nid;
+ sequences_line_out[cfg->line_outs] = seq;
+ cfg->line_outs++;
+ break;
+ case AC_JACK_SPEAKER:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
+ continue;
+ cfg->speaker_pins[cfg->speaker_outs] = nid;
+ sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
+ cfg->speaker_outs++;
+ break;
+ case AC_JACK_HP_OUT:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
+ continue;
+ cfg->hp_pins[cfg->hp_outs] = nid;
+ sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
+ cfg->hp_outs++;
+ break;
+ case AC_JACK_MIC_IN:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC);
+ break;
+ case AC_JACK_LINE_IN:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN);
+ break;
+ case AC_JACK_CD:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD);
+ break;
+ case AC_JACK_AUX:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX);
+ break;
+ case AC_JACK_SPDIF_OUT:
+ case AC_JACK_DIG_OTHER_OUT:
+ if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
+ continue;
+ cfg->dig_out_pins[cfg->dig_outs] = nid;
+ cfg->dig_out_type[cfg->dig_outs] =
+ (loc == AC_JACK_LOC_HDMI) ?
+ HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
+ cfg->dig_outs++;
+ break;
+ case AC_JACK_SPDIF_IN:
+ case AC_JACK_DIG_OTHER_IN:
+ cfg->dig_in_pin = nid;
+ if (loc == AC_JACK_LOC_HDMI)
+ cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
+ else
+ cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+ break;
+ }
+ }
+
+ /* FIX-UP:
+ * If no line-out is defined but multiple HPs are found,
+ * some of them might be the real line-outs.
+ */
+ if (!cfg->line_outs && cfg->hp_outs > 1 &&
+ !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
+ int i = 0;
+ while (i < cfg->hp_outs) {
+ /* The real HPs should have the sequence 0x0f */
+ if ((sequences_hp[i] & 0x0f) == 0x0f) {
+ i++;
+ continue;
+ }
+ /* Move it to the line-out table */
+ cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
+ sequences_line_out[cfg->line_outs] = sequences_hp[i];
+ cfg->line_outs++;
+ cfg->hp_outs--;
+ memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
+ sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
+ memmove(sequences_hp + i, sequences_hp + i + 1,
+ sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
+ }
+ memset(cfg->hp_pins + cfg->hp_outs, 0,
+ sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs));
+ if (!cfg->hp_outs)
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+
+ }
+
+ /* sort by sequence */
+ sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
+ cfg->line_outs);
+ sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
+ cfg->speaker_outs);
+ sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
+ cfg->hp_outs);
+
+ /*
+ * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
+ * as a primary output
+ */
+ if (!cfg->line_outs &&
+ !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
+ if (cfg->speaker_outs) {
+ cfg->line_outs = cfg->speaker_outs;
+ memcpy(cfg->line_out_pins, cfg->speaker_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->speaker_outs = 0;
+ memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+ cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+ } else if (cfg->hp_outs) {
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins,
+ sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+ }
+ }
+
+ reorder_outputs(cfg->line_outs, cfg->line_out_pins);
+ reorder_outputs(cfg->hp_outs, cfg->hp_pins);
+ reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
+
+ sort_autocfg_input_pins(cfg);
+
+ /*
+ * debug prints of the parsed results
+ */
+ snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
+ cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
+ cfg->line_out_pins[2], cfg->line_out_pins[3],
+ cfg->line_out_pins[4],
+ cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
+ (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
+ "speaker" : "line"));
+ snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->speaker_outs, cfg->speaker_pins[0],
+ cfg->speaker_pins[1], cfg->speaker_pins[2],
+ cfg->speaker_pins[3], cfg->speaker_pins[4]);
+ snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->hp_outs, cfg->hp_pins[0],
+ cfg->hp_pins[1], cfg->hp_pins[2],
+ cfg->hp_pins[3], cfg->hp_pins[4]);
+ snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin);
+ if (cfg->dig_outs)
+ snd_printd(" dig-out=0x%x/0x%x\n",
+ cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
+ snd_printd(" inputs:");
+ for (i = 0; i < cfg->num_inputs; i++) {
+ snd_printd(" %s=0x%x",
+ hda_get_autocfg_input_label(codec, cfg, i),
+ cfg->inputs[i].pin);
+ }
+ snd_printd("\n");
+ if (cfg->dig_in_pin)
+ snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin);
+
+ return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
+
+int snd_hda_get_input_pin_attr(unsigned int def_conf)
+{
+ unsigned int loc = get_defcfg_location(def_conf);
+ unsigned int conn = get_defcfg_connect(def_conf);
+ if (conn == AC_JACK_PORT_NONE)
+ return INPUT_PIN_ATTR_UNUSED;
+ /* Windows may claim the internal mic to be BOTH, too */
+ if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH)
+ return INPUT_PIN_ATTR_INT;
+ if ((loc & 0x30) == AC_JACK_LOC_INTERNAL)
+ return INPUT_PIN_ATTR_INT;
+ if ((loc & 0x30) == AC_JACK_LOC_SEPARATE)
+ return INPUT_PIN_ATTR_DOCK;
+ if (loc == AC_JACK_LOC_REAR)
+ return INPUT_PIN_ATTR_REAR;
+ if (loc == AC_JACK_LOC_FRONT)
+ return INPUT_PIN_ATTR_FRONT;
+ return INPUT_PIN_ATTR_NORMAL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
+
+/**
+ * hda_get_input_pin_label - Give a label for the given input pin
+ *
+ * When check_location is true, the function checks the pin location
+ * for mic and line-in pins, and set an appropriate prefix like "Front",
+ * "Rear", "Internal".
+ */
+
+static const char *hda_get_input_pin_label(struct hda_codec *codec,
+ hda_nid_t pin, bool check_location)
+{
+ unsigned int def_conf;
+ static const char * const mic_names[] = {
+ "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
+ };
+ int attr;
+
+ def_conf = snd_hda_codec_get_pincfg(codec, pin);
+
+ switch (get_defcfg_device(def_conf)) {
+ case AC_JACK_MIC_IN:
+ if (!check_location)
+ return "Mic";
+ attr = snd_hda_get_input_pin_attr(def_conf);
+ if (!attr)
+ return "None";
+ return mic_names[attr - 1];
+ case AC_JACK_LINE_IN:
+ if (!check_location)
+ return "Line";
+ attr = snd_hda_get_input_pin_attr(def_conf);
+ if (!attr)
+ return "None";
+ if (attr == INPUT_PIN_ATTR_DOCK)
+ return "Dock Line";
+ return "Line";
+ case AC_JACK_AUX:
+ return "Aux";
+ case AC_JACK_CD:
+ return "CD";
+ case AC_JACK_SPDIF_IN:
+ return "SPDIF In";
+ case AC_JACK_DIG_OTHER_IN:
+ return "Digital In";
+ default:
+ return "Misc";
+ }
+}
+
+/* Check whether the location prefix needs to be added to the label.
+ * If all mic-jacks are in the same location (e.g. rear panel), we don't
+ * have to put "Front" prefix to each label. In such a case, returns false.
+ */
+static int check_mic_location_need(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ int input)
+{
+ unsigned int defc;
+ int i, attr, attr2;
+
+ defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin);
+ attr = snd_hda_get_input_pin_attr(defc);
+ /* for internal or docking mics, we need locations */
+ if (attr <= INPUT_PIN_ATTR_NORMAL)
+ return 1;
+
+ attr = 0;
+ for (i = 0; i < cfg->num_inputs; i++) {
+ defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin);
+ attr2 = snd_hda_get_input_pin_attr(defc);
+ if (attr2 >= INPUT_PIN_ATTR_NORMAL) {
+ if (attr && attr != attr2)
+ return 1; /* different locations found */
+ attr = attr2;
+ }
+ }
+ return 0;
+}
+
+/**
+ * hda_get_autocfg_input_label - Get a label for the given input
+ *
+ * Get a label for the given input pin defined by the autocfg item.
+ * Unlike hda_get_input_pin_label(), this function checks all inputs
+ * defined in autocfg and avoids the redundant mic/line prefix as much as
+ * possible.
+ */
+const char *hda_get_autocfg_input_label(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ int input)
+{
+ int type = cfg->inputs[input].type;
+ int has_multiple_pins = 0;
+
+ if ((input > 0 && cfg->inputs[input - 1].type == type) ||
+ (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type))
+ 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,
+ has_multiple_pins);
+}
+EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
+
+/* return the position of NID in the list, or -1 if not found */
+static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+ int i;
+ for (i = 0; i < nums; i++)
+ if (list[i] == nid)
+ return i;
+ return -1;
+}
+
+/* get a unique suffix or an index number */
+static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins,
+ int num_pins, int *indexp)
+{
+ static const char * const channel_sfx[] = {
+ " Front", " Surround", " CLFE", " Side"
+ };
+ int i;
+
+ i = find_idx_in_nid_list(nid, pins, num_pins);
+ if (i < 0)
+ return NULL;
+ if (num_pins == 1)
+ return "";
+ if (num_pins > ARRAY_SIZE(channel_sfx)) {
+ if (indexp)
+ *indexp = i;
+ return "";
+ }
+ return channel_sfx[i];
+}
+
+static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
+ const struct auto_pin_cfg *cfg,
+ const char *name, char *label, int maxlen,
+ int *indexp)
+{
+ unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ int attr = snd_hda_get_input_pin_attr(def_conf);
+ const char *pfx = "", *sfx = "";
+
+ /* handle as a speaker if it's a fixed line-out */
+ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
+ name = "Speaker";
+ /* check the location */
+ switch (attr) {
+ case INPUT_PIN_ATTR_DOCK:
+ pfx = "Dock ";
+ break;
+ case INPUT_PIN_ATTR_FRONT:
+ pfx = "Front ";
+ break;
+ }
+ if (cfg) {
+ /* try to give a unique suffix if needed */
+ sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs,
+ indexp);
+ if (!sfx)
+ sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs,
+ indexp);
+ if (!sfx) {
+ /* don't add channel suffix for Headphone controls */
+ int idx = find_idx_in_nid_list(nid, cfg->hp_pins,
+ cfg->hp_outs);
+ if (idx >= 0)
+ *indexp = idx;
+ sfx = "";
+ }
+ }
+ snprintf(label, maxlen, "%s%s%s", pfx, name, sfx);
+ return 1;
+}
+
+/**
+ * snd_hda_get_pin_label - Get a label for the given I/O pin
+ *
+ * Get a label for the given pin. This function works for both input and
+ * output pins. When @cfg is given as non-NULL, the function tries to get
+ * an optimized label using hda_get_autocfg_input_label().
+ *
+ * This function tries to give a unique label string for the pin as much as
+ * possible. For example, when the multiple line-outs are present, it adds
+ * the channel suffix like "Front", "Surround", etc (only when @cfg is given).
+ * If no unique name with a suffix is available and @indexp is non-NULL, the
+ * index number is stored in the pointer.
+ */
+int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
+ const struct auto_pin_cfg *cfg,
+ char *label, int maxlen, int *indexp)
+{
+ unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ const char *name = NULL;
+ int i;
+
+ if (indexp)
+ *indexp = 0;
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+ return 0;
+
+ switch (get_defcfg_device(def_conf)) {
+ case AC_JACK_LINE_OUT:
+ return fill_audio_out_name(codec, nid, cfg, "Line Out",
+ label, maxlen, indexp);
+ case AC_JACK_SPEAKER:
+ return fill_audio_out_name(codec, nid, cfg, "Speaker",
+ label, maxlen, indexp);
+ case AC_JACK_HP_OUT:
+ return fill_audio_out_name(codec, nid, cfg, "Headphone",
+ label, maxlen, indexp);
+ case AC_JACK_SPDIF_OUT:
+ case AC_JACK_DIG_OTHER_OUT:
+ if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
+ name = "HDMI";
+ else
+ name = "SPDIF";
+ if (cfg && indexp) {
+ i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
+ cfg->dig_outs);
+ if (i >= 0)
+ *indexp = i;
+ }
+ break;
+ default:
+ if (cfg) {
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].pin != nid)
+ continue;
+ name = hda_get_autocfg_input_label(codec, cfg, i);
+ if (name)
+ break;
+ }
+ }
+ if (!name)
+ name = hda_get_input_pin_label(codec, nid, true);
+ break;
+ }
+ if (!name)
+ return 0;
+ strlcpy(label, name, maxlen);
+ return 1;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+
+int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
+ const struct hda_verb *list)
+{
+ const struct hda_verb **v;
+ snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
+ v = snd_array_new(&spec->verbs);
+ if (!v)
+ return -ENOMEM;
+ *v = list;
+ return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_gen_add_verbs);
+
+void snd_hda_gen_apply_verbs(struct hda_codec *codec)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ int i;
+ for (i = 0; i < spec->verbs.used; i++) {
+ struct hda_verb **v = snd_array_elem(&spec->verbs, i);
+ snd_hda_sequence_write(codec, *v);
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_gen_apply_verbs);
+
+void snd_hda_apply_pincfgs(struct hda_codec *codec,
+ const struct hda_pintbl *cfg)
+{
+ for (; cfg->nid; cfg++)
+ snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs);
+
+void snd_hda_apply_fixup(struct hda_codec *codec, int action)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ const char *modelname = spec->fixup_name;
+#endif
+ int depth = 0;
+
+ if (!spec->fixup_list)
+ return;
+
+ while (id >= 0) {
+ const struct hda_fixup *fix = spec->fixup_list + id;
+
+ switch (fix->type) {
+ case HDA_FIXUP_PINS:
+ if (action != HDA_FIXUP_ACT_PRE_PROBE || !fix->v.pins)
+ break;
+ snd_printdd(KERN_INFO SFX
+ "%s: Apply pincfg for %s\n",
+ codec->chip_name, modelname);
+ snd_hda_apply_pincfgs(codec, fix->v.pins);
+ break;
+ case HDA_FIXUP_VERBS:
+ if (action != HDA_FIXUP_ACT_PROBE || !fix->v.verbs)
+ break;
+ snd_printdd(KERN_INFO SFX
+ "%s: Apply fix-verbs for %s\n",
+ codec->chip_name, modelname);
+ snd_hda_gen_add_verbs(codec->spec, fix->v.verbs);
+ break;
+ case HDA_FIXUP_FUNC:
+ if (!fix->v.func)
+ break;
+ snd_printdd(KERN_INFO SFX
+ "%s: Apply fix-func for %s\n",
+ codec->chip_name, modelname);
+ fix->v.func(codec, fix, action);
+ break;
+ default:
+ snd_printk(KERN_ERR SFX
+ "%s: Invalid fixup type %d\n",
+ codec->chip_name, fix->type);
+ break;
+ }
+ if (!fix->chained)
+ break;
+ if (++depth > 10)
+ break;
+ id = fix->chain_id;
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_apply_fixup);
+
+void snd_hda_pick_fixup(struct hda_codec *codec,
+ const struct hda_model_fixup *models,
+ const struct snd_pci_quirk *quirk,
+ const struct hda_fixup *fixlist)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ const struct snd_pci_quirk *q;
+ int id = -1;
+ const char *name = NULL;
+
+ /* when model=nofixup is given, don't pick up any fixups */
+ if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
+ spec->fixup_list = NULL;
+ spec->fixup_id = -1;
+ return;
+ }
+
+ if (codec->modelname && models) {
+ while (models->name) {
+ if (!strcmp(codec->modelname, models->name)) {
+ id = models->id;
+ name = models->name;
+ break;
+ }
+ models++;
+ }
+ }
+ if (id < 0) {
+ q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+ if (q) {
+ id = q->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ name = q->name;
+#endif
+ }
+ }
+ if (id < 0) {
+ for (q = quirk; q->subvendor; q++) {
+ unsigned int vendorid =
+ q->subdevice | (q->subvendor << 16);
+ if (vendorid == codec->subsystem_id) {
+ id = q->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ name = q->name;
+#endif
+ break;
+ }
+ }
+ }
+
+ spec->fixup_id = id;
+ if (id >= 0) {
+ spec->fixup_list = fixlist;
+ spec->fixup_name = name;
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_pick_fixup);
diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h
new file mode 100644
index 00000000000..2a7889dfbd1
--- /dev/null
+++ b/sound/pci/hda/hda_auto_parser.h
@@ -0,0 +1,160 @@
+/*
+ * BIOS auto-parser helper functions for HD-audio
+ *
+ * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#ifndef __SOUND_HDA_AUTO_PARSER_H
+#define __SOUND_HDA_AUTO_PARSER_H
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+enum {
+ AUTO_PIN_MIC,
+ AUTO_PIN_LINE_IN,
+ AUTO_PIN_CD,
+ AUTO_PIN_AUX,
+ AUTO_PIN_LAST
+};
+
+enum {
+ AUTO_PIN_LINE_OUT,
+ AUTO_PIN_SPEAKER_OUT,
+ AUTO_PIN_HP_OUT
+};
+
+#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS
+#define AUTO_CFG_MAX_INS 8
+
+struct auto_pin_cfg_item {
+ hda_nid_t pin;
+ int type;
+};
+
+struct auto_pin_cfg;
+const char *hda_get_autocfg_input_label(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ int input);
+int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
+ const struct auto_pin_cfg *cfg,
+ char *label, int maxlen, int *indexp);
+
+enum {
+ INPUT_PIN_ATTR_UNUSED, /* pin not connected */
+ INPUT_PIN_ATTR_INT, /* internal mic/line-in */
+ INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */
+ INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */
+ INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */
+ INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */
+};
+
+int snd_hda_get_input_pin_attr(unsigned int def_conf);
+
+struct auto_pin_cfg {
+ int line_outs;
+ /* sorted in the order of Front/Surr/CLFE/Side */
+ hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
+ int speaker_outs;
+ hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
+ int hp_outs;
+ int line_out_type; /* AUTO_PIN_XXX_OUT */
+ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
+ int num_inputs;
+ struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS];
+ int dig_outs;
+ hda_nid_t dig_out_pins[2];
+ hda_nid_t dig_in_pin;
+ hda_nid_t mono_out_pin;
+ int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
+ int dig_in_type; /* HDA_PCM_TYPE_XXX */
+};
+
+/* 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 */
+
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ const hda_nid_t *ignore_nids,
+ unsigned int cond_flags);
+
+/* older function */
+#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
+ snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
+
+/*
+ */
+
+struct hda_gen_spec {
+ /* fix-up list */
+ int fixup_id;
+ const struct hda_fixup *fixup_list;
+ const char *fixup_name;
+
+ /* additional init verbs */
+ struct snd_array verbs;
+};
+
+
+/*
+ * Fix-up pin default configurations and add default verbs
+ */
+
+struct hda_pintbl {
+ hda_nid_t nid;
+ u32 val;
+};
+
+struct hda_model_fixup {
+ const int id;
+ const char *name;
+};
+
+struct hda_fixup {
+ int type;
+ bool chained;
+ int chain_id;
+ union {
+ const struct hda_pintbl *pins;
+ const struct hda_verb *verbs;
+ void (*func)(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action);
+ } v;
+};
+
+/* fixup types */
+enum {
+ HDA_FIXUP_INVALID,
+ HDA_FIXUP_PINS,
+ HDA_FIXUP_VERBS,
+ HDA_FIXUP_FUNC,
+};
+
+/* fixup action definitions */
+enum {
+ HDA_FIXUP_ACT_PRE_PROBE,
+ HDA_FIXUP_ACT_PROBE,
+ HDA_FIXUP_ACT_INIT,
+ HDA_FIXUP_ACT_BUILD,
+};
+
+int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
+ const struct hda_verb *list);
+void snd_hda_gen_apply_verbs(struct hda_codec *codec);
+void snd_hda_apply_pincfgs(struct hda_codec *codec,
+ const struct hda_pintbl *cfg);
+void snd_hda_apply_fixup(struct hda_codec *codec, int action);
+void snd_hda_pick_fixup(struct hda_codec *codec,
+ const struct hda_model_fixup *models,
+ const struct snd_pci_quirk *quirk,
+ const struct hda_fixup *fixlist);
+
+#endif /* __SOUND_HDA_AUTO_PARSER_H */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 7a8fcc4c15f..41ca803a1ff 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -334,78 +334,67 @@ static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid)
return NULL;
}
+/* read the connection and add to the cache */
+static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid)
+{
+ hda_nid_t list[HDA_MAX_CONNECTIONS];
+ int len;
+
+ len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list));
+ if (len < 0)
+ return len;
+ return snd_hda_override_conn_list(codec, nid, len, list);
+}
+
/**
- * snd_hda_get_conn_list - get connection list
+ * snd_hda_get_connections - copy connection list
* @codec: the HDA codec
* @nid: NID to parse
- * @listp: the pointer to store NID list
+ * @conn_list: connection list array; when NULL, checks only the size
+ * @max_conns: max. number of connections to store
*
* Parses the connection list of the given widget and stores the list
* of NIDs.
*
* Returns the number of connections, or a negative error code.
*/
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
- const hda_nid_t **listp)
+int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns)
{
struct snd_array *array = &codec->conn_lists;
- int len, err;
- hda_nid_t list[HDA_MAX_CONNECTIONS];
+ int len;
hda_nid_t *p;
bool added = false;
again:
+ mutex_lock(&codec->hash_mutex);
+ len = -1;
/* if the connection-list is already cached, read it */
p = lookup_conn_list(array, nid);
if (p) {
- if (listp)
- *listp = p + 2;
- return p[1];
+ len = p[1];
+ if (conn_list && len > max_conns) {
+ snd_printk(KERN_ERR "hda_codec: "
+ "Too many connections %d for NID 0x%x\n",
+ len, nid);
+ mutex_unlock(&codec->hash_mutex);
+ return -EINVAL;
+ }
+ if (conn_list && len)
+ memcpy(conn_list, p + 2, len * sizeof(hda_nid_t));
}
+ mutex_unlock(&codec->hash_mutex);
+ if (len >= 0)
+ return len;
if (snd_BUG_ON(added))
return -EINVAL;
- /* read the connection and add to the cache */
- len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
+ len = read_and_add_raw_conns(codec, nid);
if (len < 0)
return len;
- err = snd_hda_override_conn_list(codec, nid, len, list);
- if (err < 0)
- return err;
added = true;
goto again;
}
-EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
-
-/**
- * snd_hda_get_connections - copy connection list
- * @codec: the HDA codec
- * @nid: NID to parse
- * @conn_list: connection list array
- * @max_conns: max. number of connections to store
- *
- * Parses the connection list of the given widget and stores the list
- * of NIDs.
- *
- * Returns the number of connections, or a negative error code.
- */
-int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
- hda_nid_t *conn_list, int max_conns)
-{
- const hda_nid_t *list;
- int len = snd_hda_get_conn_list(codec, nid, &list);
-
- if (len <= 0)
- return len;
- if (len > max_conns) {
- snd_printk(KERN_ERR "hda_codec: "
- "Too many connections %d for NID 0x%x\n",
- len, nid);
- return -EINVAL;
- }
- memcpy(conn_list, list, len * sizeof(hda_nid_t));
- return len;
-}
EXPORT_SYMBOL_HDA(snd_hda_get_connections);
/**
@@ -543,6 +532,7 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
hda_nid_t *p;
int i, old_used;
+ mutex_lock(&codec->hash_mutex);
p = lookup_conn_list(array, nid);
if (p)
*p = -1; /* invalidate the old entry */
@@ -553,10 +543,12 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
for (i = 0; i < len; i++)
if (!add_conn_list(array, list[i]))
goto error_add;
+ mutex_unlock(&codec->hash_mutex);
return 0;
error_add:
array->used = old_used;
+ mutex_unlock(&codec->hash_mutex);
return -ENOMEM;
}
EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
@@ -1255,6 +1247,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
codec->addr = codec_addr;
mutex_init(&codec->spdif_mutex);
mutex_init(&codec->control_mutex);
+ mutex_init(&codec->hash_mutex);
init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32);
@@ -1264,15 +1257,9 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
- if (codec->bus->modelname) {
- codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
- if (!codec->modelname) {
- snd_hda_codec_free(codec);
- return -ENODEV;
- }
- }
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spin_lock_init(&codec->power_lock);
INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
/* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
* the caller has to power down appropriatley after initialization
@@ -1281,6 +1268,14 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
hda_keep_power_on(codec);
#endif
+ if (codec->bus->modelname) {
+ codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
+ if (!codec->modelname) {
+ snd_hda_codec_free(codec);
+ return -ENODEV;
+ }
+ }
+
list_add_tail(&codec->list, &bus->codec_list);
bus->caddr_tbl[codec_addr] = codec;
@@ -1603,6 +1598,60 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key)
return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key);
}
+/* overwrite the value with the key in the caps hash */
+static int write_caps_hash(struct hda_codec *codec, u32 key, unsigned int val)
+{
+ struct hda_amp_info *info;
+
+ mutex_lock(&codec->hash_mutex);
+ info = get_alloc_amp_hash(codec, key);
+ if (!info) {
+ mutex_unlock(&codec->hash_mutex);
+ return -EINVAL;
+ }
+ info->amp_caps = val;
+ info->head.val |= INFO_AMP_CAPS;
+ mutex_unlock(&codec->hash_mutex);
+ return 0;
+}
+
+/* query the value from the caps hash; if not found, fetch the current
+ * value from the given function and store in the hash
+ */
+static unsigned int
+query_caps_hash(struct hda_codec *codec, hda_nid_t nid, int dir, u32 key,
+ unsigned int (*func)(struct hda_codec *, hda_nid_t, int))
+{
+ struct hda_amp_info *info;
+ unsigned int val;
+
+ mutex_lock(&codec->hash_mutex);
+ info = get_alloc_amp_hash(codec, key);
+ if (!info) {
+ mutex_unlock(&codec->hash_mutex);
+ return 0;
+ }
+ if (!(info->head.val & INFO_AMP_CAPS)) {
+ mutex_unlock(&codec->hash_mutex); /* for reentrance */
+ val = func(codec, nid, dir);
+ write_caps_hash(codec, key, val);
+ } else {
+ val = info->amp_caps;
+ mutex_unlock(&codec->hash_mutex);
+ }
+ return val;
+}
+
+static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid,
+ int direction)
+{
+ if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
+ nid = codec->afg;
+ return snd_hda_param_read(codec, nid,
+ direction == HDA_OUTPUT ?
+ AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
+}
+
/**
* query_amp_caps - query AMP capabilities
* @codec: the HD-auio codec
@@ -1617,22 +1666,9 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key)
*/
u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
{
- struct hda_amp_info *info;
-
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
- if (!info)
- return 0;
- if (!(info->head.val & INFO_AMP_CAPS)) {
- if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
- nid = codec->afg;
- info->amp_caps = snd_hda_param_read(codec, nid,
- direction == HDA_OUTPUT ?
- AC_PAR_AMP_OUT_CAP :
- AC_PAR_AMP_IN_CAP);
- if (info->amp_caps)
- info->head.val |= INFO_AMP_CAPS;
- }
- return info->amp_caps;
+ return query_caps_hash(codec, nid, direction,
+ HDA_HASH_KEY(nid, direction, 0),
+ read_amp_cap);
}
EXPORT_SYMBOL_HDA(query_amp_caps);
@@ -1652,34 +1688,12 @@ EXPORT_SYMBOL_HDA(query_amp_caps);
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps)
{
- struct hda_amp_info *info;
-
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, dir, 0));
- if (!info)
- return -EINVAL;
- info->amp_caps = caps;
- info->head.val |= INFO_AMP_CAPS;
- return 0;
+ return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps);
}
EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
-static unsigned int
-query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key,
- unsigned int (*func)(struct hda_codec *, hda_nid_t))
-{
- struct hda_amp_info *info;
-
- info = get_alloc_amp_hash(codec, key);
- if (!info)
- return 0;
- if (!info->head.val) {
- info->head.val |= INFO_AMP_CAPS;
- info->amp_caps = func(codec, nid);
- }
- return info->amp_caps;
-}
-
-static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid,
+ int dir)
{
return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
}
@@ -1697,7 +1711,7 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
*/
u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
{
- return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid),
+ return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid),
read_pin_cap);
}
EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
@@ -1715,41 +1729,47 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
unsigned int caps)
{
- struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
- if (!info)
- return -ENOMEM;
- info->amp_caps = caps;
- info->head.val |= INFO_AMP_CAPS;
- return 0;
+ return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps);
}
EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps);
-/*
- * read the current volume to info
- * if the cache exists, read the cache value.
+/* read or sync the hash value with the current value;
+ * call within hash_mutex
*/
-static unsigned int get_vol_mute(struct hda_codec *codec,
- struct hda_amp_info *info, hda_nid_t nid,
- int ch, int direction, int index)
+static struct hda_amp_info *
+update_amp_hash(struct hda_codec *codec, hda_nid_t nid, int ch,
+ int direction, int index)
{
- u32 val, parm;
-
- if (info->head.val & INFO_AMP_VOL(ch))
- return info->vol[ch];
+ struct hda_amp_info *info;
+ unsigned int parm, val = 0;
+ bool val_read = false;
- parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
- parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
- parm |= index;
- val = snd_hda_codec_read(codec, nid, 0,
+ retry:
+ info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
+ if (!info)
+ return NULL;
+ if (!(info->head.val & INFO_AMP_VOL(ch))) {
+ if (!val_read) {
+ mutex_unlock(&codec->hash_mutex);
+ parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
+ parm |= direction == HDA_OUTPUT ?
+ AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
+ parm |= index;
+ val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_AMP_GAIN_MUTE, parm);
- info->vol[ch] = val & 0xff;
- info->head.val |= INFO_AMP_VOL(ch);
- return info->vol[ch];
+ val &= 0xff;
+ val_read = true;
+ mutex_lock(&codec->hash_mutex);
+ goto retry;
+ }
+ info->vol[ch] = val;
+ info->head.val |= INFO_AMP_VOL(ch);
+ }
+ return info;
}
/*
- * write the current volume in info to the h/w and update the cache
+ * write the current volume in info to the h/w
*/
static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
hda_nid_t nid, int ch, int direction, int index,
@@ -1766,7 +1786,6 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
else
parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
- info->vol[ch] = val;
}
/**
@@ -1783,10 +1802,14 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
int direction, int index)
{
struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
- if (!info)
- return 0;
- return get_vol_mute(codec, info, nid, ch, direction, index);
+ unsigned int val = 0;
+
+ mutex_lock(&codec->hash_mutex);
+ info = update_amp_hash(codec, nid, ch, direction, index);
+ if (info)
+ val = info->vol[ch];
+ mutex_unlock(&codec->hash_mutex);
+ return val;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read);
@@ -1808,15 +1831,23 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
{
struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx));
- if (!info)
- return 0;
if (snd_BUG_ON(mask & ~0xff))
mask &= 0xff;
val &= mask;
- val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
- if (info->vol[ch] == val)
+
+ mutex_lock(&codec->hash_mutex);
+ info = update_amp_hash(codec, nid, ch, direction, idx);
+ if (!info) {
+ mutex_unlock(&codec->hash_mutex);
+ return 0;
+ }
+ val |= info->vol[ch] & ~mask;
+ if (info->vol[ch] == val) {
+ mutex_unlock(&codec->hash_mutex);
return 0;
+ }
+ info->vol[ch] = val;
+ mutex_unlock(&codec->hash_mutex);
put_vol_mute(codec, info, nid, ch, direction, idx, val);
return 1;
}
@@ -2208,24 +2239,50 @@ void snd_hda_ctls_clear(struct hda_codec *codec)
/* pseudo device locking
* toggle card->shutdown to allow/disallow the device access (as a hack)
*/
-static int hda_lock_devices(struct snd_card *card)
+int snd_hda_lock_devices(struct hda_bus *bus)
{
+ struct snd_card *card = bus->card;
+ struct hda_codec *codec;
+
spin_lock(&card->files_lock);
- if (card->shutdown) {
- spin_unlock(&card->files_lock);
- return -EINVAL;
- }
+ if (card->shutdown)
+ goto err_unlock;
card->shutdown = 1;
+ if (!list_empty(&card->ctl_files))
+ goto err_clear;
+
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ int pcm;
+ for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+ struct hda_pcm *cpcm = &codec->pcm_info[pcm];
+ if (!cpcm->pcm)
+ continue;
+ if (cpcm->pcm->streams[0].substream_opened ||
+ cpcm->pcm->streams[1].substream_opened)
+ goto err_clear;
+ }
+ }
spin_unlock(&card->files_lock);
return 0;
+
+ err_clear:
+ card->shutdown = 0;
+ err_unlock:
+ spin_unlock(&card->files_lock);
+ return -EINVAL;
}
+EXPORT_SYMBOL_HDA(snd_hda_lock_devices);
-static void hda_unlock_devices(struct snd_card *card)
+void snd_hda_unlock_devices(struct hda_bus *bus)
{
+ struct snd_card *card = bus->card;
+
+ card = bus->card;
spin_lock(&card->files_lock);
card->shutdown = 0;
spin_unlock(&card->files_lock);
}
+EXPORT_SYMBOL_HDA(snd_hda_unlock_devices);
/**
* snd_hda_codec_reset - Clear all objects assigned to the codec
@@ -2239,32 +2296,21 @@ static void hda_unlock_devices(struct snd_card *card)
*/
int snd_hda_codec_reset(struct hda_codec *codec)
{
- struct snd_card *card = codec->bus->card;
- int i, pcm;
+ struct hda_bus *bus = codec->bus;
+ struct snd_card *card = bus->card;
+ int i;
- if (hda_lock_devices(card) < 0)
- return -EBUSY;
- /* check whether the codec isn't used by any mixer or PCM streams */
- if (!list_empty(&card->ctl_files)) {
- hda_unlock_devices(card);
+ if (snd_hda_lock_devices(bus) < 0)
return -EBUSY;
- }
- for (pcm = 0; pcm < codec->num_pcms; pcm++) {
- struct hda_pcm *cpcm = &codec->pcm_info[pcm];
- if (!cpcm->pcm)
- continue;
- if (cpcm->pcm->streams[0].substream_opened ||
- cpcm->pcm->streams[1].substream_opened) {
- hda_unlock_devices(card);
- return -EBUSY;
- }
- }
/* OK, let it free */
#ifdef CONFIG_SND_HDA_POWER_SAVE
- cancel_delayed_work(&codec->power_work);
- flush_workqueue(codec->bus->workq);
+ cancel_delayed_work_sync(&codec->power_work);
+ codec->power_on = 0;
+ codec->power_transition = 0;
+ codec->power_jiffies = jiffies;
+ flush_workqueue(bus->workq);
#endif
snd_hda_ctls_clear(codec);
/* relase PCMs */
@@ -2272,7 +2318,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
if (codec->pcm_info[i].pcm) {
snd_device_free(card, codec->pcm_info[i].pcm);
clear_bit(codec->pcm_info[i].device,
- codec->bus->pcm_dev_bits);
+ bus->pcm_dev_bits);
}
}
if (codec->patch_ops.free)
@@ -2297,7 +2343,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
codec->owner = NULL;
/* allow device access again */
- hda_unlock_devices(card);
+ snd_hda_unlock_devices(bus);
return 0;
}
@@ -2859,12 +2905,15 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
+ mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.iec958.status[0] = spdif->status & 0xff;
ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
+ mutex_unlock(&codec->spdif_mutex);
return 0;
}
@@ -2950,12 +2999,14 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
- hda_nid_t nid = spdif->nid;
+ struct hda_spdif_out *spdif;
+ hda_nid_t nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
+ nid = spdif->nid;
spdif->status = ucontrol->value.iec958.status[0] |
((unsigned int)ucontrol->value.iec958.status[1] << 8) |
((unsigned int)ucontrol->value.iec958.status[2] << 16) |
@@ -2977,9 +3028,12 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
+ mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
+ mutex_unlock(&codec->spdif_mutex);
return 0;
}
@@ -2999,12 +3053,14 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
- hda_nid_t nid = spdif->nid;
+ struct hda_spdif_out *spdif;
+ hda_nid_t nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
+ nid = spdif->nid;
val = spdif->ctls & ~AC_DIG1_ENABLE;
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
@@ -3092,6 +3148,9 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
+/* get the hda_spdif_out entry from the given NID
+ * call within spdif_mutex lock
+ */
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid)
{
@@ -3108,9 +3167,10 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
{
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
spdif->nid = (u16)-1;
mutex_unlock(&codec->spdif_mutex);
}
@@ -3118,10 +3178,11 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
{
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
unsigned short val;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
if (spdif->nid != nid) {
spdif->nid = nid;
val = spdif->ctls;
@@ -3486,11 +3547,14 @@ static void hda_call_codec_suspend(struct hda_codec *codec)
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D3);
#ifdef CONFIG_SND_HDA_POWER_SAVE
- snd_hda_update_power_acct(codec);
cancel_delayed_work(&codec->power_work);
+ spin_lock(&codec->power_lock);
+ snd_hda_update_power_acct(codec);
+ trace_hda_power_down(codec);
codec->power_on = 0;
codec->power_transition = 0;
codec->power_jiffies = jiffies;
+ spin_unlock(&codec->power_lock);
#endif
}
@@ -3499,6 +3563,10 @@ static void hda_call_codec_suspend(struct hda_codec *codec)
*/
static void hda_call_codec_resume(struct hda_codec *codec)
{
+ /* set as if powered on for avoiding re-entering the resume
+ * in the resume / power-save sequence
+ */
+ hda_keep_power_on(codec);
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D0);
@@ -3514,6 +3582,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
}
+ snd_hda_power_down(codec); /* flag down before returning */
}
#endif /* CONFIG_PM */
@@ -3665,7 +3734,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
}
EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
-static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid,
+ int dir)
{
unsigned int val = 0;
if (nid != codec->afg &&
@@ -3680,11 +3750,12 @@ static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid)
{
- return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid),
+ return query_caps_hash(codec, nid, 0, HDA_HASH_PARPCM_KEY(nid),
get_pcm_param);
}
-static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid,
+ int dir)
{
unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
if (!streams || streams == -1)
@@ -3696,7 +3767,7 @@ static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
{
- return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid),
+ return query_caps_hash(codec, nid, 0, HDA_HASH_PARSTR_KEY(nid),
get_stream_param);
}
@@ -3775,11 +3846,13 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
bps = 20;
}
}
+#if 0 /* FIXME: CS4206 doesn't work, which is the only codec supporting float */
if (streams & AC_SUPFMT_FLOAT32) {
formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
if (!bps)
bps = 32;
}
+#endif
if (streams == AC_SUPFMT_AC3) {
/* should be exclusive */
/* temporary hack: we have still no proper support
@@ -4283,12 +4356,18 @@ static void hda_power_work(struct work_struct *work)
container_of(work, struct hda_codec, power_work.work);
struct hda_bus *bus = codec->bus;
+ spin_lock(&codec->power_lock);
+ if (codec->power_transition > 0) { /* during power-up sequence? */
+ spin_unlock(&codec->power_lock);
+ return;
+ }
if (!codec->power_on || codec->power_count) {
codec->power_transition = 0;
+ spin_unlock(&codec->power_lock);
return;
}
+ spin_unlock(&codec->power_lock);
- trace_hda_power_down(codec);
hda_call_codec_suspend(codec);
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
@@ -4296,9 +4375,11 @@ static void hda_power_work(struct work_struct *work)
static void hda_keep_power_on(struct hda_codec *codec)
{
+ spin_lock(&codec->power_lock);
codec->power_count++;
codec->power_on = 1;
codec->power_jiffies = jiffies;
+ spin_unlock(&codec->power_lock);
}
/* update the power on/off account with the current jiffies */
@@ -4323,19 +4404,31 @@ void snd_hda_power_up(struct hda_codec *codec)
{
struct hda_bus *bus = codec->bus;
+ spin_lock(&codec->power_lock);
codec->power_count++;
- if (codec->power_on || codec->power_transition)
+ if (codec->power_on || codec->power_transition > 0) {
+ spin_unlock(&codec->power_lock);
return;
+ }
+ spin_unlock(&codec->power_lock);
+ cancel_delayed_work_sync(&codec->power_work);
+
+ spin_lock(&codec->power_lock);
trace_hda_power_up(codec);
snd_hda_update_power_acct(codec);
codec->power_on = 1;
codec->power_jiffies = jiffies;
+ codec->power_transition = 1; /* avoid reentrance */
+ spin_unlock(&codec->power_lock);
+
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
hda_call_codec_resume(codec);
- cancel_delayed_work(&codec->power_work);
+
+ spin_lock(&codec->power_lock);
codec->power_transition = 0;
+ spin_unlock(&codec->power_lock);
}
EXPORT_SYMBOL_HDA(snd_hda_power_up);
@@ -4351,14 +4444,18 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up);
*/
void snd_hda_power_down(struct hda_codec *codec)
{
+ spin_lock(&codec->power_lock);
--codec->power_count;
- if (!codec->power_on || codec->power_count || codec->power_transition)
+ if (!codec->power_on || codec->power_count || codec->power_transition) {
+ spin_unlock(&codec->power_lock);
return;
+ }
if (power_save(codec)) {
- codec->power_transition = 1; /* avoid reentrance */
+ codec->power_transition = -1; /* avoid reentrance */
queue_delayed_work(codec->bus->workq, &codec->power_work,
msecs_to_jiffies(power_save(codec) * 1000));
}
+ spin_unlock(&codec->power_lock);
}
EXPORT_SYMBOL_HDA(snd_hda_power_down);
@@ -4710,11 +4807,11 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
{
const hda_nid_t *nids = mout->dac_nids;
int chs = substream->runtime->channels;
- struct hda_spdif_out *spdif =
- snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
+ struct hda_spdif_out *spdif;
int i;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
if (mout->dig_out_nid && mout->share_spdif &&
mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
if (chs == 2 &&
@@ -4795,601 +4892,58 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup);
-/*
- * Helper for automatic pin configuration
- */
-
-static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
-{
- for (; *list; list++)
- if (*list == nid)
- return 1;
- return 0;
-}
-
-
-/*
- * Sort an associated group of pins according to their sequence numbers.
- */
-static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences,
- int num_pins)
-{
- int i, j;
- short seq;
- hda_nid_t nid;
-
- for (i = 0; i < num_pins; i++) {
- for (j = i + 1; j < num_pins; j++) {
- if (sequences[i] > sequences[j]) {
- seq = sequences[i];
- sequences[i] = sequences[j];
- sequences[j] = seq;
- nid = pins[i];
- pins[i] = pins[j];
- pins[j] = nid;
- }
- }
- }
-}
-
-
-/* add the found input-pin to the cfg->inputs[] table */
-static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid,
- int type)
-{
- if (cfg->num_inputs < AUTO_CFG_MAX_INS) {
- cfg->inputs[cfg->num_inputs].pin = nid;
- cfg->inputs[cfg->num_inputs].type = type;
- cfg->num_inputs++;
- }
-}
-
-/* sort inputs in the order of AUTO_PIN_* type */
-static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
-{
- int i, j;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- for (j = i + 1; j < cfg->num_inputs; j++) {
- if (cfg->inputs[i].type > cfg->inputs[j].type) {
- struct auto_pin_cfg_item tmp;
- tmp = cfg->inputs[i];
- cfg->inputs[i] = cfg->inputs[j];
- cfg->inputs[j] = tmp;
- }
- }
- }
-}
-
-/* Reorder the surround channels
- * ALSA sequence is front/surr/clfe/side
- * HDA sequence is:
- * 4-ch: front/surr => OK as it is
- * 6-ch: front/clfe/surr
- * 8-ch: front/clfe/rear/side|fc
- */
-static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
-{
- hda_nid_t nid;
-
- switch (nums) {
- case 3:
- case 4:
- nid = pins[1];
- pins[1] = pins[2];
- pins[2] = nid;
- break;
- }
-}
-
-/*
- * Parse all pin widgets and store the useful pin nids to cfg
- *
- * The number of line-outs or any primary output is stored in line_outs,
- * and the corresponding output pins are assigned to line_out_pins[],
- * in the order of front, rear, CLFE, side, ...
- *
- * If more extra outputs (speaker and headphone) are found, the pins are
- * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack
- * is detected, one of speaker of HP pins is assigned as the primary
- * output, i.e. to line_out_pins[0]. So, line_outs is always positive
- * if any analog output exists.
- *
- * The analog input pins are assigned to inputs array.
- * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
- * respectively.
- */
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids,
- unsigned int cond_flags)
-{
- hda_nid_t nid, end_nid;
- short seq, assoc_line_out;
- short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
- short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
- short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
- int i;
-
- memset(cfg, 0, sizeof(*cfg));
-
- memset(sequences_line_out, 0, sizeof(sequences_line_out));
- memset(sequences_speaker, 0, sizeof(sequences_speaker));
- memset(sequences_hp, 0, sizeof(sequences_hp));
- assoc_line_out = 0;
-
- codec->ignore_misc_bit = true;
- end_nid = codec->start_nid + codec->num_nodes;
- for (nid = codec->start_nid; nid < end_nid; nid++) {
- unsigned int wid_caps = get_wcaps(codec, nid);
- unsigned int wid_type = get_wcaps_type(wid_caps);
- unsigned int def_conf;
- short assoc, loc, conn, dev;
-
- /* read all default configuration for pin complex */
- if (wid_type != AC_WID_PIN)
- continue;
- /* ignore the given nids (e.g. pc-beep returns error) */
- if (ignore_nids && is_in_nid_list(nid, ignore_nids))
- continue;
-
- def_conf = snd_hda_codec_get_pincfg(codec, nid);
- if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
- AC_DEFCFG_MISC_NO_PRESENCE))
- codec->ignore_misc_bit = false;
- conn = get_defcfg_connect(def_conf);
- if (conn == AC_JACK_PORT_NONE)
- continue;
- loc = get_defcfg_location(def_conf);
- dev = get_defcfg_device(def_conf);
-
- /* workaround for buggy BIOS setups */
- if (dev == AC_JACK_LINE_OUT) {
- if (conn == AC_JACK_PORT_FIXED)
- dev = AC_JACK_SPEAKER;
- }
-
- switch (dev) {
- case AC_JACK_LINE_OUT:
- seq = get_defcfg_sequence(def_conf);
- assoc = get_defcfg_association(def_conf);
-
- if (!(wid_caps & AC_WCAP_STEREO))
- if (!cfg->mono_out_pin)
- cfg->mono_out_pin = nid;
- if (!assoc)
- continue;
- if (!assoc_line_out)
- assoc_line_out = assoc;
- else if (assoc_line_out != assoc)
- continue;
- if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
- continue;
- cfg->line_out_pins[cfg->line_outs] = nid;
- sequences_line_out[cfg->line_outs] = seq;
- cfg->line_outs++;
- break;
- case AC_JACK_SPEAKER:
- seq = get_defcfg_sequence(def_conf);
- assoc = get_defcfg_association(def_conf);
- if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
- continue;
- cfg->speaker_pins[cfg->speaker_outs] = nid;
- sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
- cfg->speaker_outs++;
- break;
- case AC_JACK_HP_OUT:
- seq = get_defcfg_sequence(def_conf);
- assoc = get_defcfg_association(def_conf);
- if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
- continue;
- cfg->hp_pins[cfg->hp_outs] = nid;
- sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
- cfg->hp_outs++;
- break;
- case AC_JACK_MIC_IN:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC);
- break;
- case AC_JACK_LINE_IN:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN);
- break;
- case AC_JACK_CD:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD);
- break;
- case AC_JACK_AUX:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX);
- break;
- case AC_JACK_SPDIF_OUT:
- case AC_JACK_DIG_OTHER_OUT:
- if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
- continue;
- cfg->dig_out_pins[cfg->dig_outs] = nid;
- cfg->dig_out_type[cfg->dig_outs] =
- (loc == AC_JACK_LOC_HDMI) ?
- HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
- cfg->dig_outs++;
- break;
- case AC_JACK_SPDIF_IN:
- case AC_JACK_DIG_OTHER_IN:
- cfg->dig_in_pin = nid;
- if (loc == AC_JACK_LOC_HDMI)
- cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
- else
- cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
- break;
- }
- }
-
- /* FIX-UP:
- * If no line-out is defined but multiple HPs are found,
- * some of them might be the real line-outs.
- */
- if (!cfg->line_outs && cfg->hp_outs > 1 &&
- !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
- int i = 0;
- while (i < cfg->hp_outs) {
- /* The real HPs should have the sequence 0x0f */
- if ((sequences_hp[i] & 0x0f) == 0x0f) {
- i++;
- continue;
- }
- /* Move it to the line-out table */
- cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
- sequences_line_out[cfg->line_outs] = sequences_hp[i];
- cfg->line_outs++;
- cfg->hp_outs--;
- memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
- sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
- memmove(sequences_hp + i, sequences_hp + i + 1,
- sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
- }
- memset(cfg->hp_pins + cfg->hp_outs, 0,
- sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs));
- if (!cfg->hp_outs)
- cfg->line_out_type = AUTO_PIN_HP_OUT;
-
- }
-
- /* sort by sequence */
- sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
- cfg->line_outs);
- sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
- cfg->speaker_outs);
- sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
- cfg->hp_outs);
-
- /*
- * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
- * as a primary output
- */
- if (!cfg->line_outs &&
- !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
- if (cfg->speaker_outs) {
- cfg->line_outs = cfg->speaker_outs;
- memcpy(cfg->line_out_pins, cfg->speaker_pins,
- sizeof(cfg->speaker_pins));
- cfg->speaker_outs = 0;
- memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
- cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
- } else if (cfg->hp_outs) {
- cfg->line_outs = cfg->hp_outs;
- memcpy(cfg->line_out_pins, cfg->hp_pins,
- sizeof(cfg->hp_pins));
- cfg->hp_outs = 0;
- memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
- cfg->line_out_type = AUTO_PIN_HP_OUT;
- }
- }
-
- reorder_outputs(cfg->line_outs, cfg->line_out_pins);
- reorder_outputs(cfg->hp_outs, cfg->hp_pins);
- reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
-
- sort_autocfg_input_pins(cfg);
-
- /*
- * debug prints of the parsed results
- */
- snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
- cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
- cfg->line_out_pins[2], cfg->line_out_pins[3],
- cfg->line_out_pins[4],
- cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
- (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
- "speaker" : "line"));
- snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
- cfg->speaker_outs, cfg->speaker_pins[0],
- cfg->speaker_pins[1], cfg->speaker_pins[2],
- cfg->speaker_pins[3], cfg->speaker_pins[4]);
- snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
- cfg->hp_outs, cfg->hp_pins[0],
- cfg->hp_pins[1], cfg->hp_pins[2],
- cfg->hp_pins[3], cfg->hp_pins[4]);
- snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin);
- if (cfg->dig_outs)
- snd_printd(" dig-out=0x%x/0x%x\n",
- cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
- snd_printd(" inputs:");
- for (i = 0; i < cfg->num_inputs; i++) {
- snd_printd(" %s=0x%x",
- hda_get_autocfg_input_label(codec, cfg, i),
- cfg->inputs[i].pin);
- }
- snd_printd("\n");
- if (cfg->dig_in_pin)
- snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin);
-
- return 0;
-}
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
-
-int snd_hda_get_input_pin_attr(unsigned int def_conf)
-{
- unsigned int loc = get_defcfg_location(def_conf);
- unsigned int conn = get_defcfg_connect(def_conf);
- if (conn == AC_JACK_PORT_NONE)
- return INPUT_PIN_ATTR_UNUSED;
- /* Windows may claim the internal mic to be BOTH, too */
- if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH)
- return INPUT_PIN_ATTR_INT;
- if ((loc & 0x30) == AC_JACK_LOC_INTERNAL)
- return INPUT_PIN_ATTR_INT;
- if ((loc & 0x30) == AC_JACK_LOC_SEPARATE)
- return INPUT_PIN_ATTR_DOCK;
- if (loc == AC_JACK_LOC_REAR)
- return INPUT_PIN_ATTR_REAR;
- if (loc == AC_JACK_LOC_FRONT)
- return INPUT_PIN_ATTR_FRONT;
- return INPUT_PIN_ATTR_NORMAL;
-}
-EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
-
/**
- * hda_get_input_pin_label - Give a label for the given input pin
+ * snd_hda_get_default_vref - Get the default (mic) VREF pin bits
*
- * When check_location is true, the function checks the pin location
- * for mic and line-in pins, and set an appropriate prefix like "Front",
- * "Rear", "Internal".
- */
-
-static const char *hda_get_input_pin_label(struct hda_codec *codec,
- hda_nid_t pin, bool check_location)
-{
- unsigned int def_conf;
- static const char * const mic_names[] = {
- "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
- };
- int attr;
-
- def_conf = snd_hda_codec_get_pincfg(codec, pin);
-
- switch (get_defcfg_device(def_conf)) {
- case AC_JACK_MIC_IN:
- if (!check_location)
- return "Mic";
- attr = snd_hda_get_input_pin_attr(def_conf);
- if (!attr)
- return "None";
- return mic_names[attr - 1];
- case AC_JACK_LINE_IN:
- if (!check_location)
- return "Line";
- attr = snd_hda_get_input_pin_attr(def_conf);
- if (!attr)
- return "None";
- if (attr == INPUT_PIN_ATTR_DOCK)
- return "Dock Line";
- return "Line";
- case AC_JACK_AUX:
- return "Aux";
- case AC_JACK_CD:
- return "CD";
- case AC_JACK_SPDIF_IN:
- return "SPDIF In";
- case AC_JACK_DIG_OTHER_IN:
- return "Digital In";
- default:
- return "Misc";
- }
-}
-
-/* Check whether the location prefix needs to be added to the label.
- * If all mic-jacks are in the same location (e.g. rear panel), we don't
- * have to put "Front" prefix to each label. In such a case, returns false.
- */
-static int check_mic_location_need(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- int input)
-{
- unsigned int defc;
- int i, attr, attr2;
-
- defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin);
- attr = snd_hda_get_input_pin_attr(defc);
- /* for internal or docking mics, we need locations */
- if (attr <= INPUT_PIN_ATTR_NORMAL)
- return 1;
-
- attr = 0;
- for (i = 0; i < cfg->num_inputs; i++) {
- defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin);
- attr2 = snd_hda_get_input_pin_attr(defc);
- if (attr2 >= INPUT_PIN_ATTR_NORMAL) {
- if (attr && attr != attr2)
- return 1; /* different locations found */
- attr = attr2;
+ * Guess the suitable VREF pin bits to be set as the pin-control value.
+ * Note: the function doesn't set the AC_PINCTL_IN_EN bit.
+ */
+unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin)
+{
+ unsigned int pincap;
+ unsigned int oldval;
+ oldval = snd_hda_codec_read(codec, pin, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ pincap = snd_hda_query_pin_caps(codec, pin);
+ pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+ /* Exception: if the default pin setup is vref50, we give it priority */
+ if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
+ return AC_PINCTL_VREF_80;
+ else if (pincap & AC_PINCAP_VREF_50)
+ return AC_PINCTL_VREF_50;
+ else if (pincap & AC_PINCAP_VREF_100)
+ return AC_PINCTL_VREF_100;
+ else if (pincap & AC_PINCAP_VREF_GRD)
+ return AC_PINCTL_VREF_GRD;
+ return AC_PINCTL_VREF_HIZ;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_default_vref);
+
+int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int val, bool cached)
+{
+ if (val) {
+ unsigned int cap = snd_hda_query_pin_caps(codec, pin);
+ if (cap && (val & AC_PINCTL_OUT_EN)) {
+ if (!(cap & AC_PINCAP_OUT))
+ val &= ~(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ else if ((val & AC_PINCTL_HP_EN) &&
+ !(cap & AC_PINCAP_HP_DRV))
+ val &= ~AC_PINCTL_HP_EN;
}
- }
- return 0;
-}
-
-/**
- * hda_get_autocfg_input_label - Get a label for the given input
- *
- * Get a label for the given input pin defined by the autocfg item.
- * Unlike hda_get_input_pin_label(), this function checks all inputs
- * defined in autocfg and avoids the redundant mic/line prefix as much as
- * possible.
- */
-const char *hda_get_autocfg_input_label(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- int input)
-{
- int type = cfg->inputs[input].type;
- int has_multiple_pins = 0;
-
- if ((input > 0 && cfg->inputs[input - 1].type == type) ||
- (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type))
- 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,
- has_multiple_pins);
-}
-EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
-
-/* return the position of NID in the list, or -1 if not found */
-static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
-{
- int i;
- for (i = 0; i < nums; i++)
- if (list[i] == nid)
- return i;
- return -1;
-}
-
-/* get a unique suffix or an index number */
-static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins,
- int num_pins, int *indexp)
-{
- static const char * const channel_sfx[] = {
- " Front", " Surround", " CLFE", " Side"
- };
- int i;
-
- i = find_idx_in_nid_list(nid, pins, num_pins);
- if (i < 0)
- return NULL;
- if (num_pins == 1)
- return "";
- if (num_pins > ARRAY_SIZE(channel_sfx)) {
- if (indexp)
- *indexp = i;
- return "";
- }
- return channel_sfx[i];
-}
-
-static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg,
- const char *name, char *label, int maxlen,
- int *indexp)
-{
- unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
- int attr = snd_hda_get_input_pin_attr(def_conf);
- const char *pfx = "", *sfx = "";
-
- /* handle as a speaker if it's a fixed line-out */
- if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
- name = "Speaker";
- /* check the location */
- switch (attr) {
- case INPUT_PIN_ATTR_DOCK:
- pfx = "Dock ";
- break;
- case INPUT_PIN_ATTR_FRONT:
- pfx = "Front ";
- break;
- }
- if (cfg) {
- /* try to give a unique suffix if needed */
- sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs,
- indexp);
- if (!sfx)
- sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs,
- indexp);
- if (!sfx) {
- /* don't add channel suffix for Headphone controls */
- int idx = find_idx_in_nid_list(nid, cfg->hp_pins,
- cfg->hp_outs);
- if (idx >= 0)
- *indexp = idx;
- sfx = "";
+ if (cap && (val & AC_PINCTL_IN_EN)) {
+ if (!(cap & AC_PINCAP_IN))
+ val &= ~(AC_PINCTL_IN_EN | AC_PINCTL_VREFEN);
}
}
- snprintf(label, maxlen, "%s%s%s", pfx, name, sfx);
- return 1;
-}
-
-/**
- * snd_hda_get_pin_label - Get a label for the given I/O pin
- *
- * Get a label for the given pin. This function works for both input and
- * output pins. When @cfg is given as non-NULL, the function tries to get
- * an optimized label using hda_get_autocfg_input_label().
- *
- * This function tries to give a unique label string for the pin as much as
- * possible. For example, when the multiple line-outs are present, it adds
- * the channel suffix like "Front", "Surround", etc (only when @cfg is given).
- * If no unique name with a suffix is available and @indexp is non-NULL, the
- * index number is stored in the pointer.
- */
-int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg,
- char *label, int maxlen, int *indexp)
-{
- unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
- const char *name = NULL;
- int i;
-
- if (indexp)
- *indexp = 0;
- if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
- return 0;
-
- switch (get_defcfg_device(def_conf)) {
- case AC_JACK_LINE_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Line Out",
- label, maxlen, indexp);
- case AC_JACK_SPEAKER:
- return fill_audio_out_name(codec, nid, cfg, "Speaker",
- label, maxlen, indexp);
- case AC_JACK_HP_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Headphone",
- label, maxlen, indexp);
- case AC_JACK_SPDIF_OUT:
- case AC_JACK_DIG_OTHER_OUT:
- if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
- name = "HDMI";
- else
- name = "SPDIF";
- if (cfg && indexp) {
- i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
- cfg->dig_outs);
- if (i >= 0)
- *indexp = i;
- }
- break;
- default:
- if (cfg) {
- for (i = 0; i < cfg->num_inputs; i++) {
- if (cfg->inputs[i].pin != nid)
- continue;
- name = hda_get_autocfg_input_label(codec, cfg, i);
- if (name)
- break;
- }
- }
- if (!name)
- name = hda_get_input_pin_label(codec, nid, true);
- break;
- }
- if (!name)
- return 0;
- strlcpy(label, name, maxlen);
- return 1;
+ if (cached)
+ return snd_hda_codec_update_cache(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ else
+ return snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
-EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+EXPORT_SYMBOL_HDA(_snd_hda_set_pin_ctl);
/**
* snd_hda_add_imux_item - Add an item to input_mux
@@ -5444,12 +4998,6 @@ int snd_hda_suspend(struct hda_bus *bus)
list_for_each_entry(codec, &bus->codec_list, list) {
if (hda_codec_is_power_on(codec))
hda_call_codec_suspend(codec);
- else /* forcibly change the power to D3 even if not used */
- hda_set_power_state(codec,
- codec->afg ? codec->afg : codec->mfg,
- AC_PWRST_D3);
- if (codec->patch_ops.post_suspend)
- codec->patch_ops.post_suspend(codec);
}
return 0;
}
@@ -5469,10 +5017,7 @@ int snd_hda_resume(struct hda_bus *bus)
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
- if (codec->patch_ops.pre_resume)
- codec->patch_ops.pre_resume(codec);
- if (snd_hda_codec_needs_resume(codec))
- hda_call_codec_resume(codec);
+ hda_call_codec_resume(codec);
}
return 0;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 56b4f74c0b1..4fc3960c859 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -704,8 +704,6 @@ struct hda_codec_ops {
unsigned int power_state);
#ifdef CONFIG_PM
int (*suspend)(struct hda_codec *codec, pm_message_t state);
- int (*post_suspend)(struct hda_codec *codec);
- int (*pre_resume)(struct hda_codec *codec);
int (*resume)(struct hda_codec *codec);
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -829,6 +827,7 @@ struct hda_codec {
struct mutex spdif_mutex;
struct mutex control_mutex;
+ struct mutex hash_mutex;
struct snd_array spdif_out;
unsigned int spdif_in_enable; /* SPDIF input enable? */
const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
@@ -861,12 +860,13 @@ struct hda_codec {
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
- unsigned int power_transition :1; /* power-state in transition */
+ int power_transition; /* power-state in transition */
int power_count; /* current (global) power refcount */
struct delayed_work power_work; /* delayed task for powerdown */
unsigned long power_on_acct;
unsigned long power_off_acct;
unsigned long power_jiffies;
+ spinlock_t power_lock;
#endif
/* codec-specific additional proc output */
@@ -911,10 +911,13 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *start_id);
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
+static inline int
+snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid)
+{
+ return snd_hda_get_connections(codec, nid, NULL, 0);
+}
int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
- const hda_nid_t **listp);
int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
const hda_nid_t *list);
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
@@ -1020,6 +1023,9 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state,
bool eapd_workaround);
+int snd_hda_lock_devices(struct hda_bus *bus);
+void snd_hda_unlock_devices(struct hda_bus *bus);
+
/*
* power management
*/
@@ -1051,12 +1057,10 @@ const char *snd_hda_get_jack_location(u32 cfg);
#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_power_up(struct hda_codec *codec);
void snd_hda_power_down(struct hda_codec *codec);
-#define snd_hda_codec_needs_resume(codec) codec->power_count
void snd_hda_update_power_acct(struct hda_codec *codec);
#else
static inline void snd_hda_power_up(struct hda_codec *codec) {}
static inline void snd_hda_power_down(struct hda_codec *codec) {}
-#define snd_hda_codec_needs_resume(codec) 1
#endif
#ifdef CONFIG_SND_HDA_PATCH_LOADER
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c19e71a94e1..2b6392be451 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -53,6 +53,8 @@
#endif
#include <sound/core.h>
#include <sound/initval.h>
+#include <linux/vgaarb.h>
+#include <linux/vga_switcheroo.h>
#include "hda_codec.h"
@@ -175,6 +177,13 @@ MODULE_DESCRIPTION("Intel HDA driver");
#define SFX "hda-intel: "
#endif
+#if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO)
+#ifdef CONFIG_SND_HDA_CODEC_HDMI
+#define SUPPORT_VGA_SWITCHEROO
+#endif
+#endif
+
+
/*
* registers
*/
@@ -472,6 +481,12 @@ struct azx {
unsigned int probing :1; /* codec probing phase */
unsigned int snoop:1;
unsigned int align_buffer_size:1;
+ unsigned int region_requested:1;
+
+ /* VGA-switcheroo setup */
+ unsigned int use_vga_switcheroo:1;
+ unsigned int init_failed:1; /* delayed init failed */
+ unsigned int disabled:1; /* disabled by VGA-switcher */
/* for debugging */
unsigned int last_cmd[AZX_MAX_CODECS];
@@ -497,6 +512,7 @@ enum {
AZX_DRIVER_NVIDIA,
AZX_DRIVER_TERA,
AZX_DRIVER_CTX,
+ AZX_DRIVER_CTHDA,
AZX_DRIVER_GENERIC,
AZX_NUM_DRIVERS, /* keep this as last entry */
};
@@ -518,6 +534,7 @@ enum {
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */
+#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -533,7 +550,23 @@ enum {
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
AZX_DCAPS_ALIGN_BUFSIZE)
-static char *driver_short_names[] __devinitdata = {
+#define AZX_DCAPS_PRESET_CTHDA \
+ (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY)
+
+/*
+ * VGA-switcher support
+ */
+#ifdef SUPPORT_VGA_SWITCHEROO
+#define DELAYED_INIT_MARK
+#define DELAYED_INITDATA_MARK
+#define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo)
+#else
+#define DELAYED_INIT_MARK __devinit
+#define DELAYED_INITDATA_MARK __devinitdata
+#define use_vga_switcheroo(chip) 0
+#endif
+
+static char *driver_short_names[] DELAYED_INITDATA_MARK = {
[AZX_DRIVER_ICH] = "HDA Intel",
[AZX_DRIVER_PCH] = "HDA Intel PCH",
[AZX_DRIVER_SCH] = "HDA Intel MID",
@@ -546,6 +579,7 @@ static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_NVIDIA] = "HDA NVidia",
[AZX_DRIVER_TERA] = "HDA Teradici",
[AZX_DRIVER_CTX] = "HDA Creative",
+ [AZX_DRIVER_CTHDA] = "HDA Creative",
[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
};
@@ -783,11 +817,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
{
struct azx *chip = bus->private_data;
unsigned long timeout;
+ unsigned long loopcounter;
int do_poll = 0;
again:
timeout = jiffies + msecs_to_jiffies(1000);
- for (;;) {
+
+ for (loopcounter = 0;; loopcounter++) {
if (chip->polling_mode || do_poll) {
spin_lock_irq(&chip->reg_lock);
azx_update_rirb(chip);
@@ -803,7 +839,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
}
if (time_after(jiffies, timeout))
break;
- if (bus->needs_damn_long_delay)
+ if (bus->needs_damn_long_delay || loopcounter > 3000)
msleep(2); /* temporary workaround */
else {
udelay(10);
@@ -951,6 +987,8 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
{
struct azx *chip = bus->private_data;
+ if (chip->disabled)
+ return 0;
chip->last_cmd[azx_command_addr(val)] = val;
if (chip->single_cmd)
return azx_single_send_cmd(bus, val);
@@ -963,6 +1001,8 @@ static unsigned int azx_get_response(struct hda_bus *bus,
unsigned int addr)
{
struct azx *chip = bus->private_data;
+ if (chip->disabled)
+ return 0;
if (chip->single_cmd)
return azx_single_get_response(bus, addr);
else
@@ -1228,6 +1268,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
spin_lock(&chip->reg_lock);
+ if (chip->disabled) {
+ spin_unlock(&chip->reg_lock);
+ return IRQ_NONE;
+ }
+
status = azx_readl(chip, INTSTS);
if (status == 0) {
spin_unlock(&chip->reg_lock);
@@ -1283,7 +1328,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
/*
* set up a BDL entry
*/
-static int setup_bdle(struct snd_pcm_substream *substream,
+static int setup_bdle(struct azx *chip,
+ struct snd_pcm_substream *substream,
struct azx_dev *azx_dev, u32 **bdlp,
int ofs, int size, int with_ioc)
{
@@ -1302,6 +1348,12 @@ static int setup_bdle(struct snd_pcm_substream *substream,
bdl[1] = cpu_to_le32(upper_32_bits(addr));
/* program the size field of the BDL entry */
chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size);
+ /* one BDLE cannot cross 4K boundary on CTHDA chips */
+ if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) {
+ u32 remain = 0x1000 - (ofs & 0xfff);
+ if (chunk > remain)
+ chunk = remain;
+ }
bdl[2] = cpu_to_le32(chunk);
/* program the IOC to enable interrupt
* only when the whole fragment is processed
@@ -1354,7 +1406,7 @@ static int azx_setup_periods(struct azx *chip,
bdl_pos_adj[chip->dev_index]);
pos_adj = 0;
} else {
- ofs = setup_bdle(substream, azx_dev,
+ ofs = setup_bdle(chip, substream, azx_dev,
&bdl, ofs, pos_adj,
!substream->runtime->no_period_wakeup);
if (ofs < 0)
@@ -1364,10 +1416,10 @@ static int azx_setup_periods(struct azx *chip,
pos_adj = 0;
for (i = 0; i < periods; i++) {
if (i == periods - 1 && pos_adj)
- ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+ ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs,
period_bytes - pos_adj, 0);
else
- ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+ ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs,
period_bytes,
!substream->runtime->no_period_wakeup);
if (ofs < 0)
@@ -1506,12 +1558,12 @@ static void azx_bus_reset(struct hda_bus *bus)
*/
/* number of codec slots for each chipset: 0 = default slots (i.e. 4) */
-static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
+static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] DELAYED_INITDATA_MARK = {
[AZX_DRIVER_NVIDIA] = 8,
[AZX_DRIVER_TERA] = 1,
};
-static int __devinit azx_codec_create(struct azx *chip, const char *model)
+static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *model)
{
struct hda_bus_template bus_temp;
int c, codecs, err;
@@ -2429,6 +2481,105 @@ static void azx_notifier_unregister(struct azx *chip)
unregister_reboot_notifier(&chip->reboot_notifier);
}
+static int DELAYED_INIT_MARK azx_first_init(struct azx *chip);
+static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip);
+
+static struct pci_dev __devinit *get_bound_vga(struct pci_dev *pci);
+
+#ifdef SUPPORT_VGA_SWITCHEROO
+static void azx_vs_set_state(struct pci_dev *pci,
+ enum vga_switcheroo_state state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct azx *chip = card->private_data;
+ bool disabled;
+
+ if (chip->init_failed)
+ return;
+
+ disabled = (state == VGA_SWITCHEROO_OFF);
+ if (chip->disabled == disabled)
+ return;
+
+ if (!chip->bus) {
+ chip->disabled = disabled;
+ if (!disabled) {
+ snd_printk(KERN_INFO SFX
+ "%s: Start delayed initialization\n",
+ pci_name(chip->pci));
+ if (azx_first_init(chip) < 0 ||
+ azx_probe_continue(chip) < 0) {
+ snd_printk(KERN_ERR SFX
+ "%s: initialization error\n",
+ pci_name(chip->pci));
+ chip->init_failed = true;
+ }
+ }
+ } else {
+ snd_printk(KERN_INFO SFX
+ "%s %s via VGA-switcheroo\n",
+ disabled ? "Disabling" : "Enabling",
+ pci_name(chip->pci));
+ if (disabled) {
+ azx_suspend(pci, PMSG_FREEZE);
+ chip->disabled = true;
+ snd_hda_lock_devices(chip->bus);
+ } else {
+ snd_hda_unlock_devices(chip->bus);
+ chip->disabled = false;
+ azx_resume(pci);
+ }
+ }
+}
+
+static bool azx_vs_can_switch(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct azx *chip = card->private_data;
+
+ if (chip->init_failed)
+ return false;
+ if (chip->disabled || !chip->bus)
+ return true;
+ if (snd_hda_lock_devices(chip->bus))
+ return false;
+ snd_hda_unlock_devices(chip->bus);
+ return true;
+}
+
+static void __devinit init_vga_switcheroo(struct azx *chip)
+{
+ struct pci_dev *p = get_bound_vga(chip->pci);
+ if (p) {
+ snd_printk(KERN_INFO SFX
+ "%s: Handle VGA-switcheroo audio client\n",
+ pci_name(chip->pci));
+ chip->use_vga_switcheroo = 1;
+ pci_dev_put(p);
+ }
+}
+
+static const struct vga_switcheroo_client_ops azx_vs_ops = {
+ .set_gpu_state = azx_vs_set_state,
+ .can_switch = azx_vs_can_switch,
+};
+
+static int __devinit register_vga_switcheroo(struct azx *chip)
+{
+ if (!chip->use_vga_switcheroo)
+ return 0;
+ /* FIXME: currently only handling DIS controller
+ * is there any machine with two switchable HDMI audio controllers?
+ */
+ return vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops,
+ VGA_SWITCHEROO_DIS,
+ chip->bus != NULL);
+}
+#else
+#define init_vga_switcheroo(chip) /* NOP */
+#define register_vga_switcheroo(chip) 0
+#endif /* SUPPORT_VGA_SWITCHER */
+
/*
* destructor
*/
@@ -2438,6 +2589,12 @@ static int azx_free(struct azx *chip)
azx_notifier_unregister(chip);
+ if (use_vga_switcheroo(chip)) {
+ if (chip->disabled && chip->bus)
+ snd_hda_unlock_devices(chip->bus);
+ vga_switcheroo_unregister_client(chip->pci);
+ }
+
if (chip->initialized) {
azx_clear_irq_pending(chip);
for (i = 0; i < chip->num_streams; i++)
@@ -2467,7 +2624,8 @@ static int azx_free(struct azx *chip)
mark_pages_wc(chip, &chip->posbuf, false);
snd_dma_free_pages(&chip->posbuf);
}
- pci_release_regions(chip->pci);
+ if (chip->region_requested)
+ pci_release_regions(chip->pci);
pci_disable_device(chip->pci);
kfree(chip->azx_dev);
kfree(chip);
@@ -2481,6 +2639,45 @@ static int azx_dev_free(struct snd_device *device)
}
/*
+ * Check of disabled HDMI controller by vga-switcheroo
+ */
+static struct pci_dev __devinit *get_bound_vga(struct pci_dev *pci)
+{
+ struct pci_dev *p;
+
+ /* check only discrete GPU */
+ switch (pci->vendor) {
+ case PCI_VENDOR_ID_ATI:
+ case PCI_VENDOR_ID_AMD:
+ case PCI_VENDOR_ID_NVIDIA:
+ if (pci->devfn == 1) {
+ p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus),
+ pci->bus->number, 0);
+ if (p) {
+ if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA)
+ return p;
+ pci_dev_put(p);
+ }
+ }
+ break;
+ }
+ return NULL;
+}
+
+static bool __devinit check_hdmi_disabled(struct pci_dev *pci)
+{
+ bool vga_inactive = false;
+ struct pci_dev *p = get_bound_vga(pci);
+
+ if (p) {
+ if (vga_default_device() && p != vga_default_device())
+ vga_inactive = true;
+ pci_dev_put(p);
+ }
+ return vga_inactive;
+}
+
+/*
* white/black-listing for position_fix
*/
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
@@ -2551,6 +2748,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
/* forced codec slots */
SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
+ /* WinFast VP200 H (Teradici) user reported broken communication */
+ SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101),
{}
};
@@ -2655,12 +2854,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
int dev, unsigned int driver_caps,
struct azx **rchip)
{
- struct azx *chip;
- int i, err;
- unsigned short gcap;
static struct snd_device_ops ops = {
.dev_free = azx_dev_free,
};
+ struct azx *chip;
+ int err;
*rchip = NULL;
@@ -2686,6 +2884,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->dev_index = dev;
INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
INIT_LIST_HEAD(&chip->pcm_list);
+ init_vga_switcheroo(chip);
chip->position_fix[0] = chip->position_fix[1] =
check_position_fix(chip, position_fix[dev]);
@@ -2713,6 +2912,53 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
}
+ if (check_hdmi_disabled(pci)) {
+ snd_printk(KERN_INFO SFX "VGA controller for %s is disabled\n",
+ pci_name(pci));
+ if (use_vga_switcheroo(chip)) {
+ snd_printk(KERN_INFO SFX "Delaying initialization\n");
+ chip->disabled = true;
+ goto ok;
+ }
+ kfree(chip);
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ err = azx_first_init(chip);
+ if (err < 0) {
+ azx_free(chip);
+ return err;
+ }
+
+ ok:
+ err = register_vga_switcheroo(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR SFX
+ "Error registering VGA-switcheroo client\n");
+ azx_free(chip);
+ return err;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_printk(KERN_ERR SFX "Error creating device [card]!\n");
+ azx_free(chip);
+ return err;
+ }
+
+ *rchip = chip;
+ return 0;
+}
+
+static int DELAYED_INIT_MARK azx_first_init(struct azx *chip)
+{
+ int dev = chip->dev_index;
+ struct pci_dev *pci = chip->pci;
+ struct snd_card *card = chip->card;
+ int i, err;
+ unsigned short gcap;
+
#if BITS_PER_LONG != 64
/* Fix up base address on ULI M5461 */
if (chip->driver_type == AZX_DRIVER_ULI) {
@@ -2724,28 +2970,23 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
#endif
err = pci_request_regions(pci, "ICH HD audio");
- if (err < 0) {
- kfree(chip);
- pci_disable_device(pci);
+ if (err < 0)
return err;
- }
+ chip->region_requested = 1;
chip->addr = pci_resource_start(pci, 0);
chip->remap_addr = pci_ioremap_bar(pci, 0);
if (chip->remap_addr == NULL) {
snd_printk(KERN_ERR SFX "ioremap error\n");
- err = -ENXIO;
- goto errout;
+ return -ENXIO;
}
if (chip->msi)
if (pci_enable_msi(pci) < 0)
chip->msi = 0;
- if (azx_acquire_irq(chip, 0) < 0) {
- err = -EBUSY;
- goto errout;
- }
+ if (azx_acquire_irq(chip, 0) < 0)
+ return -EBUSY;
pci_set_master(pci);
synchronize_irq(chip->irq);
@@ -2824,7 +3065,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
GFP_KERNEL);
if (!chip->azx_dev) {
snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n");
- goto errout;
+ return -ENOMEM;
}
for (i = 0; i < chip->num_streams; i++) {
@@ -2834,7 +3075,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
BDL_SIZE, &chip->azx_dev[i].bdl);
if (err < 0) {
snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
- goto errout;
+ return -ENOMEM;
}
mark_pages_wc(chip, &chip->azx_dev[i].bdl, true);
}
@@ -2844,13 +3085,13 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->num_streams * 8, &chip->posbuf);
if (err < 0) {
snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
- goto errout;
+ return -ENOMEM;
}
mark_pages_wc(chip, &chip->posbuf, true);
/* allocate CORB/RIRB */
err = azx_alloc_cmd_io(chip);
if (err < 0)
- goto errout;
+ return err;
/* initialize streams */
azx_init_stream(chip);
@@ -2862,14 +3103,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
/* codec detection */
if (!chip->codec_mask) {
snd_printk(KERN_ERR SFX "no codecs found!\n");
- err = -ENODEV;
- goto errout;
- }
-
- err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
- if (err <0) {
- snd_printk(KERN_ERR SFX "Error creating device [card]!\n");
- goto errout;
+ return -ENODEV;
}
strcpy(card->driver, "HDA-Intel");
@@ -2879,12 +3113,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
"%s at 0x%lx irq %i",
card->shortname, chip->addr, chip->irq);
- *rchip = chip;
return 0;
-
- errout:
- azx_free(chip);
- return err;
}
static void power_down_all_codecs(struct azx *chip)
@@ -2929,6 +3158,27 @@ static int __devinit azx_probe(struct pci_dev *pci,
goto out_free;
card->private_data = chip;
+ if (!chip->disabled) {
+ err = azx_probe_continue(chip);
+ if (err < 0)
+ goto out_free;
+ }
+
+ pci_set_drvdata(pci, card);
+
+ dev++;
+ return 0;
+
+out_free:
+ snd_card_free(card);
+ return err;
+}
+
+static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip)
+{
+ int dev = chip->dev_index;
+ int err;
+
#ifdef CONFIG_SND_HDA_INPUT_BEEP
chip->beep_mode = beep_mode[dev];
#endif
@@ -2962,25 +3212,26 @@ static int __devinit azx_probe(struct pci_dev *pci,
if (err < 0)
goto out_free;
- err = snd_card_register(card);
+ err = snd_card_register(chip->card);
if (err < 0)
goto out_free;
- pci_set_drvdata(pci, card);
chip->running = 1;
power_down_all_codecs(chip);
azx_notifier_register(chip);
- dev++;
- return err;
+ return 0;
+
out_free:
- snd_card_free(card);
+ chip->init_failed = 1;
return err;
}
static void __devexit azx_remove(struct pci_dev *pci)
{
- snd_card_free(pci_get_drvdata(pci));
+ struct snd_card *card = pci_get_drvdata(pci);
+ if (card)
+ snd_card_free(card);
pci_set_drvdata(pci, NULL);
}
@@ -3116,6 +3367,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
.driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB },
#endif
+ /* CTHDA chips */
+ { PCI_DEVICE(0x1102, 0x0010),
+ .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
+ { PCI_DEVICE(0x1102, 0x0012),
+ .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
/* Vortex86MX */
{ PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC },
/* VMware HDAudio */
@@ -3134,7 +3390,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
MODULE_DEVICE_TABLE(pci, azx_ids);
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver azx_driver = {
.name = KBUILD_MODNAME,
.id_table = azx_ids,
.probe = azx_probe,
@@ -3145,15 +3401,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_azx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_azx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_azx_init)
-module_exit(alsa_card_azx_exit)
+module_pci_driver(azx_driver);
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d68948499fb..2dd1c113a4c 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -17,6 +17,7 @@
#include <sound/jack.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_jack.h"
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index c66655cf413..8ae52465ec5 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -12,6 +12,8 @@
#ifndef __SOUND_HDA_JACK_H
#define __SOUND_HDA_JACK_H
+struct auto_pin_cfg;
+
struct hda_jack_tbl {
hda_nid_t nid;
unsigned char action; /* event action (0 = none) */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 0ec9248165b..9a096a8e0fc 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -262,6 +262,8 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
const struct hda_input_mux *imux,
struct snd_ctl_elem_value *ucontrol, hda_nid_t nid,
unsigned int *cur_val);
+int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
+ int index, int *type_index_ret);
/*
* Channel mode helper
@@ -393,72 +395,7 @@ struct hda_bus_unsolicited {
struct hda_bus *bus;
};
-/*
- * Helper for automatic pin configuration
- */
-
-enum {
- AUTO_PIN_MIC,
- AUTO_PIN_LINE_IN,
- AUTO_PIN_CD,
- AUTO_PIN_AUX,
- AUTO_PIN_LAST
-};
-
-enum {
- AUTO_PIN_LINE_OUT,
- AUTO_PIN_SPEAKER_OUT,
- AUTO_PIN_HP_OUT
-};
-
-#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS
-#define AUTO_CFG_MAX_INS 8
-
-struct auto_pin_cfg_item {
- hda_nid_t pin;
- int type;
-};
-
-struct auto_pin_cfg;
-const char *hda_get_autocfg_input_label(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- int input);
-int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg,
- char *label, int maxlen, int *indexp);
-int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
- int index, int *type_index_ret);
-
-enum {
- INPUT_PIN_ATTR_UNUSED, /* pin not connected */
- INPUT_PIN_ATTR_INT, /* internal mic/line-in */
- INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */
- INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */
- INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */
- INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */
-};
-
-int snd_hda_get_input_pin_attr(unsigned int def_conf);
-
-struct auto_pin_cfg {
- int line_outs;
- /* sorted in the order of Front/Surr/CLFE/Side */
- hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
- int speaker_outs;
- hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
- int hp_outs;
- int line_out_type; /* AUTO_PIN_XXX_OUT */
- hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
- int num_inputs;
- struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS];
- int dig_outs;
- hda_nid_t dig_out_pins[2];
- hda_nid_t dig_in_pin;
- hda_nid_t mono_out_pin;
- int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
- int dig_in_type; /* HDA_PCM_TYPE_XXX */
-};
-
+/* helper macros to retrieve pin default-config values */
#define get_defcfg_connect(cfg) \
((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
#define get_defcfg_association(cfg) \
@@ -472,19 +409,6 @@ struct auto_pin_cfg {
#define get_defcfg_misc(cfg) \
((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT)
-/* 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 */
-
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids,
- unsigned int cond_flags);
-
-/* older function */
-#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
- snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
-
/* amp values */
#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8))
@@ -502,6 +426,46 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
#define PIN_HP_AMP (AC_PINCTL_HP_EN)
+unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin);
+int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int val, bool cached);
+
+/**
+ * _snd_hda_set_pin_ctl - Set a pin-control value safely
+ * @codec: the codec instance
+ * @pin: the pin NID to set the control
+ * @val: the pin-control value (AC_PINCTL_* bits)
+ *
+ * This function sets the pin-control value to the given pin, but
+ * filters out the invalid pin-control bits when the pin has no such
+ * capabilities. For example, when PIN_HP is passed but the pin has no
+ * HP-drive capability, the HP bit is omitted.
+ *
+ * The function doesn't check the input VREF capability bits, though.
+ * Use snd_hda_get_default_vref() to guess the right value.
+ * Also, this function is only for analog pins, not for HDMI pins.
+ */
+static inline int
+snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val)
+{
+ return _snd_hda_set_pin_ctl(codec, pin, val, false);
+}
+
+/**
+ * snd_hda_set_pin_ctl_cache - Set a pin-control value safely
+ * @codec: the codec instance
+ * @pin: the pin NID to set the control
+ * @val: the pin-control value (AC_PINCTL_* bits)
+ *
+ * Just like snd_hda_set_pin_ctl() but write to cache as well.
+ */
+static inline int
+snd_hda_set_pin_ctl_cache(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int val)
+{
+ return _snd_hda_set_pin_ctl(codec, pin, val, true);
+}
+
/*
* get widget capabilities
*/
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 7143393927d..d8b2d6dee98 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -28,6 +28,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -1742,9 +1743,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
if (! ad198x_eapd_put(kcontrol, ucontrol))
return 0;
/* change speaker pin appropriately */
- snd_hda_codec_write(codec, 0x05, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->cur_eapd ? PIN_OUT : 0);
+ snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
/* toggle HP mute appropriately */
snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
HDA_AMP_MUTE,
@@ -3103,7 +3102,7 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec,
int dac_idx)
{
/* set as output */
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_set_pin_ctl(codec, nid, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
switch (nid) {
case 0x11: /* port-A - DAC 03 */
@@ -3157,6 +3156,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec)
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
int type = cfg->inputs[i].type;
+ int val;
switch (nid) {
case 0x15: /* port-C */
snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
@@ -3165,8 +3165,10 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec)
snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
break;
}
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN);
+ val = PIN_IN;
+ if (type == AUTO_PIN_MIC)
+ val |= snd_hda_get_default_vref(codec, nid);
+ snd_hda_set_pin_ctl(codec, nid, val);
if (nid != AD1988_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 09ccfabb4a1..19ae14f739c 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -26,6 +26,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
/*
*/
@@ -341,8 +342,7 @@ static int ca0110_build_pcms(struct hda_codec *codec)
static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_set_pin_ctl(codec, pin, PIN_HP);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
@@ -356,8 +356,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80);
+ snd_hda_set_pin_ctl(codec, pin, PIN_IN |
+ snd_hda_get_default_vref(codec, pin));
if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 21d91d580da..d0d3540e39e 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -30,6 +30,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#define WIDGET_CHIP_CTRL 0x15
#define WIDGET_DSP_CTRL 0x16
@@ -239,8 +240,7 @@ enum get_set {
static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_set_pin_ctl(codec, pin, PIN_HP);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
@@ -254,9 +254,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_VREF80);
+ snd_hda_set_pin_ctl(codec, pin, PIN_IN |
+ snd_hda_get_default_vref(codec, pin));
if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c83ccdba1e5..9647ed4d792 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -26,6 +26,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_jack.h"
#include <sound/tlv.h>
@@ -933,8 +934,7 @@ static void cs_automute(struct hda_codec *codec)
pin_ctl = 0;
nid = cfg->speaker_pins[i];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl);
+ snd_hda_set_pin_ctl(codec, nid, pin_ctl);
}
if (spec->gpio_eapd_hp) {
unsigned int gpio = hp_present ?
@@ -948,16 +948,14 @@ static void cs_automute(struct hda_codec *codec)
/* mute HPs if spdif jack (SENSE_B) is present */
for (i = 0; i < cfg->hp_outs; i++) {
nid = cfg->hp_pins[i];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, nid,
(spdif_present && spec->sense_b) ? 0 : PIN_HP);
}
/* SPDIF TX on/off */
if (cfg->dig_outs) {
nid = cfg->dig_out_pins[0];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, nid,
spdif_present ? PIN_OUT : 0);
}
@@ -1024,13 +1022,11 @@ static void init_output(struct hda_codec *codec)
/* set appropriate pin controls */
for (i = 0; i < cfg->line_outs; i++)
- snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, cfg->line_out_pins[i], PIN_OUT);
/* HP */
for (i = 0; i < cfg->hp_outs; i++) {
hda_nid_t nid = cfg->hp_pins[i];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_set_pin_ctl(codec, nid, PIN_HP);
if (!cfg->speaker_outs)
continue;
if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
@@ -1041,8 +1037,7 @@ static void init_output(struct hda_codec *codec)
/* Speaker */
for (i = 0; i < cfg->speaker_outs; i++)
- snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, cfg->speaker_pins[i], PIN_OUT);
/* SPDIF is enabled on presence detect for CS421x */
if (spec->hp_detect || spec->spdif_detect)
@@ -1063,14 +1058,9 @@ static void init_input(struct hda_codec *codec)
continue;
/* set appropriate pin control and mute first */
ctl = PIN_IN;
- if (cfg->inputs[i].type == AUTO_PIN_MIC) {
- unsigned int caps = snd_hda_query_pin_caps(codec, pin);
- caps >>= AC_PINCAP_VREF_SHIFT;
- if (caps & AC_PINCAP_VREF_80)
- ctl = PIN_VREF80;
- }
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+ if (cfg->inputs[i].type == AUTO_PIN_MIC)
+ ctl |= snd_hda_get_default_vref(codec, pin);
+ snd_hda_set_pin_ctl(codec, pin, ctl);
snd_hda_codec_write(codec, spec->adc_nid[i], 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_MUTE(spec->adc_idx[i]));
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index b6767b4ced4..c8fdaaefe70 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -29,6 +29,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#define NUM_PINS 11
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d906c5b74cf..3acb5824ad3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -30,6 +30,7 @@
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -66,6 +67,7 @@ struct imux_info {
};
struct conexant_spec {
+ struct hda_gen_spec gen;
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
@@ -141,6 +143,7 @@ struct conexant_spec {
unsigned int hp_laptop:1;
unsigned int asus:1;
unsigned int pin_eapd_ctrls:1;
+ unsigned int fixup_stereo_dmic:1;
unsigned int adc_switching:1;
@@ -1601,17 +1604,13 @@ static void cxt5051_update_speaker(struct hda_codec *codec)
unsigned int pinctl;
/* headphone pin */
pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0;
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x16, pinctl);
/* speaker pin */
pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1a, pinctl);
/* on ideapad there is an additional speaker (subwoofer) to mute */
if (spec->ideapad)
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1b, pinctl);
}
/* turn on/off EAPD (+ mute HP) as a master switch */
@@ -1996,8 +1995,7 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
/* Port A (HP) */
pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0;
- snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x19, pinctl);
/* Port D (HP/LO) */
pinctl = spec->cur_eapd ? spec->port_d_mode : 0;
@@ -2010,13 +2008,11 @@ static void cxt5066_update_speaker(struct hda_codec *codec)
if (!hp_port_d_present(spec))
pinctl = 0;
}
- snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1c, pinctl);
/* CLASS_D AMP */
pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1f, pinctl);
}
/* turn on/off EAPD (+ mute HP) as a master switch */
@@ -2047,8 +2043,7 @@ static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec)
/* Even though port F is the DC input, the bias is controlled on port B.
* we also leave that port as an active input (but unselected) in DC mode
* just in case that is necessary to make the bias setting take effect. */
- return snd_hda_codec_write_cache(codec, 0x1a, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ return snd_hda_set_pin_ctl_cache(codec, 0x1a,
cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index);
}
@@ -2081,14 +2076,14 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec)
}
/* disable DC (port F) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_set_pin_ctl(codec, 0x1e, 0);
/* external mic, port B */
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, 0x1a,
spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0);
/* internal mic, port C */
- snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, 0x1b,
spec->ext_mic_present ? 0 : PIN_VREF80);
}
@@ -3357,9 +3352,7 @@ static void do_automute(struct hda_codec *codec, int num_pins,
struct conexant_spec *spec = codec->spec;
int i;
for (i = 0; i < num_pins; i++)
- snd_hda_codec_write(codec, pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- on ? PIN_OUT : 0);
+ snd_hda_set_pin_ctl(codec, pins[i], on ? PIN_OUT : 0);
if (spec->pin_eapd_ctrls)
cx_auto_turn_eapd(codec, num_pins, pins, on);
}
@@ -3976,8 +3969,7 @@ static void cx_auto_init_output(struct hda_codec *codec)
if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
AC_PINCAP_HP_DRV)
val |= AC_PINCTL_HP_EN;
- snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, cfg->hp_pins[i], val);
}
mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
@@ -4030,13 +4022,11 @@ static void cx_auto_init_input(struct hda_codec *codec)
}
for (i = 0; i < cfg->num_inputs; i++) {
- unsigned int type;
+ hda_nid_t pin = cfg->inputs[i].pin;
+ unsigned int type = PIN_IN;
if (cfg->inputs[i].type == AUTO_PIN_MIC)
- type = PIN_VREF80;
- else
- type = PIN_IN;
- snd_hda_codec_write(codec, cfg->inputs[i].pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, type);
+ type |= snd_hda_get_default_vref(codec, pin);
+ snd_hda_set_pin_ctl(codec, pin, type);
}
if (spec->auto_mic) {
@@ -4063,11 +4053,9 @@ static void cx_auto_init_digital(struct hda_codec *codec)
struct auto_pin_cfg *cfg = &spec->autocfg;
if (spec->multiout.dig_out_nid)
- snd_hda_codec_write(codec, cfg->dig_out_pins[0], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, cfg->dig_out_pins[0], PIN_OUT);
if (spec->dig_in_nid)
- snd_hda_codec_write(codec, cfg->dig_in_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+ snd_hda_set_pin_ctl(codec, cfg->dig_in_pin, PIN_IN);
}
static int cx_auto_init(struct hda_codec *codec)
@@ -4084,9 +4072,9 @@ static int cx_auto_init(struct hda_codec *codec)
static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
const char *dir, int cidx,
- hda_nid_t nid, int hda_dir, int amp_idx)
+ hda_nid_t nid, int hda_dir, int amp_idx, int chs)
{
- static char name[32];
+ static char name[44];
static struct snd_kcontrol_new knew[] = {
HDA_CODEC_VOLUME(name, 0, 0, 0),
HDA_CODEC_MUTE(name, 0, 0, 0),
@@ -4096,7 +4084,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
for (i = 0; i < 2; i++) {
struct snd_kcontrol *kctl;
- knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+ knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx,
hda_dir);
knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
knew[i].index = cidx;
@@ -4115,7 +4103,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
}
#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \
- cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+ cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3)
#define cx_auto_add_pb_volume(codec, nid, str, idx) \
cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
@@ -4185,6 +4173,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
return 0;
}
+/* Returns zero if this is a normal stereo channel, and non-zero if it should
+ be split in two independent channels.
+ dest_label must be at least 44 characters. */
+static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label,
+ char *dest_label, int nid)
+{
+ struct conexant_spec *spec = codec->spec;
+ int i;
+
+ if (!spec->fixup_stereo_dmic)
+ return 0;
+
+ for (i = 0; i < AUTO_CFG_MAX_INS; i++) {
+ int def_conf;
+ if (spec->autocfg.inputs[i].pin != nid)
+ continue;
+
+ if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC)
+ return 0;
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT)
+ return 0;
+
+ /* Finally found the inverted internal mic! */
+ snprintf(dest_label, 44, "Inverted %s", label);
+ return 1;
+ }
+ return 0;
+}
+
static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
const char *label, const char *pfx,
int cidx)
@@ -4193,14 +4211,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
int i;
for (i = 0; i < spec->num_adc_nids; i++) {
+ char rightch_label[44];
hda_nid_t adc_nid = spec->adc_nids[i];
int idx = get_input_connection(codec, adc_nid, nid);
if (idx < 0)
continue;
if (codec->single_adc_amp)
idx = 0;
+
+ if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) {
+ /* Make two independent kcontrols for left and right */
+ int err = cx_auto_add_volume_idx(codec, label, pfx,
+ cidx, adc_nid, HDA_INPUT, idx, 1);
+ if (err < 0)
+ return err;
+ return cx_auto_add_volume_idx(codec, rightch_label, pfx,
+ cidx, adc_nid, HDA_INPUT, idx, 2);
+ }
return cx_auto_add_volume_idx(codec, label, pfx,
- cidx, adc_nid, HDA_INPUT, idx);
+ cidx, adc_nid, HDA_INPUT, idx, 3);
}
return 0;
}
@@ -4213,9 +4242,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx,
int i, con;
nid = spec->imux_info[idx].pin;
- if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
+ if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) {
+ char rightch_label[44];
+ if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) {
+ int err = cx_auto_add_volume_idx(codec, label, " Boost",
+ cidx, nid, HDA_INPUT, 0, 1);
+ if (err < 0)
+ return err;
+ return cx_auto_add_volume_idx(codec, rightch_label, " Boost",
+ cidx, nid, HDA_INPUT, 0, 2);
+ }
return cx_auto_add_volume(codec, label, " Boost", cidx,
nid, HDA_INPUT);
+ }
con = __select_input_connection(codec, spec->imux_info[idx].adc, nid,
&mux, false, 0);
if (con < 0)
@@ -4370,37 +4409,21 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
/*
* pin fix-up
*/
-struct cxt_pincfg {
- hda_nid_t nid;
- u32 val;
-};
-
-static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg)
-{
- for (; cfg->nid; cfg++)
- snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-
-}
-
-static void apply_pin_fixup(struct hda_codec *codec,
- const struct snd_pci_quirk *quirk,
- const struct cxt_pincfg **table)
-{
- quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
- if (quirk) {
- snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n",
- quirk->name);
- apply_pincfg(codec, table[quirk->value]);
- }
-}
-
enum {
CXT_PINCFG_LENOVO_X200,
CXT_PINCFG_LENOVO_TP410,
+ CXT_FIXUP_STEREO_DMIC,
};
+static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct conexant_spec *spec = codec->spec;
+ spec->fixup_stereo_dmic = 1;
+}
+
/* ThinkPad X200 & co with cxt5051 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
+static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
{ 0x17, 0x21a11000 }, /* dock-mic */
{ 0x19, 0x2121103f }, /* dock-HP */
@@ -4409,16 +4432,26 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
};
/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = {
+static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = {
{ 0x19, 0x042110ff }, /* HP (seq# overridden) */
{ 0x1a, 0x21a190f0 }, /* dock-mic */
{ 0x1c, 0x212140ff }, /* dock-HP */
{}
};
-static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
- [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
- [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410,
+static const struct hda_fixup cxt_fixups[] = {
+ [CXT_PINCFG_LENOVO_X200] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = cxt_pincfg_lenovo_x200,
+ },
+ [CXT_PINCFG_LENOVO_TP410] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = cxt_pincfg_lenovo_tp410,
+ },
+ [CXT_FIXUP_STEREO_DMIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_stereo_dmic,
+ },
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -4432,6 +4465,7 @@ 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, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
{}
};
@@ -4471,13 +4505,16 @@ static int patch_conexant_auto(struct hda_codec *codec)
case 0x14f15051:
add_cx5051_fake_mutes(codec);
codec->pin_amp_workaround = 1;
- apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl);
+ snd_hda_pick_fixup(codec, NULL, cxt5051_fixups, cxt_fixups);
break;
default:
codec->pin_amp_workaround = 1;
- apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl);
+ snd_hda_pick_fixup(codec, NULL, cxt5066_fixups, cxt_fixups);
+ break;
}
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
/* Show mute-led control only on HP laptops
* This is a sort of white-list: on HP laptops, EAPD corresponds
* only to the mute-LED without actualy amp function. Meanwhile,
@@ -4556,6 +4593,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_conexant_auto },
{ .id = 0x14f150b9, .name = "CX20665",
.patch = patch_conexant_auto },
+ { .id = 0x14f1510f, .name = "CX20751/2",
+ .patch = patch_conexant_auto },
+ { .id = 0x14f15110, .name = "CX20751/2",
+ .patch = patch_conexant_auto },
+ { .id = 0x14f15111, .name = "CX20753/4",
+ .patch = patch_conexant_auto },
{} /* terminator */
};
@@ -4576,6 +4619,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab");
MODULE_ALIAS("snd-hda-codec-id:14f150ac");
MODULE_ALIAS("snd-hda-codec-id:14f150b8");
MODULE_ALIAS("snd-hda-codec-id:14f150b9");
+MODULE_ALIAS("snd-hda-codec-id:14f1510f");
+MODULE_ALIAS("snd-hda-codec-id:14f15110");
+MODULE_ALIAS("snd-hda-codec-id:14f15111");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 83f345f3c96..ad319d4dc32 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1592,10 +1592,10 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
unsigned int dataDCC2, channel_id;
int i;
struct hdmi_spec *spec = codec->spec;
- struct hda_spdif_out *spdif =
- snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
+ struct hda_spdif_out *spdif;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
chs = substream->runtime->channels;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 818f90bc7d5..224410e8e9e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6,7 +6,7 @@
* Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw>
* PeiSen Hou <pshou@realtek.com.tw>
* Takashi Iwai <tiwai@suse.de>
- * Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
+ * Jonathan Woithe <jwoithe@just42.net>
*
* This driver is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -32,6 +32,7 @@
#include <sound/jack.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -66,8 +67,6 @@ struct alc_customize_define {
unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */
};
-struct alc_fixup;
-
struct alc_multi_io {
hda_nid_t pin; /* multi-io widget pin NID */
hda_nid_t dac; /* DAC to be connected */
@@ -82,19 +81,33 @@ enum {
#define MAX_VOL_NIDS 0x40
+/* make compatible with old code */
+#define alc_apply_pincfgs snd_hda_apply_pincfgs
+#define alc_apply_fixup snd_hda_apply_fixup
+#define alc_pick_fixup snd_hda_pick_fixup
+#define alc_fixup hda_fixup
+#define alc_pincfg hda_pintbl
+#define alc_model_fixup hda_model_fixup
+
+#define ALC_FIXUP_PINS HDA_FIXUP_PINS
+#define ALC_FIXUP_VERBS HDA_FIXUP_VERBS
+#define ALC_FIXUP_FUNC HDA_FIXUP_FUNC
+
+#define ALC_FIXUP_ACT_PRE_PROBE HDA_FIXUP_ACT_PRE_PROBE
+#define ALC_FIXUP_ACT_PROBE HDA_FIXUP_ACT_PROBE
+#define ALC_FIXUP_ACT_INIT HDA_FIXUP_ACT_INIT
+#define ALC_FIXUP_ACT_BUILD HDA_FIXUP_ACT_BUILD
+
+
struct alc_spec {
+ struct hda_gen_spec gen;
+
/* codec parameterization */
const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
unsigned int num_mixers;
const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
- const struct hda_verb *init_verbs[10]; /* initialization verbs
- * don't forget NULL
- * termination!
- */
- unsigned int num_init_verbs;
-
char stream_name_analog[32]; /* analog PCM stream */
const struct hda_pcm_stream *stream_analog_playback;
const struct hda_pcm_stream *stream_analog_capture;
@@ -210,11 +223,6 @@ struct alc_spec {
unsigned int pll_coef_idx, pll_coef_bit;
unsigned int coef0;
- /* fix-up list */
- int fixup_id;
- const struct alc_fixup *fixup_list;
- const char *fixup_name;
-
/* multi-io */
int multi_ios;
struct alc_multi_io multi_io[4];
@@ -319,13 +327,16 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
/* for shared I/O, change the pin-control accordingly */
if (spec->shared_mic_hp) {
+ 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 jack.
*/
- snd_hda_codec_write(codec, spec->autocfg.inputs[1].pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->cur_mux[adc_idx] ?
- PIN_VREF80 : PIN_HP);
+ if (spec->cur_mux[adc_idx])
+ val = snd_hda_get_default_vref(codec, pin) | PIN_IN;
+ else
+ val = PIN_HP;
+ snd_hda_set_pin_ctl(codec, pin, val);
spec->automute_speaker = !spec->cur_mux[adc_idx];
call_update_outputs(codec);
}
@@ -338,7 +349,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
nid = get_capsrc(spec, adc_idx);
/* no selection? */
- num_conns = snd_hda_get_conn_list(codec, nid, NULL);
+ num_conns = snd_hda_get_num_conns(codec, nid);
if (num_conns <= 1)
return 1;
@@ -376,25 +387,9 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
int auto_pin_type)
{
unsigned int val = PIN_IN;
-
- if (auto_pin_type == AUTO_PIN_MIC) {
- unsigned int pincap;
- unsigned int oldval;
- oldval = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- pincap = snd_hda_query_pin_caps(codec, nid);
- pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
- /* if the default pin setup is vref50, we give it priority */
- if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
- val = PIN_VREF80;
- else if (pincap & AC_PINCAP_VREF_50)
- val = PIN_VREF50;
- else if (pincap & AC_PINCAP_VREF_100)
- val = PIN_VREF100;
- else if (pincap & AC_PINCAP_VREF_GRD)
- val = PIN_VREFGRD;
- }
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ if (auto_pin_type == AUTO_PIN_MIC)
+ val |= snd_hda_get_default_vref(codec, nid);
+ snd_hda_set_pin_ctl(codec, nid, val);
}
/*
@@ -409,13 +404,6 @@ static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix)
spec->mixers[spec->num_mixers++] = mix;
}
-static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
-{
- if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs)))
- return;
- spec->init_verbs[spec->num_init_verbs++] = verb;
-}
-
/*
* GPIO setup tables, used in initialization
*/
@@ -517,9 +505,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
} else
val = 0;
val |= pin_bits;
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- val);
+ snd_hda_set_pin_ctl(codec, nid, val);
break;
case ALC_AUTOMUTE_AMP:
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -1200,6 +1186,16 @@ static void alc_auto_check_switches(struct hda_codec *codec)
*/
#define ALC_FIXUP_SKU_IGNORE (2)
+static void alc_fixup_sku_ignore(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->cdefine.fixup = 1;
+ spec->cdefine.sku_cfg = ALC_FIXUP_SKU_IGNORE;
+ }
+}
+
static int alc_auto_parse_customize_define(struct hda_codec *codec)
{
unsigned int ass, tmp, i;
@@ -1403,178 +1399,6 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports)
}
/*
- * Fix-up pin default configurations and add default verbs
- */
-
-struct alc_pincfg {
- hda_nid_t nid;
- u32 val;
-};
-
-struct alc_model_fixup {
- const int id;
- const char *name;
-};
-
-struct alc_fixup {
- int type;
- bool chained;
- int chain_id;
- union {
- unsigned int sku;
- const struct alc_pincfg *pins;
- const struct hda_verb *verbs;
- void (*func)(struct hda_codec *codec,
- const struct alc_fixup *fix,
- int action);
- } v;
-};
-
-enum {
- ALC_FIXUP_INVALID,
- ALC_FIXUP_SKU,
- ALC_FIXUP_PINS,
- ALC_FIXUP_VERBS,
- ALC_FIXUP_FUNC,
-};
-
-enum {
- ALC_FIXUP_ACT_PRE_PROBE,
- ALC_FIXUP_ACT_PROBE,
- ALC_FIXUP_ACT_INIT,
- ALC_FIXUP_ACT_BUILD,
-};
-
-static void alc_apply_pincfgs(struct hda_codec *codec,
- const struct alc_pincfg *cfg)
-{
- for (; cfg->nid; cfg++)
- snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-}
-
-static void alc_apply_fixup(struct hda_codec *codec, int action)
-{
- struct alc_spec *spec = codec->spec;
- int id = spec->fixup_id;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- const char *modelname = spec->fixup_name;
-#endif
- int depth = 0;
-
- if (!spec->fixup_list)
- return;
-
- while (id >= 0) {
- const struct alc_fixup *fix = spec->fixup_list + id;
- const struct alc_pincfg *cfg;
-
- switch (fix->type) {
- case ALC_FIXUP_SKU:
- if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply sku override for %s\n",
- codec->chip_name, modelname);
- spec->cdefine.sku_cfg = fix->v.sku;
- spec->cdefine.fixup = 1;
- break;
- case ALC_FIXUP_PINS:
- cfg = fix->v.pins;
- if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply pincfg for %s\n",
- codec->chip_name, modelname);
- alc_apply_pincfgs(codec, cfg);
- break;
- case ALC_FIXUP_VERBS:
- if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply fix-verbs for %s\n",
- codec->chip_name, modelname);
- add_verb(codec->spec, fix->v.verbs);
- break;
- case ALC_FIXUP_FUNC:
- if (!fix->v.func)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply fix-func for %s\n",
- codec->chip_name, modelname);
- fix->v.func(codec, fix, action);
- break;
- default:
- snd_printk(KERN_ERR "hda_codec: %s: "
- "Invalid fixup type %d\n",
- codec->chip_name, fix->type);
- break;
- }
- if (!fix->chained)
- break;
- if (++depth > 10)
- break;
- id = fix->chain_id;
- }
-}
-
-static void alc_pick_fixup(struct hda_codec *codec,
- const struct alc_model_fixup *models,
- const struct snd_pci_quirk *quirk,
- const struct alc_fixup *fixlist)
-{
- struct alc_spec *spec = codec->spec;
- const struct snd_pci_quirk *q;
- int id = -1;
- const char *name = NULL;
-
- /* when model=nofixup is given, don't pick up any fixups */
- if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
- spec->fixup_list = NULL;
- spec->fixup_id = -1;
- return;
- }
-
- if (codec->modelname && models) {
- while (models->name) {
- if (!strcmp(codec->modelname, models->name)) {
- id = models->id;
- name = models->name;
- break;
- }
- models++;
- }
- }
- if (id < 0) {
- q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
- if (q) {
- id = q->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- name = q->name;
-#endif
- }
- }
- if (id < 0) {
- for (q = quirk; q->subvendor; q++) {
- unsigned int vendorid =
- q->subdevice | (q->subvendor << 16);
- if (vendorid == codec->subsystem_id) {
- id = q->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- name = q->name;
-#endif
- break;
- }
- }
- }
-
- spec->fixup_id = id;
- if (id >= 0) {
- spec->fixup_list = fixlist;
- spec->fixup_name = name;
- }
-}
-
-/*
* COEF access helper functions
*/
static int alc_read_coef_idx(struct hda_codec *codec,
@@ -1621,8 +1445,7 @@ static void alc_auto_init_digital(struct hda_codec *codec)
pin = spec->autocfg.dig_out_pins[i];
if (!pin)
continue;
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, pin, PIN_OUT);
if (!i)
dac = spec->multiout.dig_out_nid;
else
@@ -1635,9 +1458,7 @@ static void alc_auto_init_digital(struct hda_codec *codec)
}
pin = spec->autocfg.dig_in_pin;
if (pin)
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_IN);
+ snd_hda_set_pin_ctl(codec, pin, PIN_IN);
}
/* parse digital I/Os and set up NIDs in BIOS auto-parse mode */
@@ -2068,7 +1889,6 @@ static void alc_auto_init_std(struct hda_codec *codec);
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int i;
if (spec->init_hook)
spec->init_hook(codec);
@@ -2076,8 +1896,6 @@ static int alc_init(struct hda_codec *codec)
alc_fix_pll(codec);
alc_auto_init_amp(codec, spec->init_amp);
- for (i = 0; i < spec->num_init_verbs; i++)
- snd_hda_sequence_write(codec, spec->init_verbs[i]);
alc_init_special_input_src(codec);
alc_auto_init_std(codec);
@@ -2550,6 +2368,7 @@ static struct alc_codec_rename_table rename_tbl[] = {
{ 0x10ec0269, 0xffff, 0xa023, "ALC259" },
{ 0x10ec0269, 0xffff, 0x6023, "ALC281X" },
{ 0x10ec0269, 0x00f0, 0x0020, "ALC269VC" },
+ { 0x10ec0269, 0x00f0, 0x0030, "ALC269VD" },
{ 0x10ec0887, 0x00f0, 0x0030, "ALC887-VD" },
{ 0x10ec0888, 0x00f0, 0x0030, "ALC888-VD" },
{ 0x10ec0888, 0xf0f0, 0x3020, "ALC886" },
@@ -2725,7 +2544,6 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
hda_nid_t src;
- const hda_nid_t *list;
unsigned int caps = get_wcaps(codec, nid);
int type = get_wcaps_type(caps);
@@ -2743,13 +2561,14 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
cap_nids[nums] = src;
break;
}
- n = snd_hda_get_conn_list(codec, src, &list);
+ n = snd_hda_get_num_conns(codec, src);
if (n > 1) {
cap_nids[nums] = src;
break;
} else if (n != 1)
break;
- src = *list;
+ if (snd_hda_get_connections(codec, src, &src, 1) != 1)
+ break;
}
if (++nums >= max_nums)
break;
@@ -2856,8 +2675,7 @@ static int alc_auto_create_shared_input(struct hda_codec *codec)
static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
unsigned int pin_type)
{
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
+ snd_hda_set_pin_ctl(codec, nid, pin_type);
/* unmute pin */
if (nid_has_mute(codec, nid, HDA_OUTPUT))
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
@@ -2891,7 +2709,7 @@ static void alc_auto_init_analog_input(struct hda_codec *codec)
/* mute all loopback inputs */
if (spec->mixer_nid) {
- int nums = snd_hda_get_conn_list(codec, spec->mixer_nid, NULL);
+ int nums = snd_hda_get_num_conns(codec, spec->mixer_nid);
for (i = 0; i < nums; i++)
snd_hda_codec_write(codec, spec->mixer_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
@@ -3521,7 +3339,7 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec,
if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) {
type = ALC_CTL_WIDGET_MUTE;
val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT);
- } else if (snd_hda_get_conn_list(codec, nid, NULL) == 1) {
+ } else if (snd_hda_get_num_conns(codec, nid) == 1) {
type = ALC_CTL_WIDGET_MUTE;
val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT);
} else {
@@ -3998,9 +3816,7 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output)
snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
if (output) {
- snd_hda_codec_update_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
+ snd_hda_set_pin_ctl_cache(codec, nid, PIN_OUT);
if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
@@ -4009,9 +3825,8 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output)
if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_update_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->multi_io[idx].ctl_in);
+ snd_hda_set_pin_ctl_cache(codec, nid,
+ spec->multi_io[idx].ctl_in);
}
return 0;
}
@@ -4084,7 +3899,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec)
nums = 0;
for (n = 0; n < spec->num_adc_nids; n++) {
hda_nid_t cap = spec->private_capsrc_nids[n];
- int num_conns = snd_hda_get_conn_list(codec, cap, NULL);
+ int num_conns = snd_hda_get_num_conns(codec, cap);
for (i = 0; i < imux->num_items; i++) {
hda_nid_t pin = spec->imux_pins[i];
if (pin) {
@@ -4213,7 +4028,7 @@ static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap,
if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
HDA_AMP_MUTE, 0);
- } else if (snd_hda_get_conn_list(codec, cap, NULL) > 1) {
+ } else if (snd_hda_get_num_conns(codec, cap) > 1) {
snd_hda_codec_write_cache(codec, cap, 0,
AC_VERB_SET_CONNECT_SEL, idx);
}
@@ -4427,6 +4242,25 @@ static int alc_parse_auto_config(struct hda_codec *codec,
return 1;
}
+/* common preparation job for alc_spec */
+static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
+{
+ struct alc_spec *spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ int err;
+
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+ spec->mixer_nid = mixer_nid;
+
+ err = alc_codec_rename_from_preset(codec);
+ if (err < 0) {
+ kfree(spec);
+ return err;
+ }
+ return 0;
+}
+
static int alc880_parse_auto_config(struct hda_codec *codec)
{
static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
@@ -4808,13 +4642,11 @@ static int patch_alc880(struct hda_codec *codec)
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
spec->need_dac_fix = 1;
alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl,
@@ -4890,7 +4722,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
spec->unsol_event = alc_sku_unsol_event;
- add_verb(codec->spec, alc_gpio1_init_verbs);
+ snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs);
}
}
@@ -5001,13 +4833,11 @@ static int patch_alc260(struct hda_codec *codec)
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x07);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x07;
+ spec = codec->spec;
alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -5171,8 +5001,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec,
val = snd_hda_codec_read(codec, nids[i], 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
val |= AC_PINCTL_VREF_80;
- snd_hda_codec_write(codec, nids[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, nids[i], val);
spec->keep_vref_in_automute = 1;
break;
}
@@ -5193,8 +5022,7 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
val = snd_hda_codec_read(codec, nids[i], 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
val |= AC_PINCTL_VREF_50;
- snd_hda_codec_write(codec, nids[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, nids[i], val);
}
spec->keep_vref_in_automute = 1;
}
@@ -5225,8 +5053,8 @@ static const struct alc_fixup alc882_fixups[] = {
}
},
[ALC882_FIXUP_ACER_ASPIRE_7736] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_sku_ignore,
},
[ALC882_FIXUP_ASUS_W90V] = {
.type = ALC_FIXUP_PINS,
@@ -5405,6 +5233,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+ SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G",
+ ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
@@ -5438,6 +5268,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
+ SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5473,13 +5304,11 @@ static int patch_alc882(struct hda_codec *codec)
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
switch (codec->vendor_id) {
case 0x10ec0882:
@@ -5491,10 +5320,6 @@ static int patch_alc882(struct hda_codec *codec)
break;
}
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
-
alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
alc882_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -5618,13 +5443,11 @@ static int patch_alc262(struct hda_codec *codec)
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
#if 0
/* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is
@@ -5638,13 +5461,13 @@ static int patch_alc262(struct hda_codec *codec)
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
}
#endif
- alc_auto_parse_customize_define(codec);
-
alc_fix_pll_init(codec, 0x20, 0x0a, 10);
alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ alc_auto_parse_customize_define(codec);
+
/* automatic parse from the BIOS config */
err = alc262_parse_auto_config(codec);
if (err < 0)
@@ -5707,7 +5530,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
if (err > 0) {
if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) {
add_mixer(spec, alc268_beep_mixer);
- add_verb(spec, alc268_beep_init_verbs);
+ snd_hda_gen_add_verbs(&spec->gen, alc268_beep_init_verbs);
}
}
return err;
@@ -5720,13 +5543,12 @@ static int patch_alc268(struct hda_codec *codec)
struct alc_spec *spec;
int i, has_beep, err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
/* ALC268 has no aa-loopback mixer */
+ err = alc_alloc_spec(codec, 0);
+ if (err < 0)
+ return err;
+
+ spec = codec->spec;
/* automatic parse from the BIOS config */
err = alc268_parse_auto_config(codec);
@@ -5793,6 +5615,7 @@ enum {
ALC269_TYPE_ALC269VA,
ALC269_TYPE_ALC269VB,
ALC269_TYPE_ALC269VC,
+ ALC269_TYPE_ALC269VD,
};
/*
@@ -5804,8 +5627,21 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
static const hda_nid_t alc269_ssids[] = { 0, 0x1b, 0x14, 0x21 };
static const hda_nid_t alc269va_ssids[] = { 0x15, 0x1b, 0x14, 0 };
struct alc_spec *spec = codec->spec;
- const hda_nid_t *ssids = spec->codec_variant == ALC269_TYPE_ALC269VA ?
- alc269va_ssids : alc269_ssids;
+ const hda_nid_t *ssids;
+
+ switch (spec->codec_variant) {
+ case ALC269_TYPE_ALC269VA:
+ case ALC269_TYPE_ALC269VC:
+ ssids = alc269va_ssids;
+ break;
+ case ALC269_TYPE_ALC269VB:
+ case ALC269_TYPE_ALC269VD:
+ ssids = alc269_ssids;
+ break;
+ default:
+ ssids = alc269_ssids;
+ break;
+ }
return alc_parse_auto_config(codec, alc269_ignore, ssids);
}
@@ -5822,6 +5658,11 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
static void alc269_shutup(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->codec_variant != ALC269_TYPE_ALC269VB)
+ return;
+
if ((alc_get_coef0(codec) & 0x00ff) == 0x017)
alc269_toggle_power_output(codec, 0);
if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
@@ -5833,19 +5674,24 @@ static void alc269_shutup(struct hda_codec *codec)
#ifdef CONFIG_PM
static int alc269_resume(struct hda_codec *codec)
{
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+ (alc_get_coef0(codec) & 0x00ff) == 0x018) {
alc269_toggle_power_output(codec, 0);
msleep(150);
}
codec->patch_ops.init(codec);
- if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
+ if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+ (alc_get_coef0(codec) & 0x00ff) == 0x017) {
alc269_toggle_power_output(codec, 1);
msleep(200);
}
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018)
+ if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+ (alc_get_coef0(codec) & 0x00ff) == 0x018)
alc269_toggle_power_output(codec, 1);
snd_hda_codec_resume_amp(codec);
@@ -5943,9 +5789,7 @@ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled)
{
struct hda_codec *codec = private_data;
unsigned int pinval = enabled ? 0x20 : 0x24;
- snd_hda_codec_update_cache(codec, 0x19, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinval);
+ snd_hda_set_pin_ctl_cache(codec, 0x19, pinval);
}
static void alc269_fixup_mic2_mute(struct hda_codec *codec,
@@ -6012,8 +5856,8 @@ static const struct alc_fixup alc269_fixups[] = {
}
},
[ALC269_FIXUP_SKU_IGNORE] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_sku_ignore,
},
[ALC269_FIXUP_ASUS_G73JW] = {
.type = ALC_FIXUP_PINS,
@@ -6239,21 +6083,13 @@ static void alc269_fill_coef(struct hda_codec *codec)
static int patch_alc269(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err = 0;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
-
- alc_auto_parse_customize_define(codec);
+ int err;
- err = alc_codec_rename_from_preset(codec);
+ err = alc_alloc_spec(codec, 0x0b);
if (err < 0)
- goto error;
+ return err;
+
+ spec = codec->spec;
if (codec->vendor_id == 0x10ec0269) {
spec->codec_variant = ALC269_TYPE_ALC269VA;
@@ -6270,6 +6106,9 @@ static int patch_alc269(struct hda_codec *codec)
err = alc_codec_rename(codec, "ALC3202");
spec->codec_variant = ALC269_TYPE_ALC269VC;
break;
+ case 0x0030:
+ spec->codec_variant = ALC269_TYPE_ALC269VD;
+ break;
default:
alc_fix_pll_init(codec, 0x20, 0x04, 15);
}
@@ -6283,6 +6122,8 @@ static int patch_alc269(struct hda_codec *codec)
alc269_fixup_tbl, alc269_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ alc_auto_parse_customize_define(codec);
+
/* automatic parse from the BIOS config */
err = alc269_parse_auto_config(codec);
if (err < 0)
@@ -6343,8 +6184,7 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec,
if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)))
val |= AC_PINCTL_IN_EN;
val |= AC_PINCTL_VREF_50;
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, 0x0f, val);
spec->keep_vref_in_automute = 1;
}
@@ -6398,13 +6238,11 @@ static int patch_alc861(struct hda_codec *codec)
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x15);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x15;
+ spec = codec->spec;
alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -6501,13 +6339,11 @@ static int patch_alc861vd(struct hda_codec *codec)
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -6519,7 +6355,7 @@ static int patch_alc861vd(struct hda_codec *codec)
if (codec->vendor_id == 0x10ec0660) {
/* always turn on EAPD */
- add_verb(spec, alc660vd_eapd_verbs);
+ snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs);
}
if (!spec->no_analog) {
@@ -6632,8 +6468,8 @@ static const struct alc_fixup alc662_fixups[] = {
}
},
[ALC662_FIXUP_SKU_IGNORE] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_sku_ignore,
},
[ALC662_FIXUP_HP_RP5800] = {
.type = ALC_FIXUP_PINS,
@@ -6846,27 +6682,19 @@ static const struct alc_model_fixup alc662_fixup_models[] = {
static int patch_alc662(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err = 0;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (!spec)
- return -ENOMEM;
+ int err;
- codec->spec = spec;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
/* handle multiple HPs as is */
spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
- alc_auto_parse_customize_define(codec);
-
alc_fix_pll_init(codec, 0x20, 0x04, 15);
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
-
if ((alc_get_coef0(codec) & (1 << 14)) &&
codec->bus->pci->subsystem_vendor == 0x1025 &&
spec->cdefine.platform_type == 1) {
@@ -6877,6 +6705,9 @@ static int patch_alc662(struct hda_codec *codec)
alc_pick_fixup(codec, alc662_fixup_models,
alc662_fixup_tbl, alc662_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
+ alc_auto_parse_customize_define(codec);
+
/* automatic parse from the BIOS config */
err = alc662_parse_auto_config(codec);
if (err < 0)
@@ -6926,16 +6757,12 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
*/
static int patch_alc680(struct hda_codec *codec)
{
- struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
/* ALC680 has no aa-loopback mixer */
+ err = alc_alloc_spec(codec, 0);
+ if (err < 0)
+ return err;
/* automatic parse from the BIOS config */
err = alc680_parse_auto_config(codec);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4742cac26aa..7db8228f1b8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,6 +36,7 @@
#include <sound/tlv.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -221,6 +222,7 @@ struct sigmatel_spec {
unsigned char aloopback_shift;
/* power management */
+ unsigned int power_map_bits;
unsigned int num_pwrs;
const hda_nid_t *pwr_nids;
const hda_nid_t *dac_list;
@@ -314,6 +316,9 @@ struct sigmatel_spec {
struct hda_vmaster_mute_hook vmaster_mute;
};
+#define AC_VERB_IDT_SET_POWER_MAP 0x7ec
+#define AC_VERB_IDT_GET_POWER_MAP 0xfec
+
static const hda_nid_t stac9200_adc_nids[1] = {
0x03,
};
@@ -681,8 +686,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
pinctl &= ~AC_PINCTL_VREFEN;
pinctl |= (new_vref & AC_PINCTL_VREFEN);
- error = snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+ error = snd_hda_set_pin_ctl_cache(codec, nid, pinctl);
if (error < 0)
return error;
@@ -706,8 +710,7 @@ static unsigned int stac92xx_vref_set(struct hda_codec *codec,
else
pincfg |= AC_PINCTL_IN_EN;
- error = snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg);
+ error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg);
if (error < 0)
return error;
else
@@ -2505,27 +2508,10 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
return 0;
}
-static unsigned int stac92xx_get_default_vref(struct hda_codec *codec,
- hda_nid_t nid)
-{
- unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
- pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
- if (pincap & AC_PINCAP_VREF_100)
- return AC_PINCTL_VREF_100;
- if (pincap & AC_PINCAP_VREF_80)
- return AC_PINCTL_VREF_80;
- if (pincap & AC_PINCAP_VREF_50)
- return AC_PINCTL_VREF_50;
- if (pincap & AC_PINCAP_VREF_GRD)
- return AC_PINCTL_VREF_GRD;
- return 0;
-}
-
static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
{
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_set_pin_ctl_cache(codec, nid, pin_type);
}
#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info
@@ -2594,7 +2580,7 @@ static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol,
hda_nid_t nid = kcontrol->private_value;
unsigned int vref = stac92xx_vref_get(codec, nid);
- if (vref == stac92xx_get_default_vref(codec, nid))
+ if (vref == snd_hda_get_default_vref(codec, nid))
ucontrol->value.enumerated.item[0] = 0;
else if (vref == AC_PINCTL_VREF_GRD)
ucontrol->value.enumerated.item[0] = 1;
@@ -2613,7 +2599,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol,
hda_nid_t nid = kcontrol->private_value;
if (ucontrol->value.enumerated.item[0] == 0)
- new_vref = stac92xx_get_default_vref(codec, nid);
+ new_vref = snd_hda_get_default_vref(codec, nid);
else if (ucontrol->value.enumerated.item[0] == 1)
new_vref = AC_PINCTL_VREF_GRD;
else if (ucontrol->value.enumerated.item[0] == 2)
@@ -2679,7 +2665,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
else {
unsigned int pinctl = AC_PINCTL_IN_EN;
if (io_idx) /* set VREF for mic */
- pinctl |= stac92xx_get_default_vref(codec, nid);
+ pinctl |= snd_hda_get_default_vref(codec, nid);
stac92xx_auto_set_pinctl(codec, nid, pinctl);
}
@@ -2847,7 +2833,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec,
char name[22];
if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) {
- if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
+ if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
&& nid == spec->line_switch)
control = STAC_CTL_WIDGET_IO_SWITCH;
else if (snd_hda_query_pin_caps(codec, nid)
@@ -4250,13 +4236,6 @@ static void stac_store_hints(struct hda_codec *codec)
val = snd_hda_get_bool_hint(codec, "eapd_switch");
if (val >= 0)
spec->eapd_switch = val;
- get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
- if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
- spec->gpio_mask |= spec->gpio_led;
- spec->gpio_dir |= spec->gpio_led;
- if (spec->gpio_led_polarity)
- spec->gpio_data |= spec->gpio_led;
- }
}
static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins,
@@ -4354,7 +4333,7 @@ static int stac92xx_init(struct hda_codec *codec)
unsigned int pinctl, conf;
if (type == AUTO_PIN_MIC) {
/* for mic pins, force to initialize */
- pinctl = stac92xx_get_default_vref(codec, nid);
+ pinctl = snd_hda_get_default_vref(codec, nid);
pinctl |= AC_PINCTL_IN_EN;
stac92xx_auto_set_pinctl(codec, nid, pinctl);
} else {
@@ -4390,10 +4369,18 @@ static int stac92xx_init(struct hda_codec *codec)
hda_nid_t nid = spec->pwr_nids[i];
int pinctl, def_conf;
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ def_conf = get_defcfg_connect(def_conf);
+ if (def_conf == AC_JACK_PORT_NONE) {
+ /* power off unused ports */
+ stac_toggle_power_map(codec, nid, 0);
+ continue;
+ }
/* power on when no jack detection is available */
/* or when the VREF is used for controlling LED */
if (!spec->hp_detect ||
- spec->vref_mute_led_nid == nid) {
+ spec->vref_mute_led_nid == nid ||
+ !is_jack_detectable(codec, nid)) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -4411,15 +4398,6 @@ static int stac92xx_init(struct hda_codec *codec)
stac_toggle_power_map(codec, nid, 1);
continue;
}
- def_conf = snd_hda_codec_get_pincfg(codec, nid);
- def_conf = get_defcfg_connect(def_conf);
- /* skip any ports that don't have jacks since presence
- * detection is useless */
- if (def_conf != AC_JACK_PORT_COMPLEX) {
- if (def_conf != AC_JACK_PORT_NONE)
- stac_toggle_power_map(codec, nid, 1);
- continue;
- }
if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
stac_issue_unsol_event(codec, nid);
continue;
@@ -4432,6 +4410,12 @@ static int stac92xx_init(struct hda_codec *codec)
/* sync mute LED */
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+
+ /* sync the power-map */
+ if (spec->num_pwrs)
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_IDT_SET_POWER_MAP,
+ spec->power_map_bits);
if (spec->dac_list)
stac92xx_power_down(codec);
return 0;
@@ -4460,8 +4444,7 @@ static void stac92xx_shutup_pins(struct hda_codec *codec)
struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
def_conf = snd_hda_codec_get_pincfg(codec, pin->nid);
if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)
- snd_hda_codec_write(codec, pin->nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_set_pin_ctl(codec, pin->nid, 0);
}
}
@@ -4517,9 +4500,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
pin_ctl |= flag;
if (old_ctl != pin_ctl)
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_ctl);
+ snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl);
}
static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -4528,9 +4509,7 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
if (pin_ctl & flag)
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_ctl & ~flag);
+ snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag);
}
static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
@@ -4682,14 +4661,18 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
idx = 1 << idx;
- val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff;
+ val = spec->power_map_bits;
if (enable)
val &= ~idx;
else
val |= idx;
/* power down unused output ports */
- snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val);
+ if (val != spec->power_map_bits) {
+ spec->power_map_bits = val;
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_IDT_SET_POWER_MAP, val);
+ }
}
static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid)
@@ -4866,6 +4849,11 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
struct sigmatel_spec *spec = codec->spec;
const struct dmi_device *dev = NULL;
+ if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
+ get_int_hint(codec, "gpio_led_polarity",
+ &spec->gpio_led_polarity);
+ return 1;
+ }
if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
NULL, dev))) {
@@ -4952,7 +4940,8 @@ static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
{
if (nid == codec->afg)
snd_iprintf(buffer, "Power-Map: 0x%02x\n",
- snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0));
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_IDT_GET_POWER_MAP, 0));
}
static void analog_loop_proc_hook(struct snd_info_buffer *buffer,
@@ -5009,20 +4998,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
return 0;
}
-static int stac92xx_pre_resume(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
-
- /* sync mute LED */
- if (spec->vref_mute_led_nid)
- stac_vrefout_set(codec, spec->vref_mute_led_nid,
- spec->vref_led);
- else if (spec->gpio_led)
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data);
- return 0;
-}
-
static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
@@ -5046,7 +5021,6 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
#else
#define stac92xx_suspend NULL
#define stac92xx_resume NULL
-#define stac92xx_pre_resume NULL
#define stac92xx_set_power_state NULL
#endif /* CONFIG_PM */
@@ -5592,9 +5566,6 @@ again:
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
-#ifdef CONFIG_PM
- codec->patch_ops.pre_resume = stac92xx_pre_resume;
-#endif
}
err = stac92xx_parse_auto_config(codec);
@@ -5901,9 +5872,6 @@ again:
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
-#ifdef CONFIG_PM
- codec->patch_ops.pre_resume = stac92xx_pre_resume;
-#endif
}
spec->multiout.dac_nids = spec->dac_nids;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 06214fdc948..82b368068e0 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -54,6 +54,7 @@
#include <sound/asoundef.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_jack.h"
/* Pin Widget NID */
@@ -484,7 +485,7 @@ static void activate_output_mix(struct hda_codec *codec, struct nid_path *path,
if (!path)
return;
- num = snd_hda_get_conn_list(codec, mix_nid, NULL);
+ num = snd_hda_get_num_conns(codec, mix_nid);
for (i = 0; i < num; i++) {
if (i == idx)
val = AMP_IN_UNMUTE(i);
@@ -532,8 +533,7 @@ static void init_output_pin(struct hda_codec *codec, hda_nid_t pin,
{
if (!pin)
return;
- snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
+ snd_hda_set_pin_ctl(codec, pin, pin_type);
if (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
@@ -662,12 +662,12 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
hda_nid_t nid = cfg->inputs[i].pin;
if (spec->smart51_enabled && is_smart51_pins(codec, nid))
ctl = PIN_OUT;
- else if (cfg->inputs[i].type == AUTO_PIN_MIC)
- ctl = PIN_VREF50;
- else
+ else {
ctl = PIN_IN;
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+ if (cfg->inputs[i].type == AUTO_PIN_MIC)
+ ctl |= snd_hda_get_default_vref(codec, nid);
+ }
+ snd_hda_set_pin_ctl(codec, nid, ctl);
}
/* init input-src */
@@ -1006,9 +1006,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
parm |= out_in;
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- parm);
+ snd_hda_set_pin_ctl(codec, nid, parm);
if (out_in == AC_PINCTL_OUT_EN) {
mute_aa_path(codec, 1);
notify_aa_path_ctls(codec);
@@ -1647,8 +1645,7 @@ static void toggle_output_mutes(struct hda_codec *codec, int num_pins,
parm &= ~AC_PINCTL_OUT_EN;
else
parm |= AC_PINCTL_OUT_EN;
- snd_hda_codec_write(codec, pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, parm);
+ snd_hda_set_pin_ctl(codec, pins[i], parm);
}
}
@@ -1709,8 +1706,7 @@ static void via_gpio_control(struct hda_codec *codec)
if (gpio_data == 0x02) {
/* unmute line out */
- snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0],
PIN_OUT);
if (vol_counter & 0x20) {
/* decrease volume */
@@ -1728,9 +1724,7 @@ static void via_gpio_control(struct hda_codec *codec)
}
} else if (!(gpio_data & 0x02)) {
/* mute line out */
- snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- 0);
+ snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0);
}
}
@@ -2757,8 +2751,7 @@ static void via_auto_init_dig_in(struct hda_codec *codec)
struct via_spec *spec = codec->spec;
if (!spec->dig_in_nid)
return;
- snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+ snd_hda_set_pin_ctl(codec, spec->autocfg.dig_in_pin, PIN_IN);
}
/* initialize the unsolicited events */
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 132a86e09d0..5be2e120a14 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2803,22 +2803,11 @@ static void __devexit snd_ice1712_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ice1712_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ice1712_ids,
.probe = snd_ice1712_probe,
.remove = __devexit_p(snd_ice1712_remove),
};
-static int __init alsa_card_ice1712_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ice1712_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ice1712_init)
-module_exit(alsa_card_ice1712_exit)
+module_pci_driver(ice1712_driver);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 812d10e43ae..a01a00d1cf4 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2873,7 +2873,7 @@ static int snd_vt1724_resume(struct pci_dev *pci)
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver vt1724_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_vt1724_ids,
.probe = snd_vt1724_probe,
@@ -2884,15 +2884,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_ice1724_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ice1724_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ice1724_init)
-module_exit(alsa_card_ice1724_exit)
+module_pci_driver(vt1724_driver);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index e0a4263baa2..f4e2dd4da8c 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3338,7 +3338,7 @@ static void __devexit snd_intel8x0_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver intel8x0_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_intel8x0_ids,
.probe = snd_intel8x0_probe,
@@ -3349,16 +3349,4 @@ static struct pci_driver driver = {
#endif
};
-
-static int __init alsa_card_intel8x0_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_intel8x0_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_intel8x0_init)
-module_exit(alsa_card_intel8x0_exit)
+module_pci_driver(intel8x0_driver);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index d689913a61b..fc27a6a69e7 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1324,7 +1324,7 @@ static void __devexit snd_intel8x0m_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver intel8x0m_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_intel8x0m_ids,
.probe = snd_intel8x0m_probe,
@@ -1335,16 +1335,4 @@ static struct pci_driver driver = {
#endif
};
-
-static int __init alsa_card_intel8x0m_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_intel8x0m_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_intel8x0m_init)
-module_exit(alsa_card_intel8x0m_exit)
+module_pci_driver(intel8x0m_driver);
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8fea45ab588..e69ce5f9c31 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2476,22 +2476,11 @@ static void __devexit snd_korg1212_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver korg1212_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_korg1212_ids,
.probe = snd_korg1212_probe,
.remove = __devexit_p(snd_korg1212_remove),
};
-static int __init alsa_card_korg1212_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_korg1212_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_korg1212_init)
-module_exit(alsa_card_korg1212_exit)
+module_pci_driver(korg1212_driver);
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 37598273685..ac15166bee6 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -770,22 +770,11 @@ static DEFINE_PCI_DEVICE_TABLE(lola_ids) = {
MODULE_DEVICE_TABLE(pci, lola_ids);
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver lola_driver = {
.name = KBUILD_MODNAME,
.id_table = lola_ids,
.probe = lola_probe,
.remove = __devexit_p(lola_remove),
};
-static int __init alsa_card_lola_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_lola_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_lola_init)
-module_exit(alsa_card_lola_exit)
+module_pci_driver(lola_driver);
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d94c0c292bd..d1ab4370673 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -1141,24 +1141,11 @@ static void __devexit snd_lx6464es_remove(struct pci_dev *pci)
}
-static struct pci_driver driver = {
+static struct pci_driver lx6464es_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_lx6464es_ids,
.probe = snd_lx6464es_probe,
.remove = __devexit_p(snd_lx6464es_remove),
};
-
-/* module initialization */
-static int __init mod_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit mod_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(mod_init);
-module_exit(mod_exit);
+module_pci_driver(lx6464es_driver);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 78229b0dad2..deef2139958 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2837,7 +2837,7 @@ static void __devexit snd_m3_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver m3_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_m3_ids,
.probe = snd_m3_probe,
@@ -2848,15 +2848,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_m3_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_m3_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_m3_init)
-module_exit(alsa_card_m3_exit)
+module_pci_driver(m3_driver);
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 487837c01c9..0762610c99c 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1380,22 +1380,11 @@ static void __devexit snd_mixart_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver mixart_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_mixart_ids,
.probe = snd_mixart_probe,
.remove = __devexit_p(snd_mixart_remove),
};
-static int __init alsa_card_mixart_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_mixart_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_mixart_init)
-module_exit(alsa_card_mixart_exit)
+module_pci_driver(mixart_driver);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index ade2c64bd60..8159b05ee94 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1742,7 +1742,7 @@ static void __devexit snd_nm256_remove(struct pci_dev *pci)
}
-static struct pci_driver driver = {
+static struct pci_driver nm256_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_nm256_ids,
.probe = snd_nm256_probe,
@@ -1753,16 +1753,4 @@ static struct pci_driver driver = {
#endif
};
-
-static int __init alsa_card_nm256_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_nm256_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_nm256_init)
-module_exit(alsa_card_nm256_exit)
+module_pci_driver(nm256_driver);
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index eab663eef11..610275bfbae 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -94,6 +94,7 @@ enum {
MODEL_2CH_OUTPUT,
MODEL_HG2PCI,
MODEL_XONAR_DG,
+ MODEL_XONAR_DGX,
};
static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = {
@@ -109,6 +110,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = {
{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
/* Asus Xonar DG */
{ OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG },
+ /* Asus Xonar DGX */
+ { OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX },
/* PCI 2.0 HD Audio */
{ OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT },
/* Kuroutoshikou CMI8787-HG2PCI */
@@ -827,6 +830,11 @@ static int __devinit get_oxygen_model(struct oxygen *chip,
break;
case MODEL_XONAR_DG:
chip->model = model_xonar_dg;
+ chip->model.shortname = "Xonar DG";
+ break;
+ case MODEL_XONAR_DGX:
+ chip->model = model_xonar_dg;
+ chip->model.shortname = "Xonar DGX";
break;
}
if (id->driver_data == MODEL_MERIDIAN ||
@@ -870,15 +878,4 @@ static struct pci_driver oxygen_driver = {
#endif
};
-static int __init alsa_card_oxygen_init(void)
-{
- return pci_register_driver(&oxygen_driver);
-}
-
-static void __exit alsa_card_oxygen_exit(void)
-{
- pci_unregister_driver(&oxygen_driver);
-}
-
-module_init(alsa_card_oxygen_init)
-module_exit(alsa_card_oxygen_exit)
+module_pci_driver(oxygen_driver);
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 3fdee495017..19962c6d38c 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -100,15 +100,4 @@ static struct pci_driver xonar_driver = {
.shutdown = oxygen_pci_shutdown,
};
-static int __init alsa_card_xonar_init(void)
-{
- return pci_register_driver(&xonar_driver);
-}
-
-static void __exit alsa_card_xonar_exit(void)
-{
- pci_unregister_driver(&xonar_driver);
-}
-
-module_init(alsa_card_xonar_init)
-module_exit(alsa_card_xonar_exit)
+module_pci_driver(xonar_driver);
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index 793bdf03d7e..77acd790ea4 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -1,5 +1,5 @@
/*
- * card driver for the Xonar DG
+ * card driver for the Xonar DG/DGX
*
* Copyright (c) Clemens Ladisch <clemens@ladisch.de>
*
@@ -17,8 +17,8 @@
*/
/*
- * Xonar DG
- * --------
+ * Xonar DG/DGX
+ * ------------
*
* CMI8788:
*
@@ -581,7 +581,6 @@ static void dump_cs4245_registers(struct oxygen *chip,
}
struct oxygen_model model_xonar_dg = {
- .shortname = "Xonar DG",
.longname = "C-Media Oxygen HD Audio",
.chip = "CMI8786",
.init = dg_init,
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index fd1809ab73b..0435f45e951 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1607,22 +1607,11 @@ static void __devexit pcxhr_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver pcxhr_driver = {
.name = KBUILD_MODNAME,
.id_table = pcxhr_ids,
.probe = pcxhr_probe,
.remove = __devexit_p(pcxhr_remove),
};
-static int __init pcxhr_module_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit pcxhr_module_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(pcxhr_module_init)
-module_exit(pcxhr_module_exit)
+module_pci_driver(pcxhr_driver);
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 0481d94aac9..cbeb3f77350 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1837,8 +1837,7 @@ static int snd_riptide_free(struct snd_riptide *chip)
}
if (chip->irq >= 0)
free_irq(chip->irq, chip);
- if (chip->fw_entry)
- release_firmware(chip->fw_entry);
+ release_firmware(chip->fw_entry);
release_and_free_resource(chip->res_port);
kfree(chip);
return 0;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index b4819d5e41d..46b3629dda2 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1984,22 +1984,11 @@ static void __devexit snd_rme32_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver rme32_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_rme32_ids,
.probe = snd_rme32_probe,
.remove = __devexit_p(snd_rme32_remove),
};
-static int __init alsa_card_rme32_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_rme32_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_rme32_init)
-module_exit(alsa_card_rme32_exit)
+module_pci_driver(rme32_driver);
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index ba894158e76..9b98dc40698 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -2395,22 +2395,11 @@ static void __devexit snd_rme96_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver rme96_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_rme96_ids,
.probe = snd_rme96_probe,
.remove = __devexit_p(snd_rme96_remove),
};
-static int __init alsa_card_rme96_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_rme96_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_rme96_init)
-module_exit(alsa_card_rme96_exit)
+module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b68cdec03b9..0d6930c4f4b 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
strcpy(hw->name, "HDSP hwdep interface");
hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
+ hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl;
return 0;
}
@@ -5635,22 +5636,11 @@ static void __devexit snd_hdsp_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver hdsp_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_hdsp_ids,
.probe = snd_hdsp_probe,
.remove = __devexit_p(snd_hdsp_remove),
};
-static int __init alsa_card_hdsp_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hdsp_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hdsp_init)
-module_exit(alsa_card_hdsp_exit)
+module_pci_driver(hdsp_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bc030a2088d..0a5027b9471 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6918,23 +6918,11 @@ static void __devexit snd_hdspm_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver hdspm_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_hdspm_ids,
.probe = snd_hdspm_probe,
.remove = __devexit_p(snd_hdspm_remove),
};
-
-static int __init alsa_card_hdspm_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hdspm_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hdspm_init)
-module_exit(alsa_card_hdspm_exit)
+module_pci_driver(hdspm_driver);
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index b737d1619cc..a15fc100ab0 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -2631,22 +2631,11 @@ static void __devexit snd_rme9652_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver rme9652_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_rme9652_ids,
.probe = snd_rme9652_probe,
.remove = __devexit_p(snd_rme9652_remove),
};
-static int __init alsa_card_hammerfall_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hammerfall_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hammerfall_init)
-module_exit(alsa_card_hammerfall_exit)
+module_pci_driver(rme9652_driver);
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index ff500a87f76..1552642765d 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1488,15 +1488,4 @@ static struct pci_driver sis7019_driver = {
#endif
};
-static int __init sis7019_init(void)
-{
- return pci_register_driver(&sis7019_driver);
-}
-
-static void __exit sis7019_exit(void)
-{
- pci_unregister_driver(&sis7019_driver);
-}
-
-module_init(sis7019_init);
-module_exit(sis7019_exit);
+module_pci_driver(sis7019_driver);
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 54cc802050f..baa9946bedf 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1530,22 +1530,11 @@ static void __devexit snd_sonic_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver sonicvibes_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_sonic_ids,
.probe = snd_sonic_probe,
.remove = __devexit_p(snd_sonic_remove),
};
-static int __init alsa_card_sonicvibes_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_sonicvibes_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_sonicvibes_init)
-module_exit(alsa_card_sonicvibes_exit)
+module_pci_driver(sonicvibes_driver);
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 5f1def7f45e..611983ec732 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -172,7 +172,7 @@ static void __devexit snd_trident_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver trident_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_trident_ids,
.probe = snd_trident_probe,
@@ -183,15 +183,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_trident_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_trident_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_trident_init)
-module_exit(alsa_card_trident_exit)
+module_pci_driver(trident_driver);
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 75630408c6d..b5afab48943 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2619,7 +2619,7 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver via82xx_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_via82xx_ids,
.probe = snd_via82xx_probe,
@@ -2630,15 +2630,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_via82xx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_via82xx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_via82xx_init)
-module_exit(alsa_card_via82xx_exit)
+module_pci_driver(via82xx_driver);
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5efcbcac506..59fd47ed0a3 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1223,7 +1223,7 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver via82xx_modem_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_via82xx_modem_ids,
.probe = snd_via82xx_probe,
@@ -1234,15 +1234,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_via82xx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_via82xx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_via82xx_init)
-module_exit(alsa_card_via82xx_exit)
+module_pci_driver(via82xx_modem_driver);
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 6a534bfe127..1ea1f656a5d 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -289,7 +289,7 @@ static int snd_vx222_resume(struct pci_dev *pci)
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver vx222_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_vx222_ids,
.probe = snd_vx222_probe,
@@ -300,15 +300,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_vx222_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_vx222_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_vx222_init)
-module_exit(alsa_card_vx222_exit)
+module_pci_driver(vx222_driver);
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 94ab728f5ca..9a1d01d653a 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -350,7 +350,7 @@ static void __devexit snd_card_ymfpci_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ymfpci_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ymfpci_ids,
.probe = snd_card_ymfpci_probe,
@@ -361,15 +361,4 @@ static struct pci_driver driver = {
#endif
};
-static int __init alsa_card_ymfpci_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ymfpci_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ymfpci_init)
-module_exit(alsa_card_ymfpci_exit)
+module_pci_driver(ymfpci_driver);
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index b11f82b5718..f8b01c77b29 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -433,7 +433,7 @@ probe_error:
/*
* "driver" definition
*/
-static struct platform_driver driver = {
+static struct platform_driver sh_dac_driver = {
.probe = snd_sh_dac_probe,
.remove = snd_sh_dac_remove,
.driver = {
@@ -441,4 +441,4 @@ static struct platform_driver driver = {
},
};
-module_platform_driver(driver);
+module_platform_driver(sh_dac_driver);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 91c985599d3..40b2ad1bb1c 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -35,7 +35,6 @@ source "sound/soc/blackfin/Kconfig"
source "sound/soc/davinci/Kconfig"
source "sound/soc/ep93xx/Kconfig"
source "sound/soc/fsl/Kconfig"
-source "sound/soc/imx/Kconfig"
source "sound/soc/jz4740/Kconfig"
source "sound/soc/nuc900/Kconfig"
source "sound/soc/omap/Kconfig"
@@ -48,9 +47,13 @@ source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
+source "sound/soc/ux500/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
+# generic frame-work
+source "sound/soc/generic/Kconfig"
+
endif # SND_SOC
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 2feaf376e94..70990f4017f 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -6,13 +6,13 @@ obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
+obj-$(CONFIG_SND_SOC) += generic/
obj-$(CONFIG_SND_SOC) += atmel/
obj-$(CONFIG_SND_SOC) += au1x/
obj-$(CONFIG_SND_SOC) += blackfin/
obj-$(CONFIG_SND_SOC) += davinci/
obj-$(CONFIG_SND_SOC) += ep93xx/
obj-$(CONFIG_SND_SOC) += fsl/
-obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += mid-x86/
obj-$(CONFIG_SND_SOC) += mxs/
@@ -25,3 +25,4 @@ obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
+obj-$(CONFIG_SND_SOC) += ux500/
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index b39ad356b92..7dbeef1099b 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,16 +44,8 @@
static struct snd_soc_card bf5xx_ssm2602;
-static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- unsigned int clk = 0;
- int ret = 0;
-
- pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
- params_format(params));
/*
* If you are using a crystal source which frequency is not 12MHz
* then modify the below case statement with frequency of the crystal.
@@ -61,31 +53,10 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
* If you are using the SPORT to generate clocking then this is
* where to do it.
*/
-
- switch (params_rate(params)) {
- case 8000:
- case 16000:
- case 48000:
- case 96000:
- case 11025:
- case 22050:
- case 44100:
- clk = 12000000;
- break;
- }
-
- ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+ return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000,
SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
}
-static struct snd_soc_ops bf5xx_ssm2602_ops = {
- .hw_params = bf5xx_ssm2602_hw_params,
-};
-
/* CODEC is master for BCLK and LRC in this configuration. */
#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
SND_SOC_DAIFMT_CBM_CFM)
@@ -98,7 +69,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "ssm2602.0-001b",
- .ops = &bf5xx_ssm2602_ops,
+ .init = bf5xx_ssm2602_dai_init,
.dai_fmt = BF5XX_SSM2602_DAIFMT,
},
{
@@ -108,7 +79,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "ssm2602.0-001b",
- .ops = &bf5xx_ssm2602_ops,
+ .init = bf5xx_ssm2602_dai_init,
.dai_fmt = BF5XX_SSM2602_DAIFMT,
},
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 59d8efaa17e..1e1613a438d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_ALC5632 if I2C
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS42L51 if I2C
+ select SND_SOC_CS42L52 if I2C
select SND_SOC_CS42L73 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
@@ -37,11 +38,15 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DFBMCS320
select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
+ select SND_SOC_LM49453 if I2C
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
+ select SND_SOC_MC13783 if MFD_MC13XXX
+ select SND_SOC_ML26124 if I2C
+ select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI
select SND_SOC_PCM3008
select SND_SOC_RT5631 if I2C
select SND_SOC_SGTL5000 if I2C
@@ -181,6 +186,9 @@ config SND_SOC_CQ0093VC
config SND_SOC_CS42L51
tristate
+config SND_SOC_CS42L52
+ tristate
+
config SND_SOC_CS42L73
tristate
@@ -217,6 +225,9 @@ config SND_SOC_DFBMCS320
config SND_SOC_DMIC
tristate
+config SND_SOC_LM49453
+ tristate
+
config SND_SOC_MAX98088
tristate
@@ -226,6 +237,9 @@ config SND_SOC_MAX98095
config SND_SOC_MAX9850
tristate
+config SND_SOC_OMAP_HDMI_CODEC
+ tristate
+
config SND_SOC_PCM3008
tristate
@@ -435,5 +449,11 @@ config SND_SOC_MAX9768
config SND_SOC_MAX9877
tristate
+config SND_SOC_MC13783
+ tristate
+
+config SND_SOC_ML26124
+ tristate
+
config SND_SOC_TPA6130A2
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6662eb0cdcc..fc27fec3948 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs42l51-objs := cs42l51.o
+snd-soc-cs42l52-objs := cs42l52.o
snd-soc-cs42l73-objs := cs42l73.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
@@ -25,10 +26,14 @@ snd-soc-dmic-objs := dmic.o
snd-soc-jz4740-codec-objs := jz4740.o
snd-soc-l3-objs := l3.o
snd-soc-lm4857-objs := lm4857.o
+snd-soc-lm49453-objs := lm49453.o
snd-soc-max9768-objs := max9768.o
snd-soc-max98088-objs := max98088.o
snd-soc-max98095-objs := max98095.o
snd-soc-max9850-objs := max9850.o
+snd-soc-mc13783-objs := mc13783.o
+snd-soc-ml26124-objs := ml26124.o
+snd-soc-omap-hdmi-codec-objs := omap-hdmi.o
snd-soc-pcm3008-objs := pcm3008.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-sgtl5000-objs := sgtl5000.o
@@ -121,6 +126,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
+obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o
obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
@@ -128,13 +134,17 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o
-obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
+obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o
obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
+obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
+obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
+obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 1bbad4c16d2..2023c749f23 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -26,13 +26,11 @@
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
- return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
+ return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate);
}
#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 12e3b411855..c67b50d8b31 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int word_len = 0;
-
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
/* bit size */
switch (params_format(params)) {
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index a4a6bef2c0b..13e62be4f99 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int word_len = 0, master_rate = 0;
-
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
/* bit size */
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 78e9ce48bb9..3d50fc8646b 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
static int adau1701_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
snd_pcm_format_t format;
unsigned int val;
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index ceb96ecf558..31d4483245d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int val = 0;
/* set the IEC958 bits: consumer mode, no copyright bit */
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 838ae8b22b5..618fdc30f73 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5);
int rate = params_rate(params), fs = 256;
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index c4d165a4bdd..543a12f471b 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int rate = params_rate(params), fs = 256;
u8 mode2;
@@ -517,67 +516,24 @@ static int ak4641_resume(struct snd_soc_codec *codec)
static int ak4641_probe(struct snd_soc_codec *codec)
{
- struct ak4641_platform_data *pdata = codec->dev->platform_data;
int ret;
-
- if (pdata) {
- if (gpio_is_valid(pdata->gpio_power)) {
- ret = gpio_request_one(pdata->gpio_power,
- GPIOF_OUT_INIT_LOW, "ak4641 power");
- if (ret)
- goto err_out;
- }
- if (gpio_is_valid(pdata->gpio_npdn)) {
- ret = gpio_request_one(pdata->gpio_npdn,
- GPIOF_OUT_INIT_LOW, "ak4641 npdn");
- if (ret)
- goto err_gpio;
-
- udelay(1); /* > 150 ns */
- gpio_set_value(pdata->gpio_npdn, 1);
- }
- }
-
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- goto err_register;
+ return ret;
}
/* power on device */
ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
-
-err_register:
- if (pdata) {
- if (gpio_is_valid(pdata->gpio_power))
- gpio_set_value(pdata->gpio_power, 0);
- if (gpio_is_valid(pdata->gpio_npdn))
- gpio_free(pdata->gpio_npdn);
- }
-err_gpio:
- if (pdata && gpio_is_valid(pdata->gpio_power))
- gpio_free(pdata->gpio_power);
-err_out:
- return ret;
}
static int ak4641_remove(struct snd_soc_codec *codec)
{
- struct ak4641_platform_data *pdata = codec->dev->platform_data;
-
ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
- if (pdata) {
- if (gpio_is_valid(pdata->gpio_power)) {
- gpio_set_value(pdata->gpio_power, 0);
- gpio_free(pdata->gpio_power);
- }
- if (gpio_is_valid(pdata->gpio_npdn))
- gpio_free(pdata->gpio_npdn);
- }
return 0;
}
@@ -604,6 +560,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
+ struct ak4641_platform_data *pdata = i2c->dev.platform_data;
struct ak4641_priv *ak4641;
int ret;
@@ -612,16 +569,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
if (!ak4641)
return -ENOMEM;
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power)) {
+ ret = gpio_request_one(pdata->gpio_power,
+ GPIOF_OUT_INIT_LOW, "ak4641 power");
+ if (ret)
+ goto err_out;
+ }
+ if (gpio_is_valid(pdata->gpio_npdn)) {
+ ret = gpio_request_one(pdata->gpio_npdn,
+ GPIOF_OUT_INIT_LOW, "ak4641 npdn");
+ if (ret)
+ goto err_gpio;
+
+ udelay(1); /* > 150 ns */
+ gpio_set_value(pdata->gpio_npdn, 1);
+ }
+ }
+
i2c_set_clientdata(i2c, ak4641);
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
ak4641_dai, ARRAY_SIZE(ak4641_dai));
+ if (ret != 0)
+ goto err_gpio2;
+
+ return 0;
+
+err_gpio2:
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power))
+ gpio_set_value(pdata->gpio_power, 0);
+ if (gpio_is_valid(pdata->gpio_npdn))
+ gpio_free(pdata->gpio_npdn);
+ }
+err_gpio:
+ if (pdata && gpio_is_valid(pdata->gpio_power))
+ gpio_free(pdata->gpio_power);
+err_out:
return ret;
}
static int __devexit ak4641_i2c_remove(struct i2c_client *i2c)
{
+ struct ak4641_platform_data *pdata = i2c->dev.platform_data;
+
snd_soc_unregister_codec(&i2c->dev);
+
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power)) {
+ gpio_set_value(pdata->gpio_power, 0);
+ gpio_free(pdata->gpio_power);
+ }
+ if (gpio_is_valid(pdata->gpio_npdn))
+ gpio_free(pdata->gpio_npdn);
+ }
+
return 0;
}
@@ -641,23 +644,7 @@ static struct i2c_driver ak4641_i2c_driver = {
.id_table = ak4641_i2c_id,
};
-static int __init ak4641_modinit(void)
-{
- int ret;
-
- ret = i2c_add_driver(&ak4641_i2c_driver);
- if (ret != 0)
- pr_err("Failed to register AK4641 I2C driver: %d\n", ret);
-
- return ret;
-}
-module_init(ak4641_modinit);
-
-static void __exit ak4641_exit(void)
-{
- i2c_del_driver(&ak4641_i2c_driver);
-}
-module_exit(ak4641_exit);
+module_i2c_driver(ak4641_i2c_driver);
MODULE_DESCRIPTION("SoC AK4641 driver");
MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index d47b62ddb21..1960478ce6b 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
int coeff, rate;
u16 iface;
@@ -1084,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = {
.id_table = alc5623_i2c_table,
};
-static int __init alc5623_modinit(void)
-{
- int ret;
-
- ret = i2c_add_driver(&alc5623_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "%s: can't add i2c driver", __func__);
- return ret;
- }
-
- return ret;
-}
-module_init(alc5623_modinit);
-
-static void __exit alc5623_modexit(void)
-{
- i2c_del_driver(&alc5623_i2c_driver);
-}
-module_exit(alc5623_modexit);
+module_i2c_driver(alc5623_i2c_driver);
MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index e2111e0ccad..7dd02420b36 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai,
static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
int coeff, rate;
u16 iface;
@@ -1131,7 +1130,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
i2c_set_clientdata(client, alc5632);
- alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap);
+ alc5632->regmap = devm_regmap_init_i2c(client, &alc5632_regmap);
if (IS_ERR(alc5632->regmap)) {
ret = PTR_ERR(alc5632->regmap);
dev_err(&client->dev, "regmap_init() failed: %d\n", ret);
@@ -1143,7 +1142,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
if (ret1 != 0 || ret2 != 0) {
dev_err(&client->dev,
"Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2);
- regmap_exit(alc5632->regmap);
return -EIO;
}
@@ -1152,14 +1150,12 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) {
dev_err(&client->dev,
"Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2);
- regmap_exit(alc5632->regmap);
return -EINVAL;
}
ret = alc5632_reset(alc5632->regmap);
if (ret < 0) {
dev_err(&client->dev, "Failed to issue reset\n");
- regmap_exit(alc5632->regmap);
return ret;
}
@@ -1177,7 +1173,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
if (ret < 0) {
dev_err(&client->dev, "Failed to register codec: %d\n", ret);
- regmap_exit(alc5632->regmap);
return ret;
}
@@ -1186,9 +1181,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
static __devexit int alc5632_i2c_remove(struct i2c_client *client)
{
- struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
snd_soc_unregister_codec(&client->dev);
- regmap_exit(alc5632->regmap);
return 0;
}
@@ -1209,25 +1202,7 @@ static struct i2c_driver alc5632_i2c_driver = {
.id_table = alc5632_i2c_table,
};
-static int __init alc5632_modinit(void)
-{
- int ret;
-
- ret = i2c_add_driver(&alc5632_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "%s: can't add i2c driver", __func__);
- return ret;
- }
-
- return ret;
-}
-module_init(alc5632_modinit);
-
-static void __exit alc5632_modexit(void)
-{
- i2c_del_driver(&alc5632_i2c_driver);
-}
-module_exit(alc5632_modexit);
+module_i2c_driver(alc5632_i2c_driver);
MODULE_DESCRIPTION("ASoC ALC5632 driver");
MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 1d672f52866..047917f0b8a 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
int ret;
unsigned int i;
@@ -600,10 +599,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec)
static int cs4270_soc_resume(struct snd_soc_codec *codec)
{
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
- int reg;
+ int reg, ret;
- regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
- cs4270->supplies);
+ ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
+ cs4270->supplies);
+ if (ret != 0)
+ return ret;
/* In case the device was put to hard reset during sleep, we need to
* wait 500ns here before any I2C communication. */
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index bf7141280a7..9eb01d7d58a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
int i, ret;
unsigned int ratio, val;
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index a8bf588e874..091d0193f50 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix =
static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
- 7, 0xffffff99, 0x18, adc_pcm_tlv),
+ 6, 0x19, 0x7F, adc_pcm_tlv),
SOC_DOUBLE_R("PCM Playback Switch",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
- 8, 0xffffff19, 0x18, aout_tlv),
+ 0, 0x34, 0xE4, aout_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
- 7, 0xffffff99, 0x18, adc_pcm_tlv),
+ 6, 0x19, 0x7F, adc_pcm_tlv),
SOC_DOUBLE_R("ADC Mixer Switch",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
@@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
int ret;
unsigned int i;
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
new file mode 100644
index 00000000000..a7109413aef
--- /dev/null
+++ b/sound/soc/codecs/cs42l52.c
@@ -0,0 +1,1295 @@
+/*
+ * cs42l52.c -- CS42L52 ALSA SoC audio driver
+ *
+ * Copyright 2012 CirrusLogic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ * Author: Brian Austin <brian.austin@cirrus.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/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/cs42l52.h>
+#include "cs42l52.h"
+
+struct sp_config {
+ u8 spc, format, spfs;
+ u32 srate;
+};
+
+struct cs42l52_private {
+ struct regmap *regmap;
+ struct snd_soc_codec *codec;
+ struct device *dev;
+ struct sp_config config;
+ struct cs42l52_platform_data pdata;
+ u32 sysclk;
+ u8 mclksel;
+ u32 mclk;
+ u8 flags;
+#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+ struct input_dev *beep;
+ struct work_struct beep_work;
+ int beep_rate;
+#endif
+};
+
+static const struct reg_default cs42l52_reg_defaults[] = {
+ { CS42L52_PWRCTL1, 0x9F }, /* r02 PWRCTL 1 */
+ { CS42L52_PWRCTL2, 0x07 }, /* r03 PWRCTL 2 */
+ { CS42L52_PWRCTL3, 0xFF }, /* r04 PWRCTL 3 */
+ { CS42L52_CLK_CTL, 0xA0 }, /* r05 Clocking Ctl */
+ { CS42L52_IFACE_CTL1, 0x00 }, /* r06 Interface Ctl 1 */
+ { CS42L52_ADC_PGA_A, 0x80 }, /* r08 Input A Select */
+ { CS42L52_ADC_PGA_B, 0x80 }, /* r09 Input B Select */
+ { CS42L52_ANALOG_HPF_CTL, 0xA5 }, /* r0A Analog HPF Ctl */
+ { CS42L52_ADC_HPF_FREQ, 0x00 }, /* r0B ADC HPF Corner Freq */
+ { CS42L52_ADC_MISC_CTL, 0x00 }, /* r0C Misc. ADC Ctl */
+ { CS42L52_PB_CTL1, 0x60 }, /* r0D Playback Ctl 1 */
+ { CS42L52_MISC_CTL, 0x02 }, /* r0E Misc. Ctl */
+ { CS42L52_PB_CTL2, 0x00 }, /* r0F Playback Ctl 2 */
+ { CS42L52_MICA_CTL, 0x00 }, /* r10 MICA Amp Ctl */
+ { CS42L52_MICB_CTL, 0x00 }, /* r11 MICB Amp Ctl */
+ { CS42L52_PGAA_CTL, 0x00 }, /* r12 PGAA Vol, Misc. */
+ { CS42L52_PGAB_CTL, 0x00 }, /* r13 PGAB Vol, Misc. */
+ { CS42L52_PASSTHRUA_VOL, 0x00 }, /* r14 Bypass A Vol */
+ { CS42L52_PASSTHRUB_VOL, 0x00 }, /* r15 Bypass B Vol */
+ { CS42L52_ADCA_VOL, 0x00 }, /* r16 ADCA Volume */
+ { CS42L52_ADCB_VOL, 0x00 }, /* r17 ADCB Volume */
+ { CS42L52_ADCA_MIXER_VOL, 0x80 }, /* r18 ADCA Mixer Volume */
+ { CS42L52_ADCB_MIXER_VOL, 0x80 }, /* r19 ADCB Mixer Volume */
+ { CS42L52_PCMA_MIXER_VOL, 0x00 }, /* r1A PCMA Mixer Volume */
+ { CS42L52_PCMB_MIXER_VOL, 0x00 }, /* r1B PCMB Mixer Volume */
+ { CS42L52_BEEP_FREQ, 0x00 }, /* r1C Beep Freq on Time */
+ { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */
+ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */
+ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */
+ { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */
+ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */
+ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */
+ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */
+ { CS42L52_SPKA_VOL, 0x00 }, /* r24 Speaker A Volume */
+ { CS42L52_SPKB_VOL, 0x00 }, /* r25 Speaker B Volume */
+ { CS42L52_ADC_PCM_MIXER, 0x00 }, /* r26 Channel Mixer and Swap */
+ { CS42L52_LIMITER_CTL1, 0x00 }, /* r27 Limit Ctl 1 Thresholds */
+ { CS42L52_LIMITER_CTL2, 0x7F }, /* r28 Limit Ctl 2 Release Rate */
+ { CS42L52_LIMITER_AT_RATE, 0xC0 }, /* r29 Limiter Attack Rate */
+ { CS42L52_ALC_CTL, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */
+ { CS42L52_ALC_RATE, 0x3F }, /* r2B ALC Release Rate */
+ { CS42L52_ALC_THRESHOLD, 0x3f }, /* r2C ALC Thresholds */
+ { CS42L52_NOISE_GATE_CTL, 0x00 }, /* r2D Noise Gate Ctl */
+ { CS42L52_CLK_STATUS, 0x00 }, /* r2E Overflow and Clock Status */
+ { CS42L52_BATT_COMPEN, 0x00 }, /* r2F battery Compensation */
+ { CS42L52_BATT_LEVEL, 0x00 }, /* r30 VP Battery Level */
+ { CS42L52_SPK_STATUS, 0x00 }, /* r31 Speaker Status */
+ { CS42L52_TEM_CTL, 0x3B }, /* r32 Temp Ctl */
+ { CS42L52_THE_FOLDBACK, 0x00 }, /* r33 Foldback */
+};
+
+static bool cs42l52_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42L52_CHIP:
+ case CS42L52_PWRCTL1:
+ case CS42L52_PWRCTL2:
+ case CS42L52_PWRCTL3:
+ case CS42L52_CLK_CTL:
+ case CS42L52_IFACE_CTL1:
+ case CS42L52_IFACE_CTL2:
+ case CS42L52_ADC_PGA_A:
+ case CS42L52_ADC_PGA_B:
+ case CS42L52_ANALOG_HPF_CTL:
+ case CS42L52_ADC_HPF_FREQ:
+ case CS42L52_ADC_MISC_CTL:
+ case CS42L52_PB_CTL1:
+ case CS42L52_MISC_CTL:
+ case CS42L52_PB_CTL2:
+ case CS42L52_MICA_CTL:
+ case CS42L52_MICB_CTL:
+ case CS42L52_PGAA_CTL:
+ case CS42L52_PGAB_CTL:
+ case CS42L52_PASSTHRUA_VOL:
+ case CS42L52_PASSTHRUB_VOL:
+ case CS42L52_ADCA_VOL:
+ case CS42L52_ADCB_VOL:
+ case CS42L52_ADCA_MIXER_VOL:
+ case CS42L52_ADCB_MIXER_VOL:
+ case CS42L52_PCMA_MIXER_VOL:
+ case CS42L52_PCMB_MIXER_VOL:
+ case CS42L52_BEEP_FREQ:
+ case CS42L52_BEEP_VOL:
+ case CS42L52_BEEP_TONE_CTL:
+ case CS42L52_TONE_CTL:
+ case CS42L52_MASTERA_VOL:
+ case CS42L52_MASTERB_VOL:
+ case CS42L52_HPA_VOL:
+ case CS42L52_HPB_VOL:
+ case CS42L52_SPKA_VOL:
+ case CS42L52_SPKB_VOL:
+ case CS42L52_ADC_PCM_MIXER:
+ case CS42L52_LIMITER_CTL1:
+ case CS42L52_LIMITER_CTL2:
+ case CS42L52_LIMITER_AT_RATE:
+ case CS42L52_ALC_CTL:
+ case CS42L52_ALC_RATE:
+ case CS42L52_ALC_THRESHOLD:
+ case CS42L52_NOISE_GATE_CTL:
+ case CS42L52_CLK_STATUS:
+ case CS42L52_BATT_COMPEN:
+ case CS42L52_BATT_LEVEL:
+ case CS42L52_SPK_STATUS:
+ case CS42L52_TEM_CTL:
+ case CS42L52_THE_FOLDBACK:
+ case CS42L52_CHARGE_PUMP:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool cs42l52_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case CS42L52_IFACE_CTL2:
+ case CS42L52_CLK_STATUS:
+ case CS42L52_BATT_LEVEL:
+ case CS42L52_SPK_STATUS:
+ case CS42L52_CHARGE_PUMP:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1);
+
+static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
+
+static const unsigned int limiter_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
+ 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0),
+};
+
+static const char * const cs42l52_adca_text[] = {
+ "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"};
+
+static const char * const cs42l52_adcb_text[] = {
+ "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"};
+
+static const struct soc_enum adca_enum =
+ SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5,
+ ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text);
+
+static const struct soc_enum adcb_enum =
+ SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5,
+ ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text);
+
+static const struct snd_kcontrol_new adca_mux =
+ SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum);
+
+static const struct snd_kcontrol_new adcb_mux =
+ SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum);
+
+static const char * const mic_bias_level_text[] = {
+ "0.5 +VA", "0.6 +VA", "0.7 +VA",
+ "0.8 +VA", "0.83 +VA", "0.91 +VA"
+};
+
+static const struct soc_enum mic_bias_level_enum =
+ SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+ ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
+
+static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
+
+static const struct soc_enum mica_enum =
+ SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
+ ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+
+static const struct soc_enum micb_enum =
+ SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5,
+ ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+
+static const struct snd_kcontrol_new mica_mux =
+ SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum);
+
+static const struct snd_kcontrol_new micb_mux =
+ SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum);
+
+static const char * const digital_output_mux_text[] = {"ADC", "DSP"};
+
+static const struct soc_enum digital_output_mux_enum =
+ SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6,
+ ARRAY_SIZE(digital_output_mux_text),
+ digital_output_mux_text);
+
+static const struct snd_kcontrol_new digital_output_mux =
+ SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum);
+
+static const char * const hp_gain_num_text[] = {
+ "0.3959", "0.4571", "0.5111", "0.6047",
+ "0.7099", "0.8399", "1.000", "1.1430"
+};
+
+static const struct soc_enum hp_gain_enum =
+ SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+ ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
+
+static const char * const beep_pitch_text[] = {
+ "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5",
+ "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7"
+};
+
+static const struct soc_enum beep_pitch_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4,
+ ARRAY_SIZE(beep_pitch_text), beep_pitch_text);
+
+static const char * const beep_ontime_text[] = {
+ "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s",
+ "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s",
+ "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s"
+};
+
+static const struct soc_enum beep_ontime_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0,
+ ARRAY_SIZE(beep_ontime_text), beep_ontime_text);
+
+static const char * const beep_offtime_text[] = {
+ "1.23 s", "2.58 s", "3.90 s", "5.20 s",
+ "6.60 s", "8.05 s", "9.35 s", "10.80 s"
+};
+
+static const struct soc_enum beep_offtime_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5,
+ ARRAY_SIZE(beep_offtime_text), beep_offtime_text);
+
+static const char * const beep_config_text[] = {
+ "Off", "Single", "Multiple", "Continuous"
+};
+
+static const struct soc_enum beep_config_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6,
+ ARRAY_SIZE(beep_config_text), beep_config_text);
+
+static const char * const beep_bass_text[] = {
+ "50 Hz", "100 Hz", "200 Hz", "250 Hz"
+};
+
+static const struct soc_enum beep_bass_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1,
+ ARRAY_SIZE(beep_bass_text), beep_bass_text);
+
+static const char * const beep_treble_text[] = {
+ "5 kHz", "7 kHz", "10 kHz", " 15 kHz"
+};
+
+static const struct soc_enum beep_treble_enum =
+ SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3,
+ ARRAY_SIZE(beep_treble_text), beep_treble_text);
+
+static const char * const ng_threshold_text[] = {
+ "-34dB", "-37dB", "-40dB", "-43dB",
+ "-46dB", "-52dB", "-58dB", "-64dB"
+};
+
+static const struct soc_enum ng_threshold_enum =
+ SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2,
+ ARRAY_SIZE(ng_threshold_text), ng_threshold_text);
+
+static const char * const cs42l52_ng_delay_text[] = {
+ "50ms", "100ms", "150ms", "200ms"};
+
+static const struct soc_enum ng_delay_enum =
+ SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0,
+ ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text);
+
+static const char * const cs42l52_ng_type_text[] = {
+ "Apply Specific", "Apply All"
+};
+
+static const struct soc_enum ng_type_enum =
+ SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6,
+ ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text);
+
+static const char * const left_swap_text[] = {
+ "Left", "LR 2", "Right"};
+
+static const char * const right_swap_text[] = {
+ "Right", "LR 2", "Left"};
+
+static const unsigned int swap_values[] = { 0, 1, 3 };
+
+static const struct soc_enum adca_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1,
+ ARRAY_SIZE(left_swap_text),
+ left_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new adca_mixer =
+ SOC_DAPM_ENUM("Route", adca_swap_enum);
+
+static const struct soc_enum pcma_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1,
+ ARRAY_SIZE(left_swap_text),
+ left_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new pcma_mixer =
+ SOC_DAPM_ENUM("Route", pcma_swap_enum);
+
+static const struct soc_enum adcb_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1,
+ ARRAY_SIZE(right_swap_text),
+ right_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new adcb_mixer =
+ SOC_DAPM_ENUM("Route", adcb_swap_enum);
+
+static const struct soc_enum pcmb_swap_enum =
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1,
+ ARRAY_SIZE(right_swap_text),
+ right_swap_text,
+ swap_values);
+
+static const struct snd_kcontrol_new pcmb_mixer =
+ SOC_DAPM_ENUM("Route", pcmb_swap_enum);
+
+
+static const struct snd_kcontrol_new passthrul_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0);
+
+static const struct snd_kcontrol_new passthrur_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0);
+
+static const struct snd_kcontrol_new spkl_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1);
+
+static const struct snd_kcontrol_new spkr_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1);
+
+static const struct snd_kcontrol_new hpl_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1);
+
+static const struct snd_kcontrol_new hpr_ctl =
+ SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1);
+
+static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
+
+ SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL,
+ CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv),
+
+ SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL,
+ CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv),
+
+ SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
+
+ SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
+ CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+
+ SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
+ CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
+
+ SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
+
+ SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
+ CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
+
+ SOC_ENUM("MIC Bias Level", mic_bias_level_enum),
+
+ SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL,
+ CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv),
+ SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
+ CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
+ 6, 0x7f, 0x19, ipd_tlv),
+
+ SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
+
+ SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL,
+ CS42L52_ADCB_MIXER_VOL, 7, 1, 1),
+
+ SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL,
+ CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv),
+
+ SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
+ CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
+ 6, 0x7f, 0x19, hl_tlv),
+ SOC_DOUBLE_R("PCM Mixer Switch",
+ CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
+
+ SOC_ENUM("Beep Config", beep_config_enum),
+ SOC_ENUM("Beep Pitch", beep_pitch_enum),
+ SOC_ENUM("Beep on Time", beep_ontime_enum),
+ SOC_ENUM("Beep off Time", beep_offtime_enum),
+ SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+ SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
+ SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
+ SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
+
+ SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1),
+ SOC_SINGLE_TLV("Treble Gain Volume",
+ CS42L52_TONE_CTL, 4, 15, 1, hl_tlv),
+ SOC_SINGLE_TLV("Bass Gain Volume",
+ CS42L52_TONE_CTL, 0, 15, 1, hl_tlv),
+
+ /* Limiter */
+ SOC_SINGLE_TLV("Limiter Max Threshold Volume",
+ CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv),
+ SOC_SINGLE_TLV("Limiter Cushion Threshold Volume",
+ CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv),
+ SOC_SINGLE_TLV("Limiter Release Rate Volume",
+ CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv),
+ SOC_SINGLE_TLV("Limiter Attack Rate Volume",
+ CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv),
+
+ SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0),
+ SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0),
+ SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0),
+
+ /* ALC */
+ SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL,
+ 0, 63, 0, limiter_tlv),
+ SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE,
+ 0, 63, 0, limiter_tlv),
+ SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD,
+ 5, 7, 0, limiter_tlv),
+ SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD,
+ 2, 7, 0, limiter_tlv),
+
+ SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL,
+ CS42L52_PGAB_CTL, 7, 1, 1),
+ SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL,
+ CS42L52_PGAB_CTL, 6, 1, 1),
+ SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0),
+
+ /* Noise gate */
+ SOC_ENUM("NG Type Switch", ng_type_enum),
+ SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0),
+ SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1),
+ SOC_ENUM("NG Threshold", ng_threshold_enum),
+ SOC_ENUM("NG Delay", ng_delay_enum),
+
+ SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0),
+
+ SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1),
+ SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1),
+ SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0),
+ SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0),
+ SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0),
+
+ SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0),
+ SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0),
+ SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0),
+
+ SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0),
+ SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0),
+ SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0),
+ SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0),
+
+ SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0),
+ SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0),
+
+ SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0),
+ SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0),
+
+ SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0),
+ SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0),
+
+};
+
+static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
+
+ SND_SOC_DAPM_INPUT("AIN1L"),
+ SND_SOC_DAPM_INPUT("AIN1R"),
+ SND_SOC_DAPM_INPUT("AIN2L"),
+ SND_SOC_DAPM_INPUT("AIN2R"),
+ SND_SOC_DAPM_INPUT("AIN3L"),
+ SND_SOC_DAPM_INPUT("AIN3R"),
+ SND_SOC_DAPM_INPUT("AIN4L"),
+ SND_SOC_DAPM_INPUT("AIN4R"),
+ SND_SOC_DAPM_INPUT("MICA"),
+ SND_SOC_DAPM_INPUT("MICB"),
+ SND_SOC_DAPM_SIGGEN("Beep"),
+
+ SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux),
+ SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux),
+
+ SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1),
+ SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1),
+ SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0),
+
+ SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux),
+ SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux),
+
+ SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM,
+ 0, 0, &adca_mixer),
+ SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM,
+ 0, 0, &adcb_mixer),
+
+ SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM,
+ 0, 0, &digital_output_mux),
+
+ SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_IN("AIFINL", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFINR", NULL, 0,
+ SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL,
+ 6, 0, &passthrul_ctl),
+ SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL,
+ 7, 0, &passthrur_ctl),
+
+ SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM,
+ 0, 0, &pcma_mixer),
+ SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM,
+ 0, 0, &pcmb_mixer),
+
+ SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl),
+ SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl),
+
+ SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl),
+ SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl),
+
+ SND_SOC_DAPM_OUTPUT("HPOUTA"),
+ SND_SOC_DAPM_OUTPUT("HPOUTB"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTA"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTB"),
+
+};
+
+static const struct snd_soc_dapm_route cs42l52_audio_map[] = {
+
+ {"Capture", NULL, "AIFOUTL"},
+ {"Capture", NULL, "AIFOUTL"},
+
+ {"AIFOUTL", NULL, "Output Mux"},
+ {"AIFOUTR", NULL, "Output Mux"},
+
+ {"Output Mux", "ADC", "ADC Left"},
+ {"Output Mux", "ADC", "ADC Right"},
+
+ {"ADC Left", NULL, "Charge Pump"},
+ {"ADC Right", NULL, "Charge Pump"},
+
+ {"Charge Pump", NULL, "ADC Left Mux"},
+ {"Charge Pump", NULL, "ADC Right Mux"},
+
+ {"ADC Left Mux", "Input1A", "AIN1L"},
+ {"ADC Right Mux", "Input1B", "AIN1R"},
+ {"ADC Left Mux", "Input2A", "AIN2L"},
+ {"ADC Right Mux", "Input2B", "AIN2R"},
+ {"ADC Left Mux", "Input3A", "AIN3L"},
+ {"ADC Right Mux", "Input3B", "AIN3R"},
+ {"ADC Left Mux", "Input4A", "AIN4L"},
+ {"ADC Right Mux", "Input4B", "AIN4R"},
+ {"ADC Left Mux", "PGA Input Left", "PGA Left"},
+ {"ADC Right Mux", "PGA Input Right" , "PGA Right"},
+
+ {"PGA Left", "Switch", "AIN1L"},
+ {"PGA Right", "Switch", "AIN1R"},
+ {"PGA Left", "Switch", "AIN2L"},
+ {"PGA Right", "Switch", "AIN2R"},
+ {"PGA Left", "Switch", "AIN3L"},
+ {"PGA Right", "Switch", "AIN3R"},
+ {"PGA Left", "Switch", "AIN4L"},
+ {"PGA Right", "Switch", "AIN4R"},
+
+ {"PGA Left", "Switch", "PGA MICA"},
+ {"PGA MICA", NULL, "MICA"},
+
+ {"PGA Right", "Switch", "PGA MICB"},
+ {"PGA MICB", NULL, "MICB"},
+
+ {"HPOUTA", NULL, "HP Left Amp"},
+ {"HPOUTB", NULL, "HP Right Amp"},
+ {"HP Left Amp", NULL, "Bypass Left"},
+ {"HP Right Amp", NULL, "Bypass Right"},
+ {"Bypass Left", "Switch", "PGA Left"},
+ {"Bypass Right", "Switch", "PGA Right"},
+ {"HP Left Amp", "Switch", "DAC Left"},
+ {"HP Right Amp", "Switch", "DAC Right"},
+
+ {"SPKOUTA", NULL, "SPK Left Amp"},
+ {"SPKOUTB", NULL, "SPK Right Amp"},
+
+ {"SPK Left Amp", NULL, "Beep"},
+ {"SPK Right Amp", NULL, "Beep"},
+ {"SPK Left Amp", "Switch", "Playback"},
+ {"SPK Right Amp", "Switch", "Playback"},
+
+ {"DAC Left", NULL, "Beep"},
+ {"DAC Right", NULL, "Beep"},
+ {"DAC Left", NULL, "Playback"},
+ {"DAC Right", NULL, "Playback"},
+
+ {"Output Mux", "DSP", "Playback"},
+ {"Output Mux", "DSP", "Playback"},
+
+ {"AIFINL", NULL, "Playback"},
+ {"AIFINR", NULL, "Playback"},
+
+};
+
+struct cs42l52_clk_para {
+ u32 mclk;
+ u32 rate;
+ u8 speed;
+ u8 group;
+ u8 videoclk;
+ u8 ratio;
+ u8 mclkdiv2;
+};
+
+static const struct cs42l52_clk_para clk_map_table[] = {
+ /*8k*/
+ {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0},
+
+ /*11.025k*/
+ {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /*16k*/
+ {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1},
+
+ /*22.05k*/
+ {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /* 32k */
+ {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0},
+
+ /* 44.1k */
+ {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /* 48k */
+ {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1},
+ {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1},
+
+ /* 88.2k */
+ {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+
+ /* 96k */
+ {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0},
+ {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0},
+ {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1},
+};
+
+static int cs42l52_get_clk(int mclk, int rate)
+{
+ int i, ret = 0;
+ u_int mclk1, mclk2 = 0;
+
+ for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+ if (clk_map_table[i].rate == rate) {
+ mclk1 = clk_map_table[i].mclk;
+ if (abs(mclk - mclk1) < abs(mclk - mclk2)) {
+ mclk2 = mclk1;
+ ret = i;
+ }
+ }
+ }
+ if (ret > ARRAY_SIZE(clk_map_table))
+ return -EINVAL;
+ return ret;
+}
+
+static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) {
+ cs42l52->sysclk = freq;
+ } else {
+ dev_err(codec->dev, "Invalid freq paramter\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+ u8 iface = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = CS42L52_IFACE_CTL1_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iface = CS42L52_IFACE_CTL1_SLAVE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S |
+ CS42L52_IFACE_CTL1_DAC_FMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J |
+ CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= CS42L52_IFACE_CTL1_INV_SCLK;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= CS42L52_IFACE_CTL1_INV_SCLK;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ cs42l52->config.format = iface;
+ snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format);
+
+ return 0;
+}
+
+static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (mute)
+ snd_soc_update_bits(codec, CS42L52_PB_CTL1,
+ CS42L52_PB_CTL1_MUTE_MASK,
+ CS42L52_PB_CTL1_MUTE);
+ else
+ snd_soc_update_bits(codec, CS42L52_PB_CTL1,
+ CS42L52_PB_CTL1_MUTE_MASK,
+ CS42L52_PB_CTL1_UNMUTE);
+
+ return 0;
+}
+
+static int cs42l52_pcm_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 cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ u32 clk = 0;
+ int index;
+
+ index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params));
+ if (index >= 0) {
+ cs42l52->sysclk = clk_map_table[index].mclk;
+
+ clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) |
+ (clk_map_table[index].group << CLK_32K_SR_SHIFT) |
+ (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) |
+ (clk_map_table[index].ratio << CLK_RATIO_SHIFT) |
+ clk_map_table[index].mclkdiv2;
+
+ snd_soc_write(codec, CS42L52_CLK_CTL, clk);
+ } else {
+ dev_err(codec->dev, "can't get correct mclk\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_update_bits(codec, CS42L52_PWRCTL1,
+ CS42L52_PWRCTL1_PDN_CODEC, 0);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ regcache_cache_only(cs42l52->regmap, false);
+ regcache_sync(cs42l52->regmap);
+ }
+ snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL);
+ regcache_cache_only(cs42l52->regmap, true);
+ break;
+ }
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+
+static struct snd_soc_dai_ops cs42l52_ops = {
+ .hw_params = cs42l52_pcm_hw_params,
+ .digital_mute = cs42l52_digital_mute,
+ .set_fmt = cs42l52_set_fmt,
+ .set_sysclk = cs42l52_set_sysclk,
+};
+
+static struct snd_soc_dai_driver cs42l52_dai = {
+ .name = "cs42l52",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS42L52_RATES,
+ .formats = CS42L52_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CS42L52_RATES,
+ .formats = CS42L52_FORMATS,
+ },
+ .ops = &cs42l52_ops,
+};
+
+static int cs42l52_suspend(struct snd_soc_codec *codec)
+{
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int cs42l52_resume(struct snd_soc_codec *codec)
+{
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+static int beep_rates[] = {
+ 261, 522, 585, 667, 706, 774, 889, 1000,
+ 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182
+};
+
+static void cs42l52_beep_work(struct work_struct *work)
+{
+ struct cs42l52_private *cs42l52 =
+ container_of(work, struct cs42l52_private, beep_work);
+ struct snd_soc_codec *codec = cs42l52->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int i;
+ int val = 0;
+ int best = 0;
+
+ if (cs42l52->beep_rate) {
+ for (i = 0; i < ARRAY_SIZE(beep_rates); i++) {
+ if (abs(cs42l52->beep_rate - beep_rates[i]) <
+ abs(cs42l52->beep_rate - beep_rates[best]))
+ best = i;
+ }
+
+ dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n",
+ beep_rates[best], cs42l52->beep_rate);
+
+ val = (best << CS42L52_BEEP_RATE_SHIFT);
+
+ snd_soc_dapm_enable_pin(dapm, "Beep");
+ } else {
+ dev_dbg(codec->dev, "Disabling beep\n");
+ snd_soc_dapm_disable_pin(dapm, "Beep");
+ }
+
+ snd_soc_update_bits(codec, CS42L52_BEEP_FREQ,
+ CS42L52_BEEP_RATE_MASK, val);
+
+ snd_soc_dapm_sync(dapm);
+}
+
+/* For usability define a way of injecting beep events for the device -
+ * many systems will not have a keyboard.
+ */
+static int cs42l52_beep_event(struct input_dev *dev, unsigned int type,
+ unsigned int code, int hz)
+{
+ struct snd_soc_codec *codec = input_get_drvdata(dev);
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "Beep event %x %x\n", code, hz);
+
+ switch (code) {
+ case SND_BELL:
+ if (hz)
+ hz = 261;
+ case SND_TONE:
+ break;
+ default:
+ return -1;
+ }
+
+ /* Kick the beep from a workqueue */
+ cs42l52->beep_rate = hz;
+ schedule_work(&cs42l52->beep_work);
+ return 0;
+}
+
+static ssize_t cs42l52_beep_set(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t count)
+{
+ struct cs42l52_private *cs42l52 = dev_get_drvdata(dev);
+ long int time;
+ int ret;
+
+ ret = kstrtol(buf, 10, &time);
+ if (ret != 0)
+ return ret;
+
+ input_event(cs42l52->beep, EV_SND, SND_TONE, time);
+
+ return count;
+}
+
+static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set);
+
+static void cs42l52_init_beep(struct snd_soc_codec *codec)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ cs42l52->beep = input_allocate_device();
+ if (!cs42l52->beep) {
+ dev_err(codec->dev, "Failed to allocate beep device\n");
+ return;
+ }
+
+ INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work);
+ cs42l52->beep_rate = 0;
+
+ cs42l52->beep->name = "CS42L52 Beep Generator";
+ cs42l52->beep->phys = dev_name(codec->dev);
+ cs42l52->beep->id.bustype = BUS_I2C;
+
+ cs42l52->beep->evbit[0] = BIT_MASK(EV_SND);
+ cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+ cs42l52->beep->event = cs42l52_beep_event;
+ cs42l52->beep->dev.parent = codec->dev;
+ input_set_drvdata(cs42l52->beep, codec);
+
+ ret = input_register_device(cs42l52->beep);
+ if (ret != 0) {
+ input_free_device(cs42l52->beep);
+ cs42l52->beep = NULL;
+ dev_err(codec->dev, "Failed to register beep device\n");
+ }
+
+ ret = device_create_file(codec->dev, &dev_attr_beep);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to create keyclick file: %d\n",
+ ret);
+ }
+}
+
+static void cs42l52_free_beep(struct snd_soc_codec *codec)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+
+ device_remove_file(codec->dev, &dev_attr_beep);
+ input_unregister_device(cs42l52->beep);
+ cancel_work_sync(&cs42l52->beep_work);
+ cs42l52->beep = NULL;
+
+ snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL,
+ CS42L52_BEEP_EN_MASK, 0);
+}
+#else
+static void cs42l52_init_beep(struct snd_soc_codec *codec)
+{
+}
+
+static void cs42l52_free_beep(struct snd_soc_codec *codec)
+{
+}
+#endif
+
+static int cs42l52_probe(struct snd_soc_codec *codec)
+{
+ struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ codec->control_data = cs42l52->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+ regcache_cache_only(cs42l52->regmap, true);
+
+ cs42l52_init_beep(codec);
+
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ cs42l52->sysclk = CS42L52_DEFAULT_CLK;
+ cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
+
+ /* Set Platform MICx CFG */
+ snd_soc_update_bits(codec, CS42L52_MICA_CTL,
+ CS42L52_MIC_CTL_TYPE_MASK,
+ cs42l52->pdata.mica_cfg <<
+ CS42L52_MIC_CTL_TYPE_SHIFT);
+
+ snd_soc_update_bits(codec, CS42L52_MICB_CTL,
+ CS42L52_MIC_CTL_TYPE_MASK,
+ cs42l52->pdata.micb_cfg <<
+ CS42L52_MIC_CTL_TYPE_SHIFT);
+
+ /* if Single Ended, Get Mic_Select */
+ if (cs42l52->pdata.mica_cfg)
+ snd_soc_update_bits(codec, CS42L52_MICA_CTL,
+ CS42L52_MIC_CTL_MIC_SEL_MASK,
+ cs42l52->pdata.mica_sel <<
+ CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+ if (cs42l52->pdata.micb_cfg)
+ snd_soc_update_bits(codec, CS42L52_MICB_CTL,
+ CS42L52_MIC_CTL_MIC_SEL_MASK,
+ cs42l52->pdata.micb_sel <<
+ CS42L52_MIC_CTL_MIC_SEL_SHIFT);
+
+ /* Set Platform Charge Pump Freq */
+ snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP,
+ CS42L52_CHARGE_PUMP_MASK,
+ cs42l52->pdata.chgfreq <<
+ CS42L52_CHARGE_PUMP_SHIFT);
+
+ /* Set Platform Bias Level */
+ snd_soc_update_bits(codec, CS42L52_IFACE_CTL2,
+ CS42L52_IFACE_CTL2_BIAS_LVL,
+ cs42l52->pdata.micbias_lvl);
+
+ return ret;
+}
+
+static int cs42l52_remove(struct snd_soc_codec *codec)
+{
+ cs42l52_free_beep(codec);
+ cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = {
+ .probe = cs42l52_probe,
+ .remove = cs42l52_remove,
+ .suspend = cs42l52_suspend,
+ .resume = cs42l52_resume,
+ .set_bias_level = cs42l52_set_bias_level,
+
+ .dapm_widgets = cs42l52_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets),
+ .dapm_routes = cs42l52_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map),
+
+ .controls = cs42l52_snd_controls,
+ .num_controls = ARRAY_SIZE(cs42l52_snd_controls),
+};
+
+/* Current and threshold powerup sequence Pg37 */
+static const struct reg_default cs42l52_threshold_patch[] = {
+
+ { 0x00, 0x99 },
+ { 0x3E, 0xBA },
+ { 0x47, 0x80 },
+ { 0x32, 0xBB },
+ { 0x32, 0x3B },
+ { 0x00, 0x00 },
+
+};
+
+static struct regmap_config cs42l52_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS42L52_MAX_REGISTER,
+ .reg_defaults = cs42l52_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults),
+ .readable_reg = cs42l52_readable_register,
+ .volatile_reg = cs42l52_volatile_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct cs42l52_private *cs42l52;
+ int ret;
+ unsigned int devid = 0;
+ unsigned int reg;
+
+ cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private),
+ GFP_KERNEL);
+ if (cs42l52 == NULL)
+ return -ENOMEM;
+ cs42l52->dev = &i2c_client->dev;
+
+ cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap);
+ if (IS_ERR(cs42l52->regmap)) {
+ ret = PTR_ERR(cs42l52->regmap);
+ dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+ goto err;
+ }
+
+ i2c_set_clientdata(i2c_client, cs42l52);
+
+ if (dev_get_platdata(&i2c_client->dev))
+ memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev),
+ sizeof(cs42l52->pdata));
+
+ ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch,
+ ARRAY_SIZE(cs42l52_threshold_patch));
+ if (ret != 0)
+ dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n",
+ ret);
+
+ ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg);
+ devid = reg & CS42L52_CHIP_ID_MASK;
+ if (devid != CS42L52_CHIP_ID) {
+ ret = -ENODEV;
+ dev_err(&i2c_client->dev,
+ "CS42L52 Device ID (%X). Expected %X\n",
+ devid, CS42L52_CHIP_ID);
+ goto err_regmap;
+ }
+
+ regcache_cache_only(cs42l52->regmap, true);
+
+ ret = snd_soc_register_codec(&i2c_client->dev,
+ &soc_codec_dev_cs42l52, &cs42l52_dai, 1);
+ if (ret < 0)
+ goto err_regmap;
+ return 0;
+
+err_regmap:
+ regmap_exit(cs42l52->regmap);
+
+err:
+ return ret;
+}
+
+static int cs42l52_i2c_remove(struct i2c_client *client)
+{
+ struct cs42l52_private *cs42l52 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+ regmap_exit(cs42l52->regmap);
+
+ return 0;
+}
+
+static const struct i2c_device_id cs42l52_id[] = {
+ { "cs42l52", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, cs42l52_id);
+
+static struct i2c_driver cs42l52_i2c_driver = {
+ .driver = {
+ .name = "cs42l52",
+ .owner = THIS_MODULE,
+ },
+ .id_table = cs42l52_id,
+ .probe = cs42l52_i2c_probe,
+ .remove = __devexit_p(cs42l52_i2c_remove),
+};
+
+module_i2c_driver(cs42l52_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS42L52 driver");
+MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");
+MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
new file mode 100644
index 00000000000..60985c05907
--- /dev/null
+++ b/sound/soc/codecs/cs42l52.h
@@ -0,0 +1,274 @@
+/*
+ * cs42l52.h -- CS42L52 ALSA SoC audio driver
+ *
+ * Copyright 2012 CirrusLogic, Inc.
+ *
+ * Author: Georgi Vlaev <joe@nucleusys.com>
+ * Author: Brian Austin <brian.austin@cirrus.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef __CS42L52_H__
+#define __CS42L52_H__
+
+#define CS42L52_NAME "CS42L52"
+#define CS42L52_DEFAULT_CLK 12000000
+#define CS42L52_MIN_CLK 11000000
+#define CS42L52_MAX_CLK 27000000
+#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE
+#define CS42L52_DEFAULT_MAX_CHANS 2
+#define CS42L52_SYSCLK 1
+
+#define CS42L52_CHIP_SWICTH (1 << 17)
+#define CS42L52_ALL_IN_ONE (1 << 16)
+#define CS42L52_CHIP_ONE 0x00
+#define CS42L52_CHIP_TWO 0x01
+#define CS42L52_CHIP_THR 0x02
+#define CS42L52_CHIP_MASK 0x0f
+
+#define CS42L52_FIX_BITS_CTL 0x00
+#define CS42L52_CHIP 0x01
+#define CS42L52_CHIP_ID 0xE0
+#define CS42L52_CHIP_ID_MASK 0xF8
+#define CS42L52_CHIP_REV_A0 0x00
+#define CS42L52_CHIP_REV_A1 0x01
+#define CS42L52_CHIP_REV_B0 0x02
+#define CS42L52_CHIP_REV_MASK 0x03
+
+#define CS42L52_PWRCTL1 0x02
+#define CS42L52_PWRCTL1_PDN_ALL 0x9F
+#define CS42L52_PWRCTL1_PDN_CHRG 0x80
+#define CS42L52_PWRCTL1_PDN_PGAB 0x10
+#define CS42L52_PWRCTL1_PDN_PGAA 0x08
+#define CS42L52_PWRCTL1_PDN_ADCB 0x04
+#define CS42L52_PWRCTL1_PDN_ADCA 0x02
+#define CS42L52_PWRCTL1_PDN_CODEC 0x01
+
+#define CS42L52_PWRCTL2 0x03
+#define CS42L52_PWRCTL2_OVRDB (1 << 4)
+#define CS42L52_PWRCTL2_OVRDA (1 << 3)
+#define CS42L52_PWRCTL2_PDN_MICB (1 << 2)
+#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2
+#define CS42L52_PWRCTL2_PDN_MICA (1 << 1)
+#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1
+#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0)
+#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0
+
+#define CS42L52_PWRCTL3 0x04
+#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6
+#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00
+#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02
+#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03
+#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4
+#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00
+#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02
+#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03
+#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2
+#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00
+#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02
+#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2)
+#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0)
+#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0
+#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00
+#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01
+#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02
+
+#define CS42L52_DEFAULT_OUTPUT_STATE 0x05
+#define CS42L52_PWRCTL3_CONF_MASK 0x03
+
+#define CS42L52_CLK_CTL 0x05
+#define CLK_AUTODECT_ENABLE (1 << 7)
+#define CLK_SPEED_SHIFT 5
+#define CLK_DS_MODE 0x00
+#define CLK_SS_MODE 0x01
+#define CLK_HS_MODE 0x02
+#define CLK_QS_MODE 0x03
+#define CLK_32K_SR_SHIFT 4
+#define CLK_32K 0x01
+#define CLK_NO_32K 0x00
+#define CLK_27M_MCLK_SHIFT 3
+#define CLK_27M_MCLK 0x01
+#define CLK_NO_27M 0x00
+#define CLK_RATIO_SHIFT 1
+#define CLK_R_128 0x00
+#define CLK_R_125 0x01
+#define CLK_R_132 0x02
+#define CLK_R_136 0x03
+
+#define CS42L52_IFACE_CTL1 0x06
+#define CS42L52_IFACE_CTL1_MASTER (1 << 7)
+#define CS42L52_IFACE_CTL1_SLAVE (0 << 7)
+#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6)
+#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5)
+#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5)
+#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4)
+#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2)
+#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2)
+#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2)
+#define CS42L52_IFACE_CTL1_WL_32BIT (0x00)
+#define CS42L52_IFACE_CTL1_WL_24BIT (0x01)
+#define CS42L52_IFACE_CTL1_WL_20BIT (0x02)
+#define CS42L52_IFACE_CTL1_WL_16BIT (0x03)
+#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF
+
+#define CS42L52_IFACE_CTL2 0x07
+#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6)
+#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5)
+#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4)
+#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4)
+#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3)
+#define CS42L52_IFACE_CTL2_BIAS_LVL 0x07
+
+#define CS42L52_ADC_PGA_A 0x08
+#define CS42L52_ADC_PGA_B 0x09
+#define CS42L52_ADC_SEL_SHIFT 5
+#define CS42L52_ADC_SEL_AIN1 0x00
+#define CS42L52_ADC_SEL_AIN2 0x01
+#define CS42L52_ADC_SEL_AIN3 0x02
+#define CS42L52_ADC_SEL_AIN4 0x03
+#define CS42L52_ADC_SEL_PGA 0x04
+
+#define CS42L52_ANALOG_HPF_CTL 0x0A
+#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3)
+#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0)
+
+#define CS42L52_ADC_HPF_FREQ 0x0B
+#define CS42L52_ADC_MISC_CTL 0x0C
+#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6)
+
+#define CS42L52_PB_CTL1 0x0D
+#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5
+#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00
+#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01
+#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02
+#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03
+#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04
+#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05
+#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06
+#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07
+#define CS42L52_PB_CTL1_INV_PCMB (1 << 3)
+#define CS42L52_PB_CTL1_INV_PCMA (1 << 2)
+#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1)
+#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0)
+#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD
+#define CS42L52_PB_CTL1_MUTE 3
+#define CS42L52_PB_CTL1_UNMUTE 0
+
+#define CS42L52_MISC_CTL 0x0E
+#define CS42L52_MISC_CTL_DEEMPH (1 << 2)
+#define CS42L52_MISC_CTL_DIGSFT (1 << 1)
+#define CS42L52_MISC_CTL_DIGZC (1 << 0)
+
+#define CS42L52_PB_CTL2 0x0F
+#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7)
+#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6)
+#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5)
+#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4)
+#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2)
+#define CS42L52_PB_CTL2_SPK_MONO (1 << 1)
+#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0)
+
+#define CS42L52_MICA_CTL 0x10
+#define CS42L52_MICB_CTL 0x11
+#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF
+#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6
+#define CS42L52_MIC_CTL_TYPE_MASK 0xDF
+#define CS42L52_MIC_CTL_TYPE_SHIFT 5
+
+
+#define CS42L52_PGAA_CTL 0x12
+#define CS42L52_PGAB_CTL 0x13
+#define CS42L52_PGAX_CTL_VOL_12DB 24
+#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/
+
+#define CS42L52_PASSTHRUA_VOL 0x14
+#define CS42L52_PASSTHRUB_VOL 0x15
+
+#define CS42L52_ADCA_VOL 0x16
+#define CS42L52_ADCB_VOL 0x17
+#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/
+#define CS42L52_ADCX_VOL_12DB 12
+#define CS42L52_ADCX_VOL_6DB 6
+
+#define CS42L52_ADCA_MIXER_VOL 0x18
+#define CS42L52_ADCB_MIXER_VOL 0x19
+#define CS42L52_ADC_MIXER_VOL_12DB 0x18
+
+#define CS42L52_PCMA_MIXER_VOL 0x1A
+#define CS42L52_PCMB_MIXER_VOL 0x1B
+
+#define CS42L52_BEEP_FREQ 0x1C
+#define CS42L52_BEEP_VOL 0x1D
+#define CS42L52_BEEP_TONE_CTL 0x1E
+#define CS42L52_BEEP_RATE_SHIFT 4
+#define CS42L52_BEEP_RATE_MASK 0x0F
+
+#define CS42L52_TONE_CTL 0x1F
+#define CS42L52_BEEP_EN_MASK 0x3F
+
+#define CS42L52_MASTERA_VOL 0x20
+#define CS42L52_MASTERB_VOL 0x21
+
+#define CS42L52_HPA_VOL 0x22
+#define CS42L52_HPB_VOL 0x23
+#define CS42L52_DEFAULT_HP_VOL 0xF0
+
+#define CS42L52_SPKA_VOL 0x24
+#define CS42L52_SPKB_VOL 0x25
+#define CS42L52_DEFAULT_SPK_VOL 0xF0
+
+#define CS42L52_ADC_PCM_MIXER 0x26
+
+#define CS42L52_LIMITER_CTL1 0x27
+#define CS42L52_LIMITER_CTL2 0x28
+#define CS42L52_LIMITER_AT_RATE 0x29
+
+#define CS42L52_ALC_CTL 0x2A
+#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT 7
+#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT 6
+#define CS42L52_ALC_CTL_FASTEST_ATTACK 0
+
+#define CS42L52_ALC_RATE 0x2B
+#define CS42L52_ALC_SLOWEST_RELEASE 0x3F
+
+#define CS42L52_ALC_THRESHOLD 0x2C
+#define CS42L52_ALC_MAX_RATE_SHIFT 5
+#define CS42L52_ALC_MIN_RATE_SHIFT 2
+#define CS42L52_ALC_RATE_0DB 0
+#define CS42L52_ALC_RATE_3DB 1
+#define CS42L52_ALC_RATE_6DB 2
+
+#define CS42L52_NOISE_GATE_CTL 0x2D
+#define CS42L52_NG_ENABLE_SHIFT 6
+#define CS42L52_NG_THRESHOLD_SHIFT 2
+#define CS42L52_NG_MIN_70DB 2
+#define CS42L52_NG_DELAY_SHIFT 0
+#define CS42L52_NG_DELAY_100MS 1
+
+#define CS42L52_CLK_STATUS 0x2E
+#define CS42L52_BATT_COMPEN 0x2F
+
+#define CS42L52_BATT_LEVEL 0x30
+#define CS42L52_SPK_STATUS 0x31
+#define CS42L52_SPK_STATUS_PIN_SHIFT 3
+#define CS42L52_SPK_STATUS_PIN_HIGH 1
+
+#define CS42L52_TEM_CTL 0x32
+#define CS42L52_TEM_CTL_SET 0x80
+#define CS42L52_THE_FOLDBACK 0x33
+#define CS42L52_CHARGE_PUMP 0x34
+#define CS42L52_CHARGE_PUMP_MASK 0xF0
+#define CS42L52_CHARGE_PUMP_SHIFT 4
+#define CS42L52_FIX_BITS1 0x3E
+#define CS42L52_FIX_BITS2 0x47
+
+#define CS42L52_MAX_REGISTER 0x34
+
+#endif
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 07c44b71f09..e0d45fdaa75 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -43,9 +43,6 @@ struct cs42l73_private {
};
static const struct reg_default cs42l73_reg_defaults[] = {
- { 1, 0x42 }, /* r01 - Device ID A&B */
- { 2, 0xA7 }, /* r02 - Device ID C&D */
- { 3, 0x30 }, /* r03 - Device ID E */
{ 6, 0xF1 }, /* r06 - Power Ctl 1 */
{ 7, 0xDF }, /* r07 - Power Ctl 2 */
{ 8, 0x3F }, /* r08 - Power Ctl 3 */
@@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl =
static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume",
- CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7,
- 0xffffffC1, 0x0C, hpaloa_tlv),
+ CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0,
+ 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL,
- CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv),
+ CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
- CS42L73_MICBPREPGABVOL, 5, 0xffffff35,
- 0x34, micpga_tlv),
+ CS42L73_MICBPREPGABVOL, 5, 0x34,
+ 0x24, micpga_tlv),
SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,
CS42L73_MICBPREPGABVOL, 6, 1, 1),
SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL,
- CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv),
+ CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv),
SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume",
- CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5,
- 0xE4, hl_tlv),
+ CS42L73_HLADVOL, CS42L73_HLBDVOL,
+ 0, 0x34, 0xE4, hl_tlv),
SOC_SINGLE_TLV("ADC A Boost Volume",
CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv),
SOC_SINGLE_TLV("ADC B Boost Volume",
- CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
+ CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
- SOC_SINGLE_TLV("Speakerphone Digital Playback Volume",
- CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv),
+ SOC_SINGLE_SX_TLV("Speakerphone Digital Volume",
+ CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv),
- SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume",
- CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv),
+ SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume",
+ CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv),
SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL,
CS42L73_HPBAVOL, 7, 1, 1),
@@ -568,22 +565,22 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
attn_tlv),
SOC_SINGLE_TLV("SPK-IP Mono Volume",
- CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("SPK-XSP Mono Volume",
- CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("SPK-ASP Mono Volume",
- CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("SPK-VSP Mono Volume",
- CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-IP Mono Volume",
- CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-XSP Mono Volume",
- CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-ASP Mono Volume",
- CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-VSP Mono Volume",
- CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),
SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
@@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0),
- SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0,
CS42L73_PWRCTL2, 1, 1),
- SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0,
CS42L73_PWRCTL2, 1, 1),
- SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0,
CS42L73_PWRCTL2, 3, 1),
- SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0,
CS42L73_PWRCTL2, 3, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0,
+ SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
- SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
- SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
- SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
+ SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"HL Left Mixer", NULL, "VSPIN"},
{"HL Right Mixer", NULL, "VSPIN"},
+ {"ASPINL", NULL, "ASP Playback"},
+ {"ASPINM", NULL, "ASP Playback"},
+ {"ASPINR", NULL, "ASP Playback"},
+ {"XSPINL", NULL, "XSP Playback"},
+ {"XSPINM", NULL, "XSP Playback"},
+ {"XSPINR", NULL, "XSP Playback"},
+ {"VSPIN", NULL, "VSP Playback"},
+
/* Capture Paths */
{"MIC1", NULL, "MIC1 Bias"},
{"PGA Left Mux", "Mic 1", "MIC1"},
@@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"VSPOUTL", NULL, "VSPL Output Mixer"},
{"VSPOUTR", NULL, "VSPR Output Mixer"},
+
+ {"ASP Capture", NULL, "ASPOUTL"},
+ {"ASP Capture", NULL, "ASPOUTR"},
+ {"XSP Capture", NULL, "XSPOUTL"},
+ {"XSP Capture", NULL, "XSPOUTR"},
+ {"VSP Capture", NULL, "VSPOUTL"},
+ {"VSP Capture", NULL, "VSPOUTR"},
};
struct cs42l73_mclk_div {
@@ -1091,8 +1103,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
int id = dai->id;
int mclk_coeff;
@@ -1429,25 +1440,7 @@ static struct i2c_driver cs42l73_i2c_driver = {
};
-static int __init cs42l73_modinit(void)
-{
- int ret;
- ret = i2c_add_driver(&cs42l73_i2c_driver);
- if (ret != 0) {
- pr_err("Failed to register CS42L73 I2C driver: %d\n", ret);
- return ret;
- }
- return 0;
-}
-
-module_init(cs42l73_modinit);
-
-static void __exit cs42l73_exit(void)
-{
- i2c_del_driver(&cs42l73_i2c_driver);
-}
-
-module_exit(cs42l73_exit);
+module_i2c_driver(cs42l73_i2c_driver);
MODULE_DESCRIPTION("ASoC CS42L73 driver");
MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 7843711729b..af5db708051 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -17,6 +17,7 @@
#include <linux/delay.h>
#include <linux/i2c.h>
+#include <linux/spi/spi.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/module.h>
@@ -27,6 +28,7 @@
#include <sound/tlv.h>
/* DA7210 register space */
+#define DA7210_PAGE_CONTROL 0x00
#define DA7210_CONTROL 0x01
#define DA7210_STATUS 0x02
#define DA7210_STARTUP1 0x03
@@ -146,6 +148,7 @@
#define DA7210_DAI_EN (1 << 7)
/*PLL_DIV3 bit fields */
+#define DA7210_PLL_DIV_L_MASK (0xF << 0)
#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4)
#define DA7210_PLL_BYP (1 << 6)
@@ -162,12 +165,16 @@
#define DA7210_PLL_FS_48000 (0xB << 0)
#define DA7210_PLL_FS_88200 (0xE << 0)
#define DA7210_PLL_FS_96000 (0xF << 0)
+#define DA7210_MCLK_DET_EN (0x1 << 5)
+#define DA7210_MCLK_SRM_EN (0x1 << 6)
#define DA7210_PLL_EN (0x1 << 7)
/* SOFTMUTE bit fields */
#define DA7210_RAMP_EN (1 << 6)
/* CONTROL bit fields */
+#define DA7210_REG_EN (1 << 0)
+#define DA7210_BIAS_EN (1 << 2)
#define DA7210_NOISE_SUP_EN (1 << 3)
/* IN_GAIN bit fields */
@@ -206,6 +213,47 @@
#define DA7210_OUT2_OUTMIX_L (1 << 6)
#define DA7210_OUT2_EN (1 << 7)
+struct pll_div {
+ int fref;
+ int fout;
+ u8 div1;
+ u8 div2;
+ u8 div3;
+ u8 mode; /* 0 = slave, 1 = master */
+};
+
+/* PLL dividers table */
+static const struct pll_div da7210_pll_div[] = {
+ /* for MASTER mode, fs = 44.1Khz */
+ { 12000000, 2822400, 0xE8, 0x6C, 0x2, 1}, /* MCLK=12Mhz */
+ { 13000000, 2822400, 0xDF, 0x28, 0xC, 1}, /* MCLK=13Mhz */
+ { 13500000, 2822400, 0xDB, 0x0A, 0xD, 1}, /* MCLK=13.5Mhz */
+ { 14400000, 2822400, 0xD4, 0x5A, 0x2, 1}, /* MCLK=14.4Mhz */
+ { 19200000, 2822400, 0xBB, 0x43, 0x9, 1}, /* MCLK=19.2Mhz */
+ { 19680000, 2822400, 0xB9, 0x6D, 0xA, 1}, /* MCLK=19.68Mhz */
+ { 19800000, 2822400, 0xB8, 0xFB, 0xB, 1}, /* MCLK=19.8Mhz */
+ /* for MASTER mode, fs = 48Khz */
+ { 12000000, 3072000, 0xF3, 0x12, 0x7, 1}, /* MCLK=12Mhz */
+ { 13000000, 3072000, 0xE8, 0xFD, 0x5, 1}, /* MCLK=13Mhz */
+ { 13500000, 3072000, 0xE4, 0x82, 0x3, 1}, /* MCLK=13.5Mhz */
+ { 14400000, 3072000, 0xDD, 0x3A, 0x0, 1}, /* MCLK=14.4Mhz */
+ { 19200000, 3072000, 0xC1, 0xEB, 0x8, 1}, /* MCLK=19.2Mhz */
+ { 19680000, 3072000, 0xBF, 0xEC, 0x0, 1}, /* MCLK=19.68Mhz */
+ { 19800000, 3072000, 0xBF, 0x70, 0x0, 1}, /* MCLK=19.8Mhz */
+ /* for SLAVE mode with SRM */
+ { 12000000, 2822400, 0xED, 0xBF, 0x5, 0}, /* MCLK=12Mhz */
+ { 13000000, 2822400, 0xE4, 0x13, 0x0, 0}, /* MCLK=13Mhz */
+ { 13500000, 2822400, 0xDF, 0xC6, 0x8, 0}, /* MCLK=13.5Mhz */
+ { 14400000, 2822400, 0xD8, 0xCA, 0x1, 0}, /* MCLK=14.4Mhz */
+ { 19200000, 2822400, 0xBE, 0x97, 0x9, 0}, /* MCLK=19.2Mhz */
+ { 19680000, 2822400, 0xBC, 0xAC, 0xD, 0}, /* MCLK=19.68Mhz */
+ { 19800000, 2822400, 0xBC, 0x35, 0xE, 0}, /* MCLK=19.8Mhz */
+};
+
+enum clk_src {
+ DA7210_CLKSRC_MCLK
+};
+
#define DA7210_VERSION "0.0.1"
/*
@@ -628,9 +676,12 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = {
/* Codec private data */
struct da7210_priv {
struct regmap *regmap;
+ unsigned int mclk_rate;
+ int master;
};
static struct reg_default da7210_reg_defaults[] = {
+ { 0x00, 0x00 },
{ 0x01, 0x11 },
{ 0x03, 0x00 },
{ 0x04, 0x00 },
@@ -713,10 +764,10 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
u32 dai_cfg1;
- u32 fs, bypass;
+ u32 fs, sysclk;
/* set DAI source to Left and Right ADC */
snd_soc_write(codec, DA7210_DAI_SRC_SEL,
@@ -749,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
switch (params_rate(params)) {
case 8000:
fs = DA7210_PLL_FS_8000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 11025:
fs = DA7210_PLL_FS_11025;
- bypass = 0;
+ sysclk = 2822400;
break;
case 12000:
fs = DA7210_PLL_FS_12000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 16000:
fs = DA7210_PLL_FS_16000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 22050:
fs = DA7210_PLL_FS_22050;
- bypass = 0;
+ sysclk = 2822400;
break;
case 32000:
fs = DA7210_PLL_FS_32000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 44100:
fs = DA7210_PLL_FS_44100;
- bypass = 0;
+ sysclk = 2822400;
break;
case 48000:
fs = DA7210_PLL_FS_48000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
case 88200:
fs = DA7210_PLL_FS_88200;
- bypass = 0;
+ sysclk = 2822400;
break;
case 96000:
fs = DA7210_PLL_FS_96000;
- bypass = DA7210_PLL_BYP;
+ sysclk = 3072000;
break;
default:
return -EINVAL;
@@ -795,8 +846,26 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
- snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);
+ if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) {
+ /* PLL mode, disable PLL bypass */
+ snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0);
+
+ if (!da7210->master) {
+ /* PLL slave mode, also enable SRM */
+ snd_soc_update_bits(codec, DA7210_PLL,
+ (DA7210_MCLK_SRM_EN |
+ DA7210_MCLK_DET_EN),
+ (DA7210_MCLK_SRM_EN |
+ DA7210_MCLK_DET_EN));
+ }
+ } else {
+ /* PLL bypass mode, enable PLL bypass and Auto Detection */
+ snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN,
+ DA7210_MCLK_DET_EN);
+ snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP,
+ DA7210_PLL_BYP);
+ }
/* Enable active mode */
snd_soc_update_bits(codec, DA7210_STARTUP1,
DA7210_SC_MST_EN, DA7210_SC_MST_EN);
@@ -810,17 +879,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
u32 dai_cfg1;
u32 dai_cfg3;
dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1);
dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3);
+ if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) &&
+ (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
+ return -EINVAL;
+
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
+ da7210->master = 1;
dai_cfg1 |= DA7210_DAI_MODE_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
+ da7210->master = 0;
dai_cfg1 |= DA7210_DAI_MODE_SLAVE;
break;
default:
@@ -872,10 +948,101 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute)
#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+static int da7210_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 da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case DA7210_CLKSRC_MCLK:
+ switch (freq) {
+ case 12000000:
+ case 13000000:
+ case 13500000:
+ case 14400000:
+ case 19200000:
+ case 19680000:
+ case 19800000:
+ da7210->mclk_rate = freq;
+ return 0;
+ default:
+ dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+ freq);
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
+ return -EINVAL;
+ }
+}
+
+/**
+ * da7210_set_dai_pll :Configure the codec PLL
+ * @param codec_dai : pointer to codec DAI
+ * @param pll_id : da7210 has only one pll, so pll_id is always zero
+ * @param fref : MCLK frequency, should be < 20MHz
+ * @param fout : FsDM value, Refer page 44 & 45 of datasheet
+ * @return int : Zero for success, negative error code for error
+ *
+ * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz,
+ * 19.2MHz, 19.6MHz and 19.8MHz
+ */
+static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int fref, unsigned int fout)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+
+ u8 pll_div1, pll_div2, pll_div3, cnt;
+
+ /* In slave mode, there is only one set of divisors */
+ if (!da7210->master)
+ fout = 2822400;
+
+ /* Search pll div array for correct divisors */
+ for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) {
+ /* check fref, mode and fout */
+ if ((fref == da7210_pll_div[cnt].fref) &&
+ (da7210->master == da7210_pll_div[cnt].mode) &&
+ (fout == da7210_pll_div[cnt].fout)) {
+ /* all match, pick up divisors */
+ pll_div1 = da7210_pll_div[cnt].div1;
+ pll_div2 = da7210_pll_div[cnt].div2;
+ pll_div3 = da7210_pll_div[cnt].div3;
+ break;
+ }
+ }
+ if (cnt >= ARRAY_SIZE(da7210_pll_div))
+ goto err;
+
+ /* Disable active mode */
+ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
+ /* Write PLL dividers */
+ snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1);
+ snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2);
+ snd_soc_update_bits(codec, DA7210_PLL_DIV3,
+ DA7210_PLL_DIV_L_MASK, pll_div3);
+
+ /* Enable PLL */
+ snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
+
+ /* Enable active mode */
+ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN,
+ DA7210_SC_MST_EN);
+ return 0;
+err:
+ dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref);
+ return -EINVAL;
+}
+
/* DAI operations */
static const struct snd_soc_dai_ops da7210_dai_ops = {
.hw_params = da7210_hw_params,
.set_fmt = da7210_set_dai_fmt,
+ .set_sysclk = da7210_set_dai_sysclk,
+ .set_pll = da7210_set_dai_pll,
.digital_mute = da7210_mute,
};
@@ -915,24 +1082,11 @@ static int da7210_probe(struct snd_soc_codec *codec)
return ret;
}
- /* FIXME
- *
- * This driver use fixed value here
- * And below settings expects MCLK = 12.288MHz
- *
- * When you select different MCLK, please check...
- * DA7210_PLL_DIV1 val
- * DA7210_PLL_DIV2 val
- * DA7210_PLL_DIV3 val
- * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx
- */
+ da7210->mclk_rate = 0; /* This will be set from set_sysclk() */
+ da7210->master = 0; /* This will be set from set_fmt() */
- /*
- * make sure that DA7210 use bypass mode before start up
- */
- snd_soc_write(codec, DA7210_STARTUP1, 0);
- snd_soc_write(codec, DA7210_PLL_DIV3,
- DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
+ /* Enable internal regulator & bias current */
+ snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN);
/*
* ADC settings
@@ -1007,34 +1161,13 @@ static int da7210_probe(struct snd_soc_codec *codec)
/* Enable Aux2 */
snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN);
+ /* Set PLL Master clock range 10-20 MHz, enable PLL bypass */
+ snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ |
+ DA7210_PLL_BYP);
+
/* Diable PLL and bypass it */
snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
- /*
- * If 48kHz sound came, it use bypass mode,
- * and when it is 44.1kHz, it use PLL.
- *
- * This time, this driver sets PLL always ON
- * and controls bypass/PLL mode by switching
- * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit.
- * see da7210_hw_params
- */
- snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
- snd_soc_write(codec, DA7210_PLL_DIV2, 0x99);
- snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A |
- DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
- snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
-
- /* As suggested by Dialog */
- /* unlock */
- regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B);
- regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4);
- regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01);
- regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C);
- /* re-lock */
- regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00);
- regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00);
-
/* Activate all enabled subsystem */
snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
@@ -1055,7 +1188,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
.num_dapm_routes = ARRAY_SIZE(da7210_audio_map),
};
-static struct regmap_config da7210_regmap = {
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static struct reg_default da7210_regmap_i2c_patch[] = {
+
+ /* System controller master disable */
+ { DA7210_STARTUP1, 0x00 },
+ /* Set PLL Master clock range 10-20 MHz */
+ { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ },
+
+ /* to unlock */
+ { DA7210_A_HID_UNLOCK, 0x8B},
+ { DA7210_A_TEST_UNLOCK, 0xB4},
+ { DA7210_A_PLL1, 0x01},
+ { DA7210_A_CP_MODE, 0x7C},
+ /* to re-lock */
+ { DA7210_A_HID_UNLOCK, 0x00},
+ { DA7210_A_TEST_UNLOCK, 0x00},
+};
+
+static const struct regmap_config da7210_regmap_config_i2c = {
.reg_bits = 8,
.val_bits = 8,
@@ -1066,7 +1218,6 @@ static struct regmap_config da7210_regmap = {
.cache_type = REGCACHE_RBTREE,
};
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1080,13 +1231,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, da7210);
- da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap);
+ da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c);
if (IS_ERR(da7210->regmap)) {
ret = PTR_ERR(da7210->regmap);
dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
return ret;
}
+ ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch,
+ ARRAY_SIZE(da7210_regmap_i2c_patch));
+ if (ret != 0)
+ dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_da7210, &da7210_dai, 1);
if (ret < 0) {
@@ -1119,7 +1275,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id);
/* I2C codec control layer */
static struct i2c_driver da7210_i2c_driver = {
.driver = {
- .name = "da7210-codec",
+ .name = "da7210",
.owner = THIS_MODULE,
},
.probe = da7210_i2c_probe,
@@ -1128,12 +1284,112 @@ static struct i2c_driver da7210_i2c_driver = {
};
#endif
+#if defined(CONFIG_SPI_MASTER)
+
+static struct reg_default da7210_regmap_spi_patch[] = {
+ /* Dummy read to give two pulses over nCS for SPI */
+ { DA7210_AUX2, 0x00 },
+ { DA7210_AUX2, 0x00 },
+
+ /* System controller master disable */
+ { DA7210_STARTUP1, 0x00 },
+ /* Set PLL Master clock range 10-20 MHz */
+ { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ },
+
+ /* to set PAGE1 of SPI register space */
+ { DA7210_PAGE_CONTROL, 0x80 },
+ /* to unlock */
+ { DA7210_A_HID_UNLOCK, 0x8B},
+ { DA7210_A_TEST_UNLOCK, 0xB4},
+ { DA7210_A_PLL1, 0x01},
+ { DA7210_A_CP_MODE, 0x7C},
+ /* to re-lock */
+ { DA7210_A_HID_UNLOCK, 0x00},
+ { DA7210_A_TEST_UNLOCK, 0x00},
+ /* to set back PAGE0 of SPI register space */
+ { DA7210_PAGE_CONTROL, 0x00 },
+};
+
+static const struct regmap_config da7210_regmap_config_spi = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .read_flag_mask = 0x01,
+ .write_flag_mask = 0x00,
+
+ .reg_defaults = da7210_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults),
+ .volatile_reg = da7210_volatile_register,
+ .readable_reg = da7210_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit da7210_spi_probe(struct spi_device *spi)
+{
+ struct da7210_priv *da7210;
+ int ret;
+
+ da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv),
+ GFP_KERNEL);
+ if (!da7210)
+ return -ENOMEM;
+
+ spi_set_drvdata(spi, da7210);
+ da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi);
+ if (IS_ERR(da7210->regmap)) {
+ ret = PTR_ERR(da7210->regmap);
+ dev_err(&spi->dev, "Failed to register regmap: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch,
+ ARRAY_SIZE(da7210_regmap_spi_patch));
+ if (ret != 0)
+ dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret);
+
+ ret = snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_da7210, &da7210_dai, 1);
+ if (ret < 0)
+ goto err_regmap;
+
+ return ret;
+
+err_regmap:
+ regmap_exit(da7210->regmap);
+
+ return ret;
+}
+
+static int __devexit da7210_spi_remove(struct spi_device *spi)
+{
+ struct da7210_priv *da7210 = spi_get_drvdata(spi);
+ snd_soc_unregister_codec(&spi->dev);
+ regmap_exit(da7210->regmap);
+ return 0;
+}
+
+static struct spi_driver da7210_spi_driver = {
+ .driver = {
+ .name = "da7210",
+ .owner = THIS_MODULE,
+ },
+ .probe = da7210_spi_probe,
+ .remove = __devexit_p(da7210_spi_remove)
+};
+#endif
+
static int __init da7210_modinit(void)
{
int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&da7210_i2c_driver);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&da7210_spi_driver);
+ if (ret) {
+ printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n",
+ ret);
+ }
+#endif
return ret;
}
module_init(da7210_modinit);
@@ -1143,6 +1399,9 @@ static void __exit da7210_exit(void)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&da7210_i2c_driver);
#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&da7210_spi_driver);
+#endif
}
module_exit(da7210_exit);
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 4624e752a18..85d9cabe6d5 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
uint32_t val;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec =rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
switch (params_rate(params)) {
case 8000:
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
new file mode 100644
index 00000000000..802b9f176b1
--- /dev/null
+++ b/sound/soc/codecs/lm49453.c
@@ -0,0 +1,1550 @@
+/*
+ * lm49453.c - LM49453 ALSA Soc Audio driver
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * 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.
+ *
+ * Initially based on sound/soc/codecs/wm8350.c
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+#include "lm49453.h"
+
+static struct reg_default lm49453_reg_defs[] = {
+ { 0, 0x00 },
+ { 1, 0x00 },
+ { 2, 0x00 },
+ { 3, 0x00 },
+ { 4, 0x00 },
+ { 5, 0x00 },
+ { 6, 0x00 },
+ { 7, 0x00 },
+ { 8, 0x00 },
+ { 9, 0x00 },
+ { 10, 0x00 },
+ { 11, 0x00 },
+ { 12, 0x00 },
+ { 13, 0x00 },
+ { 14, 0x00 },
+ { 15, 0x00 },
+ { 16, 0x00 },
+ { 17, 0x00 },
+ { 18, 0x00 },
+ { 19, 0x00 },
+ { 20, 0x00 },
+ { 21, 0x00 },
+ { 22, 0x00 },
+ { 23, 0x00 },
+ { 32, 0x00 },
+ { 33, 0x00 },
+ { 35, 0x00 },
+ { 36, 0x00 },
+ { 37, 0x00 },
+ { 46, 0x00 },
+ { 48, 0x00 },
+ { 49, 0x00 },
+ { 51, 0x00 },
+ { 56, 0x00 },
+ { 58, 0x00 },
+ { 59, 0x00 },
+ { 60, 0x00 },
+ { 61, 0x00 },
+ { 62, 0x00 },
+ { 63, 0x00 },
+ { 64, 0x00 },
+ { 65, 0x00 },
+ { 66, 0x00 },
+ { 67, 0x00 },
+ { 68, 0x00 },
+ { 69, 0x00 },
+ { 70, 0x00 },
+ { 71, 0x00 },
+ { 72, 0x00 },
+ { 73, 0x00 },
+ { 74, 0x00 },
+ { 75, 0x00 },
+ { 76, 0x00 },
+ { 77, 0x00 },
+ { 78, 0x00 },
+ { 79, 0x00 },
+ { 80, 0x00 },
+ { 81, 0x00 },
+ { 82, 0x00 },
+ { 83, 0x00 },
+ { 85, 0x00 },
+ { 85, 0x00 },
+ { 86, 0x00 },
+ { 87, 0x00 },
+ { 88, 0x00 },
+ { 89, 0x00 },
+ { 90, 0x00 },
+ { 91, 0x00 },
+ { 92, 0x00 },
+ { 93, 0x00 },
+ { 94, 0x00 },
+ { 95, 0x00 },
+ { 96, 0x01 },
+ { 97, 0x00 },
+ { 98, 0x00 },
+ { 99, 0x00 },
+ { 100, 0x00 },
+ { 101, 0x00 },
+ { 102, 0x00 },
+ { 103, 0x01 },
+ { 105, 0x01 },
+ { 106, 0x00 },
+ { 107, 0x01 },
+ { 107, 0x00 },
+ { 108, 0x00 },
+ { 109, 0x00 },
+ { 110, 0x00 },
+ { 111, 0x02 },
+ { 112, 0x02 },
+ { 113, 0x00 },
+ { 121, 0x80 },
+ { 122, 0xBB },
+ { 123, 0x80 },
+ { 124, 0xBB },
+ { 128, 0x00 },
+ { 130, 0x00 },
+ { 131, 0x00 },
+ { 132, 0x00 },
+ { 133, 0x0A },
+ { 134, 0x0A },
+ { 135, 0x0A },
+ { 136, 0x0F },
+ { 137, 0x00 },
+ { 138, 0x73 },
+ { 139, 0x33 },
+ { 140, 0x73 },
+ { 141, 0x33 },
+ { 142, 0x73 },
+ { 143, 0x33 },
+ { 144, 0x73 },
+ { 145, 0x33 },
+ { 146, 0x73 },
+ { 147, 0x33 },
+ { 148, 0x73 },
+ { 149, 0x33 },
+ { 150, 0x73 },
+ { 151, 0x33 },
+ { 152, 0x00 },
+ { 153, 0x00 },
+ { 154, 0x00 },
+ { 155, 0x00 },
+ { 176, 0x00 },
+ { 177, 0x00 },
+ { 178, 0x00 },
+ { 179, 0x00 },
+ { 180, 0x00 },
+ { 181, 0x00 },
+ { 182, 0x00 },
+ { 183, 0x00 },
+ { 184, 0x00 },
+ { 185, 0x00 },
+ { 186, 0x00 },
+ { 189, 0x00 },
+ { 188, 0x00 },
+ { 194, 0x00 },
+ { 195, 0x00 },
+ { 196, 0x00 },
+ { 197, 0x00 },
+ { 200, 0x00 },
+ { 201, 0x00 },
+ { 202, 0x00 },
+ { 203, 0x00 },
+ { 204, 0x00 },
+ { 205, 0x00 },
+ { 208, 0x00 },
+ { 209, 0x00 },
+ { 210, 0x00 },
+ { 211, 0x00 },
+ { 213, 0x00 },
+ { 214, 0x00 },
+ { 215, 0x00 },
+ { 216, 0x00 },
+ { 217, 0x00 },
+ { 218, 0x00 },
+ { 219, 0x00 },
+ { 221, 0x00 },
+ { 222, 0x00 },
+ { 224, 0x00 },
+ { 225, 0x00 },
+ { 226, 0x00 },
+ { 227, 0x00 },
+ { 228, 0x00 },
+ { 229, 0x00 },
+ { 230, 0x13 },
+ { 231, 0x00 },
+ { 232, 0x80 },
+ { 233, 0x0C },
+ { 234, 0xDD },
+ { 235, 0x00 },
+ { 236, 0x04 },
+ { 237, 0x00 },
+ { 238, 0x00 },
+ { 239, 0x00 },
+ { 240, 0x00 },
+ { 241, 0x00 },
+ { 242, 0x00 },
+ { 243, 0x00 },
+ { 244, 0x00 },
+ { 245, 0x00 },
+ { 248, 0x00 },
+ { 249, 0x00 },
+ { 254, 0x00 },
+ { 255, 0x00 },
+};
+
+/* codec private data */
+struct lm49453_priv {
+ struct regmap *regmap;
+ int fs_rate;
+};
+
+/* capture path controls */
+
+static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"};
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5,
+ lm49453_mic2mode_text);
+
+static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"};
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum,
+ LM49453_P0_DIGITAL_MIC1_CONFIG_REG,
+ 7, lm49453_dmic_cfg_text);
+
+static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum,
+ LM49453_P0_DIGITAL_MIC2_CONFIG_REG,
+ 7, lm49453_dmic_cfg_text);
+
+/* MUX Controls */
+static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" };
+
+static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" };
+
+static const struct soc_enum lm49453_adcl_enum =
+ SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0,
+ ARRAY_SIZE(lm49453_adcl_mux_text),
+ lm49453_adcl_mux_text);
+
+static const struct soc_enum lm49453_adcr_enum =
+ SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1,
+ ARRAY_SIZE(lm49453_adcr_mux_text),
+ lm49453_adcr_mux_text);
+
+static const struct snd_kcontrol_new lm49453_adcl_mux_control =
+ SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum);
+
+static const struct snd_kcontrol_new lm49453_adcr_mux_control =
+ SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum);
+
+static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = {
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = {
+SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0),
+SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
+};
+
+/* TLV Declarations */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
+/* Sidetone supports mono only */
+SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
+ 0, 0x3F, 0, digital_tlv),
+SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
+ 0, 0x3F, 0, digital_tlv),
+};
+
+static const struct snd_kcontrol_new lm49453_snd_controls[] = {
+ /* mic1 and mic2 supports mono only */
+ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
+ 0, digital_tlv),
+ SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
+ 0, digital_tlv),
+
+ SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
+ LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
+ LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+
+ SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
+ SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
+ SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum),
+
+ /* Capture path filter enable */
+ SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+ 0, 1, 0),
+ SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+ 1, 1, 0),
+ SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG,
+ 2, 1, 0),
+
+ SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
+ LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
+ LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
+ LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+ SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
+ LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+
+ SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
+ 0, 6, 0, digital_tlv),
+
+ SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 0, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 2, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 4, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
+ 6, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 0, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 2, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 4, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG,
+ 6, 3, 0, port_tlv),
+
+ SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG,
+ 0, 3, 0, port_tlv),
+ SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG,
+ 2, 3, 0, port_tlv),
+
+ SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG,
+ 1, 1, 0),
+ SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG,
+ 1, 1, 0),
+ SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG,
+ 2, 1, 0),
+ SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG,
+ 2, 1, 0)
+
+};
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = {
+
+ /* All end points HP,EP, LS, Lineout and Haptic */
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+ SND_SOC_DAPM_OUTPUT("EPOUT"),
+ SND_SOC_DAPM_OUTPUT("LSOUTL"),
+ SND_SOC_DAPM_OUTPUT("LSOUTR"),
+ SND_SOC_DAPM_OUTPUT("LOOUTR"),
+ SND_SOC_DAPM_OUTPUT("LOOUTL"),
+ SND_SOC_DAPM_OUTPUT("HAOUTL"),
+ SND_SOC_DAPM_OUTPUT("HAOUTR"),
+
+ SND_SOC_DAPM_INPUT("AMIC1"),
+ SND_SOC_DAPM_INPUT("AMIC2"),
+ SND_SOC_DAPM_INPUT("DMIC1DAT"),
+ SND_SOC_DAPM_INPUT("DMIC2DAT"),
+ SND_SOC_DAPM_INPUT("AUXL"),
+ SND_SOC_DAPM_INPUT("AUXR"),
+
+ SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0),
+
+ /* playback path driver enables */
+ SND_SOC_DAPM_OUT_DRV("Headset Switch",
+ LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Earpiece Switch",
+ LM49453_P0_EP_REG, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Left Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Right Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Haptic Left Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Haptic Right Switch",
+ LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0),
+
+
+ SND_SOC_DAPM_PGA("AUXL Input",
+ LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUXR Input",
+ LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0),
+
+ SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0),
+ SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0,
+ &lm49453_adcl_mux_control),
+ SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0,
+ &lm49453_adcr_mux_control),
+
+ SND_SOC_DAPM_MUX("Mic1 Input",
+ SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control),
+
+ SND_SOC_DAPM_MUX("Mic2 Input",
+ SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control),
+
+ /* AIF */
+ SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 2, 0),
+ SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 6, 0),
+
+ SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 3, 0),
+ SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0,
+ LM49453_P0_PULL_CONFIG1_REG, 7, 0),
+
+ /* Port1 TX controls */
+ SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Port2 TX controls */
+ SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Sidetone Mixer */
+ SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_sidetone_mixer_controls,
+ ARRAY_SIZE(lm49453_sidetone_mixer_controls)),
+
+ /* DAC MIXERS */
+ SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_headset_left_mixer,
+ ARRAY_SIZE(lm49453_headset_left_mixer)),
+ SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_headset_right_mixer,
+ ARRAY_SIZE(lm49453_headset_right_mixer)),
+ SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_lineout_left_mixer,
+ ARRAY_SIZE(lm49453_lineout_left_mixer)),
+ SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_lineout_right_mixer,
+ ARRAY_SIZE(lm49453_lineout_right_mixer)),
+ SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_speaker_left_mixer,
+ ARRAY_SIZE(lm49453_speaker_left_mixer)),
+ SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_speaker_right_mixer,
+ ARRAY_SIZE(lm49453_speaker_right_mixer)),
+ SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_haptic_left_mixer,
+ ARRAY_SIZE(lm49453_haptic_left_mixer)),
+ SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_haptic_right_mixer,
+ ARRAY_SIZE(lm49453_haptic_right_mixer)),
+
+ /* Capture Mixer */
+ SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx1_mixer,
+ ARRAY_SIZE(lm49453_port1_tx1_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx2_mixer,
+ ARRAY_SIZE(lm49453_port1_tx2_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx3_mixer,
+ ARRAY_SIZE(lm49453_port1_tx3_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx4_mixer,
+ ARRAY_SIZE(lm49453_port1_tx4_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx5_mixer,
+ ARRAY_SIZE(lm49453_port1_tx5_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx6_mixer,
+ ARRAY_SIZE(lm49453_port1_tx6_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx7_mixer,
+ ARRAY_SIZE(lm49453_port1_tx7_mixer)),
+ SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port1_tx8_mixer,
+ ARRAY_SIZE(lm49453_port1_tx8_mixer)),
+
+ SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port2_tx1_mixer,
+ ARRAY_SIZE(lm49453_port2_tx1_mixer)),
+ SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0,
+ lm49453_port2_tx2_mixer,
+ ARRAY_SIZE(lm49453_port2_tx2_mixer)),
+};
+
+static const struct snd_soc_dapm_route lm49453_audio_map[] = {
+ /* Port SDI mapping */
+ { "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" },
+ { "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" },
+
+ { "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" },
+ { "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" },
+
+ /* HP mapping */
+ { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HPL Mixer", "ADCL Switch", "ADC Left" },
+ { "HPL Mixer", "ADCR Switch", "ADC Right" },
+ { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "HPL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "HPL DAC", NULL, "HPL Mixer" },
+
+ { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HPR Mixer", "ADCL Switch", "ADC Left" },
+ { "HPR Mixer", "ADCR Switch", "ADC Right" },
+ { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" },
+ { "HPR Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "HPR DAC", NULL, "HPR Mixer" },
+
+ { "HPOUTL", "Headset Switch", "HPL DAC"},
+ { "HPOUTR", "Headset Switch", "HPR DAC"},
+
+ /* EP map */
+ { "EPOUT", "Earpiece Switch", "HPL DAC" },
+
+ /* Speaker map */
+ { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LSL Mixer", "ADCL Switch", "ADC Left" },
+ { "LSL Mixer", "ADCR Switch", "ADC Right" },
+ { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LSL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LSL DAC", NULL, "LSL Mixer" },
+
+ { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LSR Mixer", "ADCL Switch", "ADC Left" },
+ { "LSR Mixer", "ADCR Switch", "ADC Right" },
+ { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LSR Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LSR DAC", NULL, "LSR Mixer" },
+
+ { "LSOUTL", "Speaker Left Switch", "LSL DAC"},
+ { "LSOUTR", "Speaker Left Switch", "LSR DAC"},
+
+ /* Haptic map */
+ { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HAL Mixer", "ADCL Switch", "ADC Left" },
+ { "HAL Mixer", "ADCR Switch", "ADC Right" },
+ { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "HAL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "HAL DAC", NULL, "HAL Mixer" },
+
+ { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "HAR Mixer", "ADCL Switch", "ADC Left" },
+ { "HAR Mixer", "ADCR Switch", "ADC Right" },
+ { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "HAR Mixer", "Sideton Switch", "Sidetone" },
+
+ { "HAR DAC", NULL, "HAR Mixer" },
+
+ { "HAOUTL", "Haptic Left Switch", "HAL DAC" },
+ { "HAOUTR", "Haptic Right Switch", "HAR DAC" },
+
+ /* Lineout map */
+ { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LOL Mixer", "ADCL Switch", "ADC Left" },
+ { "LOL Mixer", "ADCR Switch", "ADC Right" },
+ { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LOL Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LOL DAC", NULL, "LOL Mixer" },
+
+ { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" },
+ { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" },
+ { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" },
+ { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" },
+ { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" },
+ { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" },
+ { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" },
+ { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" },
+
+ /* Port 2 */
+ { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" },
+ { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" },
+
+ { "LOR Mixer", "ADCL Switch", "ADC Left" },
+ { "LOR Mixer", "ADCR Switch", "ADC Right" },
+ { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" },
+ { "LOR Mixer", "Sidetone Switch", "Sidetone" },
+
+ { "LOR DAC", NULL, "LOR Mixer" },
+
+ { "LOOUTL", NULL, "LOL DAC" },
+ { "LOOUTR", NULL, "LOR DAC" },
+
+ /* TX map */
+ /* Port1 mappings */
+ { "Port1_1 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_1 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_2 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_2 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_3 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_3 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_4 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_4 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_5 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_5 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_6 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_6 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_7 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_7 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port1_8 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port1_8 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port2_1 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port2_1 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "Port2_2 Mixer", "ADCL Switch", "ADC Left" },
+ { "Port2_2 Mixer", "ADCR Switch", "ADC Right" },
+ { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" },
+ { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" },
+ { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" },
+ { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" },
+
+ { "P1_1_TX", NULL, "Port1_1 Mixer" },
+ { "P1_2_TX", NULL, "Port1_2 Mixer" },
+ { "P1_3_TX", NULL, "Port1_3 Mixer" },
+ { "P1_4_TX", NULL, "Port1_4 Mixer" },
+ { "P1_5_TX", NULL, "Port1_5 Mixer" },
+ { "P1_6_TX", NULL, "Port1_6 Mixer" },
+ { "P1_7_TX", NULL, "Port1_7 Mixer" },
+ { "P1_8_TX", NULL, "Port1_8 Mixer" },
+
+ { "P2_1_TX", NULL, "Port2_1 Mixer" },
+ { "P2_2_TX", NULL, "Port2_2 Mixer" },
+
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"},
+ { "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"},
+
+ { "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"},
+ { "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"},
+
+ { "Mic1 Input", NULL, "AMIC1" },
+ { "Mic2 Input", NULL, "AMIC2" },
+
+ { "AUXL Input", NULL, "AUXL" },
+ { "AUXR Input", NULL, "AUXR" },
+
+ /* AUX connections */
+ { "ADCL Mux", "Aux_L", "AUXL Input" },
+ { "ADCL Mux", "MIC1", "Mic1 Input" },
+
+ { "ADCR Mux", "Aux_R", "AUXR Input" },
+ { "ADCR Mux", "MIC2", "Mic2 Input" },
+
+ /* ADC connection */
+ { "ADC Left", NULL, "ADCL Mux"},
+ { "ADC Right", NULL, "ADCR Mux"},
+
+ { "DMIC1 Left", NULL, "DMIC1DAT"},
+ { "DMIC1 Right", NULL, "DMIC1DAT"},
+ { "DMIC2 Left", NULL, "DMIC2DAT"},
+ { "DMIC2 Right", NULL, "DMIC2DAT"},
+
+ /* Sidetone map */
+ { "Sidetone Mixer", NULL, "ADC Left" },
+ { "Sidetone Mixer", NULL, "ADC Right" },
+ { "Sidetone Mixer", NULL, "DMIC1 Left" },
+ { "Sidetone Mixer", NULL, "DMIC1 Right" },
+ { "Sidetone Mixer", NULL, "DMIC2 Left" },
+ { "Sidetone Mixer", NULL, "DMIC2 Right" },
+
+ { "Sidetone", "Sidetone Switch", "Sidetone Mixer" },
+};
+
+static int lm49453_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 lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+ u16 clk_div = 0;
+
+ lm49453->fs_rate = params_rate(params);
+
+ /* Setting DAC clock dividers based on substream sample rate. */
+ switch (lm49453->fs_rate) {
+ case 8000:
+ case 16000:
+ case 32000:
+ case 24000:
+ case 48000:
+ clk_div = 256;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk_div = 216;
+ break;
+ case 96000:
+ clk_div = 127;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div);
+ snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div);
+
+ return 0;
+}
+
+static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ u16 aif_val;
+ int mode = 0;
+ int clk_phase = 0;
+ int clk_shift = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aif_val = 0;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS |
+ LM49453_AUDIO_PORT1_BASIC_SYNC_MS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ mode = 1;
+ clk_phase = (1 << 5);
+ clk_shift = 1;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ mode = 1;
+ clk_phase = (1 << 5);
+ clk_shift = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
+ LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+ (aif_val | mode | clk_phase));
+
+ snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
+
+ return 0;
+}
+
+static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 pll_clk = 0;
+
+ switch (freq) {
+ case 12288000:
+ case 26000000:
+ case 19200000:
+ /* pll clk slection */
+ pll_clk = 0;
+ break;
+ case 48000:
+ case 32576:
+ /* fll clk slection */
+ pll_clk = BIT(4);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk);
+
+ return 0;
+}
+
+static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0),
+ (mute ? (BIT(1)|BIT(0)) : 0));
+ return 0;
+}
+
+static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2),
+ (mute ? (BIT(3)|BIT(2)) : 0));
+ return 0;
+}
+
+static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4),
+ (mute ? (BIT(5)|BIT(4)) : 0));
+ return 0;
+}
+
+static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4),
+ (mute ? BIT(4) : 0));
+ return 0;
+}
+
+static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute)
+{
+ snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6),
+ (mute ? (BIT(7)|BIT(6)) : 0));
+ return 0;
+}
+
+static int lm49453_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ regcache_sync(lm49453->regmap);
+
+ snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
+ LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
+ LM49453_PMC_SETUP_CHIP_EN, 0);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+/* Formates supported by LM49453 driver. */
+#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops lm49453_headset_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_hp_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_speaker_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_ls_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_haptic_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_ha_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_ep_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_ep_mute,
+};
+
+static struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
+ .hw_params = lm49453_hw_params,
+ .set_sysclk = lm49453_set_dai_sysclk,
+ .set_fmt = lm49453_set_dai_fmt,
+ .digital_mute = lm49453_lo_mute,
+};
+
+/* LM49453 dai structure. */
+static const struct snd_soc_dai_driver lm49453_dai[] = {
+ {
+ .name = "LM49453 Headset",
+ .playback = {
+ .stream_name = "Headset",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 5,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_headset_dai_ops,
+ .symmetric_rates = 1,
+ },
+ {
+ .name = "LM49453 Speaker",
+ .playback = {
+ .stream_name = "Speaker",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_speaker_dai_ops,
+ },
+ {
+ .name = "LM49453 Haptic",
+ .playback = {
+ .stream_name = "Haptic",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_haptic_dai_ops,
+ },
+ {
+ .name = "LM49453 Earpiece",
+ .playback = {
+ .stream_name = "Earpiece",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_ep_dai_ops,
+ },
+ {
+ .name = "LM49453 line out",
+ .playback = {
+ .stream_name = "Lineout",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = LM49453_FORMATS,
+ },
+ .ops = &lm49453_lineout_dai_ops,
+ },
+};
+
+static int lm49453_suspend(struct snd_soc_codec *codec)
+{
+ lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int lm49453_resume(struct snd_soc_codec *codec)
+{
+ lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+static int lm49453_probe(struct snd_soc_codec *codec)
+{
+ struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ codec->control_data = lm49453->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+/* power down chip */
+static int lm49453_remove(struct snd_soc_codec *codec)
+{
+ lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
+ .probe = lm49453_probe,
+ .remove = lm49453_remove,
+ .suspend = lm49453_suspend,
+ .resume = lm49453_resume,
+ .set_bias_level = lm49453_set_bias_level,
+ .controls = lm49453_snd_controls,
+ .num_controls = ARRAY_SIZE(lm49453_snd_controls),
+ .dapm_widgets = lm49453_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets),
+ .dapm_routes = lm49453_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map),
+ .idle_bias_off = true,
+};
+
+static const struct regmap_config lm49453_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = LM49453_MAX_REGISTER,
+ .reg_defaults = lm49453_reg_defs,
+ .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static __devinit int lm49453_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct lm49453_priv *lm49453;
+ int ret = 0;
+
+ lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv),
+ GFP_KERNEL);
+
+ if (lm49453 == NULL)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, lm49453);
+
+ lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config);
+ if (IS_ERR(lm49453->regmap)) {
+ ret = PTR_ERR(lm49453->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_lm49453,
+ lm49453_dai, ARRAY_SIZE(lm49453_dai));
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ regmap_exit(lm49453->regmap);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int __devexit lm49453_i2c_remove(struct i2c_client *client)
+{
+ struct lm49453_priv *lm49453 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+ regmap_exit(lm49453->regmap);
+ return 0;
+}
+
+static const struct i2c_device_id lm49453_i2c_id[] = {
+ { "lm49453", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id);
+
+static struct i2c_driver lm49453_i2c_driver = {
+ .driver = {
+ .name = "lm49453",
+ .owner = THIS_MODULE,
+ },
+ .probe = lm49453_i2c_probe,
+ .remove = __devexit_p(lm49453_i2c_remove),
+ .id_table = lm49453_i2c_id,
+};
+
+module_i2c_driver(lm49453_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC LM49453 driver");
+MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h
new file mode 100644
index 00000000000..a63cfa5c088
--- /dev/null
+++ b/sound/soc/codecs/lm49453.h
@@ -0,0 +1,380 @@
+/*
+ * lm49453.h - LM49453 ALSA Soc Audio drive
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * 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.
+ *
+ */
+
+#ifndef _LM49453_H
+#define _LM49453_H
+
+#include <linux/bitops.h>
+
+/* LM49453_P0 register space for page0 */
+#define LM49453_P0_PMC_SETUP_REG 0x00
+#define LM49453_P0_PLL_CLK_SEL1_REG 0x01
+#define LM49453_P0_PLL_CLK_SEL2_REG 0x02
+#define LM49453_P0_PMC_CLK_DIV_REG 0x03
+#define LM49453_P0_HSDET_CLK_DIV_REG 0x04
+#define LM49453_P0_DMIC_CLK_DIV_REG 0x05
+#define LM49453_P0_ADC_CLK_DIV_REG 0x06
+#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07
+#define LM49453_P0_PLL_HF_M_REG 0x08
+#define LM49453_P0_PLL_LF_M_REG 0x09
+#define LM49453_P0_PLL_NL_REG 0x0A
+#define LM49453_P0_PLL_N_MODL_REG 0x0B
+#define LM49453_P0_PLL_N_MODH_REG 0x0C
+#define LM49453_P0_PLL_P1_REG 0x0D
+#define LM49453_P0_PLL_P2_REG 0x0E
+#define LM49453_P0_FLL_REF_FREQL_REG 0x0F
+#define LM49453_P0_FLL_REF_FREQH_REG 0x10
+#define LM49453_P0_VCO_TARGETLL_REG 0x11
+#define LM49453_P0_VCO_TARGETLH_REG 0x12
+#define LM49453_P0_VCO_TARGETHL_REG 0x13
+#define LM49453_P0_VCO_TARGETHH_REG 0x14
+#define LM49453_P0_PLL_CONFIG_REG 0x15
+#define LM49453_P0_DAC_CLK_SEL_REG 0x16
+#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17
+
+/* Analog Mixer Input Stages */
+#define LM49453_P0_MICL_REG 0x20
+#define LM49453_P0_MICR_REG 0x21
+#define LM49453_P0_EP_REG 0x24
+#define LM49453_P0_DIS_PKVL_FB_REG 0x25
+
+/* Analog Mixer Output Stages */
+#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E
+
+/*ADC or DAC */
+#define LM49453_P0_ADC_DSP_REG 0x30
+#define LM49453_P0_DAC_DSP_REG 0x31
+
+/* EFFECTS ENABLES */
+#define LM49453_P0_ADC_FX_ENABLES_REG 0x33
+
+/* GPIO */
+#define LM49453_P0_GPIO1_REG 0x38
+#define LM49453_P0_GPIO2_REG 0x39
+#define LM49453_P0_GPIO3_REG 0x3A
+#define LM49453_P0_HAP_CTL_REG 0x3B
+#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C
+#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D
+#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E
+#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F
+
+/* DIGITAL MIXER */
+#define LM49453_P0_DMIX_CLK_SEL_REG 0x40
+#define LM49453_P0_PORT1_RX_LVL1_REG 0x41
+#define LM49453_P0_PORT1_RX_LVL2_REG 0x42
+#define LM49453_P0_PORT2_RX_LVL_REG 0x43
+#define LM49453_P0_PORT1_TX1_REG 0x44
+#define LM49453_P0_PORT1_TX2_REG 0x45
+#define LM49453_P0_PORT1_TX3_REG 0x46
+#define LM49453_P0_PORT1_TX4_REG 0x47
+#define LM49453_P0_PORT1_TX5_REG 0x48
+#define LM49453_P0_PORT1_TX6_REG 0x49
+#define LM49453_P0_PORT1_TX7_REG 0x4A
+#define LM49453_P0_PORT1_TX8_REG 0x4B
+#define LM49453_P0_PORT2_TX1_REG 0x4C
+#define LM49453_P0_PORT2_TX2_REG 0x4D
+#define LM49453_P0_STN_SEL_REG 0x4F
+#define LM49453_P0_DACHPL1_REG 0x50
+#define LM49453_P0_DACHPL2_REG 0x51
+#define LM49453_P0_DACHPR1_REG 0x52
+#define LM49453_P0_DACHPR2_REG 0x53
+#define LM49453_P0_DACLOL1_REG 0x54
+#define LM49453_P0_DACLOL2_REG 0x55
+#define LM49453_P0_DACLOR1_REG 0x56
+#define LM49453_P0_DACLOR2_REG 0x57
+#define LM49453_P0_DACLSL1_REG 0x58
+#define LM49453_P0_DACLSL2_REG 0x59
+#define LM49453_P0_DACLSR1_REG 0x5A
+#define LM49453_P0_DACLSR2_REG 0x5B
+#define LM49453_P0_DACHAL1_REG 0x5C
+#define LM49453_P0_DACHAL2_REG 0x5D
+#define LM49453_P0_DACHAR1_REG 0x5E
+#define LM49453_P0_DACHAR2_REG 0x5F
+
+/* AUDIO PORT 1 (TDM) */
+#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62
+#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63
+#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64
+#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65
+#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66
+#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67
+#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68
+#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69
+
+/* AUDIO PORT 2 */
+#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A
+#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B
+#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C
+#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D
+#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E
+#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F
+#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70
+
+/* SAMPLE RATE */
+#define LM49453_P0_PORT1_SR_LSB_REG 0x79
+#define LM49453_P0_PORT1_SR_MSB_REG 0x7A
+#define LM49453_P0_PORT2_SR_LSB_REG 0x7B
+#define LM49453_P0_PORT2_SR_MSB_REG 0x7C
+
+/* EFFECTS - HPFs */
+#define LM49453_P0_HPF_REG 0x80
+
+/* EFFECTS ADC ALC */
+#define LM49453_P0_ADC_ALC1_REG 0x82
+#define LM49453_P0_ADC_ALC2_REG 0x83
+#define LM49453_P0_ADC_ALC3_REG 0x84
+#define LM49453_P0_ADC_ALC4_REG 0x85
+#define LM49453_P0_ADC_ALC5_REG 0x86
+#define LM49453_P0_ADC_ALC6_REG 0x87
+#define LM49453_P0_ADC_ALC7_REG 0x88
+#define LM49453_P0_ADC_ALC8_REG 0x89
+#define LM49453_P0_DMIC1_LEVELL_REG 0x8A
+#define LM49453_P0_DMIC1_LEVELR_REG 0x8B
+#define LM49453_P0_DMIC2_LEVELL_REG 0x8C
+#define LM49453_P0_DMIC2_LEVELR_REG 0x8D
+#define LM49453_P0_ADC_LEVELL_REG 0x8E
+#define LM49453_P0_ADC_LEVELR_REG 0x8F
+#define LM49453_P0_DAC_HP_LEVELL_REG 0x90
+#define LM49453_P0_DAC_HP_LEVELR_REG 0x91
+#define LM49453_P0_DAC_LO_LEVELL_REG 0x92
+#define LM49453_P0_DAC_LO_LEVELR_REG 0x93
+#define LM49453_P0_DAC_LS_LEVELL_REG 0x94
+#define LM49453_P0_DAC_LS_LEVELR_REG 0x95
+#define LM49453_P0_DAC_HA_LEVELL_REG 0x96
+#define LM49453_P0_DAC_HA_LEVELR_REG 0x97
+#define LM49453_P0_SOFT_MUTE_REG 0x98
+#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99
+#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A
+#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B
+
+/*DIGITAL MIC1 */
+#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0
+#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1
+#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2
+
+/*DIGITAL MIC2 */
+#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3
+#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4
+#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5
+
+/* ADC DECIMATOR */
+#define LM49453_P0_ADC_DECIMATOR_REG 0xB6
+
+/* DAC CONFIGURE */
+#define LM49453_P0_DAC_CONFIG_REG 0xB7
+
+/* SIDETONE */
+#define LM49453_P0_STN_VOL_ADCL_REG 0xB8
+#define LM49453_P0_STN_VOL_ADCR_REG 0xB9
+#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA
+#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB
+#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC
+#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD
+
+/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */
+#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2
+#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3
+#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4
+#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5
+
+/* ADC ALC EFFECT MONITORS (Read Only) */
+#define LM49453_P0_ADC_LVLMONL_REG 0xC8
+#define LM49453_P0_ADC_LVLMONR_REG 0xC9
+#define LM49453_P0_ADC_ALCMONL_REG 0xCA
+#define LM49453_P0_ADC_ALCMONR_REG 0xCB
+#define LM49453_P0_ADC_MUTED_REG 0xCC
+#define LM49453_P0_DAC_MUTED_REG 0xCD
+
+/* HEADSET DETECT */
+#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0
+#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1
+#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2
+#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3
+#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4
+#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5
+#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6
+#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7
+#define LM49453_P0_HSD_IRQ1_REG 0xD8
+#define LM49453_P0_HSD_IRQ2_REG 0xD9
+#define LM49453_P0_HSD_IRQ3_REG 0xDA
+#define LM49453_P0_HSD_IRQ4_REG 0xDB
+#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC
+#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD
+#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE
+#define LM49453_P0_HSD_R_HPLL_REG 0xE0
+#define LM49453_P0_HSD_R_HPLH_REG 0xE1
+#define LM49453_P0_HSD_R_HPLU_REG 0xE2
+#define LM49453_P0_HSD_R_HPRL_REG 0xE3
+#define LM49453_P0_HSD_R_HPRH_REG 0xE4
+#define LM49453_P0_HSD_R_HPRU_REG 0xE5
+#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6
+#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7
+#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8
+#define LM49453_P0_HSD_RO_FINALL_REG 0xE9
+#define LM49453_P0_HSD_RO_FINALH_REG 0xEA
+#define LM49453_P0_HSD_RO_FINALU_REG 0xEB
+#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC
+#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED
+#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE
+#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4
+#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5
+
+/* I/O PULLDOWN CONFIG */
+#define LM49453_P0_PULL_CONFIG1_REG 0xF8
+#define LM49453_P0_PULL_CONFIG2_REG 0xF9
+#define LM49453_P0_PULL_CONFIG3_REG 0xFA
+
+/* RESET */
+#define LM49453_P0_RESET_REG 0xFE
+
+/* PAGE */
+#define LM49453_PAGE_REG 0xFF
+
+#define LM49453_MAX_REGISTER (0xFF+1)
+
+/* LM49453_P0_PMC_SETUP_REG (0x00h) */
+#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0))
+#define LM49453_PMC_SETUP_PLL_EN BIT(2)
+#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3)
+#define LM49453_PMC_SETUP_PLL_FLL BIT(4)
+#define LM49453_PMC_SETUP_MCLK_OVER BIT(5)
+#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6)
+#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7)
+
+/* Chip Enable bits */
+#define LM49453_CHIP_EN_SHUTDOWN 0x00
+#define LM49453_CHIP_EN 0x01
+#define LM49453_CHIP_EN_HSD_DETECT 0x02
+#define LM49453_CHIP_EN_INVALID_HSD 0x03
+
+/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */
+#define LM49453_CLK_SEL1_MCLK_SEL 0x11
+#define LM49453_CLK_SEL1_RTC_SEL 0x11
+#define LM49453_CLK_SEL1_PORT1_SEL 0x10
+#define LM49453_CLK_SEL1_PORT2_SEL 0x11
+
+/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */
+#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38
+
+/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */
+#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001
+
+/* LM49453_P0_VCO_TARGETLL_REG (0x11) */
+#define LM49453_VCO_TARGET_VAL 0x8ca0001
+
+/* LM49453_P0_ADC_DSP_REG (0x30h) */
+#define LM49453_ADC_DSP_ADC_MUTEL BIT(0)
+#define LM49453_ADC_DSP_ADC_MUTER BIT(1)
+#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2)
+#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3)
+#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4)
+#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5)
+#define LM49453_ADC_DSP_MUTE_ALL 0x3F
+
+/* LM49453_P0_DAC_DSP_REG (0x31h) */
+#define LM49453_DAC_DSP_MUTE_ALL 0xFF
+
+/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */
+#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3))
+#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3)
+#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4)
+
+/* LM49453_P0_RESET_REG (0xFEh) */
+#define LM49453_RESET_REG_RST BIT(0)
+
+/* Page select register bits (0xFF) */
+#define LM49453_PAGE0_SELECT 0x0
+#define LM49453_PAGE1_SELECT 0x1
+
+/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */
+#define LM49453_JACK_DISABLE 0x00
+#define LM49453_JACK_CONFIG1 0x01
+#define LM49453_JACK_CONFIG2 0x02
+#define LM49453_JACK_CONFIG3 0x03
+#define LM49453_JACK_CONFIG4 0x04
+#define LM49453_JACK_CONFIG5 0x05
+
+/* Page 1 REGISTERS */
+
+/* SIDETONE */
+#define LM49453_P1_SIDETONE_SA0L_REG 0x80
+#define LM49453_P1_SIDETONE_SA0H_REG 0x81
+#define LM49453_P1_SIDETONE_SAB0U_REG 0x82
+#define LM49453_P1_SIDETONE_SB0L_REG 0x83
+#define LM49453_P1_SIDETONE_SB0H_REG 0x84
+#define LM49453_P1_SIDETONE_SH0L_REG 0x85
+#define LM49453_P1_SIDETONE_SH0H_REG 0x86
+#define LM49453_P1_SIDETONE_SH0U_REG 0x87
+#define LM49453_P1_SIDETONE_SA1L_REG 0x88
+#define LM49453_P1_SIDETONE_SA1H_REG 0x89
+#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A
+#define LM49453_P1_SIDETONE_SB1L_REG 0x8B
+#define LM49453_P1_SIDETONE_SB1H_REG 0x8C
+#define LM49453_P1_SIDETONE_SH1L_REG 0x8D
+#define LM49453_P1_SIDETONE_SH1H_REG 0x8E
+#define LM49453_P1_SIDETONE_SH1U_REG 0x8F
+#define LM49453_P1_SIDETONE_SA2L_REG 0x90
+#define LM49453_P1_SIDETONE_SA2H_REG 0x91
+#define LM49453_P1_SIDETONE_SAB2U_REG 0x92
+#define LM49453_P1_SIDETONE_SB2L_REG 0x93
+#define LM49453_P1_SIDETONE_SB2H_REG 0x94
+#define LM49453_P1_SIDETONE_SH2L_REG 0x95
+#define LM49453_P1_SIDETONE_SH2H_REG 0x96
+#define LM49453_P1_SIDETONE_SH2U_REG 0x97
+#define LM49453_P1_SIDETONE_SA3L_REG 0x98
+#define LM49453_P1_SIDETONE_SA3H_REG 0x99
+#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A
+#define LM49453_P1_SIDETONE_SB3L_REG 0x9B
+#define LM49453_P1_SIDETONE_SB3H_REG 0x9C
+#define LM49453_P1_SIDETONE_SH3L_REG 0x9D
+#define LM49453_P1_SIDETONE_SH3H_REG 0x9E
+#define LM49453_P1_SIDETONE_SH3U_REG 0x9F
+#define LM49453_P1_SIDETONE_SA4L_REG 0xA0
+#define LM49453_P1_SIDETONE_SA4H_REG 0xA1
+#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2
+#define LM49453_P1_SIDETONE_SB4L_REG 0xA3
+#define LM49453_P1_SIDETONE_SB4H_REG 0xA4
+#define LM49453_P1_SIDETONE_SH4L_REG 0xA5
+#define LM49453_P1_SIDETONE_SH4H_REG 0xA6
+#define LM49453_P1_SIDETONE_SH4U_REG 0xA7
+#define LM49453_P1_SIDETONE_SA5L_REG 0xA8
+#define LM49453_P1_SIDETONE_SA5H_REG 0xA9
+#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA
+#define LM49453_P1_SIDETONE_SB5L_REG 0xAB
+#define LM49453_P1_SIDETONE_SB5H_REG 0xAC
+#define LM49453_P1_SIDETONE_SH5L_REG 0xAD
+#define LM49453_P1_SIDETONE_SH5H_REG 0xAE
+#define LM49453_P1_SIDETONE_SH5U_REG 0xAF
+
+/* CHARGE PUMP CONFIG */
+#define LM49453_P1_CP_CONFIG1_REG 0xB0
+#define LM49453_P1_CP_CONFIG2_REG 0xB1
+#define LM49453_P1_CP_CONFIG3_REG 0xB2
+#define LM49453_P1_CP_CONFIG4_REG 0xB3
+#define LM49453_P1_CP_LA_VTH1L_REG 0xB4
+#define LM49453_P1_CP_LA_VTH1M_REG 0xB5
+#define LM49453_P1_CP_LA_VTH2L_REG 0xB6
+#define LM49453_P1_CP_LA_VTH2M_REG 0xB7
+#define LM49453_P1_CP_LA_VTH3L_REG 0xB8
+#define LM49453_P1_CP_LA_VTH3H_REG 0xB9
+#define LM49453_P1_CP_CLK_DIV_REG 0xBA
+
+/* DAC */
+#define LM49453_P1_DAC_CHOP_REG 0xC0
+
+#define LM49453_CLK_SRC_MCLK 1
+#endif
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 0bb511a0388..35179e2c23c 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -24,6 +24,7 @@
#include <linux/slab.h>
#include <asm/div64.h>
#include <sound/max98095.h>
+#include <sound/jack.h>
#include "max98095.h"
enum max98095_type {
@@ -51,6 +52,8 @@ struct max98095_priv {
u8 lin_state;
unsigned int mic1pre;
unsigned int mic2pre;
+ struct snd_soc_jack *headphone_jack;
+ struct snd_soc_jack *mic_jack;
};
static const u8 max98095_reg_def[M98095_REG_CNT] = {
@@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec)
max98095_handle_bq_pdata(codec);
}
+static irqreturn_t max98095_report_jack(int irq, void *data)
+{
+ struct snd_soc_codec *codec = data;
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ unsigned int value;
+ int hp_report = 0;
+ int mic_report = 0;
+
+ /* Read the Jack Status Register */
+ value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS);
+
+ /* If ddone is not set, then detection isn't finished yet */
+ if ((value & M98095_DDONE) == 0)
+ return IRQ_NONE;
+
+ /* if hp, check its bit, and if set, clear it */
+ if ((value & M98095_HP_IN || value & M98095_LO_IN) &&
+ max98095->headphone_jack)
+ hp_report |= SND_JACK_HEADPHONE;
+
+ /* if mic, check its bit, and if set, clear it */
+ if ((value & M98095_MIC_IN) && max98095->mic_jack)
+ mic_report |= SND_JACK_MICROPHONE;
+
+ if (max98095->headphone_jack == max98095->mic_jack) {
+ snd_soc_jack_report(max98095->headphone_jack,
+ hp_report | mic_report,
+ SND_JACK_HEADSET);
+ } else {
+ if (max98095->headphone_jack)
+ snd_soc_jack_report(max98095->headphone_jack,
+ hp_report, SND_JACK_HEADPHONE);
+ if (max98095->mic_jack)
+ snd_soc_jack_report(max98095->mic_jack,
+ mic_report, SND_JACK_MICROPHONE);
+ }
+
+ return IRQ_HANDLED;
+}
+
+int max98095_jack_detect_enable(struct snd_soc_codec *codec)
+{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+ int detect_enable = M98095_JDEN;
+ unsigned int slew = M98095_DEFAULT_SLEW_DELAY;
+
+ if (max98095->pdata->jack_detect_pin5en)
+ detect_enable |= M98095_PIN5EN;
+
+ if (max98095->pdata->jack_detect_delay)
+ slew = max98095->pdata->jack_detect_delay;
+
+ ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+ return ret;
+ }
+
+ /* configure auto detection to be enabled */
+ ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+int max98095_jack_detect_disable(struct snd_soc_codec *codec)
+{
+ int ret = 0;
+
+ /* configure auto detection to be disabled */
+ ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+int max98095_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack)
+{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *client = to_i2c_client(codec->dev);
+ int ret = 0;
+
+ max98095->headphone_jack = hp_jack;
+ max98095->mic_jack = mic_jack;
+
+ /* only progress if we have at least 1 jack pointer */
+ if (!hp_jack && !mic_jack)
+ return -EINVAL;
+
+ max98095_jack_detect_enable(codec);
+
+ /* enable interrupts for headphone jack detection */
+ ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN,
+ M98095_IDDONE, M98095_IDDONE);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret);
+ return ret;
+ }
+
+ max98095_report_jack(client->irq, codec);
+ return 0;
+}
+
#ifdef CONFIG_PM
static int max98095_suspend(struct snd_soc_codec *codec)
{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+
+ if (max98095->headphone_jack || max98095->mic_jack)
+ max98095_jack_detect_disable(codec);
+
max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
@@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec)
static int max98095_resume(struct snd_soc_codec *codec)
{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *client = to_i2c_client(codec->dev);
+
max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (max98095->headphone_jack || max98095->mic_jack) {
+ max98095_jack_detect_enable(codec);
+ max98095_report_jack(client->irq, codec);
+ }
+
return 0;
}
#else
@@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec)
{
struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
struct max98095_cdata *cdata;
+ struct i2c_client *client;
int ret = 0;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
@@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec)
/* reset the codec, the DSP core, and disable all interrupts */
max98095_reset(codec);
+ client = to_i2c_client(codec->dev);
+
/* initialize private data */
max98095->sysclk = (unsigned)-1;
@@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec)
max98095->mic1pre = 0;
max98095->mic2pre = 0;
+ if (client->irq) {
+ /* register an audio interrupt */
+ ret = request_threaded_irq(client->irq, NULL,
+ max98095_report_jack,
+ IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING,
+ "max98095", codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
+ goto err_access;
+ }
+ }
+
ret = snd_soc_read(codec, M98095_0FF_REV_ID);
if (ret < 0) {
dev_err(codec->dev, "Failure reading hardware revision: %d\n",
ret);
- goto err_access;
+ goto err_irq;
}
dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
@@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec)
max98095_add_widgets(codec);
+ return 0;
+
+err_irq:
+ if (client->irq)
+ free_irq(client->irq, codec);
err_access:
return ret;
}
static int max98095_remove(struct snd_soc_codec *codec)
{
+ struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *client = to_i2c_client(codec->dev);
+
max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ if (max98095->headphone_jack || max98095->mic_jack)
+ max98095_jack_detect_disable(codec);
+
+ if (client->irq)
+ free_irq(client->irq, codec);
+
return 0;
}
diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h
index 891584a0eb0..2ebbe4e894b 100644
--- a/sound/soc/codecs/max98095.h
+++ b/sound/soc/codecs/max98095.h
@@ -175,11 +175,23 @@
/* MAX98095 Registers Bit Fields */
+/* M98095_007_JACK_AUTO_STS */
+ #define M98095_MIC_IN (1<<3)
+ #define M98095_LO_IN (1<<5)
+ #define M98095_HP_IN (1<<6)
+ #define M98095_DDONE (1<<7)
+
/* M98095_00F_HOST_CFG */
#define M98095_SEG (1<<0)
#define M98095_XTEN (1<<1)
#define M98095_MDLLEN (1<<2)
+/* M98095_013_JACK_INT_EN */
+ #define M98095_IMIC_IN (1<<3)
+ #define M98095_ILO_IN (1<<5)
+ #define M98095_IHP_IN (1<<6)
+ #define M98095_IDDONE (1<<7)
+
/* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */
#define M98095_CLKMODE_MASK 0xFF
@@ -255,6 +267,10 @@
#define M98095_EQ2EN (1<<1)
#define M98095_EQ1EN (1<<0)
+/* M98095_089_JACK_DET_AUTO */
+ #define M98095_PIN5EN (1<<2)
+ #define M98095_JDEN (1<<7)
+
/* M98095_090_PWR_EN_IN */
#define M98095_INEN (1<<7)
#define M98095_MB2EN (1<<3)
@@ -296,4 +312,10 @@
#define M98095_174_DAI1_BQ_BASE 0x74
#define M98095_17E_DAI2_BQ_BASE 0x7E
+/* Default Delay used in Slew Rate Calculation for Jack detection */
+#define M98095_DEFAULT_SLEW_DELAY 0x18
+
+extern int max98095_jack_detect(struct snd_soc_codec *codec,
+ struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack);
+
#endif
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
new file mode 100644
index 00000000000..6276e352125
--- /dev/null
+++ b/sound/soc/codecs/mc13783.c
@@ -0,0 +1,786 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.de
+ * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de
+ * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch
+ *
+ * Initial development of this code was funded by
+ * Phytec Messtechnik GmbH, http://www.phytec.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.
+ * 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 Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/mfd/mc13xxx.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+
+#include "mc13783.h"
+
+#define MC13783_AUDIO_RX0 36
+#define MC13783_AUDIO_RX1 37
+#define MC13783_AUDIO_TX 38
+#define MC13783_SSI_NETWORK 39
+#define MC13783_AUDIO_CODEC 40
+#define MC13783_AUDIO_DAC 41
+
+#define AUDIO_RX0_ALSPEN (1 << 5)
+#define AUDIO_RX0_ALSPSEL (1 << 7)
+#define AUDIO_RX0_ADDCDC (1 << 21)
+#define AUDIO_RX0_ADDSTDC (1 << 22)
+#define AUDIO_RX0_ADDRXIN (1 << 23)
+
+#define AUDIO_RX1_PGARXEN (1 << 0);
+#define AUDIO_RX1_PGASTEN (1 << 5)
+#define AUDIO_RX1_ARXINEN (1 << 10)
+
+#define AUDIO_TX_AMC1REN (1 << 5)
+#define AUDIO_TX_AMC1LEN (1 << 7)
+#define AUDIO_TX_AMC2EN (1 << 9)
+#define AUDIO_TX_ATXINEN (1 << 11)
+#define AUDIO_TX_RXINREC (1 << 13)
+
+#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2)
+#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4)
+#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6)
+#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8)
+#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10)
+#define SSI_NETWORK_CDCFSDLY(x) (1 << 11)
+#define SSI_NETWORK_DAC_SLOTS_8 (1 << 12)
+#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12)
+#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12)
+#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12)
+#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14)
+#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14)
+#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16)
+#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18)
+#define SSI_NETWORK_STDCSUMGAIN (1 << 20)
+
+/*
+ * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same
+ * register layout
+ */
+#define AUDIO_SSI_SEL (1 << 0)
+#define AUDIO_CLK_SEL (1 << 1)
+#define AUDIO_CSM (1 << 2)
+#define AUDIO_BCL_INV (1 << 3)
+#define AUDIO_CFS_INV (1 << 4)
+#define AUDIO_CFS(x) (((x) & 0x3) << 5)
+#define AUDIO_CLK(x) (((x) & 0x7) << 7)
+#define AUDIO_C_EN (1 << 11)
+#define AUDIO_C_CLK_EN (1 << 12)
+#define AUDIO_C_RESET (1 << 15)
+
+#define AUDIO_CODEC_CDCFS8K16K (1 << 10)
+#define AUDIO_DAC_CFS_DLY_B (1 << 10)
+
+struct mc13783_priv {
+ struct snd_soc_codec codec;
+ struct mc13xxx *mc13xxx;
+
+ enum mc13783_ssi_port adc_ssi_port;
+ enum mc13783_ssi_port dac_ssi_port;
+};
+
+static unsigned int mc13783_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+ unsigned int value = 0;
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ mc13xxx_reg_read(priv->mc13xxx, reg, &value);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return value;
+}
+
+static int mc13783_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return ret;
+}
+
+/* Mapping between sample rates and register value */
+static unsigned int mc13783_rates[] = {
+ 8000, 11025, 12000, 16000,
+ 22050, 24000, 32000, 44100,
+ 48000, 64000, 96000
+};
+
+static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ unsigned int rate = params_rate(params);
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) {
+ if (rate == mc13783_rates[i]) {
+ snd_soc_update_bits(codec, MC13783_AUDIO_DAC,
+ 0xf << 17, i << 17);
+ return 0;
+ }
+ }
+
+ return -EINVAL;
+}
+
+static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ unsigned int rate = params_rate(params);
+ unsigned int val;
+
+ switch (rate) {
+ case 8000:
+ val = 0;
+ break;
+ case 16000:
+ val = AUDIO_CODEC_CDCFS8K16K;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K,
+ val);
+
+ return 0;
+}
+
+static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return mc13783_pcm_hw_params_dac(substream, params, dai);
+ else
+ return mc13783_pcm_hw_params_codec(substream, params, dai);
+}
+
+static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt,
+ unsigned int reg)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+ unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV |
+ AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET;
+
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val |= AUDIO_CFS(2);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ val |= AUDIO_CFS(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ val |= AUDIO_BCL_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ val |= AUDIO_BCL_INV | AUDIO_CFS_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ val |= AUDIO_CFS_INV;
+ break;
+ }
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val |= AUDIO_C_CLK_EN;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ val |= AUDIO_CSM;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ return -EINVAL;
+ }
+
+ val |= AUDIO_C_RESET;
+
+ snd_soc_update_bits(codec, reg, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ if (dai->id == MC13783_ID_STEREO_DAC)
+ return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+ else
+ return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ int ret;
+
+ ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC);
+ if (ret)
+ return ret;
+
+ /*
+ * In synchronous mode force the voice codec into slave mode
+ * so that the clock / framesync from the stereo DAC is used
+ */
+ fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+ fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC);
+
+ return ret;
+}
+
+static int mc13783_sysclk[] = {
+ 13000000,
+ 15360000,
+ 16800000,
+ -1,
+ 26000000,
+ -1, /* 12000000, invalid for voice codec */
+ -1, /* 3686400, invalid for voice codec */
+ 33600000,
+};
+
+static int mc13783_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir,
+ unsigned int reg)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int clk;
+ unsigned int val = 0;
+ unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL;
+
+ for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) {
+ if (mc13783_sysclk[clk] < 0)
+ continue;
+ if (mc13783_sysclk[clk] == freq)
+ break;
+ }
+
+ if (clk == ARRAY_SIZE(mc13783_sysclk))
+ return -EINVAL;
+
+ if (clk_id == MC13783_CLK_CLIB)
+ val |= AUDIO_CLK_SEL;
+
+ val |= AUDIO_CLK(clk);
+
+ snd_soc_update_bits(codec, reg, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+}
+
+static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ int ret;
+
+ ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC);
+ if (ret)
+ return ret;
+
+ return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC);
+}
+
+static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+ unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK |
+ SSI_NETWORK_DAC_RXSLOT_MASK;
+
+ switch (slots) {
+ case 2:
+ val |= SSI_NETWORK_DAC_SLOTS_2;
+ break;
+ case 4:
+ val |= SSI_NETWORK_DAC_SLOTS_4;
+ break;
+ case 8:
+ val |= SSI_NETWORK_DAC_SLOTS_8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (rx_mask) {
+ case 0xfffffffc:
+ val |= SSI_NETWORK_DAC_RXSLOT_0_1;
+ break;
+ case 0xfffffff3:
+ val |= SSI_NETWORK_DAC_RXSLOT_2_3;
+ break;
+ case 0xffffffcf:
+ val |= SSI_NETWORK_DAC_RXSLOT_4_5;
+ break;
+ case 0xffffff3f:
+ val |= SSI_NETWORK_DAC_RXSLOT_6_7;
+ break;
+ default:
+ return -EINVAL;
+ };
+
+ snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int val = 0;
+ unsigned int mask = 0x3f;
+
+ if (slots != 4)
+ return -EINVAL;
+
+ if (tx_mask != 0xfffffffc)
+ return -EINVAL;
+
+ val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */
+ val |= (0x01 << 4); /* secondary timeslot TX is 1 */
+
+ snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val);
+
+ return 0;
+}
+
+static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots,
+ int slot_width)
+{
+ int ret;
+
+ ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots,
+ slot_width);
+ if (ret)
+ return ret;
+
+ ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots,
+ slot_width);
+
+ return ret;
+}
+
+static const struct snd_kcontrol_new mc1l_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0);
+
+static const struct snd_kcontrol_new mc1r_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0);
+
+static const struct snd_kcontrol_new mc2_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0);
+
+static const struct snd_kcontrol_new atx_amp_ctl =
+ SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0);
+
+
+/* Virtual mux. The chip does the input selection automatically
+ * as soon as we enable one input. */
+static const char * const adcl_enum_text[] = {
+ "MC1L", "RXINL",
+};
+
+static const struct soc_enum adcl_enum =
+ SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text);
+
+static const struct snd_kcontrol_new left_input_mux =
+ SOC_DAPM_ENUM_VIRT("Route", adcl_enum);
+
+static const char * const adcr_enum_text[] = {
+ "MC1R", "MC2", "RXINR", "TXIN",
+};
+
+static const struct soc_enum adcr_enum =
+ SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text);
+
+static const struct snd_kcontrol_new right_input_mux =
+ SOC_DAPM_ENUM_VIRT("Route", adcr_enum);
+
+static const struct snd_kcontrol_new samp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0);
+
+static const struct snd_kcontrol_new lamp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0);
+
+static const struct snd_kcontrol_new hlamp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0);
+
+static const struct snd_kcontrol_new hramp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0);
+
+static const struct snd_kcontrol_new llamp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0);
+
+static const struct snd_kcontrol_new lramp_ctl =
+ SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0);
+
+static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = {
+/* Input */
+ SND_SOC_DAPM_INPUT("MC1LIN"),
+ SND_SOC_DAPM_INPUT("MC1RIN"),
+ SND_SOC_DAPM_INPUT("MC2IN"),
+ SND_SOC_DAPM_INPUT("RXINR"),
+ SND_SOC_DAPM_INPUT("RXINL"),
+ SND_SOC_DAPM_INPUT("TXIN"),
+
+ SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl),
+ SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl),
+ SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl),
+ SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl),
+
+ SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0,
+ &left_input_mux),
+ SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0,
+ &right_input_mux),
+
+ SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0),
+
+/* Output */
+ SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("RXOUTL"),
+ SND_SOC_DAPM_OUTPUT("RXOUTR"),
+ SND_SOC_DAPM_OUTPUT("HSL"),
+ SND_SOC_DAPM_OUTPUT("HSR"),
+ SND_SOC_DAPM_OUTPUT("LSP"),
+ SND_SOC_DAPM_OUTPUT("SP"),
+
+ SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl),
+ SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl),
+ SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl),
+ SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl),
+ SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl),
+ SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl),
+ SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0),
+ SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0),
+};
+
+static struct snd_soc_dapm_route mc13783_routes[] = {
+/* Input */
+ { "MC1L Amp", NULL, "MC1LIN"},
+ { "MC1R Amp", NULL, "MC1RIN" },
+ { "MC2 Amp", NULL, "MC2IN" },
+ { "TXIN Amp", NULL, "TXIN"},
+
+ { "PGA Left Input Mux", "MC1L", "MC1L Amp" },
+ { "PGA Left Input Mux", "RXINL", "RXINL"},
+ { "PGA Right Input Mux", "MC1R", "MC1R Amp" },
+ { "PGA Right Input Mux", "MC2", "MC2 Amp"},
+ { "PGA Right Input Mux", "TXIN", "TXIN Amp"},
+ { "PGA Right Input Mux", "RXINR", "RXINR"},
+
+ { "PGA Left Input", NULL, "PGA Left Input Mux"},
+ { "PGA Right Input", NULL, "PGA Right Input Mux"},
+
+ { "ADC", NULL, "PGA Left Input"},
+ { "ADC", NULL, "PGA Right Input"},
+ { "ADC", NULL, "ADC_Reset"},
+
+/* Output */
+ { "HSL", NULL, "Headset Amp Left" },
+ { "HSR", NULL, "Headset Amp Right"},
+ { "RXOUTL", NULL, "Line out Amp Left"},
+ { "RXOUTR", NULL, "Line out Amp Right"},
+ { "SP", NULL, "Speaker Amp"},
+ { "Speaker Amp", NULL, "DAC PGA"},
+ { "LSP", NULL, "DAC PGA"},
+ { "Headset Amp Left", NULL, "DAC PGA"},
+ { "Headset Amp Right", NULL, "DAC PGA"},
+ { "Line out Amp Left", NULL, "DAC PGA"},
+ { "Line out Amp Right", NULL, "DAC PGA"},
+ { "DAC PGA", NULL, "DAC"},
+ { "DAC", NULL, "DAC_E"},
+};
+
+static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
+ "Mono", "Mono Mix"};
+
+static const struct soc_enum mc13783_enum_3d_mixer =
+ SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer),
+ mc13783_3d_mixer);
+
+static struct snd_kcontrol_new mc13783_control_list[] = {
+ SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
+ SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0),
+ SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0),
+ SOC_ENUM("3D Control", mc13783_enum_3d_mixer),
+};
+
+static int mc13783_probe(struct snd_soc_codec *codec)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ /* these are the reset values */
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027);
+ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004);
+
+ if (priv->adc_ssi_port == MC13783_SSI1_PORT)
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+ AUDIO_SSI_SEL, 0);
+ else
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC,
+ 0, AUDIO_SSI_SEL);
+
+ if (priv->dac_ssi_port == MC13783_SSI1_PORT)
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+ AUDIO_SSI_SEL, 0);
+ else
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC,
+ 0, AUDIO_SSI_SEL);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return 0;
+}
+
+static int mc13783_remove(struct snd_soc_codec *codec)
+{
+ struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ mc13xxx_lock(priv->mc13xxx);
+
+ /* Make sure VAUDIOON is off */
+ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0);
+
+ mc13xxx_unlock(priv->mc13xxx);
+
+ return 0;
+}
+
+#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
+
+#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mc13783_ops_dac = {
+ .hw_params = mc13783_pcm_hw_params_dac,
+ .set_fmt = mc13783_set_fmt_async,
+ .set_sysclk = mc13783_set_sysclk_dac,
+ .set_tdm_slot = mc13783_set_tdm_slot_dac,
+};
+
+static struct snd_soc_dai_ops mc13783_ops_codec = {
+ .hw_params = mc13783_pcm_hw_params_codec,
+ .set_fmt = mc13783_set_fmt_async,
+ .set_sysclk = mc13783_set_sysclk_codec,
+ .set_tdm_slot = mc13783_set_tdm_slot_codec,
+};
+
+/*
+ * The mc13783 has two SSI ports, both of them can be routed either
+ * to the voice codec or the stereo DAC. When two different SSI ports
+ * are used for the voice codec and the stereo DAC we can do different
+ * formats and sysclock settings for playback and capture
+ * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port
+ * forces us to use symmetric rates (mc13783-hifi).
+ */
+static struct snd_soc_dai_driver mc13783_dai_async[] = {
+ {
+ .name = "mc13783-hifi-playback",
+ .id = MC13783_ID_STEREO_DAC,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = MC13783_FORMATS,
+ },
+ .ops = &mc13783_ops_dac,
+ }, {
+ .name = "mc13783-hifi-capture",
+ .id = MC13783_ID_STEREO_CODEC,
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MC13783_RATES_RECORD,
+ .formats = MC13783_FORMATS,
+ },
+ .ops = &mc13783_ops_codec,
+ },
+};
+
+static struct snd_soc_dai_ops mc13783_ops_sync = {
+ .hw_params = mc13783_pcm_hw_params_sync,
+ .set_fmt = mc13783_set_fmt_sync,
+ .set_sysclk = mc13783_set_sysclk_sync,
+ .set_tdm_slot = mc13783_set_tdm_slot_sync,
+};
+
+static struct snd_soc_dai_driver mc13783_dai_sync[] = {
+ {
+ .name = "mc13783-hifi",
+ .id = MC13783_ID_SYNC,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = MC13783_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MC13783_RATES_RECORD,
+ .formats = MC13783_FORMATS,
+ },
+ .ops = &mc13783_ops_sync,
+ .symmetric_rates = 1,
+ }
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
+ .probe = mc13783_probe,
+ .remove = mc13783_remove,
+ .read = mc13783_read,
+ .write = mc13783_write,
+ .controls = mc13783_control_list,
+ .num_controls = ARRAY_SIZE(mc13783_control_list),
+ .dapm_widgets = mc13783_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets),
+ .dapm_routes = mc13783_routes,
+ .num_dapm_routes = ARRAY_SIZE(mc13783_routes),
+};
+
+static int mc13783_codec_probe(struct platform_device *pdev)
+{
+ struct mc13xxx *mc13xxx;
+ struct mc13783_priv *priv;
+ struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data;
+ int ret;
+
+ mc13xxx = dev_get_drvdata(pdev->dev.parent);
+
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
+ dev_set_drvdata(&pdev->dev, priv);
+ priv->mc13xxx = mc13xxx;
+ if (pdata) {
+ priv->adc_ssi_port = pdata->adc_ssi_port;
+ priv->dac_ssi_port = pdata->dac_ssi_port;
+ } else {
+ priv->adc_ssi_port = MC13783_SSI1_PORT;
+ priv->dac_ssi_port = MC13783_SSI2_PORT;
+ }
+
+ if (priv->adc_ssi_port == priv->dac_ssi_port)
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+ mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync));
+ else
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
+ mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
+
+ if (ret)
+ goto err_register_codec;
+
+ return 0;
+
+err_register_codec:
+ dev_err(&pdev->dev, "register codec failed with %d\n", ret);
+
+ return ret;
+}
+
+static int mc13783_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver mc13783_codec_driver = {
+ .driver = {
+ .name = "mc13783-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = mc13783_codec_probe,
+ .remove = __devexit_p(mc13783_codec_remove),
+};
+
+module_platform_driver(mc13783_codec_driver);
+
+MODULE_DESCRIPTION("ASoC MC13783 driver");
+MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h
new file mode 100644
index 00000000000..3a6d1993a21
--- /dev/null
+++ b/sound/soc/codecs/mc13783.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2008 Juergen Beisert, kernel@pengutronix.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.
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ */
+
+#ifndef MC13783_MIXER_H
+#define MC13783_MIXER_H
+
+#define MC13783_CLK_CLIA 1
+#define MC13783_CLK_CLIB 2
+
+#define MC13783_ID_STEREO_DAC 1
+#define MC13783_ID_STEREO_CODEC 2
+#define MC13783_ID_SYNC 3
+
+#endif /* MC13783_MIXER_H */
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
new file mode 100644
index 00000000000..22cb5bf5927
--- /dev/null
+++ b/sound/soc/codecs/ml26124.c
@@ -0,0 +1,681 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
+ *
+ * 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.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "ml26124.h"
+
+#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */
+#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */
+#define ML26124_SAI_NO_DELAY BIT(1)
+#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */
+#define ML26134_CACHESIZE 212
+#define ML26124_VMID BIT(1)
+#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_48000)
+#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+#define ML26124_NUM_REGISTER ML26134_CACHESIZE
+
+struct ml26124_priv {
+ u32 mclk;
+ u32 rate;
+ struct regmap *regmap;
+ int clk_in;
+ struct snd_pcm_substream *substream;
+};
+
+struct clk_coeff {
+ u32 mclk;
+ u32 rate;
+ u8 pllnl;
+ u8 pllnh;
+ u8 pllml;
+ u8 pllmh;
+ u8 plldiv;
+};
+
+/* ML26124 configuration */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0);
+static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
+
+static const char * const ml26124_companding[] = {"16bit PCM", "u-law",
+ "A-law"};
+
+static const struct soc_enum ml26124_adc_companding_enum
+ = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding);
+
+static const struct soc_enum ml26124_dac_companding_enum
+ = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
+
+static const struct snd_kcontrol_new ml26124_snd_controls[] = {
+ SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0,
+ 0x3f, 0, boost_vol),
+ SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0,
+ 0xff, 1, digital_tlv),
+ SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0,
+ 0xf, 1, alclvl),
+ SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0,
+ 7, 0, mingain),
+ SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4,
+ 7, 1, maxgain),
+ SOC_SINGLE_TLV("Playback Limiter Min Input Volume",
+ ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain),
+ SOC_SINGLE_TLV("Playback Limiter Max Input Volume",
+ ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain),
+ SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0,
+ 0x3f, 0, boost_vol),
+ SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0),
+ SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0),
+ SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1,
+ 1, 0),
+ SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0),
+ SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0),
+ SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0),
+ SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0),
+ SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0),
+ SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0),
+ SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0),
+ SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0),
+ SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0),
+ SOC_ENUM("DAC Companding", ml26124_dac_companding_enum),
+ SOC_ENUM("ADC Companding", ml26124_adc_companding_enum),
+};
+
+static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0),
+ SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1,
+ 0),
+ SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0),
+};
+
+/* Input mux */
+static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in",
+ "Digital MIC in", "Analog MIC Differential in"};
+
+static const struct soc_enum ml26124_insel_enum =
+ SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select);
+
+static const struct snd_kcontrol_new ml26124_input_mux_controls =
+ SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
+
+static const struct snd_kcontrol_new ml26124_line_control =
+ SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0);
+
+static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+ &ml26124_output_mixer_controls[0],
+ ARRAY_SIZE(ml26124_output_mixer_controls)),
+ SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0),
+ SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0),
+ SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0),
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &ml26124_input_mux_controls),
+ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+ &ml26124_line_control),
+ SND_SOC_DAPM_INPUT("MDIN"),
+ SND_SOC_DAPM_INPUT("MIN"),
+ SND_SOC_DAPM_INPUT("LIN"),
+ SND_SOC_DAPM_OUTPUT("SPOUT"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+};
+
+static const struct snd_soc_dapm_route ml26124_intercon[] = {
+ /* Supply */
+ {"DAC", NULL, "MCLKEN"},
+ {"ADC", NULL, "MCLKEN"},
+ {"DAC", NULL, "PLLEN"},
+ {"ADC", NULL, "PLLEN"},
+ {"DAC", NULL, "PLLOE"},
+ {"ADC", NULL, "PLLOE"},
+
+ /* output mixer */
+ {"Output Mixer", "DAC Switch", "DAC"},
+ {"Output Mixer", "Line in loopback Switch", "LIN"},
+
+ /* outputs */
+ {"LOUT", NULL, "Output Mixer"},
+ {"SPOUT", NULL, "Output Mixer"},
+ {"Line Out Enable", NULL, "LOUT"},
+
+ /* input */
+ {"ADC", NULL, "Input Mux"},
+ {"Input Mux", "Analog MIC SingleEnded in", "PGA"},
+ {"Input Mux", "Analog MIC Differential in", "PGA"},
+ {"PGA", NULL, "MIN"},
+};
+
+/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */
+static const struct clk_coeff coeff_div[] = {
+ {12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4},
+ {12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4},
+ {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4},
+};
+
+static struct reg_default ml26124_reg[] = {
+ /* CLOCK control Register */
+ {0x00, 0x00 }, /* Sampling Rate */
+ {0x02, 0x00}, /* PLL NL */
+ {0x04, 0x00}, /* PLLNH */
+ {0x06, 0x00}, /* PLLML */
+ {0x08, 0x00}, /* MLLMH */
+ {0x0a, 0x00}, /* PLLDIV */
+ {0x0c, 0x00}, /* Clock Enable */
+ {0x0e, 0x00}, /* CLK Input/Output Control */
+
+ /* System Control Register */
+ {0x10, 0x00}, /* Software RESET */
+ {0x12, 0x00}, /* Record/Playback Run */
+ {0x14, 0x00}, /* Mic Input/Output control */
+
+ /* Power Management Register */
+ {0x20, 0x00}, /* Reference Power Management */
+ {0x22, 0x00}, /* Input Power Management */
+ {0x24, 0x00}, /* DAC Power Management */
+ {0x26, 0x00}, /* SP-AMP Power Management */
+ {0x28, 0x00}, /* LINEOUT Power Management */
+ {0x2a, 0x00}, /* VIDEO Power Management */
+ {0x2e, 0x00}, /* AC-CMP Power Management */
+
+ /* Analog reference Control Register */
+ {0x30, 0x04}, /* MICBIAS Voltage Control */
+
+ /* Input/Output Amplifier Control Register */
+ {0x32, 0x10}, /* MIC Input Volume */
+ {0x38, 0x00}, /* Mic Boost Volume */
+ {0x3a, 0x33}, /* Speaker AMP Volume */
+ {0x48, 0x00}, /* AMP Volume Control Function Enable */
+ {0x4a, 0x00}, /* Amplifier Volume Fader Control */
+
+ /* Analog Path Control Register */
+ {0x54, 0x00}, /* Speaker AMP Output Control */
+ {0x5a, 0x00}, /* Mic IF Control */
+ {0xe8, 0x01}, /* Mic Select Control */
+
+ /* Audio Interface Control Register */
+ {0x60, 0x00}, /* SAI-Trans Control */
+ {0x62, 0x00}, /* SAI-Receive Control */
+ {0x64, 0x00}, /* SAI Mode select */
+
+ /* DSP Control Register */
+ {0x66, 0x01}, /* Filter Func Enable */
+ {0x68, 0x00}, /* Volume Control Func Enable */
+ {0x6A, 0x00}, /* Mixer & Volume Control*/
+ {0x6C, 0xff}, /* Record Digital Volume */
+ {0x70, 0xff}, /* Playback Digital Volume */
+ {0x72, 0x10}, /* Digital Boost Volume */
+ {0x74, 0xe7}, /* EQ gain Band0 */
+ {0x76, 0xe7}, /* EQ gain Band1 */
+ {0x78, 0xe7}, /* EQ gain Band2 */
+ {0x7A, 0xe7}, /* EQ gain Band3 */
+ {0x7C, 0xe7}, /* EQ gain Band4 */
+ {0x7E, 0x00}, /* HPF2 CutOff*/
+ {0x80, 0x00}, /* EQ Band0 Coef0L */
+ {0x82, 0x00}, /* EQ Band0 Coef0H */
+ {0x84, 0x00}, /* EQ Band0 Coef0L */
+ {0x86, 0x00}, /* EQ Band0 Coef0H */
+ {0x88, 0x00}, /* EQ Band1 Coef0L */
+ {0x8A, 0x00}, /* EQ Band1 Coef0H */
+ {0x8C, 0x00}, /* EQ Band1 Coef0L */
+ {0x8E, 0x00}, /* EQ Band1 Coef0H */
+ {0x90, 0x00}, /* EQ Band2 Coef0L */
+ {0x92, 0x00}, /* EQ Band2 Coef0H */
+ {0x94, 0x00}, /* EQ Band2 Coef0L */
+ {0x96, 0x00}, /* EQ Band2 Coef0H */
+ {0x98, 0x00}, /* EQ Band3 Coef0L */
+ {0x9A, 0x00}, /* EQ Band3 Coef0H */
+ {0x9C, 0x00}, /* EQ Band3 Coef0L */
+ {0x9E, 0x00}, /* EQ Band3 Coef0H */
+ {0xA0, 0x00}, /* EQ Band4 Coef0L */
+ {0xA2, 0x00}, /* EQ Band4 Coef0H */
+ {0xA4, 0x00}, /* EQ Band4 Coef0L */
+ {0xA6, 0x00}, /* EQ Band4 Coef0H */
+
+ /* ALC Control Register */
+ {0xb0, 0x00}, /* ALC Mode */
+ {0xb2, 0x02}, /* ALC Attack Time */
+ {0xb4, 0x03}, /* ALC Decay Time */
+ {0xb6, 0x00}, /* ALC Hold Time */
+ {0xb8, 0x0b}, /* ALC Target Level */
+ {0xba, 0x70}, /* ALC Max/Min Gain */
+ {0xbc, 0x00}, /* Noise Gate Threshold */
+ {0xbe, 0x00}, /* ALC ZeroCross TimeOut */
+
+ /* Playback Limiter Control Register */
+ {0xc0, 0x04}, /* PL Attack Time */
+ {0xc2, 0x05}, /* PL Decay Time */
+ {0xc4, 0x0d}, /* PL Target Level */
+ {0xc6, 0x70}, /* PL Max/Min Gain */
+ {0xc8, 0x10}, /* Playback Boost Volume */
+ {0xca, 0x00}, /* PL ZeroCross TimeOut */
+
+ /* Video Amplifier Control Register */
+ {0xd0, 0x01}, /* VIDEO AMP Gain Control */
+ {0xd2, 0x01}, /* VIDEO AMP Setup 1 */
+ {0xd4, 0x01}, /* VIDEO AMP Control2 */
+};
+
+/* Get sampling rate value of sampling rate setting register (0x0) */
+static inline int get_srate(int rate)
+{
+ int srate;
+
+ switch (rate) {
+ case 16000:
+ srate = 3;
+ break;
+ case 32000:
+ srate = 6;
+ break;
+ case 48000:
+ srate = 8;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return srate;
+}
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+static int ml26124_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+ int i = get_coeff(priv->mclk, params_rate(hw_params));
+
+ priv->substream = substream;
+ priv->rate = params_rate(hw_params);
+
+ if (priv->clk_in) {
+ switch (priv->mclk / params_rate(hw_params)) {
+ case 256:
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 1);
+ break;
+ case 512:
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 2);
+ break;
+ case 1024:
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 3);
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported MCLKI\n");
+ break;
+ }
+ } else {
+ snd_soc_update_bits(codec, ML26124_CLK_CTL,
+ BIT(0) | BIT(1), 0);
+ }
+
+ switch (params_rate(hw_params)) {
+ case 16000:
+ snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+ get_srate(params_rate(hw_params)));
+ snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+ coeff_div[i].pllnl);
+ snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+ coeff_div[i].pllnh);
+ snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+ coeff_div[i].pllml);
+ snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+ coeff_div[i].pllmh);
+ snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+ coeff_div[i].plldiv);
+ break;
+ case 32000:
+ snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+ get_srate(params_rate(hw_params)));
+ snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+ coeff_div[i].pllnl);
+ snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+ coeff_div[i].pllnh);
+ snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+ coeff_div[i].pllml);
+ snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+ coeff_div[i].pllmh);
+ snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+ coeff_div[i].plldiv);
+ break;
+ case 48000:
+ snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+ get_srate(params_rate(hw_params)));
+ snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+ coeff_div[i].pllnl);
+ snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+ coeff_div[i].pllnh);
+ snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+ coeff_div[i].pllml);
+ snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+ coeff_div[i].pllmh);
+ snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+ coeff_div[i].plldiv);
+ break;
+ default:
+ pr_err("%s:this rate is no support for ml26124\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (priv->substream->stream) {
+ case SNDRV_PCM_STREAM_CAPTURE:
+ snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1);
+ break;
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2);
+ break;
+ }
+
+ if (mute)
+ snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+ DVOL_CTL_DVMUTE_ON);
+ else
+ snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+ DVOL_CTL_DVMUTE_OFF);
+
+ return 0;
+}
+
+static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ unsigned char mode;
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ mode = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ mode = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode);
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ml26124_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 ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case ML26124_USE_PLLOUT:
+ priv->clk_in = ML26124_USE_PLLOUT;
+ break;
+ case ML26124_USE_MCLKI:
+ priv->clk_in = ML26124_USE_MCLKI;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ priv->mclk = freq;
+
+ return 0;
+}
+
+static int ml26124_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+ ML26124_R26_MASK, ML26124_BLT_PREAMP_ON);
+ msleep(100);
+ snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+ ML26124_R26_MASK,
+ ML26124_MICBEN_ON | ML26124_BLT_ALL_ON);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* VMID ON */
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+ ML26124_VMID, ML26124_VMID);
+ msleep(500);
+ regcache_sync(priv->regmap);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* VMID OFF */
+ snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+ ML26124_VMID, 0);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ml26124_dai_ops = {
+ .hw_params = ml26124_hw_params,
+ .digital_mute = ml26124_mute,
+ .set_fmt = ml26124_set_dai_fmt,
+ .set_sysclk = ml26124_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver ml26124_dai = {
+ .name = "ml26124-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ML26124_RATES,
+ .formats = ML26124_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = ML26124_RATES,
+ .formats = ML26124_FORMATS,},
+ .ops = &ml26124_dai_ops,
+ .symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int ml26124_suspend(struct snd_soc_codec *codec)
+{
+ ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int ml26124_resume(struct snd_soc_codec *codec)
+{
+ ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define ml26124_suspend NULL
+#define ml26124_resume NULL
+#endif
+
+static int ml26124_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+ codec->control_data = priv->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ /* Software Reset */
+ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
+ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
+
+ ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
+ .probe = ml26124_probe,
+ .suspend = ml26124_suspend,
+ .resume = ml26124_resume,
+ .set_bias_level = ml26124_set_bias_level,
+ .dapm_widgets = ml26124_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
+ .dapm_routes = ml26124_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ml26124_intercon),
+ .controls = ml26124_snd_controls,
+ .num_controls = ARRAY_SIZE(ml26124_snd_controls),
+};
+
+static const struct regmap_config ml26124_i2c_regmap = {
+ .val_bits = 8,
+ .reg_bits = 8,
+ .max_register = ML26124_NUM_REGISTER,
+ .reg_defaults = ml26124_reg,
+ .num_reg_defaults = ARRAY_SIZE(ml26124_reg),
+ .cache_type = REGCACHE_RBTREE,
+ .write_flag_mask = 0x01,
+};
+
+static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ml26124_priv *priv;
+ int ret;
+
+ priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, priv);
+
+ priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap);
+ if (IS_ERR(priv->regmap)) {
+ ret = PTR_ERR(priv->regmap);
+ dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret);
+ return ret;
+ }
+
+ return snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_ml26124, &ml26124_dai, 1);
+}
+
+static __devexit int ml26124_i2c_remove(struct i2c_client *client)
+{
+ struct ml26124_priv *priv = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+ regmap_exit(priv->regmap);
+ return 0;
+}
+
+static const struct i2c_device_id ml26124_i2c_id[] = {
+ { "ml26124", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id);
+
+static struct i2c_driver ml26124_i2c_driver = {
+ .driver = {
+ .name = "ml26124",
+ .owner = THIS_MODULE,
+ },
+ .probe = ml26124_i2c_probe,
+ .remove = __devexit_p(ml26124_i2c_remove),
+ .id_table = ml26124_i2c_id,
+};
+
+module_i2c_driver(ml26124_i2c_driver);
+
+MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>");
+MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h
new file mode 100644
index 00000000000..5ea0cbb8c46
--- /dev/null
+++ b/sound/soc/codecs/ml26124.h
@@ -0,0 +1,184 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
+ *
+ * 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.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#ifndef ML26124_H
+#define ML26124_H
+
+/* Clock Control Register */
+#define ML26124_SMPLING_RATE 0x00
+#define ML26124_PLLNL 0x02
+#define ML26124_PLLNH 0x04
+#define ML26124_PLLML 0x06
+#define ML26124_PLLMH 0x08
+#define ML26124_PLLDIV 0x0a
+#define ML26124_CLK_EN 0x0c
+#define ML26124_CLK_CTL 0x0e
+
+/* System Control Register */
+#define ML26124_SW_RST 0x10
+#define ML26124_REC_PLYBAK_RUN 0x12
+#define ML26124_MIC_TIM 0x14
+
+/* Power Mnagement Register */
+#define ML26124_PW_REF_PW_MNG 0x20
+#define ML26124_PW_IN_PW_MNG 0x22
+#define ML26124_PW_DAC_PW_MNG 0x24
+#define ML26124_PW_SPAMP_PW_MNG 0x26
+#define ML26124_PW_LOUT_PW_MNG 0x28
+#define ML26124_PW_VOUT_PW_MNG 0x2a
+#define ML26124_PW_ZCCMP_PW_MNG 0x2e
+
+/* Analog Reference Control Register */
+#define ML26124_PW_MICBIAS_VOL 0x30
+
+/* Input/Output Amplifier Control Register */
+#define ML26124_PW_MIC_IN_VOL 0x32
+#define ML26124_PW_MIC_BOST_VOL 0x38
+#define ML26124_PW_SPK_AMP_VOL 0x3a
+#define ML26124_PW_AMP_VOL_FUNC 0x48
+#define ML26124_PW_AMP_VOL_FADE 0x4a
+
+/* Analog Path Control Register */
+#define ML26124_SPK_AMP_OUT 0x54
+#define ML26124_MIC_IF_CTL 0x5a
+#define ML26124_MIC_SELECT 0xe8
+
+/* Audio Interface Control Register */
+#define ML26124_SAI_TRANS_CTL 0x60
+#define ML26124_SAI_RCV_CTL 0x62
+#define ML26124_SAI_MODE_SEL 0x64
+
+/* DSP Control Register */
+#define ML26124_FILTER_EN 0x66
+#define ML26124_DVOL_CTL 0x68
+#define ML26124_MIXER_VOL_CTL 0x6a
+#define ML26124_RECORD_DIG_VOL 0x6c
+#define ML26124_PLBAK_DIG_VOL 0x70
+#define ML26124_DIGI_BOOST_VOL 0x72
+#define ML26124_EQ_GAIN_BRAND0 0x74
+#define ML26124_EQ_GAIN_BRAND1 0x76
+#define ML26124_EQ_GAIN_BRAND2 0x78
+#define ML26124_EQ_GAIN_BRAND3 0x7a
+#define ML26124_EQ_GAIN_BRAND4 0x7c
+#define ML26124_HPF2_CUTOFF 0x7e
+#define ML26124_EQBRAND0_F0L 0x80
+#define ML26124_EQBRAND0_F0H 0x82
+#define ML26124_EQBRAND0_F1L 0x84
+#define ML26124_EQBRAND0_F1H 0x86
+#define ML26124_EQBRAND1_F0L 0x88
+#define ML26124_EQBRAND1_F0H 0x8a
+#define ML26124_EQBRAND1_F1L 0x8c
+#define ML26124_EQBRAND1_F1H 0x8e
+#define ML26124_EQBRAND2_F0L 0x90
+#define ML26124_EQBRAND2_F0H 0x92
+#define ML26124_EQBRAND2_F1L 0x94
+#define ML26124_EQBRAND2_F1H 0x96
+#define ML26124_EQBRAND3_F0L 0x98
+#define ML26124_EQBRAND3_F0H 0x9a
+#define ML26124_EQBRAND3_F1L 0x9c
+#define ML26124_EQBRAND3_F1H 0x9e
+#define ML26124_EQBRAND4_F0L 0xa0
+#define ML26124_EQBRAND4_F0H 0xa2
+#define ML26124_EQBRAND4_F1L 0xa4
+#define ML26124_EQBRAND4_F1H 0xa6
+
+/* ALC Control Register */
+#define ML26124_ALC_MODE 0xb0
+#define ML26124_ALC_ATTACK_TIM 0xb2
+#define ML26124_ALC_DECAY_TIM 0xb4
+#define ML26124_ALC_HOLD_TIM 0xb6
+#define ML26124_ALC_TARGET_LEV 0xb8
+#define ML26124_ALC_MAXMIN_GAIN 0xba
+#define ML26124_NOIS_GATE_THRSH 0xbc
+#define ML26124_ALC_ZERO_TIMOUT 0xbe
+
+/* Playback Limiter Control Register */
+#define ML26124_PL_ATTACKTIME 0xc0
+#define ML26124_PL_DECAYTIME 0xc2
+#define ML26124_PL_TARGETTIME 0xc4
+#define ML26124_PL_MAXMIN_GAIN 0xc6
+#define ML26124_PLYBAK_BOST_VOL 0xc8
+#define ML26124_PL_0CROSS_TIMOUT 0xca
+
+/* Video Amplifer Control Register */
+#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0
+#define ML26124_VIDEO_AMP_SETUP1 0xd2
+#define ML26124_VIDEO_AMP_CTL2 0xd4
+
+/* Clock select for machine driver */
+#define ML26124_USE_PLL 0
+#define ML26124_USE_MCLKI_256FS 1
+#define ML26124_USE_MCLKI_512FS 2
+#define ML26124_USE_MCLKI_1024FS 3
+
+/* Register Mask */
+#define ML26124_R0_MASK 0xf
+#define ML26124_R2_MASK 0xff
+#define ML26124_R4_MASK 0x1
+#define ML26124_R6_MASK 0xf
+#define ML26124_R8_MASK 0x3f
+#define ML26124_Ra_MASK 0x1f
+#define ML26124_Rc_MASK 0x1f
+#define ML26124_Re_MASK 0x7
+#define ML26124_R10_MASK 0x1
+#define ML26124_R12_MASK 0x17
+#define ML26124_R14_MASK 0x3f
+#define ML26124_R20_MASK 0x47
+#define ML26124_R22_MASK 0xa
+#define ML26124_R24_MASK 0x2
+#define ML26124_R26_MASK 0x1f
+#define ML26124_R28_MASK 0x2
+#define ML26124_R2a_MASK 0x2
+#define ML26124_R2e_MASK 0x2
+#define ML26124_R30_MASK 0x7
+#define ML26124_R32_MASK 0x3f
+#define ML26124_R38_MASK 0x38
+#define ML26124_R3a_MASK 0x3f
+#define ML26124_R48_MASK 0x3
+#define ML26124_R4a_MASK 0x7
+#define ML26124_R54_MASK 0x2a
+#define ML26124_R5a_MASK 0x3
+#define ML26124_Re8_MASK 0x3
+#define ML26124_R60_MASK 0xff
+#define ML26124_R62_MASK 0xff
+#define ML26124_R64_MASK 0x1
+#define ML26124_R66_MASK 0xff
+#define ML26124_R68_MASK 0x3b
+#define ML26124_R6a_MASK 0xf3
+#define ML26124_R6c_MASK 0xff
+#define ML26124_R70_MASK 0xff
+
+#define ML26124_MCLKEN BIT(0)
+#define ML26124_PLLEN BIT(1)
+#define ML26124_PLLOE BIT(2)
+#define ML26124_MCLKOE BIT(3)
+
+#define ML26124_BLT_ALL_ON 0x1f
+#define ML26124_BLT_PREAMP_ON 0x13
+
+#define ML26124_MICBEN_ON BIT(2)
+
+enum ml26124_regs {
+ ML26124_MCLK = 0,
+};
+
+enum ml26124_clk_in {
+ ML26124_USE_PLLOUT = 0,
+ ML26124_USE_MCLKI,
+};
+
+#endif
diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c
new file mode 100644
index 00000000000..1bf5c74f5f9
--- /dev/null
+++ b/sound/soc/codecs/omap-hdmi.c
@@ -0,0 +1,69 @@
+/*
+ * ALSA SoC codec driver for HDMI audio on OMAP processors.
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@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
+ *
+ */
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#define DRV_NAME "hdmi-audio-codec"
+
+static struct snd_soc_codec_driver omap_hdmi_codec;
+
+static struct snd_soc_dai_driver omap_hdmi_codec_dai = {
+ .name = "omap-hdmi-hifi",
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .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,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ },
+};
+
+static __devinit int omap_hdmi_codec_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec,
+ &omap_hdmi_codec_dai, 1);
+}
+
+static __devexit int omap_hdmi_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver omap_hdmi_codec_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+
+ .probe = omap_hdmi_codec_probe,
+ .remove = __devexit_p(omap_hdmi_codec_remove),
+};
+
+module_platform_driver(omap_hdmi_codec_driver);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 20c324c7c34..960d0e93cce 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -18,7 +18,7 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
-#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -30,6 +30,7 @@
#include "rt5631.h"
struct rt5631_priv {
+ struct regmap *regmap;
int codec_version;
int master;
int sysclk;
@@ -38,33 +39,33 @@ struct rt5631_priv {
int dmic_used_flag;
};
-static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = {
- [RT5631_SPK_OUT_VOL] = 0x8888,
- [RT5631_HP_OUT_VOL] = 0x8080,
- [RT5631_MONO_AXO_1_2_VOL] = 0xa080,
- [RT5631_AUX_IN_VOL] = 0x0808,
- [RT5631_ADC_REC_MIXER] = 0xf0f0,
- [RT5631_VDAC_DIG_VOL] = 0x0010,
- [RT5631_OUTMIXER_L_CTRL] = 0xffc0,
- [RT5631_OUTMIXER_R_CTRL] = 0xffc0,
- [RT5631_AXO1MIXER_CTRL] = 0x88c0,
- [RT5631_AXO2MIXER_CTRL] = 0x88c0,
- [RT5631_DIG_MIC_CTRL] = 0x3000,
- [RT5631_MONO_INPUT_VOL] = 0x8808,
- [RT5631_SPK_MIXER_CTRL] = 0xf8f8,
- [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00,
- [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440,
- [RT5631_SDP_CTRL] = 0x8000,
- [RT5631_MONO_SDP_CTRL] = 0x8000,
- [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010,
- [RT5631_GEN_PUR_CTRL_REG] = 0x0e00,
- [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a,
- [RT5631_MISC_CTRL] = 0x2040,
- [RT5631_DEPOP_FUN_CTRL_2] = 0x8000,
- [RT5631_SOFT_VOL_CTRL] = 0x07e0,
- [RT5631_ALC_CTRL_1] = 0x0206,
- [RT5631_ALC_CTRL_3] = 0x2000,
- [RT5631_PSEUDO_SPATL_CTRL] = 0x0553,
+static const struct reg_default rt5631_reg[] = {
+ { RT5631_SPK_OUT_VOL, 0x8888 },
+ { RT5631_HP_OUT_VOL, 0x8080 },
+ { RT5631_MONO_AXO_1_2_VOL, 0xa080 },
+ { RT5631_AUX_IN_VOL, 0x0808 },
+ { RT5631_ADC_REC_MIXER, 0xf0f0 },
+ { RT5631_VDAC_DIG_VOL, 0x0010 },
+ { RT5631_OUTMIXER_L_CTRL, 0xffc0 },
+ { RT5631_OUTMIXER_R_CTRL, 0xffc0 },
+ { RT5631_AXO1MIXER_CTRL, 0x88c0 },
+ { RT5631_AXO2MIXER_CTRL, 0x88c0 },
+ { RT5631_DIG_MIC_CTRL, 0x3000 },
+ { RT5631_MONO_INPUT_VOL, 0x8808 },
+ { RT5631_SPK_MIXER_CTRL, 0xf8f8 },
+ { RT5631_SPK_MONO_OUT_CTRL, 0xfc00 },
+ { RT5631_SPK_MONO_HP_OUT_CTRL, 0x4440 },
+ { RT5631_SDP_CTRL, 0x8000 },
+ { RT5631_MONO_SDP_CTRL, 0x8000 },
+ { RT5631_STEREO_AD_DA_CLK_CTRL, 0x2010 },
+ { RT5631_GEN_PUR_CTRL_REG, 0x0e00 },
+ { RT5631_INT_ST_IRQ_CTRL_2, 0x071a },
+ { RT5631_MISC_CTRL, 0x2040 },
+ { RT5631_DEPOP_FUN_CTRL_2, 0x8000 },
+ { RT5631_SOFT_VOL_CTRL, 0x07e0 },
+ { RT5631_ALC_CTRL_1, 0x0206 },
+ { RT5631_ALC_CTRL_3, 0x2000 },
+ { RT5631_PSEUDO_SPATL_CTRL, 0x0553 },
};
/**
@@ -96,8 +97,7 @@ static int rt5631_reset(struct snd_soc_codec *codec)
return snd_soc_write(codec, RT5631_RESET, 0);
}
-static int rt5631_volatile_register(struct snd_soc_codec *codec,
- unsigned int reg)
+static bool rt5631_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case RT5631_RESET:
@@ -111,8 +111,7 @@ static int rt5631_volatile_register(struct snd_soc_codec *codec,
}
}
-static int rt5631_readable_register(struct snd_soc_codec *codec,
- unsigned int reg)
+static bool rt5631_readable_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case RT5631_RESET:
@@ -1361,8 +1360,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk)
static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
int timesofbclk = 32, coeff;
unsigned int iface = 0;
@@ -1544,6 +1542,8 @@ static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
static int rt5631_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
+
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
@@ -1561,8 +1561,8 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3,
RT5631_PWR_FAST_VREF_CTRL,
RT5631_PWR_FAST_VREF_CTRL);
- codec->cache_only = false;
- snd_soc_cache_sync(codec);
+ regcache_cache_only(rt5631->regmap, false);
+ regcache_sync(rt5631->regmap);
}
break;
@@ -1587,7 +1587,9 @@ static int rt5631_probe(struct snd_soc_codec *codec)
unsigned int val;
int ret;
- ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+ codec->control_data = rt5631->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
@@ -1698,12 +1700,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = {
.suspend = rt5631_suspend,
.resume = rt5631_resume,
.set_bias_level = rt5631_set_bias_level,
- .reg_cache_size = RT5631_VENDOR_ID2 + 1,
- .reg_word_size = sizeof(u16),
- .reg_cache_default = rt5631_reg,
- .volatile_register = rt5631_volatile_register,
- .readable_register = rt5631_readable_register,
- .reg_cache_step = 1,
.controls = rt5631_snd_controls,
.num_controls = ARRAY_SIZE(rt5631_snd_controls),
.dapm_widgets = rt5631_dapm_widgets,
@@ -1718,6 +1714,18 @@ static const struct i2c_device_id rt5631_i2c_id[] = {
};
MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id);
+static const struct regmap_config rt5631_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 16,
+
+ .readable_reg = rt5631_readable_register,
+ .volatile_reg = rt5631_volatile_register,
+ .max_register = RT5631_VENDOR_ID2,
+ .reg_defaults = rt5631_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt5631_reg),
+ .cache_type = REGCACHE_RBTREE,
+};
+
static int rt5631_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1731,6 +1739,10 @@ static int rt5631_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, rt5631);
+ rt5631->regmap = devm_regmap_init_i2c(i2c, &rt5631_regmap_config);
+ if (IS_ERR(rt5631->regmap))
+ return PTR_ERR(rt5631->regmap);
+
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631,
rt5631_dai, ARRAY_SIZE(rt5631_dai));
return ret;
@@ -1752,17 +1764,7 @@ static struct i2c_driver rt5631_i2c_driver = {
.id_table = rt5631_i2c_id,
};
-static int __init rt5631_modinit(void)
-{
- return i2c_add_driver(&rt5631_i2c_driver);
-}
-module_init(rt5631_modinit);
-
-static void __exit rt5631_modexit(void)
-{
- i2c_del_driver(&rt5631_i2c_driver);
-}
-module_exit(rt5631_modexit);
+module_i2c_driver(rt5631_i2c_driver);
MODULE_DESCRIPTION("ASoC RT5631 driver");
MODULE_AUTHOR("flove <flove@realtek.com>");
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 8e92fb88ed0..8af6a5245b1 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = {
static struct regulator_init_data ldo_init_data = {
.constraints = {
- .min_uV = 850000,
- .max_uV = 1600000,
+ .min_uV = 1200000,
+ .max_uV = 1200000,
.valid_modes_mask = REGULATOR_MODE_NORMAL,
.valid_ops_mask = REGULATOR_CHANGE_STATUS,
},
@@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HP_OUT"),
SND_SOC_DAPM_OUTPUT("LINE_OUT"),
- SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
- mic_bias_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
+ mic_bias_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
@@ -665,8 +665,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
int channels = params_channels(params);
int i2s_ctl = 0;
@@ -809,6 +808,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
{
struct ldo_regulator *ldo;
struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+ struct regulator_config config = { };
ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
@@ -832,8 +832,11 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
ldo->codec_data = codec;
ldo->voltage = voltage;
- ldo->dev = regulator_register(&ldo->desc, codec->dev,
- init_data, ldo, NULL);
+ config.dev = codec->dev;
+ config.driver_data = ldo;
+ config.init_data = init_data;
+
+ ldo->dev = regulator_register(&ldo->desc, &config);
if (IS_ERR(ldo->dev)) {
int ret = PTR_ERR(ldo->dev);
@@ -1451,17 +1454,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
.id_table = sgtl5000_id,
};
-static int __init sgtl5000_modinit(void)
-{
- return i2c_add_driver(&sgtl5000_i2c_driver);
-}
-module_init(sgtl5000_modinit);
-
-static void __exit sgtl5000_exit(void)
-{
- i2c_del_driver(&sgtl5000_i2c_driver);
-}
-module_exit(sgtl5000_exit);
+module_i2c_driver(sgtl5000_i2c_driver);
MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>");
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index de2b20544ce..079066fef42 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -33,6 +33,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -43,8 +44,6 @@
#include "ssm2602.h"
-#define SSM2602_VERSION "0.1"
-
enum ssm2602_type {
SSM2602,
SSM2604,
@@ -53,10 +52,12 @@ enum ssm2602_type {
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
- enum snd_soc_control_type control_type;
+ struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
+ struct regmap *regmap;
+
enum ssm2602_type type;
unsigned int clk_out_pwr;
};
@@ -73,7 +74,6 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
0x0000, 0x0000
};
-#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0)
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
@@ -195,6 +195,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = {
{"ADC", NULL, "Line Input"},
};
+static const unsigned int ssm2602_rates_12288000[] = {
+ 8000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
+ .list = ssm2602_rates_12288000,
+ .count = ARRAY_SIZE(ssm2602_rates_12288000),
+};
+
+static const unsigned int ssm2602_rates_11289600[] = {
+ 8000, 44100, 88200,
+};
+
+static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
+ .list = ssm2602_rates_11289600,
+ .count = ARRAY_SIZE(ssm2602_rates_11289600),
+};
+
struct ssm2602_coeff {
u32 mclk;
u32 rate;
@@ -254,11 +272,10 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
- u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3;
int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
+ unsigned int iface;
if (substream == ssm2602->slave_substream) {
dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n");
@@ -268,31 +285,34 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
if (srate < 0)
return srate;
- snd_soc_write(codec, SSM2602_SRATE, srate);
+ regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
+ iface = 0x0;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- iface |= 0x0004;
+ iface = 0x4;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- iface |= 0x0008;
+ iface = 0x8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- iface |= 0x000c;
+ iface = 0xc;
break;
+ default:
+ return -EINVAL;
}
- snd_soc_write(codec, SSM2602_IFACE, iface);
+ regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
+ IFACE_AUDIO_DATA_LEN, iface);
return 0;
}
static int ssm2602_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_pcm_runtime *master_runtime;
@@ -322,14 +342,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
} else
ssm2602->master_substream = substream;
+ if (ssm2602->sysclk_constraints) {
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ ssm2602->sysclk_constraints);
+ }
+
return 0;
}
static void ssm2602_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
if (ssm2602->master_substream == substream)
@@ -341,14 +366,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
- struct snd_soc_codec *codec = dai->codec;
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
if (mute)
- snd_soc_update_bits(codec, SSM2602_APDIGI,
+ regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE,
APDIGI_ENABLE_DAC_MUTE);
else
- snd_soc_update_bits(codec, SSM2602_APDIGI,
+ regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE, 0);
return 0;
}
@@ -364,16 +389,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
return -EINVAL;
switch (freq) {
- case 11289600:
- case 12000000:
case 12288000:
- case 16934400:
case 18432000:
- ssm2602->sysclk = freq;
+ ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
+ break;
+ case 11289600:
+ case 16934400:
+ ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
+ break;
+ case 12000000:
+ ssm2602->sysclk_constraints = NULL;
break;
default:
return -EINVAL;
}
+ ssm2602->sysclk = freq;
} else {
unsigned int mask;
@@ -393,7 +423,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
else
ssm2602->clk_out_pwr &= ~mask;
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
}
@@ -403,8 +433,8 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
- u16 iface = 0;
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec);
+ unsigned int iface = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -455,7 +485,7 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
/* set iface */
- snd_soc_write(codec, SSM2602_IFACE, iface);
+ regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
return 0;
}
@@ -467,7 +497,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid on, osc and clkout on if enabled */
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
ssm2602->clk_out_pwr);
break;
@@ -475,13 +505,13 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
PWR_CLK_OUT_PDN | PWR_OSC_PDN);
break;
case SND_SOC_BIAS_OFF:
/* everything off */
- snd_soc_update_bits(codec, SSM2602_PWR,
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF, PWR_POWER_OFF);
break;
@@ -540,12 +570,13 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
static int ssm2602_probe(struct snd_soc_codec *codec)
{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- snd_soc_update_bits(codec, SSM2602_LOUT1V,
+ regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
- snd_soc_update_bits(codec, SSM2602_ROUT1V,
+ regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
@@ -581,27 +612,26 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret;
- pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
-
- ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type);
+ codec->control_data = ssm2602->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
- ret = ssm2602_reset(codec);
+ ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
/* set the update bits */
- snd_soc_update_bits(codec, SSM2602_LINVOL,
+ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
- snd_soc_update_bits(codec, SSM2602_RINVOL,
+ regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
/*select Line in as default input*/
- snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
+ regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
switch (ssm2602->type) {
@@ -634,9 +664,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
.set_bias_level = ssm2602_set_bias_level,
- .reg_cache_size = ARRAY_SIZE(ssm2602_reg),
- .reg_word_size = sizeof(u16),
- .reg_cache_default = ssm2602_reg,
.controls = ssm260x_snd_controls,
.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
@@ -646,6 +673,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
};
+static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
+{
+ return reg == SSM2602_RESET;
+}
+
+static const struct regmap_config ssm2602_regmap_config = {
+ .val_bits = 9,
+ .reg_bits = 7,
+
+ .max_register = SSM2602_RESET,
+ .volatile_reg = ssm2602_register_volatile,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults_raw = ssm2602_reg,
+ .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
+};
+
#if defined(CONFIG_SPI_MASTER)
static int __devinit ssm2602_spi_probe(struct spi_device *spi)
{
@@ -658,9 +702,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi)
return -ENOMEM;
spi_set_drvdata(spi, ssm2602);
- ssm2602->control_type = SND_SOC_SPI;
ssm2602->type = SSM2602;
+ ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config);
+ if (IS_ERR(ssm2602->regmap))
+ return PTR_ERR(ssm2602->regmap);
+
ret = snd_soc_register_codec(&spi->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
return ret;
@@ -701,9 +748,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
i2c_set_clientdata(i2c, ssm2602);
- ssm2602->control_type = SND_SOC_I2C;
ssm2602->type = id->driver_data;
+ ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config);
+ if (IS_ERR(ssm2602->regmap))
+ return PTR_ERR(ssm2602->regmap);
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
return ret;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 7db6fa51502..8d717f4b5a8 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
unsigned int rate;
int i, mcs = -1, ir = -1;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index df1e07ffac3..31762ebdd77 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -34,8 +34,6 @@
#include "tlv320aic23.h"
-#define AIC23_VERSION "0.1"
-
/*
* AIC23 register cache
*/
@@ -325,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 iface_reg;
int ret;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
@@ -371,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
/* set active */
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
@@ -383,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
/* deactivate */
@@ -548,8 +543,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec)
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
int ret;
- printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
-
ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 802064b5030..85944e95357 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
int fsref, divisor, wlen, pval, jval, dval, qval;
u16 reg;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8d20f6ec20f..64d2a4fa34b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -802,8 +802,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec =rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
@@ -1161,24 +1160,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
- int headset_debounce, int button_debounce)
-{
- u8 val;
-
- val = ((detect & AIC3X_HEADSET_DETECT_MASK)
- << AIC3X_HEADSET_DETECT_SHIFT) |
- ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
- << AIC3X_HEADSET_DEBOUNCE_SHIFT) |
- ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
- << AIC3X_BUTTON_DEBOUNCE_SHIFT);
-
- if (detect & AIC3X_HEADSET_DETECT_MASK)
- val |= AIC3X_HEADSET_DETECT_ENABLED;
-
- snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
-}
-
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 4587ddd0fbf..0dd41077ab7 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -62,8 +62,10 @@
#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
(((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate))))
-static void dac33_calculate_times(struct snd_pcm_substream *substream);
-static int dac33_prepare_chip(struct snd_pcm_substream *substream);
+static void dac33_calculate_times(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec);
+static int dac33_prepare_chip(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec);
enum dac33_state {
DAC33_IDLE = 0,
@@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (likely(dac33->substream)) {
- dac33_calculate_times(dac33->substream);
- dac33_prepare_chip(dac33->substream);
+ dac33_calculate_times(dac33->substream, w->codec);
+ dac33_prepare_chip(dac33->substream, w->codec);
}
break;
case SND_SOC_DAPM_POST_PMD:
@@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec)
static int dac33_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
/* Stream started, save the substream pointer */
@@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream,
static void dac33_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
dac33->substream = NULL;
@@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
/* Check parameters for validity */
@@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
* writes happens in different order, than dac33 might end up in unknown state.
* Use the known, working sequence of register writes to initialize the dac33.
*/
-static int dac33_prepare_chip(struct snd_pcm_substream *substream)
+static int dac33_prepare_chip(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned int oscset, ratioset, pwr_ctrl, reg_tmp;
u8 aictrl_a, aictrl_b, fifoctrl_a;
@@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
return 0;
}
-static void dac33_calculate_times(struct snd_pcm_substream *substream)
+static void dac33_calculate_times(struct snd_pcm_substream *substream,
+ struct snd_soc_codec *codec)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned int period_size = substream->runtime->period_size;
unsigned int rate = substream->runtime->rate;
@@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
@@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
unsigned long long t0, t1, t_now;
unsigned int time_delta, uthr;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 170cf9a8fc7..391fcfc7b63 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
static int twl4030_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
if (twl4030->master_substream) {
@@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
static void twl4030_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
if (twl4030->master_substream == substream)
@@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 mode, old_mode, format, old_format;
@@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 mode;
@@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
/* Enable voice digital filters */
twl4030_voice_enable(codec, substream->stream, 0);
@@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
u8 old_mode, mode;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index dc7509b9d53..a36e9fcdf18 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -46,17 +46,6 @@
#define TWL6040_OUTHF_0dB 0x03
#define TWL6040_OUTHF_M52dB 0x1D
-#define TWL6040_RAMP_NONE 0
-#define TWL6040_RAMP_UP 1
-#define TWL6040_RAMP_DOWN 2
-
-#define TWL6040_HSL_VOL_MASK 0x0F
-#define TWL6040_HSL_VOL_SHIFT 0
-#define TWL6040_HSR_VOL_MASK 0xF0
-#define TWL6040_HSR_VOL_SHIFT 4
-#define TWL6040_HF_VOL_MASK 0x1F
-#define TWL6040_HF_VOL_SHIFT 0
-
/* Shadow register used by the driver */
#define TWL6040_REG_SW_SHADOW 0x2F
#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1)
@@ -64,18 +53,6 @@
/* TWL6040_REG_SW_SHADOW (0x2F) fields */
#define TWL6040_EAR_PATH_ENABLE 0x01
-struct twl6040_output {
- u16 active;
- u16 left_vol;
- u16 right_vol;
- u16 left_step;
- u16 right_step;
- unsigned int step_delay;
- u16 ramp;
- struct delayed_work work;
- struct completion ramp_done;
-};
-
struct twl6040_jack_data {
struct snd_soc_jack *jack;
struct delayed_work work;
@@ -100,8 +77,6 @@ struct twl6040_data {
struct snd_soc_codec *codec;
struct workqueue_struct *workqueue;
struct mutex mutex;
- struct twl6040_output headset;
- struct twl6040_output handsfree;
};
/*
@@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec)
}
}
-/*
- * Ramp HS PGA volume to minimise pops at stream startup and shutdown.
- */
-static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec,
- unsigned int left_step, unsigned int right_step)
-{
-
- struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *headset = &priv->headset;
- int left_complete = 0, right_complete = 0;
- u8 reg, val;
-
- /* left channel */
- left_step = (left_step > 0xF) ? 0xF : left_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
- val = (~reg & TWL6040_HSL_VOL_MASK);
-
- if (headset->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < headset->left_vol) {
- if (val + left_step > headset->left_vol)
- val = headset->left_vol;
- else
- val += left_step;
-
- reg &= ~TWL6040_HSL_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN,
- (reg | (~val & TWL6040_HSL_VOL_MASK)));
- } else {
- left_complete = 1;
- }
- } else if (headset->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0x0) {
- if ((int)val - (int)left_step < 0)
- val = 0;
- else
- val -= left_step;
-
- reg &= ~TWL6040_HSL_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN, reg |
- (~val & TWL6040_HSL_VOL_MASK));
- } else {
- left_complete = 1;
- }
- }
-
- /* right channel */
- right_step = (right_step > 0xF) ? 0xF : right_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
- val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT;
-
- if (headset->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < headset->right_vol) {
- if (val + right_step > headset->right_vol)
- val = headset->right_vol;
- else
- val += right_step;
-
- reg &= ~TWL6040_HSR_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN,
- (reg | (~val << TWL6040_HSR_VOL_SHIFT)));
- } else {
- right_complete = 1;
- }
- } else if (headset->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0x0) {
- if ((int)val - (int)right_step < 0)
- val = 0;
- else
- val -= right_step;
-
- reg &= ~TWL6040_HSR_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HSGAIN,
- reg | (~val << TWL6040_HSR_VOL_SHIFT));
- } else {
- right_complete = 1;
- }
- }
-
- return left_complete & right_complete;
-}
-
-/*
- * Ramp HF PGA volume to minimise pops at stream startup and shutdown.
- */
-static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec,
- unsigned int left_step, unsigned int right_step)
-{
- struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *handsfree = &priv->handsfree;
- int left_complete = 0, right_complete = 0;
- u16 reg, val;
-
- /* left channel */
- left_step = (left_step > 0x1D) ? 0x1D : left_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN);
- reg = 0x1D - reg;
- val = (reg & TWL6040_HF_VOL_MASK);
- if (handsfree->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < handsfree->left_vol) {
- if (val + left_step > handsfree->left_vol)
- val = handsfree->left_vol;
- else
- val += left_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFLGAIN,
- reg | (0x1D - val));
- } else {
- left_complete = 1;
- }
- } else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0) {
- if ((int)val - (int)left_step < 0)
- val = 0;
- else
- val -= left_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFLGAIN,
- reg | (0x1D - val));
- } else {
- left_complete = 1;
- }
- }
-
- /* right channel */
- right_step = (right_step > 0x1D) ? 0x1D : right_step;
- reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN);
- reg = 0x1D - reg;
- val = (reg & TWL6040_HF_VOL_MASK);
- if (handsfree->ramp == TWL6040_RAMP_UP) {
- /* ramp step up */
- if (val < handsfree->right_vol) {
- if (val + right_step > handsfree->right_vol)
- val = handsfree->right_vol;
- else
- val += right_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFRGAIN,
- reg | (0x1D - val));
- } else {
- right_complete = 1;
- }
- } else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
- /* ramp step down */
- if (val > 0) {
- if ((int)val - (int)right_step < 0)
- val = 0;
- else
- val -= right_step;
-
- reg &= ~TWL6040_HF_VOL_MASK;
- twl6040_write(codec, TWL6040_REG_HFRGAIN,
- reg | (0x1D - val));
- }
- }
-
- return left_complete & right_complete;
-}
-
-/*
- * This work ramps both output PGAs at stream start/stop time to
- * minimise pop associated with DAPM power switching.
- */
-static void twl6040_pga_hs_work(struct work_struct *work)
-{
- struct twl6040_data *priv =
- container_of(work, struct twl6040_data, headset.work.work);
- struct snd_soc_codec *codec = priv->codec;
- struct twl6040_output *headset = &priv->headset;
- int i, headset_complete;
-
- /* do we need to ramp at all ? */
- if (headset->ramp == TWL6040_RAMP_NONE)
- return;
-
- /* HS PGA gain range: 0x0 - 0xf (0 - 15) */
- for (i = 0; i < 16; i++) {
- headset_complete = twl6040_hs_ramp_step(codec,
- headset->left_step,
- headset->right_step);
-
- /* ramp finished ? */
- if (headset_complete)
- break;
-
- schedule_timeout_interruptible(
- msecs_to_jiffies(headset->step_delay));
- }
-
- if (headset->ramp == TWL6040_RAMP_DOWN) {
- headset->active = 0;
- complete(&headset->ramp_done);
- } else {
- headset->active = 1;
- }
- headset->ramp = TWL6040_RAMP_NONE;
-}
-
-static void twl6040_pga_hf_work(struct work_struct *work)
-{
- struct twl6040_data *priv =
- container_of(work, struct twl6040_data, handsfree.work.work);
- struct snd_soc_codec *codec = priv->codec;
- struct twl6040_output *handsfree = &priv->handsfree;
- int i, handsfree_complete;
-
- /* do we need to ramp at all ? */
- if (handsfree->ramp == TWL6040_RAMP_NONE)
- return;
-
- /*
- * HF PGA gain range: 0x00 - 0x1d (0 - 29) */
- for (i = 0; i < 30; i++) {
- handsfree_complete = twl6040_hf_ramp_step(codec,
- handsfree->left_step,
- handsfree->right_step);
-
- /* ramp finished ? */
- if (handsfree_complete)
- break;
-
- schedule_timeout_interruptible(
- msecs_to_jiffies(handsfree->step_delay));
- }
-
-
- if (handsfree->ramp == TWL6040_RAMP_DOWN) {
- handsfree->active = 0;
- complete(&handsfree->ramp_done);
- } else
- handsfree->active = 1;
- handsfree->ramp = TWL6040_RAMP_NONE;
-}
-
-static int out_drv_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = w->codec;
- struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *out;
- struct delayed_work *work;
-
- switch (w->shift) {
- case 2: /* Headset output driver */
- out = &priv->headset;
- work = &out->work;
- /*
- * Make sure, that we do not mess up variables for already
- * executing work.
- */
- cancel_delayed_work_sync(work);
-
- out->left_step = priv->hs_left_step;
- out->right_step = priv->hs_right_step;
- out->step_delay = 5; /* 5 ms between volume ramp steps */
- break;
- case 4: /* Handsfree output driver */
- out = &priv->handsfree;
- work = &out->work;
- /*
- * Make sure, that we do not mess up variables for already
- * executing work.
- */
- cancel_delayed_work_sync(work);
-
- out->left_step = priv->hf_left_step;
- out->right_step = priv->hf_right_step;
- out->step_delay = 5; /* 5 ms between volume ramp steps */
- break;
- default:
- return -1;
- }
-
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- if (out->active)
- break;
-
- /* don't use volume ramp for power-up */
- out->ramp = TWL6040_RAMP_UP;
- out->left_step = out->left_vol;
- out->right_step = out->right_vol;
-
- queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1));
- break;
-
- case SND_SOC_DAPM_PRE_PMD:
- if (!out->active)
- break;
-
- /* use volume ramp for power-down */
- out->ramp = TWL6040_RAMP_DOWN;
- INIT_COMPLETION(out->ramp_done);
-
- queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1));
-
- wait_for_completion_timeout(&out->ramp_done,
- msecs_to_jiffies(2000));
- break;
- }
-
- return 0;
-}
-
/* set headset dac and driver power mode */
static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
{
@@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data)
return IRQ_HANDLED;
}
-static int twl6040_put_volsw(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *out = NULL;
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- int ret;
-
- /* For HS and HF we shadow the values and only actually write
- * them out when active in order to ensure the amplifier comes on
- * as quietly as possible. */
- switch (mc->reg) {
- case TWL6040_REG_HSGAIN:
- out = &twl6040_priv->headset;
- break;
- case TWL6040_REG_HFLGAIN:
- out = &twl6040_priv->handsfree;
- break;
- default:
- dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n",
- __func__, mc->reg);
- return -EINVAL;
- }
-
- out->left_vol = ucontrol->value.integer.value[0];
- out->right_vol = ucontrol->value.integer.value[1];
- if (!out->active)
- return 1;
-
- ret = snd_soc_put_volsw(kcontrol, ucontrol);
- if (ret < 0)
- return ret;
-
- return 1;
-}
-
-static int twl6040_get_volsw(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
- struct twl6040_output *out = &twl6040_priv->headset;
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
-
- switch (mc->reg) {
- case TWL6040_REG_HSGAIN:
- out = &twl6040_priv->headset;
- break;
- case TWL6040_REG_HFLGAIN:
- out = &twl6040_priv->handsfree;
- break;
- default:
- dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n",
- __func__, mc->reg);
- return -EINVAL;
- }
-
- ucontrol->value.integer.value[0] = out->left_vol;
- ucontrol->value.integer.value[1] = out->right_vol;
- return 0;
-}
-
static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
/* Playback gains */
- SOC_DOUBLE_EXT_TLV("Headset Playback Volume",
- TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw,
- twl6040_put_volsw, hs_tlv),
- SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume",
- TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1,
- twl6040_get_volsw, twl6040_put_volsw, hf_tlv),
+ SOC_DOUBLE_TLV("Headset Playback Volume",
+ TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
+ SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+ TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv),
SOC_SINGLE_TLV("Earphone Playback Volume",
TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv),
@@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
&auxr_switch_control),
/* Analog playback drivers */
- SND_SOC_DAPM_OUT_DRV_E("HF Left Driver",
- TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_OUT_DRV_E("HF Right Driver",
- TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_OUT_DRV_E("HS Left Driver",
- TWL6040_REG_HSLCTL, 2, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_OUT_DRV_E("HS Right Driver",
- TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
- out_drv_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_OUT_DRV("HF Left Driver",
+ TWL6040_REG_HFLCTL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HF Right Driver",
+ TWL6040_REG_HFRCTL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HS Left Driver",
+ TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HS Right Driver",
+ TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV_E("Earphone Driver",
TWL6040_REG_EARCTL, 0, 0, NULL, 0,
twl6040_ep_drv_event,
@@ -1339,8 +927,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
static int twl6040_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
snd_pcm_hw_constraint_list(substream->runtime, 0,
@@ -1354,8 +941,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int rate;
@@ -1391,8 +977,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream,
static int twl6040_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret;
@@ -1570,14 +1155,9 @@ static int twl6040_probe(struct snd_soc_codec *codec)
}
INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work);
- INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work);
- INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work);
mutex_init(&priv->mutex);
- init_completion(&priv->headset.ramp_done);
- init_completion(&priv->handsfree.ramp_done);
-
ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler,
0, "twl6040_irq_plug", codec);
if (ret) {
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 797b0dde2c6..6c3d43b8ee8 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute)
static int uda134x_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec =rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
struct snd_pcm_runtime *master_runtime;
@@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream,
static void uda134x_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
if (uda134x->master_substream == substream)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 4f1b23d7e40..2502214b84a 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec);
int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
@@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* set WSPLL power and divider if running from this clock */
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 3d868dc4009..7b24d6d192e 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = {
static int wl1273_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_codec *codec = dai->codec;
struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
switch (wl1273->mode) {
@@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec);
struct wl1273_core *core = wl1273->core;
unsigned int rate, width, r;
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index aefb4f89be0..e0b51e9f8b1 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -79,22 +79,65 @@ static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = {
{ "WM1250 Output", NULL, "DAC" },
};
+static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct wm1250_priv *wm1250 = snd_soc_codec_get_drvdata(dai->codec);
+
+ switch (params_rate(params)) {
+ case 8000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 1);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 1);
+ break;
+ case 16000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 0);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 1);
+ break;
+ case 32000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 1);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 0);
+ break;
+ case 64000:
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio,
+ 0);
+ gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio,
+ 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops wm1250_ev1_ops = {
+ .hw_params = wm1250_ev1_hw_params,
+};
+
static struct snd_soc_dai_driver wm1250_ev1_dai = {
.name = "wm1250-ev1",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
- .channels_max = 1,
+ .channels_max = 2,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
- .channels_max = 1,
+ .channels_max = 2,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
+ .ops = &wm1250_ev1_ops,
};
static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = {
@@ -215,23 +258,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = {
.id_table = wm1250_ev1_i2c_id,
};
-static int __init wm1250_ev1_modinit(void)
-{
- int ret = 0;
-
- ret = i2c_add_driver(&wm1250_ev1_i2c_driver);
- if (ret != 0)
- pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret);
-
- return ret;
-}
-module_init(wm1250_ev1_modinit);
-
-static void __exit wm1250_ev1_exit(void)
-{
- i2c_del_driver(&wm1250_ev1_i2c_driver);
-