From 26047e2d6bde5b2e1b791e0ec1c3234894fdf3fa Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 30 Nov 2012 11:28:55 +0100 Subject: [PATCH 01/30] ASoC: cs4271: fix sparse warning Make the flag in the pdata of type bool to fix a sparse warning. Signed-off-by: Daniel Mack Reported-by: Fengguang Wu Signed-off-by: Mark Brown --- include/sound/cs4271.h | 2 +- sound/soc/codecs/cs4271.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 6d9e15ed1dcf..dd8c48d14ed9 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,7 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int amutec_eq_bmutec:1; /* flag to enable AMUTEC=BMUTEC */ + bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2ac5fe61a96c..f07d1b7d6c69 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,7 +474,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int amutec_eq_bmutec = 0; + bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { @@ -483,7 +483,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (!of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) - amutec_eq_bmutec = 1; + amutec_eq_bmutec = true; } #endif From b8455c9f6f661fb9bcb791370478d6d15c9bf2b3 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 30 Nov 2012 11:28:56 +0100 Subject: [PATCH 02/30] ASoC: cs4271: fix property check The driver had the property check for 'cirrus,amutec_eq_bmutec' the wrong way around. That happens if you misspell the property in the bindings during tests. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f07d1b7d6c69..449a98b8874c 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -481,7 +481,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); - if (!of_get_property(codec->dev->of_node, + if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; } From 08b27848da620f206a8b6d80f26184485dd7aa40 Mon Sep 17 00:00:00 2001 From: Patrick Lai Date: Wed, 19 Dec 2012 19:36:02 -0800 Subject: [PATCH 03/30] ASoC: pcm: allow backend hardware to be freed in pause state When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d7711fce119b..cf191e6aebbe 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; From ad1937cdd59c412097ec2bb8f38c12a5640f1f9a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Dec 2012 16:17:25 +0800 Subject: [PATCH 04/30] ASoC: sta529: Fix update register bits in sta529_set_dai_fmt Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin Acked-by: Rajeev Kumar Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sta529.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4f0b2d..40c07be9b581 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } From 9bde4f0b1c83d1129a9fc8ec5b2611ba6dab1215 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 19 Dec 2012 16:05:00 +0000 Subject: [PATCH 05/30] ASoC: core: Fix SOC_DOUBLE_RANGE() macros Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 10 ++++++---- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++-- 2 files changed, 36 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 769e27c774a3..bc56738cb109 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -58,8 +58,9 @@ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -88,8 +89,9 @@ .info = snd_soc_info_volsw_range, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 91d592ff67b7..e0d4630bfb4f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2917,7 +2917,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2941,12 +2941,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2954,7 +2956,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2974,11 +2990,13 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -2988,6 +3006,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); From 2a5f431592343b78896013b055582f94c12a5049 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 21 Dec 2012 16:28:37 +0800 Subject: [PATCH 06/30] ASoC: wm2200: Fix setting dai format in wm2200_set_fmt According to the defines in wm2200.h: /* * R1284 (0x504) - Audio IF 1_5 */ We should not left shift 1 bit for fmt_val when setting dai format. Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31df77e0..a12fc2fa3971 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1626,7 +1626,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } From 5db1bc1892aa70d378ea7563852ae87f3772519b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 22 Dec 2012 10:38:14 -0200 Subject: [PATCH 07/30] ASoC: soc-core: Remove unused 'ret' variable commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced the following build warning: sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable] Remove the unused 'ret' variable. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e0d4630bfb4f..29046e750ad7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2996,7 +2996,6 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; From b50684da6cd5a9ec47ea5556bc9f5461c1d7017c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 23 Dec 2012 15:45:31 -0200 Subject: [PATCH 08/30] ASoC: sgtl5000: Fix maximum value for microphone gain sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675cd8e1c..92bbfec9b107 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP From ec20f2f8d3714cfb491a138eb4c0c720577d49e6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 21 Dec 2012 09:19:20 +0800 Subject: [PATCH 09/30] ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt() The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin Tested-by: Omair M. Abdullah Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d40a49..c0d203bfe5f0 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,7 +1218,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); From 7110a287ff2b1f3780905d1686a1a4edccb95133 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Dec 2012 23:29:42 +0800 Subject: [PATCH 10/30] ASoC: arizona: Do proper shift for setting AIF rate ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b9d0e6..38248a7a95e3 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -714,7 +714,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); From a8c02db029385fb4426e0396e108ab23cd08f384 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Dec 2012 14:05:01 +0000 Subject: [PATCH 11/30] ASoC: arizona: Correct FLL source definitions The FLL source constants were numbered as a simple enumeration but were being used in the code as direct values to be written to the registers. Renumber the constants to reflect the usage. Reported-by: Ryo Tsutsui Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 41dae1ed3b71..4deebeb07177 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 From 88ac43924b396e524288570ed3b11e8c94c1191f Mon Sep 17 00:00:00 2001 From: "MR.Swami.Reddy@ti.com" Date: Fri, 7 Dec 2012 17:00:10 +0530 Subject: [PATCH 12/30] ASoC: lm49453: Fix adc, mic and sidetone volume ranges Add adc, mic, sidetone volume ranges and appropriately added the controls. Fix the DAC HP/EP/LS/LO/HA maximum gain values. Signed-off-by: MR Swami Reddy Tested-by: Vinod Koul -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 43 +++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index c0d203bfe5f0..f34832851785 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -525,36 +525,41 @@ 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 DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 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), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_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_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -569,16 +574,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 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), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_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), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_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), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_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), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv), From 9dc754dfa78ca4ef9a117245e5ae3b9b7312d59d Mon Sep 17 00:00:00 2001 From: "MR.Swami.Reddy@ti.com" Date: Fri, 7 Dec 2012 17:06:54 +0530 Subject: [PATCH 13/30] ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B Update lm49453_reg_defs values as per LM49453 HW revision-B Signed-off-by: M R Swami Reddy Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 61 ++++++++++---------------------------- 1 file changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index f34832851785..e19490cfb3a8 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 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 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 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 }, + { 250, 0x00 }, { 255, 0x00 }, }; From 3271a4fc7daeb489bfe1730023c166065e6fb0e7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 20 Dec 2012 16:53:16 +0800 Subject: [PATCH 14/30] ASoC: cs42l52: Catch no-match case in cs42l52_get_clk In the case of no-match, return -EINVAL instead of 0. Since we assign i to ret in the for loop, ret always less than ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c69499e..9811a5478c87 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = { static int cs42l52_get_clk(int mclk, int rate) { - int i, ret = 0; + int i, ret = -EINVAL; u_int mclk1, mclk2 = 0; for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { @@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) - return -EINVAL; return ret; } From 865df9cb122d9e5ecbbb7056f2c9c64933bf8dd0 Mon Sep 17 00:00:00 2001 From: Chuansheng Liu Date: Wed, 26 Dec 2012 00:56:05 +0800 Subject: [PATCH 15/30] ASoC: core: fix the memory leak in case of device_add() failure After called device_initialize(), even device_add() returns error, we still need use the put_device() to release the reference to call rtd_release(), which will do the free() action. Signed-off-by: liu chuansheng Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 29046e750ad7..f7551c1c827c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card, INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); ret = device_add(rtd->dev); if (ret < 0) { + /* calling put_device() here to free the rtd->dev */ + put_device(rtd->dev); dev_err(card->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; From d3bf1561253383a3dbcc40afdb2b039d56093a3e Mon Sep 17 00:00:00 2001 From: Chuansheng Liu Date: Wed, 26 Dec 2012 00:57:32 +0800 Subject: [PATCH 16/30] ASoC: core: fix the memory leak in case of remove_aux_dev() When probing aux_dev, initializing is as below: device_initialize() device_add() So when remove aux_dev, we need do as below: device_del() device_put() Otherwise, the rtd_release() will not be called. So here using device_unregister() to replace device_del(), like the action in soc_remove_link_dais(). Signed-off-by: liu chuansheng Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f7551c1c827c..2370063b5824 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1556,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { device_remove_file(rtd->dev, &dev_attr_codec_reg); - device_del(rtd->dev); + device_unregister(rtd->dev); rtd->dev_registered = 0; } From 1b8d52e63c53fd271846a56b7b1e3f622fd6a0a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Dec 2012 13:14:24 +0900 Subject: [PATCH 17/30] ASoC: wm5102: Improve speaker enable performance Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 48 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 46 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 688ade080589..7a9048dad1cd 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -36,6 +36,9 @@ struct wm5102_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int spk_ena:2; + unsigned int spk_ena_pending:1; }; static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); @@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), }; +static int wm5102_spk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); + + if (arizona->rev < 1) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (!wm5102->spk_ena) { + snd_soc_write(codec, 0x4f5, 0x25a); + wm5102->spk_ena_pending = true; + } + break; + case SND_SOC_DAPM_POST_PMU: + if (wm5102->spk_ena_pending) { + msleep(75); + snd_soc_write(codec, 0x4f5, 0xda); + wm5102->spk_ena_pending = false; + wm5102->spk_ena++; + } + break; + case SND_SOC_DAPM_PRE_PMD: + wm5102->spk_ena--; + if (!wm5102->spk_ena) + snd_soc_write(codec, 0x4f5, 0x25a); + break; + case SND_SOC_DAPM_POST_PMD: + if (!wm5102->spk_ena) + snd_soc_write(codec, 0x4f5, 0x0da); + break; + } + + return 0; +} + + ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); @@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, From 0cc411b934c4317b363d1af993549f391852b980 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jan 2013 10:48:10 +0000 Subject: [PATCH 18/30] ASoC: wm2200: Remove DSP B and left justified AIF modes These are not supported. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2200.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index a12fc2fa3971..e6cefe1ac677 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: fmt_val = 0; break; - case SND_SOC_DAIFMT_DSP_B: - fmt_val = 1; - break; case SND_SOC_DAIFMT_I2S: fmt_val = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - fmt_val = 3; - break; default: dev_err(codec->dev, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); From d71753e22b24548911b377db28f80870cf50d07b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jan 2013 10:48:02 +0000 Subject: [PATCH 19/30] ASoC: arizona: Remove DSP B and left justified AIF modes These are not supported. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 38248a7a95e3..1d8bb5917594 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mode = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mode = 1; - break; case SND_SOC_DAIFMT_I2S: mode = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mode = 3; - break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); From 5f960294e2031d12f10c8488c3446fecbf59628d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jan 2013 21:06:08 +0000 Subject: [PATCH 20/30] ASoC: wm5100: Remove DSP B and left justified formats These are not supported Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm5100.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 5a5f36936235..54397a508073 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mask = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mask = 1; - break; case SND_SOC_DAIFMT_I2S: mask = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mask = 3; - break; default: dev_err(codec->dev, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); From 267f8fa2e1eef0612b2007e1f1846bcbc35cc1fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Jan 2013 21:18:12 +0000 Subject: [PATCH 21/30] ASoC: wm2000: Fix sense of speech clarity enable Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f01d63..12bcae63a7f0 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); if (wm2000->speech_clarity) - ret &= ~WM2000_SPEECH_CLARITY; - else ret |= WM2000_SPEECH_CLARITY; + else + ret &= ~WM2000_SPEECH_CLARITY; wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); From a76fefab5c82d0f51c1330e275476b2066fe7d73 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Jan 2013 19:03:17 +0000 Subject: [PATCH 22/30] ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ffc89fab96fb..7b198c38f3ef 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; + void *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + buf = kmemdup(blk->data, le32_to_cpu(blk->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { @@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } + + kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); From 41b645c8624df6ace020a8863ad1449d69140f7d Mon Sep 17 00:00:00 2001 From: Mike Dunn Date: Mon, 7 Jan 2013 13:55:12 -0800 Subject: [PATCH 23/30] ALSA: pxa27x: fix ac97 cold reset Cold reset on the pxa27x currently fails and pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44) appears in the kernel log. Through trial-and-error (the pxa270 developer's manual is mostly incoherent on the topic of ac97 reset), I got cold reset to complete by setting the WARM_RST bit in the GCR register (and later noticed that pxa3xx does this for cold reset as well). Also, a timeout loop is needed to wait for the reset to complete. Tested on a palm treo 680 machine. Signed-off-by: Mike Dunn Acked-by: Igor Grinberg Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/arm/pxa2xx-ac97-lib.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fc0ae90e5b1..1ecd0a66ecd3 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -148,6 +148,8 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { + unsigned int timeout; + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -157,8 +159,10 @@ static inline void pxa_ac97_cold_pxa27x(void) clk_enable(ac97conf_clk); udelay(5); clk_disable(ac97conf_clk); - GCR = GCR_COLD_RST; - udelay(50); + GCR = GCR_COLD_RST | GCR_WARM_RST; + timeout = 100; /* wait for the codec-ready bit to be set */ + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); } #endif From 3b4bc7bccc7857274705b05cf81a0c72cfd0b0dd Mon Sep 17 00:00:00 2001 From: Mike Dunn Date: Mon, 7 Jan 2013 13:55:13 -0800 Subject: [PATCH 24/30] ALSA: pxa27x: fix ac97 warm reset This patch fixes some code that implements a work-around to a hardware bug in the ac97 controller on the pxa27x. A bug in the controller's warm reset functionality requires that the mfp used by the controller as the AC97_nRESET line be temporarily reconfigured as a generic output gpio (AF0) and manually held high for the duration of the warm reset cycle. This is what was done in the original code, but it was broken long ago by commit fb1bf8cd ([ARM] pxa: introduce processor specific pxa27x_assert_ac97reset()) which changed the mfp to a GPIO input instead of a high output. The fix requires the ac97 controller to obtain the gpio via gpio_request_one(), with arguments that configure the gpio as an output initially driven high. Tested on a palm treo 680 machine. Reportedly, this broken code only prevents a warm reset on hardware that lacks a pull-up on the line, which appears to be the case for me. Signed-off-by: Mike Dunn Signed-off-by: Igor Grinberg Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 +++ arch/arm/mach-pxa/pxa27x.c | 4 ++-- sound/arm/pxa2xx-ac97-lib.c | 18 +++++++++++++++++- 3 files changed, 22 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/mfp-pxa27x.h b/arch/arm/mach-pxa/include/mach/mfp-pxa27x.h index a611ad3153c7..b6132aa95dc0 100644 --- a/arch/arm/mach-pxa/include/mach/mfp-pxa27x.h +++ b/arch/arm/mach-pxa/include/mach/mfp-pxa27x.h @@ -463,6 +463,9 @@ GPIO76_LCD_PCLK, \ GPIO77_LCD_BIAS +/* these enable a work-around for a hw bug in pxa27x during ac97 warm reset */ +#define GPIO113_AC97_nRESET_GPIO_HIGH MFP_CFG_OUT(GPIO113, AF0, DEFAULT) +#define GPIO95_AC97_nRESET_GPIO_HIGH MFP_CFG_OUT(GPIO95, AF0, DEFAULT) extern int keypad_set_wake(unsigned int on); #endif /* __ASM_ARCH_MFP_PXA27X_H */ diff --git a/arch/arm/mach-pxa/pxa27x.c b/arch/arm/mach-pxa/pxa27x.c index 8047ee0effc5..616cb87b6179 100644 --- a/arch/arm/mach-pxa/pxa27x.c +++ b/arch/arm/mach-pxa/pxa27x.c @@ -47,9 +47,9 @@ void pxa27x_clear_otgph(void) EXPORT_SYMBOL(pxa27x_clear_otgph); static unsigned long ac97_reset_config[] = { - GPIO113_GPIO, + GPIO113_AC97_nRESET_GPIO_HIGH, GPIO113_AC97_nRESET, - GPIO95_GPIO, + GPIO95_AC97_nRESET_GPIO_HIGH, GPIO95_AC97_nRESET, }; diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 1ecd0a66ecd3..fff7753e35c1 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include @@ -344,8 +345,21 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ + /* + * This gpio is needed for a work-around to a bug in the ac97 + * controller during warm reset. The direction and level is set + * here so that it is an output driven high when switching from + * AC97_nRESET alt function to generic gpio. + */ + ret = gpio_request_one(reset_gpio, GPIOF_OUT_INIT_HIGH, + "pxa27x ac97 reset"); + if (ret < 0) { + pr_err("%s: gpio_request_one() failed: %d\n", + __func__, ret); + goto err_conf; + } pxa27x_assert_ac97reset(reset_gpio, 0); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); @@ -388,6 +402,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe); void pxa2xx_ac97_hw_remove(struct platform_device *dev) { + if (cpu_is_pxa27x()) + gpio_free(reset_gpio); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); if (ac97conf_clk) { From d7dab4dbbb2d1b0c903378d6bade2e4ae161804e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Jan 2013 13:51:30 +0100 Subject: [PATCH 25/30] ALSA: hda - Disable runtime D3 for Intel CPT & co We've got a few bug reports that the runtime D3 results in the dead HD-audio controller. It seems that the problem is in a deeper level than the sound driver itself, so as a temporal solution, disable the feature for these controllers again. Reported-and-tested-by: Vincent Blut Reported-and-tested-by: Maurizio Avogadro Cc: [v3.7] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cca87277baf0..0b6aebacc56b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -573,9 +573,12 @@ enum { #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ /* quirks for Intel PCH */ -#define AZX_DCAPS_INTEL_PCH \ +#define AZX_DCAPS_INTEL_PCH_NOPM \ (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) + AZX_DCAPS_COUNT_LPIB_DELAY) + +#define AZX_DCAPS_INTEL_PCH \ + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci) static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, From 7ed4165e2d01bdbbb4c1086eb73eadf0f64cbbf0 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 19 Dec 2012 09:44:47 +0100 Subject: [PATCH 26/30] Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs" This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. The original patch was meant to remove clicking, but in fact caused even more clicking instead. Thanks to c4pp4 for doing most of the work with this bug. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60890bfecc19..dd798c3196ff 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM -static int conexant_suspend(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); - return 0; -} -#endif - static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; /* From 56bde0f328428f2fc6e416510d8b42de6a0cfad5 Mon Sep 17 00:00:00 2001 From: Andre Schramm Date: Wed, 9 Jan 2013 14:40:18 +0100 Subject: [PATCH 27/30] ALSA: hdspm - Fix wordclock status on AES32 Use correct bitmask for AES32 cards to determine wordclock lock state, add missing bitmask for sync check and make output of the corresponding control and /proc coherent. Signed-off-by: Andre Schramm Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e02e064d7b4..223c3d9cc69e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); */ /* status */ #define HDSPM_AES32_wcLock 0x0200000 +#define HDSPM_AES32_wcSync 0x0100000 #define HDSPM_AES32_wcFreq_bit 22 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ @@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) switch (hdspm->io_type) { case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_wcSync) - return 2; - else if (status & HDSPM_wcLock) - return 1; + if (status & HDSPM_AES32_wcLock) { + if (status & HDSPM_AES32_wcSync) + return 2; + else + return 1; + } return 0; break; @@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, unsigned int status; unsigned int status2; unsigned int timecode; + unsigned int wcLock, wcSync; int pref_syncref; char *autosync_ref; int x; @@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Status:\n"); + wcLock = status & HDSPM_AES32_wcLock; + wcSync = wcLock && (status & HDSPM_AES32_wcSync); + snd_iprintf(buffer, "Word: %s Frequency: %d\n", - (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock", + (wcLock) ? (wcSync ? "Sync " : "Lock ") : "No Lock", HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF)); for (x = 0; x < 8; x++) { From e8e7da23c9add6f636bcc631aeb4461ffb99f77f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 10 Jan 2013 11:19:14 +0530 Subject: [PATCH 28/30] ALSA: usb-audio: Make ebox44_table static Fixes the following sparse warning: sound/usb/mixer_quirks.c:1209:23: warning: symbol 'ebox44_table' was not declared. Should it be static? Signed-off-by: Sachin Kamat Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0422b1360af3..15520de1df56 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -struct std_mono_table ebox44_table[] = { +static struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, From 065380f0880dd651eb405430745926dc4747b759 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 10 Jan 2013 10:25:48 +0100 Subject: [PATCH 29/30] ALSA: hda - Add support of new codec ALC284 Added the support for a new codec ALC284, which is compatible with ALC269. Also add more codec variants to handle the SSID check properly. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 71ae23dd7103..f5196277b6e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5817,6 +5817,9 @@ enum { ALC269_TYPE_ALC269VB, ALC269_TYPE_ALC269VC, ALC269_TYPE_ALC269VD, + ALC269_TYPE_ALC280, + ALC269_TYPE_ALC282, + ALC269_TYPE_ALC284, }; /* @@ -5833,10 +5836,13 @@ static int alc269_parse_auto_config(struct hda_codec *codec) switch (spec->codec_variant) { case ALC269_TYPE_ALC269VA: case ALC269_TYPE_ALC269VC: + case ALC269_TYPE_ALC280: + case ALC269_TYPE_ALC284: ssids = alc269va_ssids; break; case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: + case ALC269_TYPE_ALC282: ssids = alc269_ssids; break; default: @@ -6400,7 +6406,8 @@ static int patch_alc269(struct hda_codec *codec) alc_auto_parse_customize_define(codec); - if (codec->vendor_id == 0x10ec0269) { + switch (codec->vendor_id) { + case 0x10ec0269: spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { case 0x0010: @@ -6425,6 +6432,20 @@ static int patch_alc269(struct hda_codec *codec) goto error; spec->init_hook = alc269_fill_coef; alc269_fill_coef(codec); + break; + + case 0x10ec0280: + case 0x10ec0290: + spec->codec_variant = ALC269_TYPE_ALC280; + break; + case 0x10ec0282: + case 0x10ec0283: + spec->codec_variant = ALC269_TYPE_ALC282; + break; + case 0x10ec0284: + case 0x10ec0292: + spec->codec_variant = ALC269_TYPE_ALC284; + break; } /* automatic parse from the BIOS config */ @@ -7129,6 +7150,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 }, + { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 }, { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", From 31be5425d795585251a3ee970319c37643e0cda2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Jan 2013 14:06:38 +0100 Subject: [PATCH 30/30] ALSA: usb-audio: Fix NULL dereference by access to non-existing substream The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-and-tested-by: Jens Axboe Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c6593101c049..d82e378d37cb 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -511,6 +511,16 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) struct snd_usb_substream *sync_subs = &subs->stream->substream[subs->direction ^ 1]; + if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA || + !subs->stream) + return snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + /* Try to find the best matching audioformat. */ list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt,