January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/16] Yet more cleanups of line6 drivers
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, here is a series of cleanup patches for line6 stuff. They are applied on top of my previous patches. The latest patchset is found in test/line6 branch of sound.git tree. Takashi Takashi Iwai (16): ALSA: line6: Minor refactoring ALSA: line6: Fix memory leak at probe error path ALSA: line6: Remove unused line6_nop_read() ALSA: line6: Reduce superfluous spinlock in midi.c ALSA: line6: Fix missing error handling in line6_pcm_acquire() ALSA: line6: Use logical OR ALSA: line6: Fix the error recovery in line6_pcm_acquire() ALSA: line6: Drop superfluous spinlock for trigger ALSA: line6: Use incremental loop ALSA: line6: Drop voodoo workarounds ALSA: line6: Rearrange PCM structure ALSA: line6: Consolidate URB unlink and sync helpers ALSA: line6: Use dev_err() ALSA: line6: Consolidate PCM stream buffer allocation and free ALSA: line6: Do clipping in volume / monitor manipulations ALSA: line6: Skip volume manipulation during silence copying sound/usb/line6/capture.c | 122 ++++++-------------------- sound/usb/line6/capture.h | 5 -- sound/usb/line6/driver.c | 157 ++++++++++++++++----------------- sound/usb/line6/driver.h | 2 - sound/usb/line6/midi.c | 18 ++-- sound/usb/line6/midi.h | 7 +- sound/usb/line6/pcm.c | 210 +++++++++++++++++++++++++++------------------ sound/usb/line6/pcm.h | 163 ++++++++++------------------------- sound/usb/line6/playback.c | 159 ++++++++++------------------------ sound/usb/line6/playback.h | 5 -- 10 files changed, 329 insertions(+), 519 deletions(-) -- 2.2.2

2 26

[alsa-devel] [patch] ASoC: Intel: remove an unused struct member
by Dan Carpenter 28 Jan '15

28 Jan '15

We never set the ->scratch pointer, so let's delete it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index eeef9a5..78daa46 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -277,7 +277,6 @@ struct sst_hsw { /* FW config */ struct sst_hsw_ipc_fw_ready fw_ready; struct sst_hsw_ipc_fw_version version; - struct sst_module *scratch; bool fw_done; struct sst_fw *sst_fw; @@ -2135,7 +2134,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); - kfree(hsw->scratch); kthread_stop(hsw->tx_thread); kfree(hsw->msg); }

2 1

[alsa-devel] [PATCH 0/5] Reduce races in line6 drivers
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, another day, another patchset. This time it mostly covers the races in PCM stream handling. This should work for most cases, but there are slightly uncovered conditions that is fairly difficult to fix. Maybe moving to the non-atomic PCM ops is one way, but it's too far ahead, ther are much more things to clean up before that. Takashi === Takashi Iwai (5): ALSA: line6: Fix racy loopback handling ALSA: line6: Clear prev_fbuf and prev_fsize properly ALSA: line6: Reorganize PCM stream handling ALSA: line6: Make common PCM pointer callback ALSA: line6: Handle error from line6_pcm_acquire() sound/usb/line6/capture.c | 113 +++------------ sound/usb/line6/capture.h | 1 - sound/usb/line6/pcm.c | 351 ++++++++++++++++++++++++++------------------- sound/usb/line6/pcm.h | 145 ++++++------------- sound/usb/line6/playback.c | 139 ++++-------------- sound/usb/line6/playback.h | 1 - sound/usb/line6/toneport.c | 17 ++- 7 files changed, 302 insertions(+), 465 deletions(-) -- 2.2.2

2 7

[alsa-devel] [PATCH 0/7 RESEND] ASoC: pcm512x: Clock master modes
by Peter Rosin 27 Jan '15

27 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! [ Note that the dt change is in patch 5/7, for those only interested in that particular bit. ] This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. This is technically not a clean resend, since I'm now sending the whole series to all parties, hopefully using the correct email addesses this time. I have also augmented the commit message of patch 1/7 according to the review comments by Lars-Peter Clausen. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 955 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1161 insertions(+), 25 deletions(-) -- 1.7.10.4

1 7

[alsa-devel] [PATCH] ASoC: rt5645: Add rt5650 codec support
by Bard Liao 27 Jan '15

27 Jan '15

This patch adds support for rt5650 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 158 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt5645.h | 15 +++++ 2 files changed, 156 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6e9cd8e..306db11 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -31,6 +31,7 @@ #include "rt5645.h" #define RT5645_DEVICE_ID 0x6308 +#define RT5650_DEVICE_ID 0x6419 #define RT5645_PR_RANGE_BASE (0xff + 1) #define RT5645_PR_SPACING 0x100 @@ -59,6 +60,10 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) +static const struct reg_default rt5650_init_list[] = { + {0xf6, 0x0100}, +}; + static const struct reg_default rt5645_reg[] = { { 0x00, 0x0000 }, { 0x01, 0xc8c8 }, @@ -86,6 +91,7 @@ static const struct reg_default rt5645_reg[] = { { 0x2a, 0x5656 }, { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, { 0x2f, 0x1002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, @@ -192,6 +198,8 @@ static const struct reg_default rt5645_reg[] = { { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, { 0xdd, 0x001b }, + { 0xdf, 0x0008 }, + { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, { 0xe7, 0x0200 }, { 0xec, 0xb300 }, @@ -241,6 +249,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) case RT5645_IRQ_CTRL3: case RT5645_INT_IRQ_ST: case RT5645_IL_CMD: + case RT5650_4BTN_IL_CMD1: case RT5645_VENDOR_ID: case RT5645_VENDOR_ID1: case RT5645_VENDOR_ID2: @@ -286,6 +295,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_STO_DAC_MIXER: case RT5645_MONO_DAC_MIXER: case RT5645_DIG_MIXER: + case RT5650_A_DAC_SOUR: case RT5645_DIG_INF1_DATA: case RT5645_PDM_OUT_CTRL: case RT5645_REC_L1_MIXER: @@ -376,6 +386,8 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_IL_CMD: case RT5645_IL_CMD2: case RT5645_IL_CMD3: + case RT5650_4BTN_IL_CMD1: + case RT5650_4BTN_IL_CMD2: case RT5645_DRC1_HL_CTRL1: case RT5645_DRC2_HL_CTRL1: case RT5645_ADC_MONO_HP_CTRL1: @@ -1005,6 +1017,44 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); +/* MX-2d [3] [2] */ +static const char * const rt5650_a_dac1_src[] = { + "DAC1", "Stereo DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_L_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_l_mux = + SOC_DAPM_ENUM("A DAC1 L source", rt5650_a_dac1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_R_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_r_mux = + SOC_DAPM_ENUM("A DAC1 R source", rt5650_a_dac1_r_enum); + +/* MX-2d [1] [0] */ +static const char * const rt5650_a_dac2_src[] = { + "Stereo DAC Mixer", "Mono DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_L_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_l_mux = + SOC_DAPM_ENUM("A DAC2 L source", rt5650_a_dac2_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_R_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_r_mux = + SOC_DAPM_ENUM("A DAC2 R source", rt5650_a_dac2_r_enum); + /* MX-2F [13:12] */ static const char * const rt5645_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "VAD_ADC" @@ -1149,11 +1199,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: hp_amp_power(codec, 1); /* headphone unmute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK | - RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1173,12 +1228,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: /* headphone mute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, - RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | - RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1572,6 +1631,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_l_mux), + SND_SOC_DAPM_MUX("A DAC1 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_r_mux), + SND_SOC_DAPM_MUX("A DAC2 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_l_mux), + SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_r_mux), +}; + static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1777,13 +1847,9 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, - { "DAC L1", NULL, "Stereo DAC MIXL" }, { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R1", NULL, "Stereo DAC MIXR" }, { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC L2", NULL, "Mono DAC MIXL" }, { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R2", NULL, "Mono DAC MIXR" }, { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, { "SPK MIXL", "BST1 Switch", "BST1" }, @@ -1872,6 +1938,30 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPOR", NULL, "SPK amp" }, }; +static const struct snd_soc_dapm_route rt5650_specific_dapm_routes[] = { + { "A DAC1 L Mux", "DAC1", "DAC1 MIXL"}, + { "A DAC1 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC1 R Mux", "DAC1", "DAC1 MIXR"}, + { "A DAC1 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + + { "A DAC2 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC2 L Mux", "Mono DAC Mixer", "Mono DAC MIXL"}, + { "A DAC2 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + { "A DAC2 R Mux", "Mono DAC Mixer", "Mono DAC MIXR"}, + + { "DAC L1", NULL, "A DAC1 L Mux" }, + { "DAC R1", NULL, "A DAC1 R Mux" }, + { "DAC L2", NULL, "A DAC2 L Mux" }, + { "DAC R2", NULL, "A DAC2 R Mux" }, +}; + +static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC R2", NULL, "Mono DAC MIXR" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2287,6 +2377,22 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645->codec = codec; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5645: + snd_soc_dapm_add_routes(&codec->dapm, + rt5645_specific_dapm_routes, + ARRAY_SIZE(rt5645_specific_dapm_routes)); + break; + case CODEC_TYPE_RT5650: + snd_soc_dapm_new_controls(&codec->dapm, + rt5650_specific_dapm_widgets, + ARRAY_SIZE(rt5650_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5650_specific_dapm_routes, + ARRAY_SIZE(rt5650_specific_dapm_routes)); + break; + } + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); @@ -2418,6 +2524,7 @@ static const struct regmap_config rt5645_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5650", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); @@ -2450,9 +2557,18 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); - if (val != RT5645_DEVICE_ID) { + + switch (val) { + case RT5645_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5645; + break; + case RT5650_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5650; + break; + default: dev_err(&i2c->dev, - "Device with ID register %x is not rt5645\n", val); + "Device with ID register %x is not rt5645 or rt5650\n", + val); return -ENODEV; } @@ -2463,6 +2579,14 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ARRAY_SIZE(rt5650_init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", + ret); + } + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index a815e36..7454231 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -47,6 +47,7 @@ #define RT5645_STO_DAC_MIXER 0x2a #define RT5645_MONO_DAC_MIXER 0x2b #define RT5645_DIG_MIXER 0x2c +#define RT5650_A_DAC_SOUR 0x2d #define RT5645_DIG_INF1_DATA 0x2f /* Mixer - PDM */ #define RT5645_PDM_OUT_CTRL 0x31 @@ -150,6 +151,8 @@ #define RT5645_IL_CMD 0xdb #define RT5645_IL_CMD2 0xdc #define RT5645_IL_CMD3 0xdd +#define RT5650_4BTN_IL_CMD1 0xdf +#define RT5650_4BTN_IL_CMD2 0xe0 #define RT5645_DRC1_HL_CTRL1 0xe7 #define RT5645_DRC2_HL_CTRL1 0xe9 #define RT5645_MUTI_DRC_CTRL1 0xea @@ -472,6 +475,12 @@ #define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) #define RT5645_DAC_L2_DAC_R_VOL_SFT 4 +/* Analog DAC1/2 Input Source Control (0x2d) */ +#define RT5650_A_DAC1_L_IN_SFT 3 +#define RT5650_A_DAC1_R_IN_SFT 2 +#define RT5650_A_DAC2_L_IN_SFT 1 +#define RT5650_A_DAC2_R_IN_SFT 0 + /* Digital Interface Data Control (0x2f) */ #define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) #define RT5645_IF1_ADC2_IN_SFT 15 @@ -2175,6 +2184,11 @@ enum { RT5645_DMIC_DATA_GPIO11, }; +enum { + CODEC_TYPE_RT5645, + CODEC_TYPE_RT5650, +}; + struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; @@ -2184,6 +2198,7 @@ struct rt5645_priv { struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; + int codec_type; int sysclk; int sysclk_src; int lrck[RT5645_AIFS]; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Intel: Used lock version to update shim registers
by Jie Yang 27 Jan '15

27 Jan '15

We need hold lock each time updating shirm registers, otherwise, we may set unexpected values to them when they are set in different thread at different time sequence. The notification work will be scheduled in global work queue, which won't hold this sst->spinlock itself, so here we need change to use the lock version to update shim registers. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index eeef9a5..e18cb07 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -653,11 +653,11 @@ static void hsw_notification_work(struct work_struct *work) } /* tell DSP that notification has been handled */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); + sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: ts3a227e: Add dts property that allows to specify micbias voltage
by Anatol Pomozov 27 Jan '15

27 Jan '15

The voltage controls key press threshold. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/ts3a227e.txt | 5 ++++ sound/soc/codecs/ts3a227e.c | 27 ++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt index e8bf23e..a836881 100644 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -13,6 +13,11 @@ Required properties: - interrupt-parent: The parent interrupt controller - interrupts: Interrupt number for /INT pin from the 227e +Optional properies: + - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). + Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage + 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V + Default value is "1" (2.2V). Examples: diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 54d431c..5ecfa31 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -79,6 +79,10 @@ static const int ts3a227e_buttons[] = { /* TS3A227E_REG_SETTING_2 0x05 */ #define KP_ENABLE 0x04 +/* TS3A227E_REG_SETTING_3 0x06 */ +#define MICBIAS_SETTING_SFT (3) +#define MICBIAS_SETTING_MASK (0x7 << MICBIAS_SETTING_SFT) + /* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ #define TYPE_3_POLE 0x01 #define TYPE_4_POLE_OMTP 0x02 @@ -248,6 +252,21 @@ static const struct regmap_config ts3a227e_regmap_config = { .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), }; +static int ts3a227e_parse_dt(struct ts3a227e *ts3a227e, struct device_node *np) +{ + u32 micbias; + int err; + + err = of_property_read_u32(np, "ti,micbias", &micbias); + if (!err) { + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, + MICBIAS_SETTING_MASK, + (micbias & 0x07) << MICBIAS_SETTING_SFT); + } + + return 0; +} + static int ts3a227e_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -266,6 +285,14 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, if (IS_ERR(ts3a227e->regmap)) return PTR_ERR(ts3a227e->regmap); + if (dev->of_node) { + ret = ts3a227e_parse_dt(ts3a227e, dev->of_node); + if (ret) { + dev_err(dev, "Failed to parse device tree: %d\n", ret); + return ret; + } + } + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "TS3A227E", ts3a227e); -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH] ASoC: ts3a227e: Remap keys to match Android headset specification
by Anatol Pomozov 27 Jan '15

27 Jan '15

ts3a227e datasheet says typical key resistance is key1 50 Om key2 135 Om key3 240 Om key4 470 Om The android headset specification expect buttons impedance: A (MEDIA) 0-70 Om D (VOICECOMMAND) 110-180 Om B (VOLUMEUP) 210-290 Om C (VOLUMEDOWN) 360-680 Om Thus key mapping should be key1 - MEDIA key2 - VOICECOMMAND key3 - VOLUMEUP key3 - VOLUMEDOWN Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/ts3a227e.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 9f2dced..54d431c 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -221,9 +221,9 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ts3a227e->jack = jack; ts3a227e_jack_report(ts3a227e); -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH] ASoC: wm8731: init mutex in i2c init path
by Manuel Lauss 27 Jan '15

27 Jan '15

The I2C init path forgot to init the mutex, leading to an oops when controls are accessed. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Fixes an oops on bootup when alsactl tries to restore codec parameters on my MIPS DB1300 board. sound/soc/codecs/wm8731.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b4..b115ed8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -717,6 +717,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- 2.2.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH v2 1/2] ASoC: atmel_ssc_dai: fix start event for I2S mode
by Bo Shen 27 Jan '15

27 Jan '15

According to the I2S specification information as following: - WS = 0, channel 1 (left) - WS = 1, channel 2 (right) So, the start event should be TF/RF falling edge. Reported-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: None sound/soc/atmel/atmel_ssc_dai.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e..e691aab 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); -- 2.3.0.rc0

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