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

[alsa-devel] randconfig build error with next-20150528, in sound/core/seq/seq_info.c
by Jim Davis 28 May '15

28 May '15

Building with the attached random configuration file, sound/core/seq/seq_info.c:61:12: error: redefinition of ‘snd_seq_info_init’ int __init snd_seq_info_init(void) ^ In file included from sound/core/seq/seq_info.c:26:0: sound/core/seq/seq_info.h:36:19: note: previous definition of ‘snd_seq_info_init ’ was here static inline int snd_seq_info_init(void) { return 0; } ^ sound/core/seq/seq_info.c:77:12: error: redefinition of ‘snd_seq_info_done’ int __exit snd_seq_info_done(void) ^ In file included from sound/core/seq/seq_info.c:26:0: sound/core/seq/seq_info.h:37:19: note: previous definition of ‘snd_seq_info_done’ was here static inline int snd_seq_info_done(void) { return 0; }

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[alsa-devel] [RFC 0/5] Add a gpio jack device
by Dylan Reid 28 May '15

28 May '15

The first three changes add a gpio audio jack device. This device can be used on systems that report headphone or mic plug through GPIOS. There can be 0-N of these devices created per board each can report one of several events. For example, this allows for a single jack for HP/Mic and a separate jack for line out. The last two patches are an example of the jack being used on Acer's Tegra Chromebook, those can be considered separately and might not be worth the churn. Dylan Reid (5): ALSA: Add jack types to dt-bindings ASoC: jack - make add_gpiods accept pre-filled descriptors ASoC: Add GPIO based jack device ASoC: tegra_max98090: Change nyan to use gpio-jack ARM: tegra: nyan: specify gpio-audio-jack device .../devicetree/bindings/sound/gpio-audio-jack.txt | 39 +++++ arch/arm/boot/dts/tegra124-nyan.dtsi | 22 ++- include/dt-bindings/sound/audio-jack-events.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/gpio-audio-jack.c | 191 +++++++++++++++++++++ sound/soc/soc-jack.c | 9 +- sound/soc/tegra/tegra_max98090.c | 107 +++--------- 8 files changed, 291 insertions(+), 92 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/gpio-audio-jack.txt create mode 100644 include/dt-bindings/sound/audio-jack-events.h create mode 100644 sound/soc/codecs/gpio-audio-jack.c -- 2.2.1.62.g3f15098

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[alsa-devel] [PATCH] ASoC: wm_adsp: Dump DSP_SCRATCH1 on DSP shutdown
by Richard Fitzgerald 28 May '15

28 May '15

The DSP_SCRATCH1 register is used by firmwares to hold diagnostic information. Include this in the shutdown message - it can be useful for debugging. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 14 +++++++++++++- 1 files changed, 13 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f6642c1..9b7c03c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -121,6 +121,11 @@ #define ADSP2_WDMA_CONFIG_2 0x31 #define ADSP2_RDMA_CONFIG_1 0x34 +#define ADSP2_SCRATCH0 0x40 +#define ADSP2_SCRATCH1 0x41 +#define ADSP2_SCRATCH2 0x42 +#define ADSP2_SCRATCH3 0x43 + /* * ADSP2 Control */ @@ -1880,6 +1885,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; + unsigned int scratch1 = 0xFFFFFFFF; int ret; switch (event) { @@ -1898,6 +1904,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + /* Capture DSP_SCRATCH1, it can be useful for analysis */ + ret = regmap_read(dsp->regmap, dsp->base + ADSP2_SCRATCH1, + &scratch1); + if (ret) + adsp_err(dsp, "Failed to read SCRATCH1 %d\n", ret); + dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, @@ -1935,7 +1947,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, kfree(alg_region); } - adsp_dbg(dsp, "Shutdown complete\n"); + adsp_info(dsp, "Shutdown complete (SCRATCH1:0x%x)\n", scratch1); break; default: -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm_adsp: Dump scratch registers on DSP shutdown
by Richard Fitzgerald 28 May '15

28 May '15

The SCRATCH registers are used by firmwares to hold diagnostic information. Log this during shutdown to assist analysis and debug of firmwares. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f6642c1..6e7a7000 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -121,6 +121,11 @@ #define ADSP2_WDMA_CONFIG_2 0x31 #define ADSP2_RDMA_CONFIG_1 0x34 +#define ADSP2_SCRATCH0 0x40 +#define ADSP2_SCRATCH1 0x41 +#define ADSP2_SCRATCH2 0x42 +#define ADSP2_SCRATCH3 0x43 + /* * ADSP2 Control */ @@ -364,6 +369,25 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, } } +static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) +{ + u16 scratch[4]; + int ret; + + ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, + scratch, sizeof(scratch)); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + return; + } + + adsp_info(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", + be16_to_cpu(scratch[0]), + be16_to_cpu(scratch[1]), + be16_to_cpu(scratch[2]), + be16_to_cpu(scratch[3])); +} + static int wm_coeff_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1898,6 +1922,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + /* Log firmware state, it can be useful for analysis */ + wm_adsp2_show_fw_status(dsp); + dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: Intel: remove unused function hsw_pcm_free_modules()
by Jie Yang 28 May '15

28 May '15

Remove the unused function hsw_pcm_free_modules() to fix the compling warning: sound/soc/intel/haswell/sst-haswell-pcm.c:923:13: warning: 'sw_pcm_free_modules' defined but not used [-Wunused-function] static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index bd96629..1557e37 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -920,21 +920,6 @@ err: return -ENODEV; } -static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) -{ - struct sst_hsw *hsw = pdata->hsw; - struct hsw_pcm_data *pcm_data; - int i; - - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - sst_hsw_runtime_module_free(pcm_data->runtime); - } - if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { - sst_hsw_runtime_module_free(pdata->runtime_waves); - } -} - static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Fix noise generator gain TLV
by Richard Fitzgerald 28 May '15

28 May '15

The Arizona codec drivers had an incorrect dB scaling for the noise generator gain that started at 0dB and went upwards. Actually the highest setting is 0dB. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 0c6d1bc..d476221 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -42,7 +42,7 @@ struct wm5102_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); -static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const struct wm_adsp_region wm5102_dsp1_regions[] = { diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fbaeddb..3ee6cfd 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -167,7 +167,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); -static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); #define WM5110_NG_SRC(name, base) \ diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index a4d1177..e7c81ba 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -40,7 +40,7 @@ struct wm8997_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); -static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); static const struct reg_default wm8997_sysclk_reva_patch[] = { -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: hda - Drop unused fields from struct hda_codec_preset
by Takashi Iwai 28 May '15

28 May '15

It's very unlikely that we'd need these fields out of sudden. Let's drop them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 10 +--------- sound/pci/hda/hda_codec.h | 4 ---- 2 files changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 00aa31c5f08e..d5ac25cc7fee 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -27,15 +27,7 @@ static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv) u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id; for (preset = driver->preset; preset->id; preset++) { - u32 mask = preset->mask; - - if (preset->afg && preset->afg != codec->core.afg) - continue; - if (preset->mfg && preset->mfg != codec->core.mfg) - continue; - if (!mask) - mask = ~0; - if (preset->id == (id & mask) && + if (preset->id == id && (!preset->rev || preset->rev == codec->core.revision_id)) { codec->preset = preset; return 1; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index ed7e9cfd699a..6c572588f7e5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -82,11 +82,7 @@ struct hda_bus { */ struct hda_codec_preset { unsigned int id; - unsigned int mask; - unsigned int subs; - unsigned int subs_mask; unsigned int rev; - hda_nid_t afg, mfg; const char *name; int (*patch)(struct hda_codec *codec); }; -- 2.4.2

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[alsa-devel] [PATCH 1/2] ASoC: atmel: simplify Kconfig
by Alexandre Belloni 28 May '15

28 May '15

Enclose the options in if SND_ATMEL_SOC ... endif to remove the dependency. Also remove the useless description for SND_ATMEL_SOC_SSC. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Kconfig | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index e7d08806f3e9..93abe4e6d596 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -6,27 +6,22 @@ config SND_ATMEL_SOC the ATMEL SSC interface. You will also need to select the audio interfaces to support below. +if SND_ATMEL_SOC + config SND_ATMEL_SOC_PDC tristate - depends on SND_ATMEL_SOC config SND_ATMEL_SOC_DMA tristate - depends on SND_ATMEL_SOC select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC tristate - depends on SND_ATMEL_SOC - help - Say Y or M if you want to add support for codecs the - ATMEL SSC interface. You will also needs to select the individual - machine drivers to support below. config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" depends on ARCH_AT91 || COMPILE_TEST - depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI + depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -37,7 +32,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_ATMEL_SOC_WM8904 tristate "Atmel ASoC driver for boards using WM8904 codec" depends on ARCH_AT91 || COMPILE_TEST - depends on ATMEL_SSC && SND_ATMEL_SOC && I2C + depends on ATMEL_SSC && I2C select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8904 @@ -48,10 +43,11 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" depends on ARCH_AT91 || COMPILE_TEST - depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI + depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. +endif -- 2.1.4

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[alsa-devel] [PATCH] ALSA: hda - Disable Headphone Mic boost for ALC662
by David Henningsson 28 May '15

28 May '15

When headphone mic boost is above zero, some 10 - 20 second delay might occur before the headphone mic is operational. Therefore disable the headphone mic boost control (recording gain is sufficient even without it). (Note: this patch is not about the headset mic, it's about the less common mic-in only mode.) BugLink: https://bugs.launchpad.net/bugs/1454235 Suggested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0fa455..da2d371 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4240,6 +4240,11 @@ static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + + /* Disable boost for mic-in permanently. (This code is only called + from quirks that guarantee that the headphone is at NID 0x1b.) */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000); + snd_hda_override_wcaps(codec, 0x1b, get_wcaps(codec, 0x1b) & ~AC_WCAP_IN_AMP); } else alc_fixup_headset_mode(codec, fix, action); } -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Disable Headphone Mic boost for ALC662
by David Henningsson 28 May '15

28 May '15

When headphone mic boost is above zero, some 10 - 20 second delay might occur before the headphone mic is operational. Therefore disable the headphone mic boost control (recording gain is sufficient even without it). (Note: this patch is not about the headset mic, it's about the less common mic-in only mode.) BugLink: https://bugs.launchpad.net/bugs/1454235 Suggested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0fa455..87605fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4240,6 +4240,10 @@ static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + + /* Disable boost for mic-in permanently */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000); + snd_hda_override_wcaps(codec, 0x1b, get_wcaps(codec, 0x1b) & ~AC_WCAP_IN_AMP); } else alc_fixup_headset_mode(codec, fix, action); } -- 1.9.1

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