May 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm_adsp: Use adsp_err/warn instead of dev_err/warn
by Charles Keepax 27 May '14

27 May '14

We have defines for adsp messages best to consistently use them. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 37 +++++++++++++++++-------------------- 1 files changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff15eec..0600271 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1543,16 +1543,16 @@ static void wm_adsp2_boot_work(struct work_struct *work) ret = regmap_read(dsp->regmap, dsp->base + ADSP2_CLOCKING, &val); if (ret != 0) { - dev_err(dsp->dev, "Failed to read clocking: %d\n", ret); + adsp_err(dsp, "Failed to read clocking: %d\n", ret); return; } if ((val & ADSP2_CLK_SEL_MASK) >= 3) { ret = regulator_enable(dsp->dvfs); if (ret != 0) { - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); + adsp_err(dsp, + "Failed to enable supply: %d\n", + ret); return; } @@ -1560,9 +1560,9 @@ static void wm_adsp2_boot_work(struct work_struct *work) 1800000, 1800000); if (ret != 0) { - dev_err(dsp->dev, - "Failed to raise supply: %d\n", - ret); + adsp_err(dsp, + "Failed to raise supply: %d\n", + ret); return; } } @@ -1672,15 +1672,15 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret = regulator_set_voltage(dsp->dvfs, 1200000, 1800000); if (ret != 0) - dev_warn(dsp->dev, - "Failed to lower supply: %d\n", - ret); + adsp_warn(dsp, + "Failed to lower supply: %d\n", + ret); ret = regulator_disable(dsp->dvfs); if (ret != 0) - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); + adsp_err(dsp, + "Failed to enable supply: %d\n", + ret); } list_for_each_entry(ctl, &dsp->ctl_list, list) @@ -1732,28 +1732,25 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); - dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); + adsp_err(adsp, "Failed to get DCVDD: %d\n", ret); return ret; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { - dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", - ret); + adsp_err(adsp, "Failed to enable DCVDD: %d\n", ret); return ret; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { - dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", - ret); + adsp_err(adsp, "Failed to initialise DVFS: %d\n", ret); return ret; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { - dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", - ret); + adsp_err(adsp, "Failed to disable DCVDD: %d\n", ret); return ret; } } -- 1.7.2.5

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[alsa-devel] [PATCH v5] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 27 May '14

27 May '14

Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no longer a valid pointer. Add the remaining register entries into sgtl5000_reg_defaults[] and remove sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and make the code simpler. Tested on a im53-qsb board. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v4: - Remove sgtl5000_restore_regs() as suggested by Lars-Peter sound/soc/codecs/sgtl5000.c | 75 +++++++++------------------------------------ 1 file changed, 15 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..3d39f0b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -1068,71 +1083,11 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) return 0; } -/* - * restore all sgtl5000 registers, - * since a big hole between dap and regular registers, - * we will restore them respectively. - */ -static int sgtl5000_restore_regs(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - u16 reg; - - /* restore regular registers */ - for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - - /* These regs should restore in particular order */ - if (reg == SGTL5000_CHIP_ANA_POWER || - reg == SGTL5000_CHIP_CLK_CTRL || - reg == SGTL5000_CHIP_LINREG_CTRL || - reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_REF_CTRL) - continue; - - snd_soc_write(codec, reg, cache[reg]); - } - - /* restore dap registers */ - for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); - - /* - * restore these regs according to the power setting sequence in - * sgtl5000_set_power_regs() and clock setting sequence in - * sgtl5000_set_clock(). - * - * The order of restore is: - * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after - * SGTL5000_CHIP_ANA_POWER PLL bits set - * 2. SGTL5000_CHIP_LINREG_CTRL should be set before - * SGTL5000_CHIP_ANA_POWER LINREG_D restored - * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, - * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored - */ - snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); - - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); - return 0; -} - static int sgtl5000_resume(struct snd_soc_codec *codec) { /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Restore registers by cached in memory */ - sgtl5000_restore_regs(codec); return 0; } #else -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: jack: Fix multiple definition of `snd_soc_jack_add_gpiods'
by Jarkko Nikula 27 May '14

27 May '14

Commit f025d3b9c64e ("ASoC: jack: Add support for GPIO descriptor defined jack pins") caused build error when CONFIG_GPIOLIB is not set: sound/include/sound/soc.h:470: multiple definition of `snd_soc_jack_add_gpiods' sound/soc/soc-core.o:sound/include/sound/soc.h:470: first defined here make[2]: *** [sound/soc/snd-soc-core.o] Error 1 make[2]: *** Waiting for unfinished jobs.... make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Fix this by marking snd_soc_jack_add_gpiods() as static inline in soc.h. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 98dca42487bc..15ee2f71b92e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -464,9 +464,10 @@ static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, return 0; } -int snd_soc_jack_add_gpiods(struct device *gpiod_dev, - struct snd_soc_jack *jack, - int count, struct snd_soc_jack_gpio *gpios) +static inline int snd_soc_jack_add_gpiods(struct device *gpiod_dev, + struct snd_soc_jack *jack, + int count, + struct snd_soc_jack_gpio *gpios) { return 0; } -- 2.0.0.rc2

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[alsa-devel] [PATCH] ASoC: jack: Clarify GPIO descriptor lookup in struct snd_soc_jack_gpio doc
by Jarkko Nikula 27 May '14

27 May '14

Clarify struct snd_soc_jack_gpio documentation for the idx and name fields. Because name is passed as connection ID to gpiod_get_index() when using GPIO descriptor defined jack pins it is not only used as a label in debugfs but also as function name lookup in systems that support functions names for GPIOs. Clarify also idx since the index is within the function of the GPIO consumer device and not within the device itself only. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 15ee2f71b92e..72e056529745 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -599,9 +599,11 @@ struct snd_soc_jack_zone { * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * * @gpio: legacy gpio number - * @idx: gpio descriptor index within the GPIO consumer device + * @idx: gpio descriptor index within the function of the GPIO + * consumer device * @gpiod_dev GPIO consumer device - * @name: gpio name + * @name: gpio name. Also as connection ID for the GPIO consumer + * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms -- 2.0.0.rc2

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[alsa-devel] [PATCH 1/1] sound/aoa/codecs/onyx.c: use static const for texts
by Fabian Frederick 27 May '14

27 May '14

'texts' is only used as source in strcpy Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiway(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- This is untested. sound/aoa/codecs/onyx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index f01bffb..401107b 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -241,7 +241,7 @@ static struct snd_kcontrol_new inputgain_control = { static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Line-In", "Microphone" }; + static const char * const texts[] = { "Line-In", "Microphone" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; -- 1.9.1

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[alsa-devel] [PATCH] pcm: route: Correctly close slave pcm when no matching chmap is found
by David Henningsson 27 May '14

27 May '14

This patch fixes a bug where the slave pcm was not correctly closed on some error conditions, such as not finding a matching chmap. Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_route.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 49567ea..751e36f 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -1429,8 +1429,10 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (tt_chmap) { err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); free(tt_chmap); - if (err < 0) + if (err < 0) { + snd_pcm_close(spcm); return err; + } } err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); -- 1.9.1

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[alsa-devel] [PATCH 1/5] ALSA: hda - Add fixup_forced flag
by Hui Wang 27 May '14

27 May '14

From: David Henningsson <david.henningsson(a)canonical.com> The "fixup_forced" flag will indicate whether a specific fixup (or nofixup) has been set by the user, to override the driver's default. This flag will help future patches. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 9 ++++++--- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 90d2fda..36961ab 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -852,15 +852,17 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { codec->fixup_list = NULL; codec->fixup_id = -1; + codec->fixup_forced = 1; return; } if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { - id = models->id; - name = models->name; - break; + codec->fixup_id = models->id; + codec->fixup_name = models->name; + codec->fixup_forced = 1; + return; } models++; } @@ -889,6 +891,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } + codec->fixup_forced = 0; codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a423313..5825aa1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -402,6 +402,7 @@ struct hda_codec { /* fix-up list */ int fixup_id; + unsigned int fixup_forced:1; /* fixup explicitly set by user */ const struct hda_fixup *fixup_list; const char *fixup_name; -- 1.8.1.2

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[alsa-devel] [PATCH] hdajackretask: Make sure codecs do not show up twice under 3.15 kernel
by David Henningsson 27 May '14

27 May '14

On a 3.15 based kernel, both "hdaudioC*D*" and "hwC*D*" shows up under /sys/class/sound, and pointing to the same object. Make sure we only match on the hwC*D* name to avoid duplicate entries. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/sysfs-pin-configs.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index 4044989..5bae0f9 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -20,8 +20,10 @@ int get_codec_name_list(codec_name_t* names, int entries) const gchar* dirname = g_dir_read_name(sysdir); if (!dirname) break; + /* Split e g "hwC0D1" into "hw", "0" and "1" */ cd = g_strsplit_set(dirname, "CD", 9); ok = g_strv_length(cd) == 3; + ok &= strcmp(cd[0], "hw") == 0; if (ok) { gchar* filetest = g_strdup_printf("/sys/class/sound/%s/init_pin_configs", dirname); ok = g_file_test(filetest, G_FILE_TEST_IS_REGULAR); -- 1.9.1

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[alsa-devel] [PATCH v4 00/17] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 27 May '14

27 May '14

Hi, This series is a cleanup of the fsl-ssi driver. v4 includes the changes I made to fix the baudclock handling. I moved the baudclock enable/disable function calls into hw_params()/hw_free(). To avoid inconsistent reference counters for the baudclock I use a variable baudclk_streams which describes the active streams to be able to decide if we have to enable/disable the clock and set the rate. I currently don't have a board to test i2s-master, maybe I am able to test it next week. The new patch which includes these changes is "ASoC: fsl-ssi: Fix baudclock handling". Best regards, Markus Changes in v4: - New patch to fix baudclock handling (enable/disable/set_rate) Changes in v3: - Some new patches to improve/fix i2s master mode by Sascha - baudclock is enabled/disabled in startup/shutdown now - bitclock setup moved to a seperate function (not set_dai_sysclk) - Regmap config changed to NATIVE now. Markus Pargmann (12): ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Sascha Hauer (5): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: Move fsl_ssi_set_dai_sysclk above fsl_ssi_hw_params ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 3 +- sound/soc/fsl/fsl_ssi.c | 1259 +++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 112 +++- sound/soc/fsl/fsl_ssi_dbg.c | 163 ++++++ 5 files changed, 815 insertions(+), 723 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.9.1

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[alsa-devel] [PATCH v2] ALSA: hda: fix tegra build
by Arnd Bergmann 27 May '14

27 May '14

When CONFIG_PM is disabled, the CONFIG_SND_HDA_POWER_SAVE_DEFAULT symbol does not get defined, which causes a build error for the hda-tegra driver: hda/hda_tegra.c:80:25: error: 'CONFIG_SND_HDA_POWER_SAVE_DEFAULT' undeclared here (not in a function) static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; ^ /git/arm-soc/sound/pci/hda/hda_tegra.c:235:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function] static void hda_tegra_disable_clocks(struct hda_tegra *data) ^ This works around the problem by not referencing that macro when CONFIG_PM is disabled. Instead, we assume that it's disabled unconditionally and cannot be enabled at runtime. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- Same patch, but this time with correct Signed-off-by line. diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 086fd06..a366ba9 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -77,10 +77,14 @@ struct hda_tegra { void __iomem *regs; }; +#ifdef CONFIG_PM static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); +#else +static int power_save = 0; +#endif /* * DMA page allocation ops.

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