May 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm2000: Add register readability information
by Mark Brown 31 May '12

31 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a75c376..4531c55 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -691,9 +691,39 @@ static int wm2000_resume(struct snd_soc_codec *codec) #define wm2000_resume NULL #endif +static bool wm2000_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM2000_REG_SYS_START: + case WM2000_REG_SPEECH_CLARITY: + case WM2000_REG_SYS_WATCHDOG: + case WM2000_REG_ANA_VMID_PD_TIME: + case WM2000_REG_ANA_VMID_PU_TIME: + case WM2000_REG_CAT_FLTR_INDX: + case WM2000_REG_CAT_GAIN_0: + case WM2000_REG_SYS_STATUS: + case WM2000_REG_SYS_MODE_CNTRL: + case WM2000_REG_SYS_START0: + case WM2000_REG_SYS_START1: + case WM2000_REG_ID1: + case WM2000_REG_ID2: + case WM2000_REG_REVISON: + case WM2000_REG_SYS_CTL1: + case WM2000_REG_SYS_CTL2: + case WM2000_REG_ANC_STAT: + case WM2000_REG_IF_CTL: + return true; + default: + return false; + } +} + static const struct regmap_config wm2000_regmap = { .reg_bits = 8, .val_bits = 8, + + .max_register = WM2000_REG_IF_CTL, + .readable_reg = wm2000_readable_reg, }; static int wm2000_probe(struct snd_soc_codec *codec) -- 1.7.10

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[alsa-devel] [PATCH] ASoC: ac97: Remove empty remove() function
by Mark Brown 31 May '12

31 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ac97.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2023c74..ea06b83 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -91,11 +91,6 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return 0; } -static int ac97_soc_remove(struct snd_soc_codec *codec) -{ - return 0; -} - #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { @@ -119,7 +114,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .write = ac97_write, .read = ac97_read, .probe = ac97_soc_probe, - .remove = ac97_soc_remove, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, }; -- 1.7.10

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Re: [alsa-devel] [PATCH] netwinder: encapsulate CPLD hardware locking and access
by David Woodhouse 31 May '12

31 May '12

On Thu, 2012-05-31 at 12:15 +0200, Christian Dietrich wrote: > I hope my draft of the preemption disabling is not to much crap. I would > appreciate some comments. And please be aware, that i don't own this > hardware, and haven't tested it on real hardware therefore. Looks relatively sane at first glance; thanks. -- dwmw2

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[alsa-devel] [PATCH 1/2] ASoC: core: Add single controls with specified range of values
by Adam Thomson 31 May '12

31 May '12

Control type added for cases where a specific range of values within a register are required for control. Added convenience macro: SOC_SINGLE_RANGE_TLV Added accessor implementations: snd_soc_info_volsw_range snd_soc_put_volsw_range snd_soc_get_volsw_range Signed-off-by: Michal Hajduk <Michal.Hajduk(a)diasemi.com> Signed-off-by: Adam Thomson <Adam.Thomson(a)diasemi.com> --- include/sound/soc.h | 16 +++++++++ sound/soc/soc-core.c | 89 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 105 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c703871..2096c60 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -67,6 +67,16 @@ {.reg = xreg, .rreg = xreg, \ .shift = xshift, .rshift = xshift, \ .max = xmax, .min = xmin} } +#define SOC_SINGLE_RANGE_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .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} } #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, \ @@ -460,6 +470,12 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b37ee80..1a1aba4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2790,6 +2790,95 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); /** + * snd_soc_info_volsw_range - single mixer info callback with range. + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. + */ +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); + +/** + * snd_soc_put_volsw_range - single mixer put value callback with range. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int val, val_mask; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + return snd_soc_update_bits_locked(codec, reg, val_mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); + +/** + * snd_soc_get_volsw_range - single mixer get callback with range + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + + ucontrol->value.integer.value[0] = + ((snd_soc_read(codec, reg) >> shift) & mask) - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); + +/** * snd_soc_limit_volume - Set new limit to an existing volume control. * * @codec: where to look for the control -- 1.7.0.4

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[alsa-devel] [PATCH 0/2] DA732x codec support
by Adam Thomson 31 May '12

31 May '12

These patches add new controls to ASoC core and a driver for the DA732x audio codec. Adam Thomson (2): ASoC: core: Add single controls with specified range of values ASoC: codecs: Add DA732x codec driver include/sound/soc.h | 16 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da732x.c | 1614 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da732x.h | 133 ++++ sound/soc/codecs/da732x_reg.h | 650 +++++++++++++++++ sound/soc/soc-core.c | 89 +++ 7 files changed, 2508 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/da732x.c create mode 100644 sound/soc/codecs/da732x.h create mode 100644 sound/soc/codecs/da732x_reg.h

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Re: [alsa-devel] kmemleak findings about USB audio recording on 3.3.2
by Clemens Ladisch 31 May '12

31 May '12

Alexander E. Patrakov wrote: > On my home computer with linux-3.3.2, kmemleak found a lot of copies of > this (not only from pulseaudio, but also from wine): > > unreferenced object 0xffff88021afaed40 (size 16): > comm "pulseaudio", pid 839, jiffies 4294711382 (age 31392.844s) > hex dump (first 16 bytes): > 80 3e 00 00 c0 5d 00 00 00 7d 00 00 02 00 00 00 .>...]...}...... > backtrace: > [<ffffffff811620d7>] create_object+0x127/0x2b0 > [<ffffffff81636d81>] kmemleak_alloc+0x21/0x50 > [<ffffffff81157bc3>] __kmalloc+0x103/0x1e0 > [<ffffffffa02f650b>] snd_usb_pcm_open+0x3bb/0x4c0 [snd_usb_audio] Please try this patch. --8<---------------------------------------------------------------->8-- ALSA: usb-audio: fix rate_list memory leak The array of sample rates is reallocated every time when opening the PCM device, but was freed only once when unplugging the device. Reported-by: "Alexander E. Patrakov" <patrakov(a)gmail.com> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -783,6 +783,9 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, int count = 0, needs_knot = 0; int err; + kfree(subs->rate_list.list); + subs->rate_list.list = NULL; + list_for_each_entry(fp, &subs->fmt_list, list) { if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) return 0;

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[alsa-devel] [PATCH 1/1] ALSA: hdspm - Work around broken DDS value on PCI RME MADI
by Adrian Knoth 31 May '12

31 May '12

On PCI RME MADI cards, the PLL register does not contain the proper value, so the calculated system_sample_rate is wrong. In this case, we simply return the cached rate from struct hdspm. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f27a98e..f3cf8d0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1988,6 +1988,13 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ); rate = hdspm_calc_dds_value(hdspm, period); + if (rate > 207000) { + /* Unreasonable high sample rate as seen on PCI MADI cards. + * Use the cached value instead. + */ + rate = hdspm->system_sample_rate; + } + return rate; } -- 1.7.10

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Change Class-D amplifier gain control name
by Jarkko Nikula 30 May '12

30 May '12

ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output gain control as a playback control. Fix this by changing the name as "Class-D Playback Volume". Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4f..58ef59d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0); static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = - SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); + SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = -- 1.7.10

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Re: [alsa-devel] [PATCH 4/5] ASoC: codecs: Add AB8500 codec-driver
by Mark Brown 30 May '12

30 May '12

On Thu, May 24, 2012 at 03:26:38PM +0200, Ola Lilja wrote: > +static void show_regulator_status(struct device *dev) > +{ > + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); > + struct ab8500_codec_drvdata_dbg *dbg = &drvdata->dbg; > + > + dev_dbg(dev, "%s: Regulator-status:\n", __func__); > + dev_dbg(dev, "%s: V-AUD: %s\n", __func__, > + (regulator_is_enabled(dbg->vaud) > 0) ? > + "On" : "Off"); > + dev_dbg(dev, "%s: V-AMIC1: %s\n", __func__, > + (regulator_is_enabled(dbg->vamic1) > 0) ? > + "On" : "Off"); > + dev_dbg(dev, "%s: V-AMIC2: %s\n", __func__, > + (regulator_is_enabled(dbg->vamic2) > 0) ? > + "On" : "Off"); > + dev_dbg(dev, "%s: V-DMIC: %s\n", __func__, > + (regulator_is_enabled(dbg->vdmic) > 0) ? > + "On" : "Off"); What problems are you finding when you try to use the debug infrastructure in both the regulator API and DAPM to discover the state of the regulators? > + /* Clocks */ > + SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), > + SND_SOC_DAPM_CLOCK_SUPPLY("gpio.1"), This looks wrong - audioclk looks reasonable but gpio.1 looks like a board-specific name which shouldn't be encoded into the driver. > + if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) { > + dev_err(codec->dev, > + "%s: ERROR: This control supports '%s' only!\n", > + __func__, enum_sid_state[SID_APPLY_FIR]); > + return 0; > + } I'd expect this to return an error... > + status = snd_soc_dapm_force_enable_pin(&codec->dapm, > + "ANC Configure Input"); > + if (status < 0) { > + dev_err(dev, > + "%s: ERROR: Failed to enable power (status = %d)!\n", > + __func__, status); > + goto cleanup; > + } > + snd_soc_dapm_sync(&codec->dapm); > + > + mutex_lock(&codec->mutex); Your locking looks bad here. Nothing ensures that something doesn't come along and undo the force enable. Looking at the code this is the only function that fiddles with the input but there's still a race where one writer might exit the mutex section and disable the pin while a second enters the mutex section. > +static int filter_control_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct filter_control *fc = > + (struct filter_control *)kcontrol->private_value; > + unsigned int i; > + > + for (i = 0; i < fc->count; i++) > + ucontrol->value.integer.value[i] = fc->value[i]; > + > + return 0; > +} > + > +static int filter_control_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct filter_control *fc = > + (struct filter_control *)kcontrol->private_value; > + unsigned int i; > + > + for (i = 0; i < fc->count; i++) > + fc->value[i] = ucontrol->value.integer.value[i]; These don't seem to be locked? > +int ab8500_audio_init_audioblock(struct snd_soc_codec *codec) static. Lots of other functions in the rest of the driver have the same issue. > +static int ab8500_codec_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) > +{ > + dev_dbg(dai->codec->dev, "%s Enter.\n", __func__); > + > + return 0; > +} Remove empty functions. > + default: > + dev_err(dai->codec->dev, > + "%s: ERROR: Unsupported INV mask 0x%x\n", > + __func__, fmt & SND_SOC_DAIFMT_INV_MASK); > + return -EINVAL; > + break; The break is redundant. > + /* Only 16 bit slot width is supported at the moment in TDM mode */ > + if (slot_width != 16) { > + dev_err(dai->codec->dev, > + "%s: ERROR: Unsupported slot_width %d.\n", > + __func__, slot_width); > + return -EINVAL; > + } You've got code which supports other widths... > +static struct snd_soc_codec_driver ab8500_codec_driver = { > + .probe = ab8500_codec_probe, > + .read = ab8500_codec_read_reg, > + .write = ab8500_codec_write_reg, > + .reg_cache_size = 0, no need to init things to zero or NULL in static structs. > +#define PRE_PMU_POST_PMD (SND_SOC_DAPM_PRE_PMU | \ > + SND_SOC_DAPM_POST_PMD) You shouldn't define stuff like this in your driver!

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Re: [alsa-devel] [PATCH 2/2 V2] ASoC: fix pxa-ssp compiling issue under mach-mmp
by Mark Brown 30 May '12

30 May '12

On Fri, May 25, 2012 at 11:12:08AM +0800, Qiao Zhou wrote: > pxa-ssp.c uses API like cpu_is_pxa3xx(), cpu_is_pxa2xx(), which is > defined under arch-pxa architecture, and drivers under mach-mmp > can't find it. so just use ssp->type to replace that API. This doesn't apply to for-3.5 (or Linus' tree I suspect). Can you please regenerate against that?

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