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

[alsa-devel] [PATCH] ASoC: core: Fix SOC_DOUBLE_RANGE() macros
by Mark Brown 19 Dec '12

19 Dec '12

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 <broonie(a)opensource.wolfsonmicro.com> --- 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 4a27822..1e4ea78 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 b1738d5..d4732ac 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2922,7 +2922,7 @@ 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; @@ -2946,12 +2946,14 @@ 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) @@ -2959,7 +2961,21 @@ 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); @@ -2979,11 +2995,13 @@ 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; @@ -2993,6 +3011,16 @@ 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); -- 1.7.10.4

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[alsa-devel] Newbie help with WM8962 and tinyalsa
by Randi Eckstein 19 Dec '12

19 Dec '12

If anyone has an explanation on how to use tinymix to play on a wm8962, I'd really appreciate it. Thanks, Randi Eckstein

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 19 Dec '12

19 Dec '12

On Wed, Dec 19, 2012 at 06:36:37PM +0800, Chuansheng Liu wrote: > But some devices has been set to STANDY bias directly during device probing, > such as cs42l73_probe(): > cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > Then it will cause runtime_get() not be called but laterly runtime_put() will > be called. Also found some other uppaired cases. This is just a bug in the driver, if it's idle_bias_off then it really should be starting in _OFF or at the very least starting actually in _STANDBY (including taking the runtime reference) rather than partially in _STANDBY. > So here add new flag get_runtime, and the logic will be: > 1/ when device is from off to non-off bias, runtime_get() will be called if not yet; > 2/ When device is off bias, runtime_put() will be called if runtime_get() has > been called; This is really not a good idea at all, it's just adding new special cases and making the code more obscure.

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[alsa-devel] Speaker not working; muted?
by Martin Michlmayr 19 Dec '12

19 Dec '12

I've a problem with audio on my HP 2560p laptop (using snd_hda_intel): the speaker no longer works. However, I successfully get sound on a headphone. I've attached the output of alsa-info. It would be great if someone could take a look at let me know if this is a bug in the driver or wrong settings. Background: I used ALSA successfully in the past to play audio with the in-built speaker. At some point, I used aumix (which requires OSS modules) to turn the volume to zero. I believe I turned the volume back up again. I also rebooted my laptop at some point. I don't know if it's related to using aumix or something else, but at some point the in-built speaker stopped working... I can still listen to audio using a headphone. Another observation is that my laptop has a speaker mute button. When I reboot my laptop, and the BIOS or grub is running, I see that the mute button is activated. However, pressing the button doesn't change anything... when the Linux kernel loads, the mute button is still on, but when ALSA starts the button changes colour: it seems to me that ALSA thinks that mute is off while in reality it is on. -- Martin Michlmayr http://www.cyrius.com/

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add noise gate hold time enumeration
by Mark Brown 18 Dec '12

18 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++++++ sound/soc/codecs/arizona.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a85a68f..89c0433 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -323,6 +323,15 @@ const struct soc_enum arizona_lhpf4_mode = arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); +static const char *arizona_ng_hold_text[] = { + "30ms", "120ms", "250ms", "500ms", +}; + +const struct soc_enum arizona_ng_hold = + SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, + 4, arizona_ng_hold_text); +EXPORT_SYMBOL_GPL(arizona_ng_hold); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index db0c8dd..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -176,6 +176,8 @@ extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; +extern const struct soc_enum arizona_ng_hold; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: arizona: Correct FLL source definitions
by Mark Brown 18 Dec '12

18 Dec '12

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 <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 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 7f22b4f..68dfae5 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 -- 1.7.10.4

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[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of more Haswell HDMI codec pins
by mengdong.lin＠intel.com 18 Dec '12

18 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec may report invalid connection list entries on all three pins. But our previous patch only fixes the HDMI pin, not the DP pin, so audio playbak via DP can still fail. This patch adds fixup for Haswell to set proper connection list entries for the other 2 pins including the DP pin. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 42fdf14..73d298a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1710,8 +1710,14 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, if (vendor_param == -1 || !(vendor_param & 0x02)) return; + snd_hda_override_conn_list(codec, 0x05, 3, list); + hdmi_read_pin_conn(codec, 1); + snd_hda_override_conn_list(codec, 0x06, 3, list); hdmi_read_pin_conn(codec, 1); + + snd_hda_override_conn_list(codec, 0x07, 3, list); + hdmi_read_pin_conn(codec, 1); } -- 1.7.9.5

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Re: [alsa-devel] [pulseaudio-discuss] Front mic not working properly
by Raymond Yau 18 Dec '12

18 Dec '12

>> > >> > I'm totally stuck with a sound problem I fail to debug on my own. >> > >> > My setup: >> > Asus P5Q Motherboard with Realtec ALC1200 soundchip. Stereo speakers plugged in to the rear output and headphone speaker/mic combo plugged in to the front jacks. System: Ubuntu 12.10 >> > Desired behaviour: Sound is played back on headphones AND rear speaker. Front mic should "just-work" Asus P5Q specification Realtek ALC1200, 8-channel High Definition Audio CODEC - Support Jack-Detection, Multi-streaming, and Front Panel Jack-Retasking Alt_playback is still not enabled to support multi streaming playback on realtec codecs >> > Problem 2: >> > This is the biggest problem. >> > As soon as I plug in the front mic, Inside gnome-alsamixer the volume for master, surround,center,lfe start to jump between on and off. Why is the Front mic jack controlling these values? Do you mean BIOS did not setup pin default correctly after asus-p5q model was removed? http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6f… > How can I debug this behaviour? Post the output of alsa-info.sh > I just want them to stay the same, if the mic is plugged in or not. >> > Inside the gnome volume settings, the output port starts to jump between "Analog Output" and "Analog Headphones" >> >> This look like problem in Jack detection of the headphone and Mic Jack of front audio panel >> >> alsamixer -c0 >> >> Does the playback switch change rapidly on and off in alsamixer ? >> >> Do your computer chassis have a hda front audio panel and you connect them to the front audio connector of the motherboard according to the user manual ? >> >> Does it help if you disabled the automute switch ? >> > Thanks a lot. Your input helped me to at least work around the problem. My chassis has a HDA and an AC97 connector. The HDA connector was plugged in and correctly configured in BIOS. > I now use the AC97 connector (and changed the BIOS settings) and everything is working properly. Checking the front audio panel specification of your computer chassis Setting front audio panel to AC97 in BIOS of Asus motherboard change the miscellaneous bit of pin default of front HP and Mic Jack （i.e. misc = NO_PRESENCE) The alsa driver won't enable the unsolicited event on these nodes > > Regarding your questions: Disabling automute didn't help. And yes, the playback switch toggled rapidly. http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Try disable trigger sense by codec->no_trigger_sense = 1; > > One thing I didn't test: I mistakenly plugged the rear speaker into the front channel jack (plugging the green connector into a green jack seemed so obvious). I don't know is this is the same channel as the headphone connector on the front panel... > > In AC97 mode it doesn't matter if I use the front channel jack or the rear channel jack (both at the rear of the computer) to connect my main speakers.

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[alsa-devel] [PATCH 0/2] A few fixes for Reatek codecs
by Takashi Iwai 17 Dec '12

17 Dec '12

Hi, here is a patch series for fixing some obvious errors I've overseen for Realtek codecs. It'll be queued for 3.8-rc1. Takashi

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