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

[alsa-devel] [Resend PATCH] ASoC: wm8904: remove redundant code
by Bo Shen 19 Sep '12

19 Sep '12

The core_intercon register two times for wm8904 Remove one Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index dc4262e..7c8df52 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1185,8 +1185,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, core_intercon, - ARRAY_SIZE(core_intercon)); snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); snd_soc_dapm_add_routes(dapm, dac_intercon, -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - avoid non-standard "Docking" name in mixers
by David Henningsson 19 Sep '12

19 Sep '12

The standard name (and what PulseAudio picks up) is "Dock Mic", not "Docking Mic" or "Docking-Station". Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_analog.c | 14 +++++++------- sound/pci/hda/patch_conexant.c | 4 ++-- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a82cce..cdd43ea 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1810,7 +1810,7 @@ static const struct hda_input_mux ad1981_hp_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, - { "Docking-Station", 0x1 }, + { "Dock Mic", 0x1 }, { "Mix", 0x2 }, }, }; @@ -1834,8 +1834,8 @@ static const struct snd_kcontrol_new ad1981_hp_mixers[] = { */ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), /* FIXME: does this laptop have analog CD connection? */ @@ -3565,7 +3565,7 @@ static const struct hda_amp_list ad1884_loopbacks[] = { static const char * const ad1884_slave_vols[] = { "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958", + "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958", NULL }; @@ -3626,7 +3626,7 @@ static const struct hda_input_mux ad1984_thinkpad_capture_source = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, { "Mix", 0x3 }, - { "Docking-Station", 0x4 }, + { "Dock Mic", 0x4 }, }, }; @@ -3655,8 +3655,8 @@ static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7d70210..b871b013 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1710,8 +1710,8 @@ static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; -- 1.7.9.5

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Re: [alsa-devel] [PATCH v4 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 19 Sep '12

19 Sep '12

Hi Samuel, On 09/19/2012 03:24 AM, Samuel Ortiz wrote: > Hi Peter, > > On Mon, Sep 10, 2012 at 01:46:18PM +0300, Peter Ujfalusi wrote: >> Hello, >> >> Generated on top of: >> git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap > I applied the first 8 patches, but I'd like to get Mark's ACK for the rest of > the serie. > Unless you're expecting Mark to take the whole thing, in which case you can > add my: > > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> > > to the MFD patches. > Please let me know how you'd like to proceed. If Mark is still OK with it I think it would be better to queue this series via ASoC so we can keep them in one place (also I think patch 14 needs patch 6 in order to not break x86/x86_64 allmodconfig). On the other hand if they go separately (1-8 via MFD, 9-14 via ASoC) they will meet in linux-next where this matters the most. Thank you, Péter

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[alsa-devel] Need help with WM8960
by Timur Tabi 19 Sep '12

19 Sep '12

Mark, I'm trying to port some old Linux code to upstream for a third-party board that uses a Freescale P1022 and a Wolfson WM8960. I've got audio working, but it's using a hack from the third-party code, and I don't know how to do implement the change properly, mostly because I don't understand DAPM. The first problem is that I get this: wm8960 0-001a: No platform data supplied What kind of platform data is the driver expecting, and how do I get it that data? I don't seem to need any platform data to make the code work. Second, I uploaded the hack to wm8960.c at http://pastebin.com/RwVy6cQg You can see that he disables the DAPM controls for DACL, DACR, ADCL, and ADCR, and forces those bits set in the wm8960_probe function. There's also a comment that documents some limitations. What is the proper way to implement this change? Third, in order to actually hear audio, I had to unmute the "Left Output Mixer PCM" and "Right Output Mixer PCM". Is this something that the machine driver should do automatically? And if so, how? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH] ASoC: imx-audmux: remove null check of audmux_base in audmux_read_file
by Richard Zhao 19 Sep '12

19 Sep '12

When audmux_read_file is called, it means the driver is already initialised successfully, so we don't need to check audmux_base. It also fix smatch warning: sound/soc/fsl/imx-audmux.c:78 audmux_read_file() warn: possible memory leak of 'buf' Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e7c800e..524ce62 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -74,9 +74,6 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (!audmux_base) - return -ENOSYS; - if (audmux_clk) clk_prepare_enable(audmux_clk); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: imx-pcm-dma: check kzalloc return value in function snd_imx_open
by Richard Zhao 19 Sep '12

19 Sep '12

It fixed smatch warning: sound/soc/fsl/imx-pcm-dma.c:112 snd_imx_open() error: potential null dereference 'dma_data'. (kzalloc returns null) Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/fsl/imx-pcm-dma.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 48f9d88..a23505a 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,6 +109,9 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); + if (!dma_data) + return -ENOMEM; + dma_data->peripheral_type = dma_params->shared_peripheral ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Avoid recalculating the bitmask for SOC_ENUM controls
by Lars-Peter Clausen 19 Sep '12

19 Sep '12

For ENUM controls the bitmask is calculated based on the number of items. Currently this is done each time the control is accessed. And while the performance impact of this should be negligible we can easily do better. The roundup_pow_of_two macro performs the same calculation which is currently done manually, but it is also possible to use this macro with compile time constants and so it can be used to initialize static data. So we can use it to initialize the mask field of a ENUM control during its declaration. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 +++- sound/soc/codecs/twl4030.c | 8 +++----- sound/soc/soc-core.c | 16 ++++++---------- sound/soc/soc-dapm.c | 22 ++++++++-------------- 4 files changed, 20 insertions(+), 30 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 313b766..91244a0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -20,6 +20,7 @@ #include <linux/interrupt.h> #include <linux/kernel.h> #include <linux/regmap.h> +#include <linux/log2.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/compress_driver.h> @@ -160,7 +161,8 @@ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .max = xmax, .texts = xtexts } + .max = xmax, .texts = xtexts, \ + .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0} #define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts) #define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc..2548f5c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -999,7 +999,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; - unsigned short mask, bitmask; + unsigned short mask; if (twl4030->configured) { dev_err(codec->dev, @@ -1007,18 +1007,16 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits(codec, e->reg, mask, val); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5b0713..9a6daf9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2413,16 +2413,14 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] - = (val >> e->shift_l) & (bitmask - 1); + = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2443,19 +2441,17 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; - unsigned int mask, bitmask; + unsigned int mask; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits_locked(codec, e->reg, mask, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3b8ddb4..873e6e7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -358,12 +358,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrol_news[i].private_value; - int val, item, bitmask; + int val, item; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = soc_widget_read(w, e->reg); - item = (val >> e->shift_l) & (bitmask - 1); + item = (val >> e->shift_l) & e->mask; p->connect = 0; for (i = 0; i < e->max; i++) { @@ -2680,15 +2678,13 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(widget->codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2712,22 +2708,20 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; - unsigned int mask, bitmask; + unsigned int mask; struct snd_soc_dapm_update update; int wi; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); -- 1.7.10.4

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[alsa-devel] [PATCH v2 0/3] Allow suspend/resume with USB audio.
by Dylan Reid 18 Sep '12

18 Sep '12

Problem being addressed: shell one$ aplay -Dhw:<usbcard> file.wav shell two$ echo mem > /sys/power/state After resume, aplay attempts resume, then restart, then after a timeout gets a write error and no audio will be played until it is restarted. During suspend/resume, if the USB port remains powered the audio device will still be present upon resume. The interface and endpoint have to be reconfigured however. These patches move the configuration from hw_params to prepare causing re-configuration during resume. Is this the right way to fix this? Or is there an easier way I am missing? Thanks for looking. v2: fix unused variable warnings. Dylan Reid (3): ALSA: usb-audio: set period_bytes in substream. ALSA: usb-audio: Don't require hw_params in endpoint. ALSA: usb-audio: Move configuration to prepare. sound/usb/card.h | 2 + sound/usb/endpoint.c | 29 +++++++----- sound/usb/endpoint.h | 5 ++- sound/usb/pcm.c | 121 +++++++++++++++++++++++++++++++-------------------- 4 files changed, 97 insertions(+), 60 deletions(-) -- 1.7.12.146.g16d26b1

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[alsa-devel] [PATCH 0/3] Allow suspend/resume with USB audio.
by Dylan Reid 18 Sep '12

18 Sep '12

Problem being addressed: shell one$ aplay -Dhw:<usbcard> file.wav shell two$ echo mem > /sys/power/state After resume, aplay attempts resume, then restart, then after a timeout gets a write error and no audio will be played until it is restarted. During suspend/resume, if the USB port remains powered the audio device will still be present upon resume. The interface and endpoint have to be reconfigured however. These patches move the configuration from hw_params to prepare causing re-configuration during resume. Is this the right way to fix this? Or is there an easier way I am missing? Thanks for looking. Dylan Reid (3): ALSA: usb-audio: set period_bytes in substream. ALSA: usb-audio: Don't require hw_params in endpoint. ALSA: usb-audio: Move configuration to prepare. sound/usb/card.h | 2 + sound/usb/endpoint.c | 29 +++++++----- sound/usb/endpoint.h | 5 ++- sound/usb/pcm.c | 121 +++++++++++++++++++++++++++++++-------------------- 4 files changed, 97 insertions(+), 60 deletions(-) -- 1.7.12.146.g16d26b1

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[alsa-devel] what is wrong?
by Eva Lovi 18 Sep '12

18 Sep '12

Hello, Does Alsa support this soundcard? card 0: I82801AAICH [Intel 82801AA-ICH], device 0: Intel ICH [Intel 82801AA-ICH] I tried aplay, but I don't hear anything. In amixer I increased the volume, but I still don't hear anything. What's the problem? Eva Lovi

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