December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for v3.13
by Mark Brown 13 Dec '13

13 Dec '13

The following changes since commit 374b105797c3d4f29c685f3be535c35f5689b30e: Linux 3.13-rc3 (2013-12-06 09:34:04 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc3 for you to fetch changes up to e20ab019e28dcf09c2727aa69e2a073ed66718b3: Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/tegra' and 'asoc/fix/wm8962' into asoc-linus (2013-12-12 00:30:04 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few driver and error handling fixes plus a fix to ensure that we mute streams when we should. The Atmel trigger addition is a fix to ensure that we do the correct sequence of interactions with the hardware. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dma' into asoc-linus Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/tegra' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver Stephen Warren (2): ASoC: tegra: fix uninitialized variables in set_fmt ASoC: don't leak on error in snd_dmaengine_pcm_register sound/soc/atmel/atmel_ssc_dai.c | 30 ++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/wm8962.c | 13 ++++++++++++ sound/soc/fsl/imx-wm8962.c | 2 -- sound/soc/soc-generic-dmaengine-pcm.c | 38 +++++++++++++++++++++++++---------- sound/soc/soc-pcm.c | 5 +++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 ++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 9 files changed, 84 insertions(+), 28 deletions(-)

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Re: [alsa-devel] [PATCH] ASOC:DAPM: extend dapm kcontrol to support runtime route update
by Mark Brown 13 Dec '13

13 Dec '13

On Thu, Dec 12, 2013 at 02:56:08AM -0800, Nenghua Cao wrote: Don't top post. Please also fix your mailer to word wrap within paragaphs, it makes your mails much more legible. > For common mix/mux which aren't related with DPCM, it doesn't need > to do DPCM update. And it still can use the old controls. But for > the mix/mux which impacts fe<->be link, they need to do this works. > For them, they call use the new set of controls. I think it can > make our code has better backward-compatibility. How does this improve backwards compatibility?

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[alsa-devel] [PATCH V2] ASOC:DAPM: update DPCM runtime when mixer/mux changes
by Nenghua Cao 13 Dec '13

13 Dec '13

From: Nenghua Cao <nhcao(a)marvell.com> V2 -> V1: Removes the flag and update DPCM always when mixer/mux changes, according to Takashi's and Mark's suggestion. >> Usually backward compatibility is concerned when something new breaks >> the existing ones. In your case, always updating DPCM would work, >> too, even without an extra flag; it's just suboptimal. >Right, and if we have separate versions then at some point we'll end up >having to define DPCM versions of everything which is going to get >tedious and error prone. V1 -> V0: refined patch, and add flag into soc_mixer_control and soc_enum instead of the alsa core, according to Takashi's suggestion. > You don't need to extend the ALSA core API for such a purpose. It's > ASoC-specific, so rather extend struct soc_mixer_control instead, if > such a new flag is really needed. Nenghua Cao (1): ASOC:DAPM: update DPCM runtime when mixer/mux changes sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 1 files changed, 24 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] ASoC: omap-pcm: Lower the dma coherent mask to 32bits
by Peter Ujfalusi 13 Dec '13

13 Dec '13

The underlying API dma_coerce_mask_and_coherent() checks the requested bits mask against the size of dma_addr_t which is 32bits on OMAP. Because of the previously used 64bits mask audio was not probing after commit c9bd5e6 (and 4dcfa6007). 32bits for the DMA_BIT_MASK looks to be the correct one to use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, This is the fix for OMAP audio for 3.13-rc2+. The dma coherent changes went in between -rc1 and -rc2 which broke things because omap-pcm was using 64bits for dma coherent mask. Regards, Peter sound/soc/omap/omap-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b8fa986..01d59d0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -202,7 +202,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; -- 1.8.4.5

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[alsa-devel] [PATCH V1] ASOC:DAPM: extend dapm kcontrol to support runtime route update
by Nenghua Cao 13 Dec '13

13 Dec '13

From: Nenghua Cao <nhcao(a)marvell.com> V1 -> V0: refined patch, and add flag into soc_mixer_control and soc_enum instead of the alsa core, according to Takashi's suggestion. >You don't need to extend the ALSA core API for such a purpose. It's >ASoC-specific, so rather extend struct soc_mixer_control instead, if >such a new flag is really needed. nhcao (1): ASOC:DAPM: extend dapm kcontrol to support runtime route update include/sound/soc-dapm.h | 14 ++++++++++++++ include/sound/soc.h | 19 +++++++++++++++++++ sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 3 files changed, 57 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] alsa-lib: minor malloc changes
by Olivier Langlois 13 Dec '13

13 Dec '13

replace malloc + memset with calloc since calloc may skip the memset if returned memory comes directly from sbrk() use malloc instead of malloc if first thing done with allocated memory is to flip all bits to 1. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/alisp/alisp.c | 3 +-- src/pcm/pcm_ladspa.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/src/alisp/alisp.c b/src/alisp/alisp.c index 1796c33..3c61bb1 100644 --- a/src/alisp/alisp.c +++ b/src/alisp/alisp.c @@ -3202,12 +3202,11 @@ int alsa_lisp(struct alisp_cfg *cfg, struct alisp_instance **_instance) struct alisp_object *p, *p1; int i, j, retval = 0; - instance = (struct alisp_instance *)malloc(sizeof(struct alisp_instance)); + instance = (struct alisp_instance *)calloc(1, sizeof(struct alisp_instance)); if (instance == NULL) { nomem(); return -ENOMEM; } - memset(instance, 0, sizeof(struct alisp_instance)); instance->verbose = cfg->verbose && cfg->vout; instance->warning = cfg->warning && cfg->wout; instance->debug = cfg->debug && cfg->dout; diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 0a9c52e..9ce5242 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -1326,7 +1326,7 @@ static int snd_pcm_ladspa_parse_bindings(snd_pcm_ladspa_plugin_t *lplug, count = (unsigned int)(channel + 1); } if (count > 0) { - array = (unsigned int *)calloc(count, sizeof(unsigned int)); + array = (unsigned int *)malloc(count * sizeof(unsigned int)); if (! array) return -ENOMEM; memset(array, 0xff, count * sizeof(unsigned int)); -- 1.8.5.1

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[alsa-devel] [PATCH] ALSA: hda - Add Dell headset detection quirk for three laptop models
by Hui Wang 13 Dec '13

13 Dec '13

On the Dell machines with codec whose Subsystem Id is 0x10280610, 0x10280629 or 0x1028063e, no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. The codecs on these machines belong to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c5f3e1b..3578f11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4300,11 +4300,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: tegra: update module reset list for Tegra124
by Stephen Warren 12 Dec '13

12 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra124 adds a number of extra modules into the configlink bus, which must be taken out of reset before the bus is used. Update the AHUB driver to know about these extra modules (the AHUB HW module hosts the configlink bus). Based-on-work-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Based-on-work-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This patch depends on "ASoC: tegra: use reset framework" to compile, which is ack'd and slated to go through a (large) topic branch in the Tegra tree. So, we can either: a) Merge that Tegra topic branch into the ASoC tree, then apply this. Note that I haven't created the topic branch yet, since I'm still waiting for DMA dependencies to be applied. b) Apply this change to the Tegra tree too. This change isn't directly related to the changes in the Tegra tree; it just makes use of the new reset controller feature that's introduced there. .../bindings/sound/nvidia,tegra30-ahub.txt | 9 +++++++++ sound/soc/tegra/tegra30_ahub.c | 20 ++++++++++++++++---- 2 files changed, 25 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt index 32de7ec789aa..946e2ac46091 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -30,6 +30,15 @@ Required properties: Tegra114 and later additionally require: - amx - adx + Tegra124 and later additionally require: + - amx1 + - adx1 + - afc0 + - afc1 + - afc2 + - afc3 + - afc4 + - afc5 - ranges : The bus address mapping for the configlink register bus. Can be empty since the mapping is 1:1. - dmas : Must contain an entry for each entry in clock-names. diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 342cd4fff0a5..d6f4c9940e0c 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -335,9 +335,13 @@ EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); #define MOD_LIST_MASK_TEGRA30 BIT(0) #define MOD_LIST_MASK_TEGRA114 BIT(1) +#define MOD_LIST_MASK_TEGRA124 BIT(2) #define MOD_LIST_MASK_TEGRA30_OR_LATER \ - (MOD_LIST_MASK_TEGRA30 | MOD_LIST_MASK_TEGRA114) + (MOD_LIST_MASK_TEGRA30 | MOD_LIST_MASK_TEGRA114 | \ + MOD_LIST_MASK_TEGRA124) +#define MOD_LIST_MASK_TEGRA114_OR_LATER \ + (MOD_LIST_MASK_TEGRA114 | MOD_LIST_MASK_TEGRA124) static const struct { const char *rst_name; @@ -352,8 +356,16 @@ static const struct { { "dam1", MOD_LIST_MASK_TEGRA30_OR_LATER }, { "dam2", MOD_LIST_MASK_TEGRA30_OR_LATER }, { "spdif", MOD_LIST_MASK_TEGRA30_OR_LATER }, - { "amx", MOD_LIST_MASK_TEGRA114 }, - { "adx", MOD_LIST_MASK_TEGRA114 }, + { "amx", MOD_LIST_MASK_TEGRA114_OR_LATER }, + { "adx", MOD_LIST_MASK_TEGRA114_OR_LATER }, + { "amx1", MOD_LIST_MASK_TEGRA124 }, + { "adx1", MOD_LIST_MASK_TEGRA124 }, + { "afc0", MOD_LIST_MASK_TEGRA124 }, + { "afc1", MOD_LIST_MASK_TEGRA124 }, + { "afc2", MOD_LIST_MASK_TEGRA124 }, + { "afc3", MOD_LIST_MASK_TEGRA124 }, + { "afc4", MOD_LIST_MASK_TEGRA124 }, + { "afc5", MOD_LIST_MASK_TEGRA124 }, }; #define LAST_REG(name) \ @@ -492,7 +504,7 @@ static struct tegra30_ahub_soc_data soc_data_tegra114 = { }; static struct tegra30_ahub_soc_data soc_data_tegra124 = { - .mod_list_mask = MOD_LIST_MASK_TEGRA114, + .mod_list_mask = MOD_LIST_MASK_TEGRA124, .set_audio_cif = tegra124_ahub_set_cif, }; -- 1.8.1.5

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[alsa-devel] snd-usb driver
by Lucas Takejame 12 Dec '13

12 Dec '13

Hello guys, I'm new here. The company that i work for is developing a LV2 host pedal board (http://portalmod.com/en/index.html) and my job now is to find if I can improve in anyway our usb driver (which is the kernel's snd-usb driver). I know the code is already optimized but I was hoping that, since we only use one specific sound card and we won't need a generic driver, I could change something in the code to optimize it latency wise. I have only a brief knowledge on the USB protocol and even less on writing drivers, so I'm here first to ask if it's possible to improve latency by changing the driver's code and, if possible, if you guys could give me some directions on how to do it. Thanks in advance, Lucas

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[alsa-devel] how to create alsa nodes?
by loody 12 Dec '13

12 Dec '13

Dear all: In destop x86 system, the udev will automatically create the nodes in snd directory, such as pcmC0D0c, pcmC0D1c, etc. I try to let the usb sound device work on mips platform, but I have no idea what is the proper way to create these nodes. Should I 1. create them one by one by hand from copying the attributes in my PC (if I have 2 or more usb sound devices, I have to first plug them in pc and monitor the nodes) 2. smarter way to do so? -- Regards, miloody

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