May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Intel: Fix printk formatting
by Joonas Lahtinen 11 May '16

11 May '16

Format number after 0x in hex. Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Joonas Lahtinen <joonas.lahtinen(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c..994256b 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { -- 2.5.5

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[alsa-devel] [PATCH RFT] ASoC: max98371 Remove duplicate entry in max98371_reg
by Axel Lin 11 May '16

11 May '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 21fae09..cf0a39b 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -107,7 +107,6 @@ static const struct reg_default max98371_reg[] = { { 0x39, 0x00 }, { 0x3A, 0x00 }, { 0x3B, 0x00 }, - { 0x3B, 0x00 }, { 0x3C, 0x00 }, { 0x3D, 0x00 }, { 0x3E, 0x00 }, -- 2.5.0

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[alsa-devel] [PATCH v5 0/4] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 11 May '16

11 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 5 of the patch: * rebased to the current master of git://git.kernel.org/pub/scm/linux/kernel/git/nsekhar/linux-davinci.git * independent changes submitted separately (3/7 and 7/7 of the version 4) * 5/7 of the version 4 removed as already applied * corrected mcbsp0 and mcbsp1 nodes in the DTS file: using correct address and sound@ prefix

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[alsa-devel] [PATCH alsa-lib] pcm: dmix: Fix doubly resume of slave PCM
by Takashi Iwai 11 May '16

11 May '16

The dmix plugin and co may trigger the resume for each instance in snd_pcm_direct_resume(). It means that the slave PCM gets resumed or re-prepared/started by each opened dmix stream, and this may end up with the doubly triggers even though the slave PCM has been already resumed by another dmix stream. For avoiding this conflicts, check the slave PCM state and resume only when it's still in the suspended state. Meanwhile we keep the shadow state updated no matter whether the slave was triggered or not. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e28738b0de96..ac082f1a73b2 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -841,6 +841,12 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) int err; snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + /* resume only when the slave PCM is still in suspended state */ + if (snd_pcm_state(dmix->spcm) != SND_PCM_STATE_SUSPENDED) { + err = 0; + goto out; + } + err = snd_pcm_resume(dmix->spcm); if (err == -ENOSYS) { /* FIXME: error handling? */ @@ -848,6 +854,7 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) snd_pcm_start(dmix->spcm); err = 0; } + out: dmix->state = snd_pcm_state(dmix->spcm); snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); return err; -- 2.8.2

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[alsa-devel] ASoC: DAPM and differential inputs
by Jeremy McDermond 10 May '16

10 May '16

I’ve been working on the tlv320aic32x4 driver for a while now to add some features for some projects I’m working on using the chip. I’m trying to wrap my head around DAPM and getting it to work with the variety of inputs on the aic32x4. Basically, the codec has a set of registers that control the positive and negative inputs to each of the right and left input mixers. The only setting is what value of resistor to connect between the input pin and a particular mixer (none, 10k, 20k, 40k). For example, you can have a differential pair between IN1_R and IN2_R, or between IN1_R and IN3_R, or single ended by setting the negative pin to the common. So, there’s a lot of flexibility and not just a differential “mode”. The registers would even allow you to set up two negative pins, although I’m not sure why that would be useful. I’m currently setting up DAPM so that both the positive and negative pins are wired to a single mixer. This doesn’t seem to be working since only the negative pin seems to cause the path to be enabled. Do I have to set up another “Negative Right” and “Negative Left” mixer to handle the negative pins? Am I missing something else? -- Jeremy McDermond (NH6Z) Xenotropic Systems mcdermj(a)xenotropic.com

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[alsa-devel] [PATCH] MAINTAINERS: Add myself as reviewer of FSL/NXP SoC sound drivers
by Fabio Estevam 10 May '16

10 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> I would like to help reviewing FSL/NXP SoC sound drivers. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 43b85c1..f896605 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4846,6 +4846,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi <timur(a)tabi.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> +R: Fabio Estevam <fabio.estevam(a)nxp.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 1.9.1

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[alsa-devel] [PATCH] ASoC: simple-card: Add pm callbacks to platform driver
by Peter Ujfalusi 10 May '16

10 May '16

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, we have moved most of the TI boards to use simple-card and I have noticed that we have quite bad regression because of it caused by simple-card not having the pm ops configured. This will render audio to not recover after suspend/resume when the audio IP is loosing context. Regards, Peter sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c6111e7a6d93..52a1e1ff3c62 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -658,6 +658,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 2.8.1

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Do not allow multiple streams in one direction
by Peter Ujfalusi 10 May '16

10 May '16

Make sure that the user can not start multiple streams with the same direction. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c4cd8e4fff9a..50523e744483 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1252,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; -- 2.8.1

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[alsa-devel] [PATCH 0/4] ASoC: davinci-mcasp: Master mode clock divider correction
by Peter Ujfalusi 10 May '16

10 May '16

Hi, If the McASP is used as clock master and the reference clock is AUXCLK we can have additional level of divider. The BCLK divider is limited to maximum 32, if the desired bclk can not be reached with this, the AUXCLK divider also needs to be used. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Use defines for clkdiv IDs ASoC: davinci-mcasp: Change __davinci_mcasp_set_clkdiv() first parameter ASoC: davinci-mcasp: Restructure the davinci_mcasp_calc_clk_div() ASoC: davinci-mcasp: Calculate AUXCLK divider when setting up master clocks sound/soc/davinci/davinci-mcasp.c | 82 +++++++++++++++++++++++++-------------- sound/soc/davinci/davinci-mcasp.h | 5 +++ 2 files changed, 57 insertions(+), 30 deletions(-) -- 2.8.2

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[alsa-devel] [PATCH] ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS
by Kuninori Morimoto 10 May '16

10 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_dmapp_get_id() returns 0xFF as error code if system used strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is prohibited number. In order not to use prohibited number, this patch indicates error message and returns 0x00 (same as SSI00) in error case. Special thanks to Dung-san. Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8f..6bc93cb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } -- 1.9.1

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