September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rsnd: add rsnd_dma_alloc()
by Kuninori Morimoto 21 Sep '17

21 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car sound DMA will be used from SSI/SRC. dma.c doesn't alloc DMA handler in .probe timing, because we don't know what kind of DMA transfer will be used then. Thus, SSI/SRC have *rsnd_mod for DMA. rsnd_dma_attach() will allocate it and attach it to system. It will be PIO mode if it can't alloc DMA handler. In case of MIX is used, rsnd_dma_attach() will be called twice from SSI. To avoid duplicate allocation, current rsnd_dma_attach() is checking allocated DMA handler. This DMA related operation is a little bit difficult to understand. This patch adds new rsnd_dma_alloc() and separates allocation and attach for readable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 60 +++++++++++++++++++++++++++---------------------- 1 file changed, 33 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 041ec10..17220c9 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -753,14 +753,15 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, } } -int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, - struct rsnd_mod **dma_mod) +static int rsnd_dma_alloc(struct rsnd_dai_stream *io, struct rsnd_mod *mod, + struct rsnd_mod **dma_mod) { struct rsnd_mod *mod_from = NULL; struct rsnd_mod *mod_to = NULL; struct rsnd_priv *priv = rsnd_io_to_priv(io); struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dma *dma; struct rsnd_mod_ops *ops; enum rsnd_mod_type type; int (*attach)(struct rsnd_dai_stream *io, struct rsnd_dma *dma, @@ -800,40 +801,45 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, type = RSND_MOD_AUDMA; } - if (!(*dma_mod)) { - struct rsnd_dma *dma; + dma = devm_kzalloc(dev, sizeof(*dma), GFP_KERNEL); + if (!dma) + return -ENOMEM; - dma = devm_kzalloc(dev, sizeof(*dma), GFP_KERNEL); - if (!dma) - return -ENOMEM; + *dma_mod = rsnd_mod_get(dma); - *dma_mod = rsnd_mod_get(dma); + ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, + rsnd_mod_get_status, type, dma_id); + if (ret < 0) + return ret; - ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, - rsnd_mod_get_status, type, dma_id); - if (ret < 0) - return ret; + dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", + rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), + rsnd_mod_name(mod_from), rsnd_mod_id(mod_from), + rsnd_mod_name(mod_to), rsnd_mod_id(mod_to)); + + ret = attach(io, dma, mod_from, mod_to); + if (ret < 0) + return ret; - dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", - rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), - rsnd_mod_name(mod_from), rsnd_mod_id(mod_from), - rsnd_mod_name(mod_to), rsnd_mod_id(mod_to)); + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + dma->mod_from = mod_from; + dma->mod_to = mod_to; + + return 0; +} + +int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, + struct rsnd_mod **dma_mod) +{ + if (!(*dma_mod)) { + int ret = rsnd_dma_alloc(io, mod, dma_mod); - ret = attach(io, dma, mod_from, mod_to); if (ret < 0) return ret; - - dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); - dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); - dma->mod_from = mod_from; - dma->mod_to = mod_to; } - ret = rsnd_dai_connect(*dma_mod, io, type); - if (ret < 0) - return ret; - - return 0; + return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } int rsnd_dma_probe(struct rsnd_priv *priv) -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: Handle return value of devm_kasprintf
by Arvind Yadav 21 Sep '17

21 Sep '17

devm_kasprintf() can fail here and we must check its return value. Arvind Yadav (3): [PATCH 1/3] ASoC: davinci-mcasp: Handle return value of devm_kasprintf [PATCH 2/3] ASoC: fsl-asoc-card: Handle return value of devm_kasprintf [PATCH 3/3] ASoC: omap-hdmi-audio: Handle return value of devm_kasprintf sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ sound/soc/fsl/fsl-asoc-card.c | 2 ++ sound/soc/omap/omap-hdmi-audio.c | 3 +++ 3 files changed, 17 insertions(+) -- 1.9.1

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[alsa-devel] [PATCH v2 2/3] ASoC: fsl-asoc-card: Handle return value of devm_kasprintf
by Arvind Yadav 21 Sep '17

21 Sep '17

devm_kasprintf() can fail here and we must check its return value. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2: Set return 'ret' to -ENOMEM. sound/soc/fsl/fsl-asoc-card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2db4d0c..a389885 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -639,6 +639,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); + if (!priv->dai_link[0].codec_name) { + ret = -ENOMEM; + goto asrc_fail; + } } priv->dai_link[0].platform_of_node = cpu_np; -- 1.9.1

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[alsa-devel] How to play music through 3 sound cards simultaneously by aplay
by 韩 21 Sep '17

21 Sep '17

There are three wm8960 codecs in my paltform. I want to test playing music of these three cards at the same time. But I do not know how to realize this idea by alsa tool aplay. Can someone give me some hints? Best regards, Yuejuan

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[alsa-devel] [asoc:topic/wm97xx 4/8] sound/soc/codecs/wm9713.c:27:30: fatal error: sound/ac97/codec.h: No such file or directory
by kbuild test robot 20 Sep '17

20 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm97xx head: 5aa3b03ad0712e3964a0ae2788f61f08da814bfe commit: 9bd400cadae05a26c3a22dd8393081cb96d0e26a [4/8] ASoC: wm9713: add ac97 new bus support config: frv-allmodconfig (attached as .config) compiler: frv-linux-gcc (GCC) 6.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 9bd400cadae05a26c3a22dd8393081cb96d0e26a # save the attached .config to linux build tree make.cross ARCH=frv All errors (new ones prefixed by >>): >> sound/soc/codecs/wm9713.c:27:30: fatal error: sound/ac97/codec.h: No such file or directory #include <sound/ac97/codec.h> ^ compilation terminated. vim +27 sound/soc/codecs/wm9713.c > 27 #include <sound/ac97/codec.h> 28 #include <sound/ac97/compat.h> 29 #include <sound/initval.h> 30 #include <sound/pcm_params.h> 31 #include <sound/tlv.h> 32 #include <sound/soc.h> 33 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/7] Refactor audio device tree handling
by Charles Keepax 20 Sep '17

20 Sep '17

Move the audio device tree binding out into its own documentation file, as is already done for the regulator and extcon bindings. Then add support for a new device tree property allowing the volume limits on the CODEC to be configured. It should be possible to put the patches through their separate trees. Thanks, Charles Charles Keepax (7): ASoC: arizona: Add new common Arizona init function ASoC: arizona: Add handling for audio related device tree entries ASoC: arizona: Add audio device tree bindings mfd: arizona: Remove audio related device tree code mfd: arizona: Remove audio bindings from MFD binding document ASoC: arizona: Add support for setting the output volume limits ASoC: arizona: Add device tree binding doc for volume limits Documentation/devicetree/bindings/mfd/arizona.txt | 40 +---- .../devicetree/bindings/sound/wlf,arizona.txt | 53 +++++++ MAINTAINERS | 1 + drivers/mfd/arizona-core.c | 132 +--------------- include/linux/mfd/arizona/pdata.h | 3 + sound/soc/codecs/arizona.c | 166 ++++++++++++++++++++- sound/soc/codecs/arizona.h | 6 +- sound/soc/codecs/cs47l24.c | 14 +- sound/soc/codecs/wm5102.c | 14 +- sound/soc/codecs/wm5110.c | 14 +- sound/soc/codecs/wm8997.c | 15 +- sound/soc/codecs/wm8998.c | 11 +- 12 files changed, 288 insertions(+), 181 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,arizona.txt -- 2.11.0

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[alsa-devel] alsa-lib and floating point support
by Ricard Wanderlof 20 Sep '17

20 Sep '17

We recently came across an issue with the alsa-lib softvol plugin, which requires hardware floating-point (i.e. --with-softfloat not enabled during build) support in order to use anything but a set of default values (min gain -51 dB, max 0 dB, and 256 gain steps). With no hardware floating point support, the plugin just prints an error message when loaded. static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, ... #ifndef HAVE_SOFT_FLOAT ... double v = (pow(10.0, db / 20.0) * (double)(1 << VOL_SCALE_SHIFT)); ... #else SNDERR("Cannot handle the given dB range and resolution"); return -EINVAL; #endif } ... I'm trying to understand why it is critical to have hardware floating point support in this case. The pow() function is very resource hungry, true, but the function is only called when the plugin is loaded, and not for instance on every sample while the stream is running, so on the whole I would expect the impact to be minimal. Is there some other rationale that I'm missing? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] ASoC: rl6231: make arrays div and pd static const, reduces object code size
by Colin King 20 Sep '17

20 Sep '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the read-only arrays div and pd on the stack, instead make them static const. Makes the object code smaller by 210 bytes: Before: text data bss dec hex filename 2869 720 0 3589 e05 sound/soc/codecs/rl6231.o After: text data bss dec hex filename 2495 880 0 3375 d2f sound/soc/codecs/rl6231.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rl6231.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 7b447d0b173a..974a9040651d 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -71,7 +71,7 @@ EXPORT_SYMBOL_GPL(rl6231_get_pre_div); */ int rl6231_calc_dmic_clk(int rate) { - int div[] = {2, 3, 4, 6, 8, 12}; + static const int div[] = {2, 3, 4, 6, 8, 12}; int i; if (rate < 1000000 * div[0]) { @@ -189,7 +189,8 @@ EXPORT_SYMBOL_GPL(rl6231_pll_calc); int rl6231_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.14.1

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[alsa-devel] [PATCH RESEND 0/3] Intel-rt5670 updates
by Pierre-Louis Bossart 20 Sep '17

20 Sep '17

[when I started cleaning-up some patches reviewed by Andy Shevchenko, I realized this initial batch has not been merged, although both Takashi and Vinod reviewed them, see http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125331.… no change except for the addition of Takashi's Reviewed-by and Vinod's Acked-by. I don't mean to be pushy or anything, just checking if something is missing ] Instead of submitting all my patches at ones, I broke them by platforms/codecs. This series is only for Intel-rt5670 platforms. It fixes a mistake on the Dell Venue 5585 - corrected with the help from Realtek and Dell who kindly provided schematics and data sheet information needed to fix non-working DMIC capture. Since there are still some reports of non-functional audio on some platforms, I added quirks that can be changed from the kernel command line (and later added as DMI-based quirks). The last two patches are minor corrections to fix ACPI HID naming issues and use a more meaningful card name. Pierre-Louis Bossart (3): ASoC: rt5670: refactor DMI quirks and fix Dell Venue settings ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name sound/soc/codecs/rt5670.c | 124 +++++++++++++++++++++++++++----- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++- 2 files changed, 112 insertions(+), 22 deletions(-) -- 2.11.0

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[alsa-devel] [asoc:topic/wm97xx 5/8] sound/soc/codecs/wm9712.c:23:30: fatal error: sound/ac97/codec.h: No such file or directory
by kbuild test robot 20 Sep '17

20 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm97xx head: 5aa3b03ad0712e3964a0ae2788f61f08da814bfe commit: 2ed1a8e0ce8db6d36f849526db61ce3c85a9f8d1 [5/8] ASoC: wm9712: add ac97 new bus support config: arm-multi_v7_defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 2ed1a8e0ce8db6d36f849526db61ce3c85a9f8d1 # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): >> sound/soc/codecs/wm9712.c:23:30: fatal error: sound/ac97/codec.h: No such file or directory #include <sound/ac97/codec.h> ^ compilation terminated. vim +23 sound/soc/codecs/wm9712.c > 23 #include <sound/ac97/codec.h> 24 #include <sound/ac97/compat.h> 25 #include <sound/initval.h> 26 #include <sound/soc.h> 27 #include <sound/tlv.h> 28 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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