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

[alsa-devel] [PATCH 0/7] constify snd_soc_ops structures
by Julia Lawall 17 Oct '16

17 Oct '16

Constify snd_soc_ops structures. --- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-wm8962.c | 2 +- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-scu-card.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- sound/soc/intel/boards/bxt_rt298.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++-- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 6 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 6 +++--- sound/soc/intel/boards/skl_rt286.c | 4 ++-- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 6 +++--- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- 32 files changed, 46 insertions(+), 46 deletions(-)

2 8

[alsa-devel] PCM params and Sub unit direction:Invalid argument
by vbs 17 Oct '16

17 Oct '16

Hello gentlemen, I have a problem using libasound (1.1.0-0ubuntu1) in a program (Reicast emulator) with my system. When trying to initialize the ALSA device in the code by invoking "snd_pcm_hw_params" I get "Invalid argument". I stripped the problem down to a simple demo program: http://pastebin.com/DMQrcYft When I run it I get: Unable to set hw parameters: Invalid argument So the call to "snd_pcm_hw_params" fails on my system. I am using Lubuntu x64 16.04 on a NUC D54250. This is my .asoundrc: pcm.softvol { type softvol slave { pcm "hw:0,3" } control { name "Master" card 0 } } pcm.!default { type plug slave.pcm "softvol" } I created a softvol device to have a master volume mixer (some programs rely on it). This is the output of "aplay -L": http://pastebin.com/k0qkfNnd Two points I have noticed: 1. The problem disappears when not using the softvol device. When I replace my .asoundrc with this, then it works: pcm.!default { type hw card 0 device 3 } So the problems might be related to the defined softvol device. 2. In the sample program there is this code: int dir; ... rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); The variable dir is not initialized. But AFAIK that parameter in the function "snd_pcm_hw_params_set_rate_near" is also used as an input value. But the uninitialized variable will just have an undefined state. So this looks fishy to me. When I change the definition to "int dir=0" then it actually works! With "int dir=1" the problem happens again. I would think this is just a bug in the code but the strange thing is that the official example does it in the same way (calling "snd_pcm_hw_params_set_buffer_time_near" with an uninitialized dir variable): http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html <http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html> How can that unintialized variable be correct if it really is used as an input value? Maybe someone has thoughts on this? Is the code correct? Especially regarding the unintialized variable? Are there maybe problems with my .asoundrc softvol device? Big thanks in advance guys! Cheers, Robin

1 0

[alsa-devel] [PATCH 1/3] ASoC: ak4641: Remove unused DAI ID defines
by Lars-Peter Clausen 15 Oct '16

15 Oct '16

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. This driver was merged after the change of DAIs are matched was made and the IDs were never used and presumably never will be, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4641.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/ak4641.h b/sound/soc/codecs/ak4641.h index 4a26324..38b9e93 100644 --- a/sound/soc/codecs/ak4641.h +++ b/sound/soc/codecs/ak4641.h @@ -38,10 +38,4 @@ #define AK4641_CACHEREGNUM 0x14 - - -#define AK4641_DAI_HIFI 0 -#define AK4641_DAI_VOICE 1 - - #endif -- 2.1.4

1 2

[alsa-devel] [PATCH] ASoC: es8328: Remove some unused defines
by Lars-Peter Clausen 15 Oct '16

15 Oct '16

The es8328 drivers seems to have used the wm8753 driver as a template. Its header file contains a few defines that are copied verbatim from the wm8753 header that do not really apply to this driver and are unused. So remove those defines. This seems safe as they deal with PLL and clock divider settings and the es8328 driver neither implements a set_pll() nor a set_clkdiv() callback. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/es8328.h | 37 ------------------------------------- 1 file changed, 37 deletions(-) diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 1a736e7..8930322 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -278,43 +278,6 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_REG_MAX 0x35 -#define ES8328_PLL1 0 -#define ES8328_PLL2 1 - -/* clock inputs */ -#define ES8328_MCLK 0 -#define ES8328_PCMCLK 1 - -/* clock divider id's */ -#define ES8328_PCMDIV 0 -#define ES8328_BCLKDIV 1 -#define ES8328_VXCLKDIV 2 - -/* PCM clock dividers */ -#define ES8328_PCM_DIV_1 (0 << 6) -#define ES8328_PCM_DIV_3 (2 << 6) -#define ES8328_PCM_DIV_5_5 (3 << 6) -#define ES8328_PCM_DIV_2 (4 << 6) -#define ES8328_PCM_DIV_4 (5 << 6) -#define ES8328_PCM_DIV_6 (6 << 6) -#define ES8328_PCM_DIV_8 (7 << 6) - -/* BCLK clock dividers */ -#define ES8328_BCLK_DIV_1 (0 << 7) -#define ES8328_BCLK_DIV_2 (1 << 7) -#define ES8328_BCLK_DIV_4 (2 << 7) -#define ES8328_BCLK_DIV_8 (3 << 7) - -/* VXCLK clock dividers */ -#define ES8328_VXCLK_DIV_1 (0 << 6) -#define ES8328_VXCLK_DIV_2 (1 << 6) -#define ES8328_VXCLK_DIV_4 (2 << 6) -#define ES8328_VXCLK_DIV_8 (3 << 6) -#define ES8328_VXCLK_DIV_16 (4 << 6) - -#define ES8328_DAI_HIFI 0 -#define ES8328_DAI_VOICE 1 - #define ES8328_1536FS 1536 #define ES8328_1024FS 1024 #define ES8328_768FS 768 -- 2.1.4

1 0

Re: [alsa-devel] [FFADO-devel] Clicking issues with the output of an Onyx Blackbird
by Jordi Torres 15 Oct '16

15 Oct '16

> (CCed to alsa-devel) I sent this issue to alsa mailist 6 days ago but I supouse that hit the spam filter. Even I published an alsa-info http://www.alsa-project.org/db/?f=cb6745b6d0acdb5d9375096ec8dc008f6ff49cd7 When I read the wiki about reporting bugs, I understood that kernel related bugs goes to bugzilla.kernel.org and user-land bugs goes to the devel mail-list. So I supouse that I should sent this issue to the bugzilla as it's considered a kernel bug > Well, could I request you to gather information of packet sequence? Here is the trace of packets: https://drive.google.com/file/d/0Bz44Dr5G2FeJSXBHMzc5WTFmOWs/view?usp=shari… The trace includes the opening of alsa device and playing a sine wave for 30 seconds My linux box has two firewire ports I don't know the details about the host controller but could be feasible sniff the connection between the other computer with official drivers and the DAW? Could that be useful? PD: Resent due to using the wrong sender address

2 2

[alsa-devel] struct snd_soc_codec_driver nested and duplicated fields
by Christian N 15 Oct '16

15 Oct '16

Hi I'm tryng to understand serveral links between asoc structures to avoid bad codec/machine driver implementation (TEF663x for iMX6). I saw that several structures have pointer to structures which one could have pointer to the first one, and serveral fields are duplicated. Some examples snd_soc_codec_driver->probe snd_soc_codec_driver->snd_soc_dapm_widget but snd_soc_codec_driver->snd_soc_component_driver->probe (duplicated?) snd_soc_codec_driver->snd_soc_component_driver->snd_soc_dapm_widget (duplicated?) and snd_soc_codec_driver->snd_soc_component_driver->snd_soc_dapm_widget->snd_soc_dapm_context-> snd_soc_component (hey is recursive???!!!!!) I would like figure out why several fields are duplicated in structures that could be nested. Is a compatibility issue? thanks in advance

2 1

[alsa-devel] [TDM help please] multiplex out a stereo I2S line
by Matt Flax 14 Oct '16

14 Oct '16

Hi there, Problem is that I need multiplex all 6 channels out a stereo I2S line. I have a working system ... the CPU thinks it is stereo I2S, but the codec thinks it is TDM multichannel. When I aplay a pre-interleaved wav file, it works ! Check the video here : https://www.kickstarter.com/projects/1250664710/1299781340?token=e465b540 Problem is that I need to multiplex all 6 channels down the stereo I2S line. I have tried to do this with the asoundrc by attempting to declare the same device 3 times and interleaving there, however have failed. I am happy to alter my machine driver to implement this ... Is there a way to present 3 stereo subdevices to the user, where each of these subdevcies is interleaved to form the TDM before sending to the codec ? Is there perhaps a better or more elegant way to go about this ? thanks Matt

1 1

Re: [alsa-devel] inconsistent reference counting for snd-soc-skl
by Subhransu S. Prusty 13 Oct '16

13 Oct '16

On Thu, Oct 13, 2016 at 10:19:37PM +0900, Takashi Sakamoto wrote: > Hi, > > I find an issue of snd-soc-skl due to inconsistent reference counting. > See this call graph. > > skl_probe() > ->skl_first_init() > ->skl_i915_init() > ->snd_hdac_display_power() (CONFIG_SND_SOC_HDAC_HDMI is enabled) > ->bus->i915_power_refcount++ > ->all error pathes > ->skl_free() > ->snd_hda_i915_exit() > ->WARN_ON(!bus->i915_power_refcount) > > This might be also hit by rmmod(8). I see this bug in kernel 4.8 and > pre-4.9. Thanks Takashi for reporting. We will fix this. Regards, Subhransu > > > Regards > > Takashi Sakamoto --

2 1

Re: [alsa-devel] [FFADO-devel] Clicking issues with the output of an Onyx Blackbird
by Jordi Torres 13 Oct '16

13 Oct '16

> (CCed to alsa-devel) I sent this issue to alsa mailist 6 days ago but I supouse that hit the spam filter. Even I published an alsa-info http://www.alsa-project.org/db/?f=cb6745b6d0acdb5d9375096ec8dc008f6ff49cd7 When I read the wiki about reporting bugs, I understood that kernel related bugs goes to bugzilla.kernel.org and user-land bugs goes to the devel mail-list. So I supouse that I should sent this issue to the bugzilla as it's considered a kernel bug > Well, could I request you to gather information of packet sequence? Here is the trace of packets: https://drive.google.com/file/d/0Bz44Dr5G2FeJSXBHMzc5WTFmOWs/view?usp=shari… The trace includes the opening of alsa device and playing a sine wave for 30 seconds My linux box has two firewire ports I don't know the details about the host controller but could be feasible sniff the connection between the other computer with official drivers and the DAW? Could that be useful?

1 0

[alsa-devel] Applied "ASoC: cs4270: fix DAPM stream name mismatch" to the asoc tree
by Mark Brown 13 Oct '16

13 Oct '16

The patch ASoC: cs4270: fix DAPM stream name mismatch has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From aa5f920993bda2095952177eea79bc8e58ae6065 Mon Sep 17 00:00:00 2001 From: murray foster <mrafoster(a)gmail.com> Date: Sun, 9 Oct 2016 13:28:45 -0700 Subject: [PATCH] ASoC: cs4270: fix DAPM stream name mismatch Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. [Errors must be at probe time not compile time -- broonie] Signed-off-by: Murray Foster <mrafoster(a)gmail.com> Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/cs4270.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 18baea2f7d65..84f86745c30e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.9.3

1 0