May 2020 - Alsa-devel - mailman.alsa-project.org

Difference between PlaybackMixer and PlaybackMixerElem
by Curtis Malainey 22 May '20

22 May '20

Hello Jaroslav, Just a quick question, what is the difference between PlaybackMixer and PlaybackMixerElem? I can see that one can be parsed using snd_use_case_parse _selem_id? Is that the only difference? Thanks Curtis

1 0

[PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Split be_hw_params_fixup function
by Lukasz Majczak 21 May '20

21 May '20

Split be_hw_params_fixup function for different codecs as current common function, leads to crash while trying to get snd_soc_dpcm with container_of() macro in kabylake_ssp_fixup(). The crash call path looks as below: soc_pcm_hw_params() snd_soc_dai_hw_params(codec_dai, substream, &codec_params); rtd->dai_link->be_hw_params_fixup(rtd, params) kabylake_ssp_fixup() In this case, codec_params is just a copy of an internal structure and is not embedded into struct snd_soc_dpcm thus we cannot use container_of() on it. Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 130 ++++++++++++------ 1 file changed, 85 insertions(+), 45 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..12a9983979e0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -171,8 +171,8 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, - { "codec1_in", NULL, "ssp0 Rx" }, - { "ssp0 Rx", NULL, "AIF1 Capture" }, + { "codec1_in", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "AIF1 Capture" }, /* IV feedback path */ { "codec0_fb_in", NULL, "ssp0 Rx"}, @@ -328,42 +328,52 @@ static const struct snd_soc_ops kabylake_rt5663_fe_ops = { .startup = kbl_fe_startup, }; -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static void kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params, snd_pcm_format_t pcm_fmt) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, pcm_fmt); +} + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int kabylake_ssp1_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; + channels->min = channels->max = 2; else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + channels->min = channels->max = 4; return 0; } @@ -400,20 +410,6 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, int ret = 0, j; for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { @@ -433,10 +429,37 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } +static int kabylake_dmic01_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0, 8, 16); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set sysclk err: %d\n", ret); + return ret; + } + + return ret; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; +static struct snd_soc_ops kabylake_dmic01_ops = { + .hw_params = kabylake_dmic01_hw_params, +}; + + static const unsigned int channels_dmic[] = { 4, }; @@ -507,14 +530,19 @@ SND_SOC_DAILINK_DEF(ssp0_pin, SND_SOC_DAILINK_DEF(ssp0_codec, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(dmic01_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic01_codec, + DAILINK_COMP_ARRAY( + COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + SND_SOC_DAILINK_DEF(idisp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); SND_SOC_DAILINK_DEF(idisp1_codec, @@ -618,9 +646,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp0_fixup, .dpcm_playback = 1, - .dpcm_capture = 1, .ops = &kabylake_ssp0_ops, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), }, @@ -632,12 +659,25 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp1_fixup, .ops = &kabylake_rt5663_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, + { + .name = "dmic01", + .id = 2, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_dmic_fixup, + .dpcm_capture = 1, + .ops = &kabylake_dmic01_ops, + SND_SOC_DAILINK_REG(dmic01_pin, dmic01_codec, platform), + }, { .name = "iDisp1", .id = 3, base-commit: a4f6fc98cd2fa1774bcaeb248c67156ef9402a56 -- 2.25.1

3 5

[PATCH 0/2] powerpc: Remove support for ppc405/440 Xilinx platforms
by Michal Simek 21 May '20

21 May '20

Hi, recently we wanted to update xilinx intc driver and we found that function which we wanted to remove is still wired by ancient Xilinx PowerPC platforms. Here is the thread about it. https://lore.kernel.org/linux-next/48d3232d-0f1d-42ea-3109-f44bbabfa2e8@xil… I have been talking about it internally and there is no interest in these platforms and it is also orphan for quite a long time. None is really running/testing these platforms regularly that's why I think it makes sense to remove them also with drivers which are specific to this platform. U-Boot support was removed in 2017 without anybody complain about it https://github.com/Xilinx/u-boot-xlnx/commit/98f705c9cefdfdba62c069821bbba1… Based on current ppc/next. If anyone has any objection about it, please let me know. Thanks, Michal Michal Simek (2): sound: ac97: Remove sound driver for ancient platform powerpc: Remove Xilinx PPC405/PPC440 support Documentation/devicetree/bindings/xilinx.txt | 143 -- Documentation/powerpc/bootwrapper.rst | 28 +- MAINTAINERS | 6 - arch/powerpc/Kconfig.debug | 2 +- arch/powerpc/boot/Makefile | 7 +- arch/powerpc/boot/dts/Makefile | 1 - arch/powerpc/boot/dts/virtex440-ml507.dts | 406 ------ arch/powerpc/boot/dts/virtex440-ml510.dts | 466 ------- arch/powerpc/boot/ops.h | 1 - arch/powerpc/boot/serial.c | 5 - arch/powerpc/boot/uartlite.c | 79 -- arch/powerpc/boot/virtex.c | 97 -- arch/powerpc/boot/virtex405-head.S | 31 - arch/powerpc/boot/wrapper | 8 - arch/powerpc/configs/40x/virtex_defconfig | 75 - arch/powerpc/configs/44x/virtex5_defconfig | 74 - arch/powerpc/configs/ppc40x_defconfig | 8 - arch/powerpc/configs/ppc44x_defconfig | 8 - arch/powerpc/include/asm/xilinx_intc.h | 16 - arch/powerpc/include/asm/xilinx_pci.h | 21 - arch/powerpc/kernel/cputable.c | 39 - arch/powerpc/platforms/40x/Kconfig | 31 - arch/powerpc/platforms/40x/Makefile | 1 - arch/powerpc/platforms/40x/virtex.c | 54 - arch/powerpc/platforms/44x/Kconfig | 37 - arch/powerpc/platforms/44x/Makefile | 2 - arch/powerpc/platforms/44x/virtex.c | 60 - arch/powerpc/platforms/44x/virtex_ml510.c | 30 - arch/powerpc/platforms/Kconfig | 4 - arch/powerpc/sysdev/Makefile | 2 - arch/powerpc/sysdev/xilinx_intc.c | 88 -- arch/powerpc/sysdev/xilinx_pci.c | 132 -- arch/powerpc/xmon/ppc-dis.c | 6 - arch/powerpc/xmon/ppc-opc.c | 23 - arch/powerpc/xmon/ppc.h | 5 - drivers/char/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 2 +- sound/drivers/Kconfig | 12 - sound/drivers/Makefile | 2 - sound/drivers/ml403-ac97cr.c | 1298 ------------------ 40 files changed, 7 insertions(+), 3305 deletions(-) delete mode 100644 arch/powerpc/boot/dts/virtex440-ml507.dts delete mode 100644 arch/powerpc/boot/dts/virtex440-ml510.dts delete mode 100644 arch/powerpc/boot/uartlite.c delete mode 100644 arch/powerpc/boot/virtex.c delete mode 100644 arch/powerpc/boot/virtex405-head.S delete mode 100644 arch/powerpc/configs/40x/virtex_defconfig delete mode 100644 arch/powerpc/configs/44x/virtex5_defconfig delete mode 100644 arch/powerpc/include/asm/xilinx_intc.h delete mode 100644 arch/powerpc/include/asm/xilinx_pci.h delete mode 100644 arch/powerpc/platforms/40x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex_ml510.c delete mode 100644 arch/powerpc/sysdev/xilinx_intc.c delete mode 100644 arch/powerpc/sysdev/xilinx_pci.c delete mode 100644 sound/drivers/ml403-ac97cr.c -- 2.26.0

10 29

[PATCH 1/2] soundwire: intel: use a single module
by Bard Liao 21 May '20

21 May '20

From: Rander Wang <rander.wang(a)intel.com> It's not clear why we have two modules for the Intel controller/master support when there is a single Kconfig. This adds complexity for no good reason, the two parts need to work together anyways. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/Makefile | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index 7319918e0aec..4f6094767212 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -16,12 +16,9 @@ soundwire-cadence-objs := cadence_master.o obj-$(CONFIG_SOUNDWIRE_CADENCE) += soundwire-cadence.o #Intel driver -soundwire-intel-objs := intel.o +soundwire-intel-objs := intel.o intel_init.o obj-$(CONFIG_SOUNDWIRE_INTEL) += soundwire-intel.o -soundwire-intel-init-objs := intel_init.o -obj-$(CONFIG_SOUNDWIRE_INTEL) += soundwire-intel-init.o - #Qualcomm driver soundwire-qcom-objs := qcom.o obj-$(CONFIG_SOUNDWIRE_QCOM) += soundwire-qcom.o -- 2.17.1

3 6

ASoC dailink capture/playback flags
by Pierre-Louis Bossart 21 May '20

21 May '20

Hi, Some of us at Intel are confused with multiple definitions of capture/playback dailink flags (credits to Rander Wang for reporting this in https://github.com/thesofproject/linux/pull/2070). struct snd_soc_dai_link { [...] /* For unidirectional dai links */ unsigned int playback_only:1; unsigned int capture_only:1; [...] /* DPCM capture and Playback support */ unsigned int dpcm_capture:1; unsigned int dpcm_playback:1; [...] }; And of course when you start looking at some machine drivers, there are confusions, e.g. below with DPCM front-ends using one or the other set, and with copy-paste and partial diffs, this propagates without being consistent or being noticed: [GLK_DPCM_AUDIO_HS_PB] = { .name = "Glk Audio Headset Playback", .stream_name = "Headset Audio", .dpcm_playback = 1, <<<< .nonatomic = 1, .dynamic = 1, SND_SOC_DAILINK_REG(system2, dummy, platform), }, [GLK_DPCM_AUDIO_ECHO_REF_CP] = { .name = "Glk Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, .capture_only = 1, <<<< should this be .dpcm_capture = 1? .nonatomic = 1, .dynamic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, [GLK_DPCM_AUDIO_REF_CP] = { .name = "Glk Audio Reference cap", .stream_name = "Refcap", .init = NULL, .dpcm_capture = 1, <<<< .nonatomic = 1, .dynamic = 1, .ops = &geminilake_refcap_ops, SND_SOC_DAILINK_REG(reference, dummy, platform), }, So here are the questions: - when using DPCM, is there an expectation to use dpcm_ flags only? - should we instead use playback/capture_only when only one of the two dpcm_ flags is set? - should we flags errors if we ever encounter cases with e.g. dpcm_playback = true and capture_only = true? - do we actually need two sets of definitions? There are very few users of the .playback_only and .capture_only flags and only a single place where it's checked in soc-pcm.c Thanks for your feedback -Pierre

3 2

latency.c randomly gives persistent scratchy audio
by GitHub issues - opened 20 May '20

20 May '20

alsa-project/alsa-lib issue #53 was opened from archenemies: I had been intending to modify the example program test/latency.c in the alsa-lib distribution to build a filter. However, I found the original program itself was unable to produce a good transfer of audio. I am using it like ./latency -C plughw:3 -P hw:0,7 -r 44100 -m 256 -M 512 -c 2 -s 3000 but I also tried higher numbers (2048 and 4096) for -m and -M and this didn't help. This command should theoretically work to transfer audio samples between the capture device, hw:3 which is a USB audio card with 3.5mm input, and the playback device hw:0,7 which is an HDMI monitor with builtin speakers. The second device has no volume control in alsamixer. Most of the times when I run the above command, there is a scratchy sound which can be heard here mpv http://ix.io/2mPY In the recording, whenever the music stops and starts again, it's because I killed the 'latency' program and restarted it. You can hear that there are different levels of distortion from one invocation to the next. The signal is coming from a digital audio player which is plugged into the USB sound card via analog 3.5mm. Nothing else changes about the setup, I am just stopping and starting "./latency". The same level of scratchiness usually persists throughout the running of the program, however, each time the program is restarted a new level of scratchiness is chosen seemingly at random. It's almost like every time the program is started, a threshold is chosen uniformly at random, such that whenever the audio signal exceeds this threshold then it gets clipped. The other circumstances would seem to rule out buffer underflow or crystal drift, and there is no warning of dropped samples. I made some experiments, and even the effect of dropping every 10th sample sounds less distorted than this. But if it is from crystal drift, it should be more like every 1/1000th sample getting dropped. The sound that I am hearing is more like the result of zeroing or thresholding all the samples which exceed a certain amplitude, and a significant number of samples must be affected to make it sound this bad. I do not notice this problem with "ecasound -i .. -o .." or with Pulseaudio's "module-loopback", which are both able to produce clear audio. As I uderstand it, ecasound does not do resampling to account for crystal drift. Pulseaudio's module-loopback does do adaptive resampling. I'm including the command that I use, and the output. I don't know what some of those words mean - tstamp_mode, start_threshold, silence_threshold, period_event. Anyway, please let me know if it is possible to make this program behave more like Pulseaudio or Ecasound. I'll be happy to assist with reproduction issues and so on. $ ./latency -P hw:0,7 -C plughw:3 -r 44100 -m 4096 -M 16384 -c 2 -s 3000 !!!Scheduler set to Round Robin with priority 99 FAILED!!! Playback device is hw:0,7 Capture device is plughw:3 Parameters are 44100Hz, S16_LE, 2 channels, non-blocking mode Poll mode: no Loop limit is 132300000 frames, minimum latency = 4096, maximum latency = 16384 Hardware PCM card 0 'HD-Audio Generic' device 7 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 2048 period_time : 46439 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 2048 period_event : 0 start_threshold : 2147483647 stop_threshold : 4096 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 2048 period_time : 46439 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 2048 period_event : 0 start_threshold : 2147483647 stop_threshold : 4096 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Hardware PCM card 3 'USB Audio Device' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 2048 period_time : 46439 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 2048 period_event : 0 start_threshold : 2147483647 stop_threshold : 4096 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 Trying latency 4096 frames, 92879.819us, 92.879819ms (10.7666Hz) The Linux distribution is Arch, the kernel is 5.6.10-arch1-1, x86_64 GNU/Linux. Thank you. Issue URL : https://github.com/alsa-project/alsa-lib/issues/53 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: mmp-sspa: Fix the error handling in probe()
by Lubomir Rintel 20 May '20

20 May '20

If we fail after pm_runtime_enable(), we fail to undo it. Same with clk_prepare_enable(). Let's order them after all things that can fail. Fixes: 7d98cc648253 ("ASoC: mmp-sspa: Add support for the runtime power management") Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> --- sound/soc/pxa/mmp-sspa.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 60bf6fe05315..77a6d2998570 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -467,6 +467,7 @@ static const struct snd_soc_component_driver mmp_sspa_component = { static int asoc_mmp_sspa_probe(struct platform_device *pdev) { struct sspa_priv *sspa; + int ret; sspa = devm_kzalloc(&pdev->dev, sizeof(struct sspa_priv), GFP_KERNEL); @@ -518,8 +519,6 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) return PTR_ERR(sspa->sysclk); } } - pm_runtime_enable(&pdev->dev); - clk_prepare_enable(sspa->audio_clk); platform_set_drvdata(pdev, sspa); sspa->playback_dma_data.maxburst = 4; @@ -529,16 +528,21 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) sspa->playback_dma_data.addr = 0x80 + SSPA_D; if (pdev->dev.of_node) { - int ret; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, &mmp_pcm_config, 0); if (ret) return ret; } - return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, - &mmp_sspa_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); + if (ret) + return ret; + + pm_runtime_enable(&pdev->dev); + clk_prepare_enable(sspa->audio_clk); + + return 0; } static int asoc_mmp_sspa_remove(struct platform_device *pdev) -- 2.26.2

2 1

[PATCH 0/3] soundwire: add sysfs support
by Bard Liao 20 May '20

20 May '20

Add soundwire sysfs support. Pierre-Louis Bossart (3): soundwire: disco: s/ch/channels/ soundwire: master: add sysfs support soundwire: add Slave sysfs support .../ABI/testing/sysfs-bus-soundwire-master | 23 ++ .../ABI/testing/sysfs-bus-soundwire-slave | 91 ++++++ drivers/soundwire/Makefile | 3 +- drivers/soundwire/bus.c | 1 + drivers/soundwire/bus.h | 1 + drivers/soundwire/bus_type.c | 9 +- drivers/soundwire/master.c | 84 +++++ drivers/soundwire/mipi_disco.c | 11 +- drivers/soundwire/sysfs_local.h | 14 + drivers/soundwire/sysfs_slave.c | 215 +++++++++++++ drivers/soundwire/sysfs_slave_dpn.c | 300 ++++++++++++++++++ include/linux/soundwire/sdw.h | 8 +- 12 files changed, 748 insertions(+), 12 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-soundwire-master create mode 100644 Documentation/ABI/testing/sysfs-bus-soundwire-slave create mode 100644 drivers/soundwire/sysfs_local.h create mode 100644 drivers/soundwire/sysfs_slave.c create mode 100644 drivers/soundwire/sysfs_slave_dpn.c -- 2.17.1

3 6

[PATCH 1/5] drivers/powerpc: Replace _ALIGN_UP() by ALIGN()
by Christophe Leroy 20 May '20

20 May '20

_ALIGN_UP() is specific to powerpc ALIGN() is generic and does the same Replace _ALIGN_UP() by ALIGN() Signed-off-by: Christophe Leroy <christophe.leroy(a)c-s.fr> --- drivers/ps3/ps3-lpm.c | 6 +++--- drivers/vfio/pci/vfio_pci_nvlink2.c | 2 +- drivers/video/fbdev/ps3fb.c | 4 ++-- sound/ppc/snd_ps3.c | 2 +- 4 files changed, 7 insertions(+), 7 deletions(-) diff --git a/drivers/ps3/ps3-lpm.c b/drivers/ps3/ps3-lpm.c index 83c45659bc9d..064b5884ba13 100644 --- a/drivers/ps3/ps3-lpm.c +++ b/drivers/ps3/ps3-lpm.c @@ -1096,8 +1096,8 @@ int ps3_lpm_open(enum ps3_lpm_tb_type tb_type, void *tb_cache, lpm_priv->tb_cache_internal = NULL; lpm_priv->tb_cache = NULL; } else if (tb_cache) { - if (tb_cache != (void *)_ALIGN_UP((unsigned long)tb_cache, 128) - || tb_cache_size != _ALIGN_UP(tb_cache_size, 128)) { + if (tb_cache != (void *)ALIGN((unsigned long)tb_cache, 128) + || tb_cache_size != ALIGN(tb_cache_size, 128)) { dev_err(sbd_core(), "%s:%u: unaligned tb_cache\n", __func__, __LINE__); result = -EINVAL; @@ -1116,7 +1116,7 @@ int ps3_lpm_open(enum ps3_lpm_tb_type tb_type, void *tb_cache, result = -ENOMEM; goto fail_malloc; } - lpm_priv->tb_cache = (void *)_ALIGN_UP( + lpm_priv->tb_cache = (void *)ALIGN( (unsigned long)lpm_priv->tb_cache_internal, 128); } diff --git a/drivers/vfio/pci/vfio_pci_nvlink2.c b/drivers/vfio/pci/vfio_pci_nvlink2.c index ed20d73cc27c..65c61710c0e9 100644 --- a/drivers/vfio/pci/vfio_pci_nvlink2.c +++ b/drivers/vfio/pci/vfio_pci_nvlink2.c @@ -67,7 +67,7 @@ static size_t vfio_pci_nvgpu_rw(struct vfio_pci_device *vdev, * * This is not fast path anyway. */ - sizealigned = _ALIGN_UP(posoff + count, PAGE_SIZE); + sizealigned = ALIGN(posoff + count, PAGE_SIZE); ptr = ioremap_cache(data->gpu_hpa + posaligned, sizealigned); if (!ptr) return -EFAULT; diff --git a/drivers/video/fbdev/ps3fb.c b/drivers/video/fbdev/ps3fb.c index 834f63edf700..9df78fb77267 100644 --- a/drivers/video/fbdev/ps3fb.c +++ b/drivers/video/fbdev/ps3fb.c @@ -44,7 +44,7 @@ #define GPU_CMD_BUF_SIZE (2 * 1024 * 1024) #define GPU_FB_START (64 * 1024) #define GPU_IOIF (0x0d000000UL) -#define GPU_ALIGN_UP(x) _ALIGN_UP((x), 64) +#define GPU_ALIGN_UP(x) ALIGN((x), 64) #define GPU_MAX_LINE_LENGTH (65536 - 64) #define GPU_INTR_STATUS_VSYNC_0 0 /* vsync on head A */ @@ -1015,7 +1015,7 @@ static int ps3fb_probe(struct ps3_system_bus_device *dev) } #endif - max_ps3fb_size = _ALIGN_UP(GPU_IOIF, 256*1024*1024) - GPU_IOIF; + max_ps3fb_size = ALIGN(GPU_IOIF, 256*1024*1024) - GPU_IOIF; if (ps3fb_videomemory.size > max_ps3fb_size) { dev_info(&dev->core, "Limiting ps3fb mem size to %lu bytes\n", max_ps3fb_size); diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 6d2a33b8faa0..b8161a08f2ca 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -926,7 +926,7 @@ static int snd_ps3_driver_probe(struct ps3_system_bus_device *dev) PAGE_SHIFT, /* use system page size */ 0, /* dma type; not used */ NULL, - _ALIGN_UP(SND_PS3_DMA_REGION_SIZE, PAGE_SIZE)); + ALIGN(SND_PS3_DMA_REGION_SIZE, PAGE_SIZE)); dev->d_region->ioid = PS3_AUDIO_IOID; ret = ps3_dma_region_create(dev->d_region); -- 2.25.0

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imx8mm fsl_sai clock config failing
by Tim Harvey 19 May '20

19 May '20

Greetings, I'm trying to get imx8mm sai working with a tlv320aic3x codec on a new board and am finding that the sai3 bus clock is getting derived from the wrong source leading hw_params() to fail with 'fsl-sai 30030000.sai: failed to derive required Tx rate: 3072000' In comparison to the imx8mm-evk I find it's clock gets configured as desired yet I have the same device-tree configuration for sai3. Here is my board's dts: / { ... sound { compatible = "simple-audio-card"; simple-audio-card,name = "tlv320-audio"; simple-audio-card,format = "i2s"; simple-audio-card,frame-master = <&cpudai>; simple-audio-card,bitclock-master = <&cpudai>; simple-audio-card,widgets = "Line", "Left Line Out Jack", "Line", "Right Line Out Jack"; simple-audio-card,routing = "Left Line Out Jack", "LLOUT", "Right Line Out Jack", "RLOUT"; cpudai: simple-audio-card,cpu { sound-dai = <&sai3>; dai-tdm-slot-num = <2>; dai-tdm-slot-width = <32>; }; simple-audio-card,codec { sound-dai = <&tlv320aic3105>; clocks = <&clk IMX8MM_CLK_SAI3_ROOT>; }; }; }; &i2c2 { ... tlv320aic3105: codec@18 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_tlv>; compatible = "ti,tlv320aic3x"; reg = <0x18>; reset-gpios = <&gpio4 4 GPIO_ACTIVE_LOW>; ai3x-micbias-vg = <2>; /* MICBIAS_2_5V */ /* Regulators */ DRVDD-supply = <&reg_3p3v>; AVDD-supply = <&reg_3p3v>; IOVDD-supply = <&reg_3p3v>; DVDD-supply = <&reg_1p8v>; }; } &sai3 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai3>; assigned-clocks = <&clk IMX8MM_CLK_SAI3>; assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>; assigned-clock-rates = <24576000>; status = "okay"; }; and /sys/kernel/debug/clk/clk_summary from my board: osc_24m 7 10 0 24000000 0 0 50000 ... audio_pll1_ref_sel 0 0 0 24000000 0 0 50000 audio_pll1 0 0 0 650000000 0 0 50000 audio_pll1_bypass 0 0 0 650000000 0 0 50000 audio_pll1_out 0 0 0 650000000 0 0 50000 sai3 0 0 0 25000000 0 0 50000 sai3_root_clk 0 0 0 25000000 0 0 50000 ^^^^ note 25000000 instead of requested 24576000 And Here is the imx8mm-evk dts: / { ... wm8524: audio-codec { #sound-dai-cells = <0>; compatible = "wlf,wm8524"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_gpio_wlf>; wlf,mute-gpios = <&gpio5 21 GPIO_ACTIVE_LOW>; }; sound-wm8524 { compatible = "simple-audio-card"; simple-audio-card,name = "wm8524-audio"; simple-audio-card,format = "i2s"; simple-audio-card,frame-master = <&cpudai>; simple-audio-card,bitclock-master = <&cpudai>; simple-audio-card,widgets = "Line", "Left Line Out Jack", "Line", "Right Line Out Jack"; simple-audio-card,routing = "Left Line Out Jack", "LINEVOUTL", "Right Line Out Jack", "LINEVOUTR"; cpudai: simple-audio-card,cpu { sound-dai = <&sai3>; dai-tdm-slot-num = <2>; dai-tdm-slot-width = <32>; }; simple-audio-card,codec { sound-dai = <&wm8524>; clocks = <&clk IMX8MM_CLK_SAI3_ROOT>; }; }; }; &sai3 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai3>; assigned-clocks = <&clk IMX8MM_CLK_SAI3>; assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>; assigned-clock-rates = <24576000>; status = "okay"; }; and /sys/kernel/debug/clk/clk_summary from imx8mm_evk: osc_24m 7 10 0 24000000 0 0 50000 ... audio_pll1_ref_sel 0 0 0 24000000 0 0 50000 audio_pll1 0 0 0 393215995 0 0 50000 audio_pll1_bypass 0 0 0 393215995 0 0 50000 audio_pll1_out 0 0 0 393215995 0 0 50000 sai3 0 0 0 24576000 0 0 50000 sai3_root_clk 0 0 0 24576000 0 0 50000 ^^^^ note 24576000 as requested Any ideas what I'm doing wrong here? Note that even though the imx8mm-evk appears to configure its clock correctly I still have not been able to play audio out the wm8524 and the system appears to hang during playback as if BCLK/MCLK were not clocking. Perhaps there is something not supported yet upstream with regards to IMX8MM SAI? Best Regards, Tim

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