August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH] ASoC: dapm: Invalidate only paths reachable for a given stream
by Szymon Mielczarek 12 Aug '19

12 Aug '19

By resetting the cached number of endpoints for all card's widgets we may overwrite previously cached values for other streams. The situation may happen especially when running streams simultaneously. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)linux.intel.com> --- sound/soc/soc-dapm.c | 50 ++++++++++++++++++++++++++++++++------------ 1 file changed, 37 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d09bdca63c62..10819b3e0b98 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1128,6 +1128,34 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, return 0; } +/* + * Recursively reset the cached number of inputs or outputs for the specified + * widget and all widgets that can be reached via incoming or outcoming paths + * from the widget. + */ +static void invalidate_paths_ep(struct snd_soc_dapm_widget *widget, + enum snd_soc_dapm_direction dir) +{ + enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); + struct snd_soc_dapm_path *path; + + widget->endpoints[dir] = -1; + + snd_soc_dapm_widget_for_each_path(widget, rdir, path) { + if (path->weak || path->is_supply) + continue; + + if (path->walking) + return; + + if (path->connect) { + path->walking = 1; + invalidate_paths_ep(path->node[dir], dir); + path->walking = 0; + } + } +} + /* * Common implementation for is_connected_output_ep() and * is_connected_input_ep(). The function is inlined since the combined size of @@ -1257,21 +1285,17 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - /* - * For is_connected_{output,input}_ep fully discover the graph we need - * to reset the cached number of inputs and outputs. - */ - list_for_each_entry(w, &card->widgets, list) { - w->endpoints[SND_SOC_DAPM_DIR_IN] = -1; - w->endpoints[SND_SOC_DAPM_DIR_OUT] = -1; - } - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets, + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w = dai->playback_widget; + invalidate_paths_ep(w, SND_SOC_DAPM_DIR_OUT); + paths = is_connected_output_ep(w, &widgets, custom_stop_condition); - else - paths = is_connected_input_ep(dai->capture_widget, &widgets, + } else { + w = dai->capture_widget; + invalidate_paths_ep(w, SND_SOC_DAPM_DIR_IN); + paths = is_connected_input_ep(w, &widgets, custom_stop_condition); + } /* Drop starting point */ list_del(widgets.next); -- 2.17.1

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[alsa-devel] Applied "ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec" to the asoc tree
by Mark Brown 12 Aug '19

12 Aug '19

The patch ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec has been applied to the asoc tree at https://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 abe51c351827e0086dad079dfe02918fecdf4830 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 12 Aug 2019 12:52:25 +0300 Subject: [PATCH] ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec The DAC and ADC path of the codec is independent, have dedicated LRCK (FS) and BCK for DAC/ADC. They can be configured to use different format, TDM slots and slot_width if needed. Move these parameters under dedicated io_params structure and manage them independently based on the dai. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/20190812095226.18870-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 125 ++++++++++++++++-------------------- 1 file changed, 54 insertions(+), 71 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e84a1509fe65..75fa8e9ee894 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -44,18 +44,25 @@ static const char *const pcm3168a_supply_names[PCM3168A_NUM_SUPPLIES] = { "VCCDA2" }; +#define PCM3168A_DAI_DAC 0 +#define PCM3168A_DAI_ADC 1 + +/* ADC/DAC side parameters */ +struct pcm3168a_io_params { + bool master_mode; + unsigned int fmt; + int tdm_slots; + u32 tdm_mask; + int slot_width; +}; + struct pcm3168a_priv { struct regulator_bulk_data supplies[PCM3168A_NUM_SUPPLIES]; struct regmap *regmap; struct clk *scki; - bool adc_master_mode; - bool dac_master_mode; unsigned long sysclk; - unsigned int adc_fmt; - unsigned int dac_fmt; - int tdm_slots; - u32 tdm_mask[2]; - int slot_width; + + struct pcm3168a_io_params io_params[2]; }; static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" }; @@ -308,8 +315,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, - unsigned int format, bool dac) +static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); @@ -356,43 +362,31 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } - if (dac) { + if (dai->id == PCM3168A_DAI_DAC) { reg = PCM3168A_DAC_PWR_MST_FMT; mask = PCM3168A_DAC_FMT_MASK; shift = PCM3168A_DAC_FMT_SHIFT; - pcm3168a->dac_master_mode = master_mode; - pcm3168a->dac_fmt = fmt; } else { reg = PCM3168A_ADC_MST_FMT; mask = PCM3168A_ADC_FMTAD_MASK; shift = PCM3168A_ADC_FMTAD_SHIFT; - pcm3168a->adc_master_mode = master_mode; - pcm3168a->adc_fmt = fmt; } + pcm3168a->io_params[dai->id].master_mode = master_mode; + pcm3168a->io_params[dai->id].fmt = fmt; + regmap_update_bits(pcm3168a->regmap, reg, mask, fmt << shift); return 0; } -static int pcm3168a_set_dai_fmt_dac(struct snd_soc_dai *dai, - unsigned int format) -{ - return pcm3168a_set_dai_fmt(dai, format, true); -} - -static int pcm3168a_set_dai_fmt_adc(struct snd_soc_dai *dai, - unsigned int format) -{ - return pcm3168a_set_dai_fmt(dai, format, false); -} - static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + struct pcm3168a_io_params *io_params = &pcm3168a->io_params[dai->id]; if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) { dev_err(component->dev, @@ -408,22 +402,25 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return -EINVAL; } - if (pcm3168a->tdm_slots && pcm3168a->tdm_slots != slots) { + if (io_params->tdm_slots && io_params->tdm_slots != slots) { dev_err(component->dev, "Not matching slots %d vs %d\n", - pcm3168a->tdm_slots, slots); + io_params->tdm_slots, slots); return -EINVAL; } - if (pcm3168a->slot_width && pcm3168a->slot_width != slot_width) { + if (io_params->slot_width && io_params->slot_width != slot_width) { dev_err(component->dev, "Not matching slot_width %d vs %d\n", - pcm3168a->slot_width, slot_width); + io_params->slot_width, slot_width); return -EINVAL; } - pcm3168a->tdm_slots = slots; - pcm3168a->slot_width = slot_width; - pcm3168a->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; - pcm3168a->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; + io_params->tdm_slots = slots; + io_params->slot_width = slot_width; + /* Ignore the not relevant mask for the DAI/direction */ + if (dai->id == PCM3168A_DAI_DAC) + io_params->tdm_mask = tx_mask; + else + io_params->tdm_mask = rx_mask; return 0; } @@ -434,7 +431,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - bool tx, master_mode; + struct pcm3168a_io_params *io_params = &pcm3168a->io_params[dai->id]; + bool master_mode; u32 val, mask, shift, reg; unsigned int rate, fmt, ratio, max_ratio; unsigned int tdm_slots; @@ -444,23 +442,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, ratio = pcm3168a->sysclk / rate; - tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - if (tx) { + if (dai->id == PCM3168A_DAI_DAC) { max_ratio = PCM3168A_NUM_SCKI_RATIOS_DAC; reg = PCM3168A_DAC_PWR_MST_FMT; mask = PCM3168A_DAC_MSDA_MASK; shift = PCM3168A_DAC_MSDA_SHIFT; - master_mode = pcm3168a->dac_master_mode; - fmt = pcm3168a->dac_fmt; } else { max_ratio = PCM3168A_NUM_SCKI_RATIOS_ADC; reg = PCM3168A_ADC_MST_FMT; mask = PCM3168A_ADC_MSAD_MASK; shift = PCM3168A_ADC_MSAD_SHIFT; - master_mode = pcm3168a->adc_master_mode; - fmt = pcm3168a->adc_fmt; } + master_mode = io_params->master_mode; + fmt = io_params->fmt; + for (i = 0; i < max_ratio; i++) { if (pcm3168a_scki_ratios[i] == ratio) break; @@ -471,8 +467,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (pcm3168a->slot_width) - slot_width = pcm3168a->slot_width; + if (io_params->slot_width) + slot_width = io_params->slot_width; else slot_width = params_width(params); @@ -497,8 +493,8 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (pcm3168a->tdm_slots) - tdm_slots = pcm3168a->tdm_slots; + if (io_params->tdm_slots) + tdm_slots = io_params->tdm_slots; else tdm_slots = params_channels(params); @@ -534,7 +530,7 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(pcm3168a->regmap, reg, mask, val); - if (tx) { + if (dai->id == PCM3168A_DAI_DAC) { mask = PCM3168A_DAC_FMT_MASK; shift = PCM3168A_DAC_FMT_SHIFT; } else { @@ -552,20 +548,13 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int fmt; unsigned int sample_min; unsigned int channel_max; unsigned int channel_maxs[] = { - 6, /* rx */ - 8 /* tx */ + 8, /* DAC */ + 6 /* ADC */ }; - if (tx) - fmt = pcm3168a->dac_fmt; - else - fmt = pcm3168a->adc_fmt; - /* * Available Data Bits * @@ -578,7 +567,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, * I2S * LEFT_J */ - switch (fmt) { + switch (pcm3168a->io_params[dai->id].fmt) { case PCM3168A_FMT_RIGHT_J: sample_min = 16; channel_max = 2; @@ -588,7 +577,7 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, case PCM3168A_FMT_DSP_A: case PCM3168A_FMT_DSP_B: sample_min = 24; - channel_max = channel_maxs[tx]; + channel_max = channel_maxs[dai->id]; break; default: sample_min = 24; @@ -600,8 +589,8 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, sample_min, 32); /* Allow all channels in multi DIN/DOUT mode */ - if (pcm3168a->tdm_slots == 2) - channel_max = channel_maxs[tx]; + if (pcm3168a->io_params[dai->id].tdm_slots == 2) + channel_max = channel_maxs[dai->id]; snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -609,26 +598,19 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { +static const struct snd_soc_dai_ops pcm3168a_dai_ops = { .startup = pcm3168a_startup, - .set_fmt = pcm3168a_set_dai_fmt_dac, + .set_fmt = pcm3168a_set_dai_fmt, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, .digital_mute = pcm3168a_digital_mute, .set_tdm_slot = pcm3168a_set_tdm_slot, }; -static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { - .startup = pcm3168a_startup, - .set_fmt = pcm3168a_set_dai_fmt_adc, - .set_sysclk = pcm3168a_set_dai_sysclk, - .hw_params = pcm3168a_hw_params, - .set_tdm_slot = pcm3168a_set_tdm_slot, -}; - static struct snd_soc_dai_driver pcm3168a_dais[] = { { .name = "pcm3168a-dac", + .id = PCM3168A_DAI_DAC, .playback = { .stream_name = "Playback", .channels_min = 1, @@ -636,10 +618,11 @@ static struct snd_soc_dai_driver pcm3168a_dais[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = PCM3168A_FORMATS }, - .ops = &pcm3168a_dac_dai_ops + .ops = &pcm3168a_dai_ops }, { .name = "pcm3168a-adc", + .id = PCM3168A_DAI_ADC, .capture = { .stream_name = "Capture", .channels_min = 1, @@ -647,7 +630,7 @@ static struct snd_soc_dai_driver pcm3168a_dais[] = { .rates = SNDRV_PCM_RATE_8000_96000, .formats = PCM3168A_FORMATS }, - .ops = &pcm3168a_adc_dai_ops + .ops = &pcm3168a_dai_ops }, }; -- 2.20.1

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[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: quirks and DMIC corrections
by Pierre-Louis Bossart 12 Aug '19

12 Aug '19

Extend quirk for GeminiLake-based Chromebooks and correct missing dependencies for DMIC support (align with all other machine drivers). Keyon Jie (3): ASoC: Intel: sof-rt5682: add dmic dapm widget to support dmic PCM ASoC: Intel: skl-hda-dsp-generic: add dependency to dmic driver ASoC: Intel: skl-hda-dsp-generic: add dmic dapm widget and route Sathya Prakash M R (1): ASoC: Intel: boards: Match Product Family instead of product sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++++ sound/soc/intel/boards/sof_rt5682.c | 7 +++++-- 3 files changed, 10 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: pcm3168a: DAC/ADC independence and right_j fix
by Peter Ujfalusi 12 Aug '19

12 Aug '19

Hi, The first patch makes sure that the DAC and ADC side of the codec can be used and configured independently from each other. The second patch allows the reconfiguration of tdm_slots and slot_width. It is needed in case the codec is configured as slave and right_j whisch is the only mode when it can support 16 and 24 bit audio, but 16 bit requires 16 slot_width while 24 bit can use 48 or 32 slot_width. The codec driver must allow runtime reconfiguration of the tdm_slots and slot_width for this purpose. Regards, Peter --- Peter Ujfalusi (2): ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec ASoC: pcm3168a: Allow reconfiguration of tdm_slots and slot_width sound/soc/codecs/pcm3168a.c | 129 ++++++++++++++---------------------- 1 file changed, 50 insertions(+), 79 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: Protect hw_params callback against race
by Peter Ujfalusi 12 Aug '19

12 Aug '19

If the playback and capture of the same McASP is connected to different dai link (non duplex PCM links, like with pcm3168a codec) then there is a high probability of race between the two direction leaving McASP in a confused state. Protect the hw_params() with a mutex to make sure that this is not happening. The concurrent execution of hw_params for capture and playback can be easily triggered with custom .asoundrc file (for pcm3168a codec): pcm.dmixed8 { type dmix ipc_key 2048 ipc_perm 0666 slave { pcm "hw:0,0" format S24_LE channels 8 rate 96000 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 6 6 7 7 } } pcm.cpb-headset-1 { type plug slave.pcm dmixed8 hint { show on description "Headset 1 jack" } ttable.0.0 1 ttable.1.4 1 } pcm.dsnooped6 { type dsnoop ipc_key 2049 ipc_perm 0666 slave { pcm "hw:0,1" format S24_LE channels 6 rate 96000 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 } } pcm.cpb-mic-1 { type plug slave.pcm "dsnooped6" hint { show on description "Microphone 1 jack" } ttable.0.0 1 ttable.1.3 1 } Then running: arecord -D cpb-mic-1 -f S24_LE -c2 -r48000 | aplay -D cpb-headset-1 -f S24_LE -c2 -r48000 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 7aa3c32e4a49..fe7a0b3572e2 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -111,6 +111,7 @@ struct davinci_mcasp { u32 channels; int max_format_width; u8 active_serializers[2]; + struct mutex mutex; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; @@ -1169,6 +1170,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + mutex_lock(&mcasp->mutex); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: @@ -1197,12 +1200,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, default: printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); - return -EINVAL; + ret = -EINVAL; + goto out; } ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); if (ret) - return ret; + goto out; /* * If mcasp is BCLK master, and a BCLK divider was not provided by @@ -1223,7 +1227,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, ret = mcasp_common_hw_param(mcasp, substream->stream, period_size * channels, channels); if (ret) - return ret; + goto out; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) ret = mcasp_dit_hw_param(mcasp, params_rate(params)); @@ -1232,7 +1236,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, channels); if (ret) - return ret; + goto out; davinci_config_channel_size(mcasp, word_length); @@ -1242,7 +1246,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, mcasp->max_format_width = word_length; } - return 0; +out: + mutex_unlock(&mcasp->mutex); + + return ret; } static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, @@ -2335,6 +2342,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret) return -EINVAL; + mutex_init(&mcasp->mutex); + ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v4 0/7] Allwinner H6 SPDIF support
by Clément Péron 12 Aug '19

12 Aug '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Actually all these features are unused and only a bit for flushing the TX fifo is required. Also this series requires the DMA working on H6, a first version has been submitted by Jernej Škrabec[1] but has not been accepted yet. [1] https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 Changes since v3: - rename reg_fctl_ftx to val_fctl_ftx - rebase this series on sound-next - fix dt-bindings due to change in sound-next - change node name sound_spdif to sound-spdif Changes since v2: - Split quirks and H6 support patch - Add specific section for quirks comment Changes since v1: - Remove H3 compatible - Add TX fifo bit flush quirks - Add H6 bindings in SPDIF driver Clément Péron (7): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Move quirks to the top ASoC: sun4i-spdif: Add TX fifo bit flush quirks ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../sound/allwinner,sun4i-a10-spdif.yaml | 1 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 ++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 49 ++++++++++++++++--- 5 files changed, 87 insertions(+), 6 deletions(-) -- 2.20.1

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[alsa-devel] Is period_size bytes per period or frames per period?
by GitHub issues - edited 12 Aug '19

12 Aug '19

alsa-project/alsa-lib issue #8 was edited from crocket: According to https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html period_size in dmix is bytes per period. According to aplay --dump-hw-params -D dmix ... period_size is frames per period. Which one is right? Issue URL : https://github.com/alsa-project/alsa-lib/issues/8 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] Is period_size bytes per period or frames per period?
by GitHub issues - opened 12 Aug '19

12 Aug '19

alsa-project/alsa-lib issue #8 was opened from crocket: According to https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html period_size in dmix is bytes per period. According to aplay --dump-hw-params period_size is frames per period. Which one is right? Issue URL : https://github.com/alsa-project/alsa-lib/issues/8 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH v4 0/9]ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 12 Aug '19

12 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. As the LibreELEC project is progressing extremely well then there has been some activity getting surround sound working and this is included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. I still need to investigate the FIFO syncing issues which i've not had a chance to change or address the concerns that broonie and wens brought up. This change has been moved to the top of the patch stack. BR, CK --- v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. Marcus Cooper (9): ASoC: sun4i-i2s: Fix sun8i tx channel offset mask ASoC: sun4i-i2s: Add offset to RX channel select ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Reduce quirks for sun8i-h3 ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Add multichannel functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Adjust regmap settings sound/soc/sunxi/sun4i-i2s.c | 242 ++++++++++++++++++++++++------------ 1 file changed, 164 insertions(+), 78 deletions(-) -- 2.21.0

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[alsa-devel] [asoc:for-5.4 175/205] include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t'
by kbuild test robot 11 Aug '19

11 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: 83ee7ab1627b7597bbb9acc26b37e9f96fff40c9 commit: abf31feea26c0f412a191c83f408311a0de7435c [175/205] ASoC: Intel: Update request-reply IPC model config: i386-randconfig-a004-201932 (attached as .config) compiler: gcc-4.9 (Debian 4.9.2-10+deb8u1) 4.9.2 reproduce: git checkout abf31feea26c0f412a191c83f408311a0de7435c # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc/intel/haswell/sst-haswell-ipc.c:9: sound/soc/intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc/intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^ >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc/intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ -- In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc//intel/haswell/sst-haswell-ipc.c:9: sound/soc//intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc//intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^ >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc//intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ vim +82 include/linux/dynamic_debug.h 923abb9d797ba0 Gal Pressman 2019-05-01 75 923abb9d797ba0 Gal Pressman 2019-05-01 76 extern __printf(3, 4) 923abb9d797ba0 Gal Pressman 2019-05-01 77 void __dynamic_ibdev_dbg(struct _ddebug *descriptor, 923abb9d797ba0 Gal Pressman 2019-05-01 78 const struct ib_device *ibdev, 923abb9d797ba0 Gal Pressman 2019-05-01 79 const char *fmt, ...); 923abb9d797ba0 Gal Pressman 2019-05-01 80 2bdde670beedf7 Rasmus Villemoes 2019-03-07 81 #define DEFINE_DYNAMIC_DEBUG_METADATA(name, fmt) \ c0d2af63786394 Joe Perches 2012-10-18 @82 static struct _ddebug __aligned(8) \ 07613b0b5ef857 Jason Baron 2011-10-04 83 __attribute__((section("__verbose"))) name = { \ 07613b0b5ef857 Jason Baron 2011-10-04 84 .modname = KBUILD_MODNAME, \ 07613b0b5ef857 Jason Baron 2011-10-04 85 .function = __func__, \ 07613b0b5ef857 Jason Baron 2011-10-04 86 .filename = __FILE__, \ 07613b0b5ef857 Jason Baron 2011-10-04 87 .format = (fmt), \ 07613b0b5ef857 Jason Baron 2011-10-04 88 .lineno = __LINE__, \ 07613b0b5ef857 Jason Baron 2011-10-04 89 .flags = _DPRINTK_FLAGS_DEFAULT, \ 2bdde670beedf7 Rasmus Villemoes 2019-03-07 90 _DPRINTK_KEY_INIT \ 07613b0b5ef857 Jason Baron 2011-10-04 91 } 07613b0b5ef857 Jason Baron 2011-10-04 92 :::::: The code at line 82 was first introduced by commit :::::: c0d2af637863940b1a4fb208224ca7acb905c39f dynamic_debug: Remove unnecessary __used :::::: TO: Joe Perches <joe(a)perches.com> :::::: CC: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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