January 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

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[alsa-devel] Disable DAPM for click avoidance ?
by Ricard Wanderlof 27 Mar '18

27 Mar '18

I'm having an issue with a codec where the normal DAPM procedure of disabling most of the codec when there is no stream ongoing results in large clicks when a playback stream starts and stops, because the output jumps from 0V when the output is disabled to over a volt when the output is enabled. There appears to be a 'slow stop/start' feature in the codec, however, it takes about one second to ramp up and down the output voltage, and that is too long a delay when starting a stream. In our usecase the power consumtion of the codec is negligable compared to the rest of the electronics, so leaving the codec running at all times is perfectly feasible. Is there any way from a machine driver to disable the DAPM switching, i.e. leave parts of the codec running at all times? In this specific case, there are two internal PM controls which enable the output. One of them is controlled by an output mixer element (SND_SOC_DAPM_MIXER_E), and the other one from set_bias_level(). I've had a look in the core code, but it seems that the whole business of changing the bias level when a stream is started is always performed and that there is no way to say "Set the bias level to ON and leave it there". An alternative would be to augment the codec driver with an 'always on' feature, which effectively always enables PM for certain parts of the chip. But it seems like the wrong place to do this type of thing. Another use for this feature is that we have had a lot of problems over the years with buggy I2S interfaces misbehaving when streams are started and stopped, and there would be an advantage to letting the AIF run the whole time after startup. /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 - JACK PCM plugin] jack: Do not Xrun the ALSA buffer
by twischer＠de.adit-jv.com 21 Mar '18

21 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> when the JACK thread is requesting too many audio frames Playback: Without this commit the ALSA audio buffer will be played with endless repeats as long as the user application has not provided new audio data. Therefore this garbage will be played as long as the user application has not called snd_pcm_stop() after an Xrun. With this fix the rest of the JACK buffer will be filled with silence. Capture: Without this commit the audio data in the ALSA buffer would be overwritten. With this commit the new data from the JACK buffer will not be copied. Therefore the existing data in the ALSA buffer will not be overwritten. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130986.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c22a5d0..a4f35f4 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -54,6 +54,68 @@ typedef struct { static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); + +static inline snd_pcm_uframes_t snd_pcm_jack_playback_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr + io->buffer_size - appl_ptr; + if (avail < 0) + avail += jack->boundary; + else if ((snd_pcm_uframes_t) avail >= jack->boundary) + avail -= jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_capture_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr - appl_ptr; + if (avail < 0) + avail += jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + return (io->stream == SND_PCM_STREAM_PLAYBACK) ? + snd_pcm_jack_playback_avail(io, hw_ptr, appl_ptr) : + snd_pcm_jack_capture_avail(io, hw_ptr, appl_ptr); +} + +static inline snd_pcm_uframes_t snd_pcm_jack_hw_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + /* available data/space which can be transfered by the user application */ + const snd_pcm_uframes_t user_avail = snd_pcm_jack_avail(io, hw_ptr, + appl_ptr); + + if (user_avail > io->buffer_size) { + /* there was an Xrun */ + return 0; + } + /* available data/space which can be transfered by the DMA */ + return io->buffer_size - user_avail; +} + static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { static char buf[32]; @@ -154,7 +216,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; snd_pcm_uframes_t hw_ptr; - const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; unsigned int channel; @@ -164,40 +225,57 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) jack->areas[channel].first = 0; jack->areas[channel].step = jack->sample_bits; } - - if (io->state != SND_PCM_STATE_RUNNING) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) { - for (channel = 0; channel < io->channels; channel++) - snd_pcm_area_silence(&jack->areas[channel], 0, nframes, io->format); - return 0; - } - } hw_ptr = jack->hw_ptr; - areas = snd_pcm_ioplug_mmap_areas(io); - - while (xfer < nframes) { - snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; - snd_pcm_uframes_t cont = io->buffer_size - offset; - - if (cont < frames) - frames = cont; + if (io->state == SND_PCM_STATE_RUNNING) { + const snd_pcm_channel_area_t *areas = snd_pcm_ioplug_mmap_areas(io); + + while (xfer < nframes) { + snd_pcm_uframes_t frames = nframes - xfer; + snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; + snd_pcm_uframes_t cont = io->buffer_size - offset; + snd_pcm_uframes_t hw_avail = + snd_pcm_jack_hw_avail(io, hw_ptr, + io->appl_ptr); + + /* stop copying if there is no more data available */ + if (hw_avail <= 0) + break; + + /* split the snd_pcm_area_copy() function into two parts + * if the data to copy passes the buffer wrap around + */ + if (cont < frames) + frames = cont; + + if (hw_avail < frames) + frames = hw_avail; + + for (channel = 0; channel < io->channels; channel++) { + if (io->stream == SND_PCM_STREAM_PLAYBACK) + snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); + else + snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + } - for (channel = 0; channel < io->channels; channel++) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) - snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); - else - snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + hw_ptr += frames; + if (hw_ptr >= jack->boundary) + hw_ptr -= jack->boundary; + xfer += frames; } - - hw_ptr += frames; - if (hw_ptr >= jack->boundary) - hw_ptr -= jack->boundary; - xfer += frames; } jack->hw_ptr = hw_ptr; + /* check if requested frames were copied */ + if (xfer < nframes) { + /* always fill the not yet written JACK buffer with silence */ + if (io->stream == SND_PCM_STREAM_PLAYBACK) { + const snd_pcm_uframes_t samples = nframes - xfer; + for (channel = 0; channel < io->channels; channel++) + snd_pcm_area_silence(&jack->areas[channel], xfer, samples, io->format); + } + } + pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ return 0; -- 2.7.4

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[alsa-devel] [PATCH v1 0/6] ALSA: usb: UAC3. Add support for Basic Audio Device (BADD)
by Jorge Sanjuan 19 Mar '18

19 Mar '18

This adds functionality for the Basic Audio Device (BADD) subset that is defined in the USB Audio Class 3 (UAC3). In a BADD compliant driver, all Class Specific descriptors are inferred and so they should all be generated in the *host* as the device wont expose those. This also applies to the new HIGH CAPABILITY descriptor as the GET request is not to be supported in a BADD compliant driver. The new class requires the device to have 3 usb configurations as follows: 1: Legacy Mode: UAC1 or UAC2. 2: BADD device with a prefined topology. (Minimum). 3: UAC3 device for more detailed description or more complex devices that can't be covered by the BADD profile. This patch series also includes some minor fixes to sound/usb. It has been tested with an actual UAC3 compliant device in different host machines and all configs (UAC1/2, BADD and UAC3) work. Also, this has been implemented on top of the the patch which adds UAC3 support to the usb sound card driver: commit ddd452d7b04b86fb5f9285a19ac54deca9264ac1 Author: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Date: Tue Nov 7 04:01:20 2017 +0200 Jorge Sanjuan (6): usb: audio: Add bi-directional terminal types. usb: audio: Fix variable length field to be variable. ALSA: usb: UAC3. Add initial mixer support. ALSA: usb: Only get AudioControl header for UAC1 class. ALSA: usb: Use Class Specific EP for UAC3 devices. ALSA: usb: ADC3: Add initial BADD spec support include/linux/usb/audio-v3.h | 2 +- include/uapi/linux/usb/audio.h | 21 +- sound/usb/Makefile | 3 +- sound/usb/badd.c | 518 +++++++++++++++++++++++++++++++++++++++++ sound/usb/badd.h | 14 ++ sound/usb/card.c | 25 +- sound/usb/mixer.c | 24 ++ sound/usb/stream.c | 165 ++++++++----- sound/usb/usbaudio.h | 1 + 9 files changed, 702 insertions(+), 71 deletions(-) create mode 100644 sound/usb/badd.c create mode 100644 sound/usb/badd.h -- 2.11.0

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[alsa-devel] [PATCH] ASoC: mt8173-rt5650: fix child-node lookup
by Johan Hovold 12 Mar '18

12 Mar '18

This driver used the wrong OF-helper when looking up the optional capture-codec child node during probe. Instead of searching just children of the sound node, a tree-wide depth-first search starting at the unrelated platform node was done. Not only could this end up matching an unrelated node or no node at all; the platform node could also be prematurely freed since of_find_node_by_name() drops a reference to its first argument. This particular pattern has been observed leading to crashes after probe deferrals in other drivers. Fix this by dropping the broken call to of_find_node_by_name() and keeping only the second, correct lookup using of_get_child_by_name() while taking care not to bail out if the optional node is missing. Note that this also addresses two capture-codec node-reference leaks (one for each of the original helper calls). Compile tested only. Fixes: d349caeb0510 ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") Cc: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 40ebefd625c1..679fc8bea0a3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -272,15 +272,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) } mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; - if (of_find_node_by_name(platform_node, "codec-capture")) { - np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); - if (!np) { - dev_err(&pdev->dev, - "%s: Can't find codec-capture DT node\n", - __func__); - return -EINVAL; - } + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (np) { ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + of_node_put(np); if (ret < 0) { dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", -- 2.13.6

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[alsa-devel] [PATCH] sound/tlv320dac33: Add device tree support
by Pavel Machek 05 Mar '18

05 Mar '18

This adds device tree support to tlv320dac33.c. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/Documentation/devicetree/bindings/sound/tlv320dac33.txt b/Documentation/devicetree/bindings/sound/tlv320dac33.txt new file mode 100644 index 0000000..6cbd311 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320dac33.txt @@ -0,0 +1,32 @@ +Texas Instruments - tlv320dac33 Codec module + +The tlv320dac33 serial control bus communicates through I2C protocols. + +Required properties: + +- compatible - "ti,tlv320dac33" +- reg - I2C slave address + +Optional properties: + +- power-gpios - gpio pin to power the device, active high + +- interrupts - The interrupt output from the device. +- interrupt-parent - The parent interrupt controller. + +- ti,keep-bclk - Keep the BCLK running in FIFO modes +- ti,burst-bclkdiv - BCLK divider value in burst mode + +Example: + +tlv320dac33: audio-codec@19 { + compatible = "ti,tlv320dac33"; + reg = <0x19>; + + interrupt-parent = <&gpio2>; + interrupts = <21 1>; /* gpio_53, IRQF_TRIGGER_RISING */ + power-gpio = <&gpio2 28 0>; /* gpio_60 */ + + ti,keep-bclk; + ti,burst-bclkdiv = <3>; +}; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5b94a15..2a0fe06 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +32,7 @@ #include <linux/interrupt.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +108,7 @@ struct tlv320dac33_priv { unsigned int nsample; /* burst read amount from host */ int mode1_latency; /* latency caused by the i2c writes in * us */ - u8 burst_bclkdiv; /* BCLK divider value in burst mode */ + u32 burst_bclkdiv; /* BCLK divider value in burst mode */ unsigned int burst_rate; /* Interface speed in Burst modes */ int keep_bclk; /* Keep the BCLK continuously running @@ -1484,16 +1487,11 @@ static struct snd_soc_dai_driver dac33_dai = { static int dac33_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_platform_data *pdata = client->dev.platform_data; struct tlv320dac33_priv *dac33; + struct device_node *np = client->dev.of_node; int ret, i; - if (client->dev.platform_data == NULL) { - dev_err(&client->dev, "Platform data not set\n"); - return -ENODEV; - } - pdata = client->dev.platform_data; - dac33 = devm_kzalloc(&client->dev, sizeof(struct tlv320dac33_priv), GFP_KERNEL); if (dac33 == NULL) @@ -1505,10 +1503,26 @@ static int dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); - dac33->power_gpio = pdata->power_gpio; - dac33->burst_bclkdiv = pdata->burst_bclkdiv; - dac33->keep_bclk = pdata->keep_bclk; - dac33->mode1_latency = pdata->mode1_latency; + if (pdata) { + dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; + dac33->keep_bclk = pdata->keep_bclk; + dac33->mode1_latency = pdata->mode1_latency; + } else if (np) { + ret = of_get_named_gpio(np, "power-gpios", 0); + if (ret >= 0) + dac33->power_gpio = ret; + else + dac33->power_gpio = -1; + + if (of_property_read_bool(np, "ti,keep-bclk")) + dac33->keep_bclk = true; + + of_property_read_u32(np, "ti,burst-bclkdiv", &dac33->burst_bclkdiv); + } else { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } if (!dac33->mode1_latency) dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; @@ -1574,9 +1588,16 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); +static const struct of_device_id tlv320dac33_of_match[] = { + { .compatible = "ti,tlv320dac33", }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_of_match); + static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", + .of_match_table = of_match_ptr(tlv320dac33_of_match), }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake fixes
by Guneshwor Singh 01 Mar '18

01 Mar '18

This includes fixes for Skylake+ driver which are improving widget type check, fixes for simultaneous wake-on-voice (WoV) and capture use-case, fix to improve firmware download latency and lastly, fix to use uuid instead of module id for library modules. Guneshwor Singh (1): ASoC: Intel: Skylake: Refine skl widget type check Pawse, GuruprasadX (2): ASoC: Intel: Skylake: Add BE DAI for WoV usecase ASoC: Intel: board: Add BE DAI link for WoV and update DAPM machine map. Sanyog Kale (1): ASoC: Intel: Skylake: Disable clock and power gating during fw/lib download Sriram Periyasamy (1): ASoC: Intel: Skylake: Find module id from UUID for bind params sound/soc/intel/boards/bxt_rt298.c | 15 +++++ sound/soc/intel/skylake/skl-messages.c | 7 ++- sound/soc/intel/skylake/skl-pcm.c | 20 +++++- sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-topology.c | 107 +++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++- sound/soc/intel/skylake/skl.c | 21 +++++++ sound/soc/intel/skylake/skl.h | 4 ++ 8 files changed, 176 insertions(+), 14 deletions(-) -- 2.16.1

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[alsa-devel] [PATCH 0/7] Let Atmel use simple-audio-card
by Ladislav Michl 01 Mar '18

01 Mar '18

Alexandre, following patchset is sent as a bare minimum I need to use MAX9867 codec in master mode on AT91SAM9G20 based board. Codec's MCLK is driven by PCK0 and SSC is wired as follows: RK0 <-- BCLK RF0 <-- LRCLK TD0 --> SDIN RD0 <-- SDOUT Following DT framgment was used: i2c-gpio-0 { codec: max9867@18 { compatible = "maxim,max9867"; reg = <0x18>; #sound-dai-cells = <0>; }; }; sound { compatible = "simple-audio-card"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_pck0_as_mck>; simple-audio-card,name = "simear4-max9867"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&codec_dai>; simple-audio-card,frame-master = <&codec_dai>; simple-audio-card,routing = "LINE_IN", "Line In Jack", "Speaker", "HPOUT"; simple-audio-card,widgets = "Line", "Line In Jack", "Speaker", "Speaker"; cpu_dai: simple-audio-card,cpu { sound-dai = <&ssc0>; }; codec_dai: simple-audio-card,codec { sound-dai = <&codec>; clocks = <&pck0>; clock-frequency = <12384000>; }; }; &ssc0 { status = "okay"; pinctrl-0 = <&pinctrl_board_ssc0_tx &pinctrl_ssc0_rx>; atmel,clk-from-rk-pin; #sound-dai-cells = <0>; }; Codec code itself seems to be unused since it was merged and I'm wondering whenever it worked at all. A comment from codec author would help a lot, just to be sure I did not break anything. By no means this code is ready to be merged (only except those patches with commit log and after carefull consideration :)), I'm posting it as a base for discussion. Thank you, ladis Ladislav Michl (7): ASoC: atmel: Remove redundant dev_err() call in probe function ASoC: atmel_ssc_dai: Fix TCMR settings in I2S slave mode ASoC: simple_card_utils: Set clock frequency ASoC: max9867: Show Kconfig entry ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix BSEL value in master mode. ASoC: max9867: Take chip out of shutdown sound/soc/atmel/atmel-classd.c | 7 +-- sound/soc/atmel/atmel-pdmic.c | 7 +-- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/max9867.c | 110 +++------------------------------- sound/soc/generic/simple-card-utils.c | 4 ++ 6 files changed, 19 insertions(+), 114 deletions(-) -- 2.15.1

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[alsa-devel] [PATCH - JACK PCM plugin] jack: Use boundary as hw_ptr wrap around
by twischer＠de.adit-jv.com 24 Feb '18

24 Feb '18

From: Timo Wischer <twischer(a)de.adit-jv.com> instead of using buffer_size as wrap around. This is required to detect Xruns. It is also required to allow the JACK thread to processes the whole ALSA audio buffer at once without calling snd_pcm_avail_update() in between. For example when the hw_ptr will be updated with hw_ptr += buffer_size and it is using the buffer_size as wrap around hw_ptr %= buffer_size would result in the same value as before the add operation. Due to that the user application would not recognize that the complete audio buffer was copied. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130942.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 3aed332..c22a5d0 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -40,6 +40,7 @@ typedef struct { char **port_names; unsigned int num_ports; + snd_pcm_uframes_t boundary; unsigned int hw_ptr; unsigned int sample_bits; snd_pcm_uframes_t min_avail; @@ -130,6 +131,21 @@ static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, static snd_pcm_sframes_t snd_pcm_jack_pointer(snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; + + /* ALSA library is calulating the delta between the last pointer and + * the current one. + * Normally it is expecting a value between 0 and buffer_size. + * The following example would result in an negative delta + * which would result in a hw_ptr which will be reduced. + * last_hw = jack->boundary - io->buffer_size + * hw = 0 + * But we cannot use + * return jack->hw_ptr % io->buffer_size; + * because in this case an update of + * hw_ptr += io->buffer_size + * would not be recognized by the ALSA library. + * Therefore we are using jack->boundary as the wrap around. + */ return jack->hw_ptr; } @@ -162,7 +178,7 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) while (xfer < nframes) { snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = hw_ptr; + snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; snd_pcm_uframes_t cont = io->buffer_size - offset; if (cont < frames) @@ -176,7 +192,8 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) } hw_ptr += frames; - hw_ptr %= io->buffer_size; + if (hw_ptr >= jack->boundary) + hw_ptr -= jack->boundary; xfer += frames; } jack->hw_ptr = hw_ptr; @@ -200,6 +217,8 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) err = snd_pcm_sw_params_current(io->pcm, swparams); if (err == 0) { snd_pcm_sw_params_get_avail_min(swparams, &jack->min_avail); + /* get boundary for available calulation */ + snd_pcm_sw_params_get_boundary(swparams, &jack->boundary); } /* deactivate jack connections if this is XRUN recovery */ -- 2.7.4

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