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

[alsa-devel] [PATCH] sound: stm32: replace "%p" with "%pK"
by Benjamin Gaignard 09 Jul '18

09 Jul '18

The format specifier "%p" can leak kernel addresses. Use "%pK" instead. Signed-off-by: Benjamin Gaignard <benjamin.gaignard(a)st.com> --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index db73fef3e500..0e9373064032 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) unsigned int old_pos = priv->pos; unsigned int cur_size = size; - dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n", __func__, &pcm_buff[priv->pos], priv->pos, size); if ((priv->pos + size) > buff_size) { -- 2.15.0

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[alsa-devel] How to add new sound card with ASoC generic driver
by Riccardo Casagrande 09 Jul '18

09 Jul '18

Hello, I'm new to ALSA and linux in general, I need help from who knows well ALSA due to the tight schedule of the project. I'm working on a Yocto project on NXP i.MX6Quad. The hardware architetture is this: the i.MAX6Quad has 6 output connected to 3 ST FDA2100 and 2 input from ADC. In the first step what I need is simply to enable 6 output and send I2S audio to FDAs. In the second step is required a clock for ADC and enable input ports. I spent days/weeks reading docs, examples etc without success. I understood that working on device tree file and setting a simple audio card with generic codec properly is the answer, but my knowledge is almost nothing. I followed these docs/thread https://elixir.bootlin.com/linux/latest/source/Documentation/devicetree/bin… https://community.nxp.com/thread/475428 but compiling I have syntax errors on device tree file. There is someone who can help me step by step? Thanks in advance Riccardo

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[alsa-devel] Binding volume and mute switch
by Ladislav Michl 09 Jul '18

09 Jul '18

Hello, to provide almost self descriptive controls to alsamixer I'd like to achive something like this: http://ladislav.michl.sweb.cz/alsamixer1.png Master volume is muted using switch coupled with master volume control, microphone is muted using switch coupled with microphone volume control, etc... For mic and line input is works as expected, but for sidetone and master volume I had to use "proper" names for mixer controls, see bellow: /* Input mixer */ static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { SOC_DAPM_DOUBLE("Mic Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), SOC_DAPM_DOUBLE("Line Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), }; /* Output mixer */ static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { SOC_DAPM_DOUBLE_R("Line Bypass Switch", MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), }; /* Sidetone mixer */ static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), }; /* Line out switch */ static const struct snd_kcontrol_new max9867_line_out_control = SOC_DAPM_DOUBLE_R("Switch", MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), SND_SOC_DAPM_INPUT("LINL"), SND_SOC_DAPM_INPUT("LINR"), SND_SOC_DAPM_MIXER("Left Input Mixer", MAX9867_PWRMAN, 6, 1, max9867_input_mixer_controls, ARRAY_SIZE(max9867_input_mixer_controls)), SND_SOC_DAPM_MIXER("Right Input Mixer", MAX9867_PWRMAN, 5, 1, max9867_input_mixer_controls, ARRAY_SIZE(max9867_input_mixer_controls)), SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, max9867_sidetone_mixer_controls, ARRAY_SIZE(max9867_sidetone_mixer_controls)), SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, max9867_output_mixer_controls, ARRAY_SIZE(max9867_output_mixer_controls)), SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, &max9867_line_out_control), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { {"Left Input Mixer", "Mic Switch", "MICL"}, {"Right Input Mixer", "Mic Switch", "MICR"}, {"Left Input Mixer", "Line Switch", "LINL"}, {"Right Input Mixer", "Line Switch", "LINR"}, {"ADCL", NULL, "Left Input Mixer"}, {"ADCR", NULL, "Right Input Mixer"}, {"Digital", "Sidetone Switch", "ADCL"}, {"Digital", "Sidetone Switch", "ADCR"}, {"DACL", NULL, "Digital"}, {"DACR", NULL, "Digital"}, {"Output Mixer", "Line Bypass Switch", "LINL"}, {"Output Mixer", "Line Bypass Switch", "LINR"}, {"Output Mixer", NULL, "DACL"}, {"Output Mixer", NULL, "DACR"}, {"Master Playback", "Switch", "Output Mixer"}, {"LOUT", NULL, "Master Playback"}, {"ROUT", NULL, "Master Playback"}, }; With other control names than "Digital" for sidetone digital mixer, sidetone switch is shown separately from sidetone volume. The same for master volume switch, which is bind with master volume only once its name is "Master Playback". Is it intended to work this way or there is some other, more elegant way how to bind mute switch with volume control. (above solution seem too fragile to me) Thank you, ladis

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[alsa-devel] Hiding some of the controls defined in codec driver from machine driver context
by Giedrius Trainavičius 09 Jul '18

09 Jul '18

Hello, Is it possible to 'hide' some of the controls of the codec in the machine driver? The codec I'm using exposes many advanced controls which are useful for board developers, but not for the end user, so I'd like to define my own 'simplified' controls, and take care of board specific logic internally in the machine driver. I found this old discussion on the topic: https://www.spinics.net/lists/alsa-devel/msg36386.html - I'm targeting kernel 4.14, maybe there's already some means to achieve this? Thank you! Giedrius.

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Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Lukas Wunner 09 Jul '18

09 Jul '18

On Mon, Jul 09, 2018 at 08:52:33AM +0000, Qu, Jim wrote: > Now, I found the audio device will auto suspend even if the gpu is active, > and if I plug in a HDMI device it also do not resume back. > > 1. Did you encounter similar issue before? > 2. audio will auto suspend as default at beginning of boot, is it expect > behaviour? Yes, that's expected. Once you start streaming audio to attached displays, user space opens the codec device and this causes the HDA controller to runtime resume. If the discrete GPU is suspended at that moment, it will be powered on and kept powered on as long as user space is streaming audio. Do you need to runtime resume the HDA controller even if user space isn't streaming audio? Why, and in which situation exactly? You're saying above that the HDA controller isn't runtime resumed on hotplug of a display. Is that necessary to retrieve ELD or something? I'm not sure if there's code in the HDA driver to acquire a runtime PM ref on HPD, but maybe it's necessary. Or maybe the code is there but somehow no HPD interrupt is received by the HDA driver? Thanks, Lukas

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[alsa-devel] Getting a notification of snd_kcontrol value change, defined in codec, in machine driver
by Giedrius Trainavičius 08 Jul '18

08 Jul '18

Hey, What would be the best way in machine driver to learn about changes of alsa mixer control values? The controls are defined in the codec driver. The intended use case would be to use the current volume setting for the codec on the board to forward to a dedicated microcontroller for visual feedback. I am aware that this could be bridged over via a user space program, but I will have to forward the current PCM stream format to the MCU anyway from hw_params, so ideally I would like all of this functionality contained in the machine driver. Thank you, Giedrius.

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[alsa-devel] [PATCH] slimbus: stream: Fix htmldocs warnings
by Srinivas Kandagatla 07 Jul '18

07 Jul '18

This patch fixes below warning during building htmldoc: slimbus.h:352: warning: Function parameter or member 'name' not described in 'slim_stream_runtime' This patch also removes documentation for state variable in struct slim_stream_runtime which was redundant and removed. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/slimbus/slimbus.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 411880f3fc0a..32ac4046154f 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -325,9 +325,8 @@ enum slim_transport_protocol { /** * struct slim_stream_runtime - SLIMbus stream runtime instance * - * @dev: Name of the stream + * @name: Name of the stream * @dev: SLIM Device instance associated with this stream - * @state: state of stream * @direction: direction of stream * @prot: Transport protocol used in this stream * @rate: Data rate of samples * -- 2.16.2

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[alsa-devel] Bad header guard in alsa-lib
by Guillaume Blanc 06 Jul '18

06 Jul '18

Hello, While building alsa-lib, I got a redefinition error caused by a bad header guard in pcm_plugin.h. I assume this is a mistake and not a weird include trick, attached is a patch to fix this. Best regards, Guillaume Blanc

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[alsa-devel] [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 06 Jul '18

06 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 4 * nau8810_mclk_scaler[i] / 10 Addresses-Coverity-ID: 1357595 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index bfd74b8..e7fd0b2 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,7 @@ static int nau88l0_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda/realtek - two more lenovo models need fixup of MIC_LOCATION
by Hui Wang 06 Jul '18

06 Jul '18

We have two new lenovo desktop models which need to apply the fixup of ALC294_FIXUP_LENOVO_MIC_LOCATION, and they have the same pin cfg as the machine with subsystem id:0x17aa3136, now use the pincfg table to apply the fixup for them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63154b9..666713e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6556,7 +6556,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), - SND_PCI_QUIRK(0x17aa, 0x3136, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), @@ -6822,6 +6821,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x02a11040}, {0x1b, 0x01014020}, {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, + {0x14, 0x90170110}, + {0x19, 0x02a11020}, + {0x1a, 0x02a11030}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: nau8824: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 06 Jul '18

06 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd1445..468d514 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

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[alsa-devel] [PATCH] regmap: slimbus: add support to raw read/writes
by Srinivas Kandagatla 05 Jul '18

05 Jul '18

SLIMbus supports upto 16 bytes in value management messages, so add support to raw read/writes upto 16 bytes. Also useful for paged register access on SLIMbus interfaced codecs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/base/regmap/regmap-slimbus.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap-slimbus.c b/drivers/base/regmap/regmap-slimbus.c index 91d501eda8a9..fc07518666b2 100644 --- a/drivers/base/regmap/regmap-slimbus.c +++ b/drivers/base/regmap/regmap-slimbus.c @@ -31,11 +31,30 @@ static int regmap_slimbus_byte_reg_write(void *context, unsigned int reg, return slim_writeb(sdev, reg, val); } +static int regmap_slimbus_write(void *context, const void *data, size_t count) +{ + struct slim_device *sdev = context; + + return slim_write(sdev, *(u16 *)data, count - 2, (u8 *)data + 2); +} + +static int regmap_slimbus_read(void *context, const void *reg, size_t reg_size, + void *val, size_t val_size) +{ + struct slim_device *sdev = context; + + return slim_read(sdev, *(u16 *)reg, val_size, val); +} + static struct regmap_bus regmap_slimbus_bus = { .reg_write = regmap_slimbus_byte_reg_write, .reg_read = regmap_slimbus_byte_reg_read, - .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, - .val_format_endian_default = REGMAP_ENDIAN_LITTLE, + .max_raw_write = 16, + .max_raw_read = 16, + .write = regmap_slimbus_write, + .read = regmap_slimbus_read, + .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, + .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; static const struct regmap_bus *regmap_get_slimbus(struct slim_device *slim, -- 2.16.2

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[alsa-devel] Can't detect "XRUNs"
by Raphaël Dingé 05 Jul '18

05 Jul '18

Hi, I don't get audio drop notification "XRUNs" in /proc/kmsg. Not sure if this is the right mailing list as maybe my question is more kernel related, but I'll try a shot here. I'm currently making a proof of concept to see if I can run a quite CPU hungry audio application running without audio drops over time given a set of tests. In particular I want to automate playback buffer underruns detection so that I don't have to sit and listen closely when they happen. With some very extreme tests, I can hear that there are audio drops all over the place. I have 2 audio cards in the system, one is audio over HDMI, and the other one is an USB audio card that supports audio class definition. In both case the sound is working as I would expect. The system itself is an Intel NUC i7. I configured the kernel with CONFIG_SND_DEBUG=y, CONFIG_SND_PCM_XRUN_DEBUG=y and CONFIG_SND_VERBOSE_PROCFS=y as well as echo 5 > /proc/asound/cardX/pcmYp/xrun_debug with X and Y the card and pcm stream of interest, and so hw:X,Y when opening the device with snd_pcm_open. Then I watch the output of the ksyslog interface using cat /proc/kmsg When an audio drop occurs, there is nothing about XRUNs or whatever it is called. I don't even know what kind of log message it should look like, and can't find it on google as well. With audio over HDMI, I have some occurence of ALSA: PCM: [Q] Lost interrupts? that seems more of less related to audio drops, but not always. With the USB audio card, I have nothing. I tried to write to the xrun_injection proc file in the stream, but nothing happens as well. Finally I log any occurence of snd_pcm_writei returning a negative number, but that never happens as well. Am I missing something completely? Thanks for any information on that matter, Raphael

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[alsa-devel] [PATCH v2 0/2] slimbus: Add Stream Support
by Srinivas Kandagatla 05 Jul '18

05 Jul '18

Thanks to Vinod and Stephen for reviewing v1 patchset This v2 patchset adds basic stream support for SLIMbus devices and controllers. Mostly inspired by soundwire stream patches. But slimbus stream is much simpler compared to soundwire >From slim_device side, we have below 6 new apis. slim_stream_allocate() - allocating runtime slim stream slim_stream_prepare() - to configure runtime stream with config slim_stream_enable() - enable channels/ports for data slim_stream_disable() - disable channels/ports. slim_stream_unprepare() - un configure runtime stream slim_stream_free() - free all associated memory. >From Controller side: Data channel Management and reconfiguration messages are applicable for all the controllers which are inline with SLIMbus specs. However light weight controller like NGD which have user specific implementation of some messages need to be invoked instead of standard message commands. For usecases like this simple enable/disable stream apis are provided. Assumptions: 1> Current design assumes that the channel and ports are statically allocated to the device during SoC integration, which is the case with all the Qualcomm SoCs. 2> One-to-One mapping between Port and Channel, SLIMBus versions earlier than 2.0 has only one endpoint per port. Current patchset can be extended to support more than one endpoints per port. 3> Only audio usecase, This patchset only supports Isochronous and Push/Pull transport protocols, which are sufficient for audio use cases. 4> DSP does all the data handling for the allocated channels. Which is true for Qcom SoCs. TODO: Bandwidth management. Dependency: This patchset has dependency on the NGD driver https://patchwork.kernel.org/patch/10474959/ Tested this patchset with WCD9335 codec playback and capture on DB820c using 4.18-rc1 with qdsp6 + slim tx patches merged in sound.git. I have pushed my working branch to [1] incase someone want to try. [1]:https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimb… Changes since v1 (https://lwn.net/Articles/758100/): - removed redundant state machine checks for audio usecases, suggested by Vinod. - fixed few typos. - removed irqsave/restore spinlock variant spotted by Vinod. - general cleanups suggested by Stephen and Vinod. thanks, srini Srinivas Kandagatla (2): slimbus: stream: add stream support slimbus: ngd: add stream support Documentation/driver-api/slimbus.rst | 5 + drivers/slimbus/Makefile | 2 +- drivers/slimbus/core.c | 2 + drivers/slimbus/qcom-ngd-ctrl.c | 149 ++++++++++- drivers/slimbus/slimbus.h | 189 ++++++++++++++ drivers/slimbus/stream.c | 477 +++++++++++++++++++++++++++++++++++ include/linux/slimbus.h | 46 ++++ 7 files changed, 867 insertions(+), 3 deletions(-) create mode 100644 drivers/slimbus/stream.c -- 2.16.2

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[alsa-devel] [PATCH - JACK IO plug v2 0/3] snd_pcm_drain support
by twischer＠de.adit-jv.com 05 Jul '18

05 Jul '18

Now in a clean manner as composed patches. pcm: ioplug: Provide avail helper function --------- No changes jack: Avoid call to snd_pcm_avail_update() from JACK thread --------- v2 uses snd_pcm_uframes_t as type for avail and does not check for avail < 0 becasue snd_pcm_ioplug_avail() will never retrun a value < 0 jack: Update poll_fd also in draining state --------- This patch is depending on [1]. Therefore it was rebased on top of [1]. Additionally the commit message was extended. v1 of this patch can be found in [2]. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137775.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137769.html

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[alsa-devel] [PATCH] ASoC: fsl_spdif: Use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 05 Jul '18

05 Jul '18

Add suffix ULL to constant 64 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: rate[index] * txclk_df * 64 Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87..740b90d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- 2.7.4

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[alsa-devel] [PATCH 1/1] ASoC: rsnd: cmd: Add missing newline to debug message
by jiada_wang＠mentor.com 05 Jul '18

05 Jul '18

From: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> To comply with the style of all kernel messages, add newline to the end of every message. Fixes: 70fb10529f61 ("ASoC: rsnd: add MIX (Mixer) support") Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/cmd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index d8043ad33540..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -86,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); -- 2.17.0

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[alsa-devel] [PATCH] ASoC: pxa-ssp: add support for an external clock in devicetree
by Daniel Mack 05 Jul '18

05 Jul '18

Allow setting a clock called 'extclk' in the device of the ssp-dai device. If specified, this clock will be set to the mclk rate from the DAI's .set_sysclk() callback. The DAI will also configure itself to use that external clock. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../bindings/sound/mrvl,pxa-ssp.txt | 8 ++++++ sound/soc/pxa/pxa-ssp.c | 25 +++++++++++++++++++ 2 files changed, 33 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt index efd3fb1f68d2..feef39b4a4fd 100644 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -5,6 +5,14 @@ Required properties: compatible Must be "mrvl,pxa-ssp-dai" port A phandle reference to a PXA ssp upstream device +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + Example: /* upstream device */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..c43fe3b9c79c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; @@ -205,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -773,6 +789,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { -- 2.17.1

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[alsa-devel] [PATCH v3] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 04 Jul '18

04 Jul '18

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 12 +++++++++--- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..82a41e4d7170 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,18 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; + /* use wait time from substream if available */ + if (substream->wait_time) + wait_time = substream->wait_time; + else + wait_time = 10 * 1000; /* 10 secs */ + if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->period_size * 2 / + (runtime->rate / 1000); wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time = msecs_to_jiffies(wait_time); } for (;;) { -- 2.17.1

2 2

[alsa-devel] [PATCH] ASoC: nau8825: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 04 Jul '18

04 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea49..b9fed99 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: Use snd_pcm_stop_xrun() helper
by Takashi Iwai 04 Jul '18

04 Jul '18

Hi, this is a series of trivial fixes to replace some open codes with the standard snd_pcm_stop_xrun() helper. Functionally same, but it addresses the potential stream lock issues automagically, too. Takashi === Takashi Iwai (4): ASoC: davinci: Use snd_pcm_stop_xrun() helper ASoC: qcom: Use snd_pcm_stop_xrun() helper ASoC: sti: Use snd_pcm_stop_xrun() helper ASoC: stm32: Use snd_pcm_stop_xrun() helper sound/soc/davinci/davinci-mcasp.c | 16 ++++------------ sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sti/uniperif_player.c | 6 +++--- sound/soc/sti/uniperif_reader.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 ++----- 5 files changed, 11 insertions(+), 22 deletions(-) -- 2.18.0

2 8

[alsa-devel] [PATCH 0/6] ASoC: qdsp6: Add missing slimbus capture support
by Srinivas Kandagatla 04 Jul '18

04 Jul '18

This patchset adds support to slimbus tx ports plus 3 trival fixes to do with closing port when its not really open. slim tx support invoving add dais and routing, rest of the dsp configuration commands are already available in mainline. slim tx is verified with wcd9335 codec on DB820c. Thanks, srini Srinivas Kandagatla (6): ASoC: qdsp6: q6afe: Add missing slimbus capture ports ASoC: qdsp6: q6afe-dai: add support to slim tx dais ASoC: qdsp6: q6routing: add slim rx routings ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels ASoC: qdsp6: q6afe-dai: do not close port if its not opened ASoC: qdsp6: q6asm-dai: do not close port if its not opened sound/soc/qcom/qdsp6/q6afe-dai.c | 179 +++++++++++++++++++++++++++++++++++---- sound/soc/qcom/qdsp6/q6afe.c | 21 +++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++ 4 files changed, 216 insertions(+), 16 deletions(-) -- 2.16.2

2 12

[alsa-devel] [PATCH - PCM rate v2 1/1] pcm: rate: Add error check for snd_pcm_avail_update()
by twischer＠de.adit-jv.com 04 Jul '18

04 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without these changes a negative error code returned by snd_pcm_avail_update() will be not handled correctly. With this patch the returned error code of snd_pcm_avail_update() will be returned by snd_pcm_rate_avail_update(). Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Compared to v1[1] of this patch only the changelog and the subject has changed. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137838.html diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 4e0c7ca..031a43d 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -970,9 +970,12 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) { snd_pcm_rate_t *rate = pcm->private_data; snd_pcm_t *slave = rate->gen.slave; - snd_pcm_uframes_t slave_size; + snd_pcm_sframes_t slave_size; slave_size = snd_pcm_avail_update(slave); + if (slave_size < 0) + return slave_size; + if (pcm->stream == SND_PCM_STREAM_CAPTURE) goto _capture; snd_pcm_rate_sync_hwptr(pcm); -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: pcm: trace XRUN event at injection, too
by Takashi Iwai 04 Jul '18

04 Jul '18

The PCM xrun injection triggers directly snd_pcm_stop() without the standard xrun handler, hence it's not recorded on the event buffer. Ditto for snd_pcm_stop_xrun() call and SNDRV_PCM_IOCTL_XRUN ioctl. They are inconvenient from the debugging POV. Let's make them to trigger XRUN via the standard helper more consistently. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm.c | 2 +- sound/core/pcm_lib.c | 7 ++++--- sound/core/pcm_local.h | 2 ++ sound/core/pcm_native.c | 8 ++++---- 4 files changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c352bfb973cc..6f037a4b8b52 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -497,7 +497,7 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, snd_pcm_stream_lock_irq(substream); runtime = substream->runtime; if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irq(substream); } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..c1d2e8e1fc6b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -153,7 +153,8 @@ EXPORT_SYMBOL(snd_pcm_debug_name); dump_stack(); \ } while (0) -static void xrun(struct snd_pcm_substream *substream) +/* call with stream lock held */ +void __snd_pcm_xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -201,7 +202,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, } } else { if (avail >= runtime->stop_threshold) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } } @@ -297,7 +298,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } if (pos >= runtime->buffer_size) { diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 7a499d02df6c..c515612969a4 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -65,4 +65,6 @@ static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} #endif +void __snd_pcm_xrun(struct snd_pcm_substream *substream); + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cecc79772c94..20174d0c0527 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1337,13 +1337,12 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) { unsigned long flags; - int ret = 0; snd_pcm_stream_lock_irqsave(substream, flags); if (snd_pcm_running(substream)) - ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irqrestore(substream, flags); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); @@ -1591,7 +1590,8 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) result = 0; /* already there */ break; case SNDRV_PCM_STATE_RUNNING: - result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); + result = 0; break; default: result = -EBADFD; -- 2.18.0

1 1

[alsa-devel] [PATCH - PCM rate 1/1] pcm: rate: Use signed variable for return of snd_pcm_avail_update
by twischer＠de.adit-jv.com 04 Jul '18

04 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this changes a negative error code returned by snd_pcm_avail_update() will be not handled correctly Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 4e0c7ca..031a43d 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -970,9 +970,12 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) { snd_pcm_rate_t *rate = pcm->private_data; snd_pcm_t *slave = rate->gen.slave; - snd_pcm_uframes_t slave_size; + snd_pcm_sframes_t slave_size; slave_size = snd_pcm_avail_update(slave); + if (slave_size < 0) + return slave_size; + if (pcm->stream == SND_PCM_STREAM_CAPTURE) goto _capture; snd_pcm_rate_sync_hwptr(pcm); -- 2.7.4

2 1