January 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ASoC: mediatek: mt8192-mt6359: add format constraints for RT5682
by Tzung-Bi Shih 25 Jan '21

25 Jan '21

MT8192 determines the I2S clock rates according to the sampling rates. There is only 1 set of I2S in between MT8192 and RT5682. If playing and capturing via RT5682 in different sampling rates, the I2S data will be corrupted. Adds format constraints to the corresponding DAI links to make sure the sampling rates are symmetric. Fixes: 18b13ff23fab ("ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from previous version: - Separated from the series [1] per [2]. - Updated commit message for more details. [1]: https://patchwork.kernel.org/project/alsa-devel/list/?series=419769 [2]: https://patchwork.kernel.org/project/alsa-devel/patch/20210122100742.369912… .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 49 +++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index cc0fc72305d2..c368236d8a3a 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -410,6 +410,53 @@ static const struct snd_soc_ops mt8192_mt6359_rt1015_rt5682_capture1_ops = { .startup = mt8192_mt6359_rt1015_rt5682_cap1_startup, }; +static int +mt8192_mt6359_rt5682_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int channels[] = { + 1, 2 + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + static const unsigned int rates[] = { + 48000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channels failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8192_mt6359_rt5682_ops = { + .startup = mt8192_mt6359_rt5682_startup, +}; + /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -657,6 +704,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8192_mt6359_rt5682_ops, SND_SOC_DAILINK_REG(playback3), }, { @@ -730,6 +778,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8192_mt6359_rt5682_ops, SND_SOC_DAILINK_REG(capture2), }, { -- 2.30.0.280.ga3ce27912f-goog

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[PATCH 0/2] ASoC: SOF: partial fix to Kconfig issues
by Pierre-Louis Bossart 25 Jan '21

25 Jan '21

We've had several reports of broken dependencies. The 'right' fix is to revisit the module dependencies as suggested by Arnd Bergmann. This is WIP at https://github.com/thesofproject/linux/pull/2683. Since this is taking longer than expected, I am only sharing quick fixes for now. Pierre-Louis Bossart (2): ASoC: SOF: Intel: soundwire: fix select/depend unmet dependencies ASoC: SOF: SND_INTEL_DSP_CONFIG dependency sound/soc/sof/intel/Kconfig | 3 ++- sound/soc/sof/sof-acpi-dev.c | 11 ++++++----- sound/soc/sof/sof-pci-dev.c | 10 ++++++---- 3 files changed, 14 insertions(+), 10 deletions(-) -- 2.25.1

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ASoC: simple-card, fsl-sai, ak4458, imx-ak4458
by Eliot Blennerhassett 25 Jan '21

25 Jan '21

Greetings, late last year I worked on a project using $SUBJECT. That has been suspended at least for now, so I'm writing these notes to my future self, or anyone else who may want to tread the same path. I'm happy to expand on explanation, or work on patches if anyone is interested. Background: Normally, fsl-sai is used with the platform-specific imx-ak4458 (afaik only in vendor tree). However, our project wished to be able use pcm1789 which has no imx specific driver so I started trying to use simple-card with the generic ak4458 driver. I encountered a some of problems doing this: ------------------------------------------------------------------------ 1) Reset polarity of ak4458. When imx-ak4458 is used, the platform driver handles the codec reset specified in DT compatible = "fsl,imx-audio-ak4458"; ak4458,pdn-gpio = <&gpio4 20 GPIO_ACTIVE_LOW>; Used here. Afaics gpio_set_value sets the raw value given, ignoring the polarity specified by the DT? gpio_set_value_cansleep(priv->pdn_gpio, 0); usleep_range(1000, 2000); gpio_set_value_cansleep(priv->pdn_gpio, 1); usleep_range(1000, 2000); The codec driver reset code is not used. When simple-card is used, the codec driver handles reset, specified like: compatible = "asahi-kasei,ak4458"; reset-gpios = <&gpio4 20 GPIO_ACTIVE_LOW>; and used here (inverse for power on) static void ak4458_power_off(struct ak4458_priv *ak4458) { if (ak4458->reset_gpiod) { gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); usleep_range(1000, 2000); } } My understanding of gpiod functions is that value being set is the *logical* value of the GPIO. I.e. setting an active low gpio to 0 will set the hardware pin high. So it appears that the power_off and power_on functions are doing the opposite of what is intended. This is borne out by my hardware working correctly when the ------------------------------------------------------------------------ 2) Clock rate setting with simple-card When simple-card is used and DT specifies mclk fs: simple-audio-card,mclk-fs = <256>; asoc_simple_hw_params() calls snd_soc_dai_set_sysclk(..., clk_id=0, ...) The hard-coded clk_id=0 doesn't work with fsl-sai, which requires clk_id==1. For my testing purposes I changed the hard-coded value, but I think the proper solution could be to add a DT property to specify the clk_id (default=0) ? ------------------------------------------------------------------------ 3) Memory mapped stream access by aplay does not work. This precludes use of alsa plugins e.g. dmix I have found no reason or solution for this so far ------------------------------------------------------------------------ 4) Unable to get multiple source clocks working with fsl-sai With a single assigned-clock, switching between 48k and 44k1 clock rate families is accomplished by reparenting the root clock to the appropriate audio pll clock. &sai2 { assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>; assigned-clocks = <&clk IMX8MM_CLK_SAI2>; assigned-clock-rates = <12288000>; ... }; However if two of the sai mclks could be set to 48k and 44k1 derived rates respectively, the clock reparenting would not be required, and fsl_sai_set_bclk() would search the mclks and choose the appropriate mclk for the requested rate. DT would be something like this: &sai2 { assigned-clock-parents = <&clk IMX8MM_AUDIO_PLL1_OUT>, <&clk IMX8MM_AUDIO_PLL2_OUT>; assigned-clocks = <&clk IMX8MM_CLK_SAI2>, <&clk IMX8MM_CLK_SAI1>; assigned-clock-rates = <12288000>, <11289600>; }; This setup doesn't work as I hoped it would, don't know why not. regards -- Eliot Blennerhassett

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[PATCH] ALSA: fireface: fix info leak in hwdep_read()
by Dan Carpenter 25 Jan '21

25 Jan '21

If "ff->dev_lock_changed" has not changed and "count" is too large then this will copy data beyond the end of the struct to user space. Fixes: f656edd5fb33 ("ALSA: fireface: add hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/fireface/ff-hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-hwdep.c b/sound/firewire/fireface/ff-hwdep.c index 4b2e0dff5ddb..b84dde609a03 100644 --- a/sound/firewire/fireface/ff-hwdep.c +++ b/sound/firewire/fireface/ff-hwdep.c @@ -35,12 +35,12 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (ff->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (ff->dev_lock_count > 0); ff->dev_lock_changed = false; - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&ff->lock); -- 2.29.2

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[PATCH] ALSA: hda: intel-dsp-config: add PCI id for TGL-H
by Kai Vehmanen 25 Jan '21

25 Jan '21

From: Bard Liao <bard.liao(a)intel.com> Adding PCI id for TGL-H. Like for other TGL platforms, SOF is used if Soundwire codecs or PCH-DMIC is detected. Signed-off-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Xiuli Pan <xiuli.pan(a)intel.com> Reviewed-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a0d070c60c9..c45686172517 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,6 +307,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa0c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x43c8, + }, #endif /* Elkhart Lake */ -- 2.29.2

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[PATCH] ALSA: hda: Assign boolean values to a bool variable
by Jiapeng Zhong 25 Jan '21

25 Jan '21

Fix the following coccicheck warnings: ./sound/pci/hda/hda_intel.c:2309:3-23: WARNING: Assignment of 0/1 to bool variable. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Zhong <abaci-bugfix(a)linux.alibaba.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5a50d3a..24e9762 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2304,7 +2304,7 @@ static int azx_probe_continue(struct azx *chip) /* HSW/BDW controllers need this power */ if (CONTROLLER_IN_GPU(pci)) - hda->need_i915_power = 1; + hda->need_i915_power = true; } /* Request display power well for the HDA controller or codec. For -- 1.8.3.1

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[PATCH v2] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-750
by Olivia Mackintosh 24 Jan '21

24 Jan '21

Re-add the sample-rate quirk originally implemented by Dmitry Panchenko for the DJM-900NXS2 but for the DJM-750. This commit only adds it again for the DJM-750 since I can only verify that it is absolutely required for this device. Other models may need this patch but I'm hesitant to add them as I cannot test. The 'wIndex' is passed literally in the function call as we cannot derive this from 'fmt->sync_ep' due to it not being set. We can change this if and when we revisit implicit FB. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef1..b49a66f863f3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,23 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + u16 windex) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate&0xff; + sr[1] = (cur_rate>>8)&0xff; + sr[2] = (cur_rate>>16)&0xff; + usb_set_interface(subs->dev, 0, 1); + // we should derive windex from fmt-sync_ep but it's not set + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, windex, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1500,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ + pioneer_djm_set_format_quirk(subs, 0x0086); + break; } } -- 2.30.0

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[PATCH 0/7] ALSA: add virtio sound driver
by Anton Yakovlev 24 Jan '21

24 Jan '21

This series implements a driver part of the virtio sound device specification v8 [1]. The driver supports PCM playback and capture substreams, jack and channel map controls. A message-based transport is used to write/read PCM frames to/from a device. The series is based (and was actually tested) on Linus's master branch [2], on top of commit 1e2a199f6ccd ("Merge tag 'spi-fix-v5.11-rc4' of ...") As a device part was used OpenSynergy proprietary implementation. Any comments are very welcome. [1] https://lists.oasis-open.org/archives/virtio-dev/202003/msg00185.html [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git Anton Yakovlev (7): uapi: virtio_ids: add a sound device type ID from OASIS spec uapi: virtio_snd: add the sound device header file ALSA: virtio: add virtio sound driver ALSA: virtio: handling control and I/O messages for the PCM device ALSA: virtio: PCM substream operators ALSA: virtio: introduce jack support ALSA: virtio: introduce device suspend/resume support MAINTAINERS | 8 + include/uapi/linux/virtio_ids.h | 1 + include/uapi/linux/virtio_snd.h | 361 +++++++++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/virtio/Kconfig | 10 + sound/virtio/Makefile | 13 + sound/virtio/virtio_card.c | 593 ++++++++++++++++++++++++++++++++ sound/virtio/virtio_card.h | 121 +++++++ sound/virtio/virtio_chmap.c | 237 +++++++++++++ sound/virtio/virtio_ctl_msg.c | 293 ++++++++++++++++ sound/virtio/virtio_ctl_msg.h | 122 +++++++ sound/virtio/virtio_jack.c | 255 ++++++++++++++ sound/virtio/virtio_pcm.c | 582 +++++++++++++++++++++++++++++++ sound/virtio/virtio_pcm.h | 132 +++++++ sound/virtio/virtio_pcm_msg.c | 317 +++++++++++++++++ sound/virtio/virtio_pcm_ops.c | 524 ++++++++++++++++++++++++++++ 17 files changed, 3573 insertions(+), 1 deletion(-) create mode 100644 include/uapi/linux/virtio_snd.h create mode 100644 sound/virtio/Kconfig create mode 100644 sound/virtio/Makefile create mode 100644 sound/virtio/virtio_card.c create mode 100644 sound/virtio/virtio_card.h create mode 100644 sound/virtio/virtio_chmap.c create mode 100644 sound/virtio/virtio_ctl_msg.c create mode 100644 sound/virtio/virtio_ctl_msg.h create mode 100644 sound/virtio/virtio_jack.c create mode 100644 sound/virtio/virtio_pcm.c create mode 100644 sound/virtio/virtio_pcm.h create mode 100644 sound/virtio/virtio_pcm_msg.c create mode 100644 sound/virtio/virtio_pcm_ops.c -- 2.30.0

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jack: use program_invocation_short_name
by GitHub pull_request - opened 24 Jan '21

24 Jan '21

alsa-project/alsa-plugins pull request #18 was opened from ArsenArsen: this results in more identifiable names in jack patchbays and control programs Request URL : https://github.com/alsa-project/alsa-plugins/pull/18 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/18.patch Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-750
by Olivia Mackintosh 24 Jan '21

24 Jan '21

Re-add the sample-rate quirk originally implemented by Dmitry Panchenko for the DJM-900NXS2 but for the DJM-750. This commit only adds it for the DJM-750 since I can only verify that it is absolutely required for this device. Other models may need this patch but I'm hesitant to add them as I cannot test. The 'wIndex' is taken from `fmt->sync_ep` as different devices have a different sync endpoint which is specified in 'quirks-table.h'. Adding other devices should only require adding an additional case. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef1..517346054fb7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,22 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate&0xff; + sr[1] = (cur_rate>>8)&0xff; + sr[2] = (cur_rate>>16)&0xff; + usb_set_interface(subs->dev, 0, 1); + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, fmt->sync_ep, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1499,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ + pioneer_djm_set_format_quirk(subs, fmt); + break; } } -- 2.30.0

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