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

[alsa-devel] [PATCH v7 0/8] Fix year 2038 issue for sound subsystem
by Arnd Bergmann 19 Oct '21

19 Oct '21

This is a series I worked on with Baolin in 2017 and 2018, but we never quite managed to finish up the last pieces. During the ALSA developer meetup at ELC-E 2018 in Edinburgh, a decision was made to go with this approach for keeping best compatibility with existing source code, and then I failed to follow up by resending the patches. Now I have patches for all remaining time_t uses in the kernel, so it's absolutely time to revisit them. I have done more review of the patches myself and found a couple of minor issues that I have fixed up, otherwise the series is still the same as before. Conceptually, the idea of these patches is: - 64-bit applications should see no changes at all, neither compile-time nor run-time. - 32-bit code compiled with a 64-bit time_t currently does not work with ALSA, and requires kernel changes and/or sound/asound.h changes - Most 32-bit code using these interfaces will work correctly on a modified kernel, with or without the uapi header changes. - 32-bit code using SNDRV_TIMER_IOCTL_TREAD requires the updated header file for 64-bit time_t support - 32-bit i386 user space with 64-bit time_t is broken for SNDRV_PCM_IOCTL_STATUS, SNDRV_RAWMIDI_IOCTL_STATUS and SNDRV_PCM_IOCTL_SYNC_PTR because of i386 alignment. This is also addressed by the updated uapi header. - PCM mmap is currently supported on native x86 kernels (both 32-bit and 64-bit) but not for compat mode. This series breaks the 32-bit native mmap support for 32-bit time_t, but instead allows it for 64-bit time_t on both native and compat kernels. This seems to be the best trade-off, as mmap support is optional already, and most 32-bit code runs in compat mode anyway. - I've tried to avoid breaking compilation of 32-bit code as much as possible. Anything that does break however is likely code that is already broken on 64-bit time_t and needs source changes to fix them. I hope I addressed all review comments by now, so please pull this for linux-5.6. A git branch with the same contents is available for testing at [1]. Arnd [1] https://git.kernel.org/pub/scm/linux/kernel/git/arnd/playground.git y2038-alsa-v7 [2] https://lore.kernel.org/lkml/CAK8P3a2Os66+iwQYf97qh05W2JP8rmWao8zmKoHiXqVHv… Changes since v6: (Arnd): - Add a patch to update the API versions - Hide a timespec reference in #ifndef __KERNEL__ to remove the last reference to time_t - Use a more readable way to do padding and describe it in the changelog - Rebase to linux-5.5-rc1, changing include/sound/soc-component.h and sound/drivers/aloop.c as needed. Changes since v5 (Arnd): - Rebased to linux-5.4-rc4 - Updated to completely remove timespec and time_t references from alsa - found and fixed a few bugs Changes since v4 (Baolin): - Add patch 5 to change trigger_tstamp member of struct snd_pcm_runtime. - Add patch 8 to change internal timespec. - Add more explanation in commit message. - Use ktime_get_real_ts64() in patch 6. - Split common code out into a separate function in patch 6. - Fix tu->tread bug in patch 6 and remove #if __BITS_PER_LONG == 64 macro. Changes since v3: - Move struct snd_pcm_status32 to pcm.h file. - Modify comments and commit message. - Add new patch2 ~ patch6. Changes since v2: - Renamed all structures to make clear. - Remove CONFIG_X86_X32 macro and introduced new compat_snd_pcm_status64_x86_32. Changes since v1: - Add one macro for struct snd_pcm_status_32 which only active in 32bits kernel. - Convert pcm_compat.c to use struct snd_pcm_status_64. - Convert pcm_native.c to use struct snd_pcm_status_64. --- Arnd Bergmann (3): ALSA: move snd_pcm_ioctl_sync_ptr_compat into pcm_native.c ALSA: add new 32-bit layout for snd_pcm_mmap_status/control ALSA: bump uapi version numbers Baolin Wang (6): ALSA: Replace timespec with timespec64 ALSA: Avoid using timespec for struct snd_timer_status ALSA: Avoid using timespec for struct snd_ctl_elem_value ALSA: Avoid using timespec for struct snd_pcm_status ALSA: Avoid using timespec for struct snd_rawmidi_status ALSA: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 74 ++++++-- include/sound/soc-component.h | 4 +- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 145 +++++++++++++-- sound/core/pcm.c | 12 +- sound/core/pcm_compat.c | 282 ++++++++---------------------- sound/core/pcm_lib.c | 38 ++-- sound/core/pcm_native.c | 226 +++++++++++++++++++++--- sound/core/rawmidi.c | 132 +++++++++++--- sound/core/rawmidi_compat.c | 87 +++------ sound/core/timer.c | 229 ++++++++++++++++++------ sound/core/timer_compat.c | 62 +------ sound/drivers/aloop.c | 2 +- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 15 files changed, 817 insertions(+), 494 deletions(-) -- 2.20.0

5 46

[REGRESSION] "ALSA: HDA: Early Forbid of runtime PM" broke my laptop's internal audio
by Michael Catanzaro 17 Aug '21

17 Aug '21

Hi, On my System76 Gazelle Pro (gazp9) laptop, internal audio broke after upgrading to kernel 5.10. The laptop's speakers produce no sound. Audio from headphones still works fine. A quick test is to visit GNOME System Settings -> Sound -> Output, select Test, click Front Left or Front Right and notice there is no sound. I tested a mainline kernel revision from yesterday (5.11-rc5+) and the regression is not yet fixed. I bisected the regression to "ALSA: HDA: Early Forbid of runtime PM": https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… I created a downstream bug report here, including some bits from dmesg that may or may not be relevant in comment #2: https://bugzilla.redhat.com/show_bug.cgi?id=1918933 Audio devices: $ lspci | grep -i audio 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) OS: Fedora 33 I'm happy to provide any additional required info or test patches if you CC me, since I'm not a kernel developer and am not subscribed to these lists. Thanks.

6 16

[PATCH] ALSA: hda: Enable runtime PM when codec probe fails
by Kai-Heng Feng 02 Jul '21

02 Jul '21

When codec probe fails, it doesn't enable runtime suspend, and can prevent graphics card from getting powered down: [ 4.280991] snd_hda_intel 0000:01:00.1: no codecs initialized $ cat /sys/bus/pci/devices/0000:01:00.1/power/runtime_status active So enable runtime PM when codec probe fails, to let graphics card be able to runtime suspend again. Merge azx_probe_continue() into azx_probe() and just let probe fail for this case could be a better approach. However that's a much bigger task so let's settle with a quirk workaround. BugLink: https://bugs.launchpad.net/bugs/1907212 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6852668f1bcb..3fd920069268 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2328,7 +2328,7 @@ static int azx_probe_continue(struct azx *chip) if (bus->codec_mask) { err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); if (err < 0) - goto out_free; + goto out_enable_rpm; } #ifdef CONFIG_SND_HDA_PATCH_LOADER @@ -2360,6 +2360,7 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); +out_enable_rpm: if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); -- 2.29.2

8 32

alsamixer: use background color instead of COLOR_BLACK
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #77 was opened from r-burns: Hi! I have a cosmetic "improvement" I'd like to share. Scare quotes because this is possibly opinionated, but I'd still like to know what you think. I think on typical terminals, COLOR_BLACK is the same or quite similar to the background color, so this does not make much of a difference. But on "solarized" color schemes and similar, the COLOR_BLACK may be quite different from the background color, so in my opinion it looks odd to have the alsamixer panel stand out this way. For example, I'm using the "Nord" color scheme on a pitch black #000000 background, and here's how the mixer looks with/without this patch (blank terminal, htop, and ncmpcpp also provided for comparison): ![image](https://user-images.githubusercontent.com/52847440/107131140-dbf46500-6888-11eb-938a-9a3258bf7184.png) As you can see, the current alsamixer in the center looks a bit strange and sticks out against my chosen background color and the other TUIs I'm using. Htop and ncmpcpp agree on using the builtin `-1` background color for their own backgrounds, but alsamixer uses COLOR_BLACK which stands out from the actual background. The alsamixer on the bottom right is using this patch, and blends in seamlessly. Again, this is just my opinion! Feel free to close if this is wrong or superfluous. Nonetheless, I thought I'd share this patch, in case anyone else feels the same way. Request URL : https://github.com/alsa-project/alsa-utils/pull/77 Patch URL : https://github.com/alsa-project/alsa-utils/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-utils

2 1

ALSA 1.2.4 release
by Jaroslav Kysela 31 May '21

31 May '21

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: FTP: ftp://ftp.alsa-project.org/pub HTTP: https://www.alsa-project.org/files/pub Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.3.2_v1.2.4 All files on the primary ALSA site have also GPG signature now. The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 1

[PATCH 00/14] ASoC: Intel/SOF: extend run-time driver selection to ACPI devices
by Pierre-Louis Bossart 26 Apr '21

26 Apr '21

The module snd-intel-dspcfg, suggested by Jaroslav last year, currently provide the means to select a PCI driver at run-time, based on quirks, recommendations or user selection via a kernel parameter. This capability removed a lot of confusions in distributions and removed the need for recompilations to select legacy HDaudio, SST or SOF drivers. This patchset extends the concept to ACPI devices. This was driven by the desire to at some point deprecate the Atom/SST driver for Baytrail and Cherrytrail, which is no longer maintained by Intel. By having the SOF driver enabled by distributions for Baytrail/Cherrytrail, we can enable more end-user tests and make the transition easier for distributions (likely in 2021 at this point). This patchset provides the same solution for Broadwell, mainly to have a single build for all Intel platforms. SOF on Broadwell remains an option not recommended for distributions, as long as the 'catpt' driver is maintained there is no burning desire to make SOF the default on the three Broadwell-based platforms with the DSP enabled. Pierre-Louis Bossart (14): ASoC: Intel: broadwell: add missing pm_ops ASoC: Intel: bdw-rt5677: add missing pm_ops ALSA: hda: intel-dsp-config: add helper for ACPI DSP driver selection ASoC: soc-acpi: add helper to identify parent driver. ASoC: Intel: boards: byt/cht: set card and driver name at run time ASoC: Intel: byt/cht: set pm ops dynamically ASoC: SOF: acpi: add dynamic selection of DSP driver ASoC: Intel: Atom: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and Atom/SST drivers ALSA: hda: intel-dsp-config: add Broadwell ACPI DSP driver selection ASoC: Intel: broadwell: set card and driver name dynamically ASoC: Intel: catpt: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and catpt drivers ALSA: hda: intel-dsp-config: ignore dsp_driver parameter for PCI legacy devices include/sound/intel-dsp-config.h | 7 ++ include/sound/soc-acpi.h | 6 + sound/hda/intel-dsp-config.c | 111 +++++++++++++++++++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/atom/sst/sst_acpi.c | 8 ++ sound/soc/intel/boards/bdw-rt5650.c | 17 ++- sound/soc/intel/boards/bdw-rt5677.c | 18 ++- sound/soc/intel/boards/broadwell.c | 20 ++-- sound/soc/intel/boards/bytcht_cx2072x.c | 27 +++-- sound/soc/intel/boards/bytcht_da7213.c | 27 +++-- sound/soc/intel/boards/bytcht_es8316.c | 29 +++-- sound/soc/intel/boards/bytcr_rt5640.c | 30 +++-- sound/soc/intel/boards/bytcr_rt5651.c | 27 +++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 29 +++-- sound/soc/intel/boards/cht_bsw_nau8824.c | 29 +++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 38 ++++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 29 +++-- sound/soc/intel/catpt/device.c | 12 ++ sound/soc/sof/intel/Kconfig | 33 +++--- sound/soc/sof/sof-acpi-dev.c | 14 ++- 20 files changed, 392 insertions(+), 121 deletions(-) -- 2.25.1

6 54

[PATCH] ASoC: fsl: imx-pcm-dma: Don't request dma channel in probe
by Shengjiu Wang 22 Apr '21

22 Apr '21

There are two requirements that we need to move the request of dma channel from probe to open. - When dma device binds with power-domains, the power will be enabled when we request dma channel. If the request of dma channel happen on probe, then the power-domains will be always enabled after kernel boot up, which is not good for power saving, so we need to move the request of dma channel to .open(); - With FE-BE case, if the dma channel is requested in probe, then there will be below issue, which is caused by that the dma channel will be requested duplicately [ 638.906268] sysfs: cannot create duplicate filename '/devices/soc0/soc/2000000.bus/2000000.spba-bus/2024000.esai/dma:tx' [ 638.919061] CPU: 1 PID: 673 Comm: aplay Not tainted 5.7.0-rc1-12956-gfc64b2585593 #287 [ 638.927113] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 638.933690] [<c0110dd8>] (unwind_backtrace) from [<c010b8ec>] (show_stack+0x10/0x14) [ 638.941464] [<c010b8ec>] (show_stack) from [<c0557fc0>] (dump_stack+0xe4/0x118) [ 638.948808] [<c0557fc0>] (dump_stack) from [<c032aeb4>] (sysfs_warn_dup+0x50/0x64) [ 638.956406] [<c032aeb4>] (sysfs_warn_dup) from [<c032b1a8>] (sysfs_do_create_link_sd+0xc8/0xd4) [ 638.965134] [<c032b1a8>] (sysfs_do_create_link_sd) from [<c05dc668>] (dma_request_chan+0xb0/0x210) [ 638.974120] [<c05dc668>] (dma_request_chan) from [<c05dc7d0>] (dma_request_slave_channel+0x8/0x14) [ 638.983111] [<c05dc7d0>] (dma_request_slave_channel) from [<c09d5548>] (fsl_asrc_dma_hw_params+0x1e0/0x438) [ 638.992881] [<c09d5548>] (fsl_asrc_dma_hw_params) from [<c09c1654>] (soc_pcm_hw_params+0x4a0/0x6a8) [ 639.001952] [<c09c1654>] (soc_pcm_hw_params) from [<c09c39d4>] (dpcm_fe_dai_hw_params+0x70/0xe4) [ 639.010765] [<c09c39d4>] (dpcm_fe_dai_hw_params) from [<c099b274>] (snd_pcm_hw_params+0x158/0x418) [ 639.019750] [<c099b274>] (snd_pcm_hw_params) from [<c099c5a0>] (snd_pcm_ioctl+0x734/0x183c) [ 639.028129] [<c099c5a0>] (snd_pcm_ioctl) from [<c029ff94>] (ksys_ioctl+0x2ac/0xb98) [ 639.035812] [<c029ff94>] (ksys_ioctl) from [<c0100080>] (ret_fast_syscall+0x0/0x28) [ 639.043490] Exception stack(0xec529fa8 to 0xec529ff0) [ 639.048565] 9fa0: bee84650 01321870 00000004 c25c4111 bee84650 0002000f [ 639.056766] 9fc0: bee84650 01321870 01321820 00000036 00001f40 00000000 0002c2f8 00000003 [ 639.064964] 9fe0: b6f483fc bee8451c b6ee2655 b6e1dcf8 [ 639.070339] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-pcm-dma.c | 173 +++++++++++++++++++++++++++++++++--- 1 file changed, 159 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 04a9bc749016..dae53b384df4 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -11,6 +11,7 @@ #include <linux/dmaengine.h> #include <linux/types.h> #include <linux/module.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,24 +30,168 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = filter, -}; +static int imx_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_dmaengine_dai_dma_data *dma_data; + struct dma_slave_config config; + struct dma_chan *chan; + int ret = 0; -int imx_pcm_dma_init(struct platform_device *pdev, size_t size) + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + chan = snd_dmaengine_pcm_get_chan(substream); + if (!chan) + return -EINVAL; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + if (!dma_data) + return -EINVAL; + + snd_dmaengine_pcm_set_config_from_dai_data(substream, + dma_data, + &config); + return dmaengine_slave_config(chan, &config); +} + +static int imx_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) { - struct snd_dmaengine_pcm_config *config; + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static snd_pcm_uframes_t imx_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + return snd_dmaengine_pcm_pointer(substream); +} + +static int imx_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + return snd_dmaengine_pcm_trigger(substream, cmd); +} + +static int imx_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_dmaengine_dai_dma_data *dma_data; + struct device *dev = component->dev; + struct snd_pcm_hardware hw; + struct dma_chan *chan; + int ret; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "failed to set pcm hw params periods\n"); + return ret; + } + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + if (!dma_data) + return -EINVAL; + + chan = dma_request_slave_channel(cpu_dai->dev, tx ? "tx" : "rx"); + if (!chan) { + /* Try to request channel using compat_filter_fn */ + chan = snd_dmaengine_pcm_request_channel(filter, + dma_data->filter_data); + if (!chan) + return -ENXIO; + } - config = devm_kzalloc(&pdev->dev, - sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL); - if (!config) - return -ENOMEM; - *config = imx_dmaengine_pcm_config; + ret = snd_dmaengine_pcm_open(substream, chan); + if (ret) + goto pcm_open_fail; - return devm_snd_dmaengine_pcm_register(&pdev->dev, - config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(chan->device->dev); + hw.buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE; + hw.fifo_size = dma_data->fifo_size; + + /* Refine the hw according to caps of DMA. */ + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &hw, + chan); + if (ret < 0) + goto refine_runtime_hwparams_fail; + + snd_soc_set_runtime_hwparams(substream, &hw); + + /* Support allocate memory from IRAM */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_IRAM, + chan->device->dev, + hw.buffer_bytes_max, + &substream->dma_buffer); + if (ret < 0) + goto alloc_pagas_fail; + + return 0; + +alloc_pagas_fail: +refine_runtime_hwparams_fail: + snd_dmaengine_pcm_close(substream); +pcm_open_fail: + dma_release_channel(chan); + + return ret; +} + +static int imx_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + + return snd_dmaengine_pcm_close_release_chan(substream); +} + +static int imx_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + + return dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); +} + +static const struct snd_soc_component_driver imx_pcm_component = { + .name = "imx-pcm-dma", + .pcm_construct = imx_pcm_new, + .open = imx_pcm_open, + .close = imx_pcm_close, + .hw_params = imx_pcm_hw_params, + .hw_free = imx_pcm_hw_free, + .trigger = imx_pcm_trigger, + .pointer = imx_pcm_pointer, +}; + +int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +{ + return devm_snd_soc_register_component(&pdev->dev, + &imx_pcm_component, NULL, 0); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -- 2.21.0

5 17

[PATCH v6 0/5] Enable rk3066a HDMI sound
by Johan Jonker 22 Mar '21

22 Mar '21

Update the rk3066a HDMI documents with a #sound-dai-cells property. Include the code for sound in the HDMI driver. Add a simple-sound-card compatible node to rk3066a.dtsi, because I2S0 and HDMI TX are connected internally. And as last enable rk3066a HDMI sound in the rk3066a-mk808.dts file. make ARCH=arm CROSS_COMPILE=/usr/bin/arm-linux-gnueabi- -j4 cp ./arch/arm/boot/zImage ../zImage-dtb cat ./arch/arm/boot/dts/rk3066a-mk808.dtb >> ../zImage-dtb ../tools/rkcrc -k ../zImage-dtb ../mk808.img sudo ../tools/rkflashtool w 0x4000 0x8000 < ../mk808.img sudo ../tools/rkflashtool b Changed v6: remove patches that are applied to linux-next add platform_device_unregister() restyle Changed v5: removed unused variable fill frame structure Johan Jonker (4): dt-bindings: display: add #sound-dai-cells property to rockchip rk3066 hdmi ARM: dts: rockchip: rk3066a: add #sound-dai-cells to hdmi node ARM: dts: rockchip: add hdmi-sound node to rk3066a.dtsi ARM: dts: rockchip: enable hdmi_sound and i2s0 for rk3066a-mk808 Zheng Yang (1): drm: rockchip: add sound support to rk3066 hdmi driver .../display/rockchip/rockchip,rk3066-hdmi.yaml | 4 + arch/arm/boot/dts/rk3066a-mk808.dts | 8 + arch/arm/boot/dts/rk3066a.dtsi | 17 ++ drivers/gpu/drm/rockchip/Kconfig | 2 + drivers/gpu/drm/rockchip/rk3066_hdmi.c | 277 ++++++++++++++++++++- 5 files changed, 307 insertions(+), 1 deletion(-) -- 2.11.0

2 7

[PATCH 0/3] Use clocks property in a device node
by Sameer Pujar 15 Mar '21

15 Mar '21

It is recommended to not specifiy clocks property in an endpoint subnode. This series moves clocks to device node. However after moving the clocks to device node, the audio playback or capture fails. The specified clock is not actually getting enabled and hence the failure is seen. There seems to be a bug in simple-card-utils.c where clock handle is not assigned when parsing clocks from device node. Fix the same and revert original change which actually added clocks property in endpoint subnode. Also update Jetson AGX Xavier DT where the usage is found. Sameer Pujar (3): ASoC: simple-card-utils: Fix device module clock Revert "ASoC: audio-graph-card: Add clocks property to endpoint node" arm64: tegra: Move clocks from RT5658 endpoint to device node .../devicetree/bindings/sound/audio-graph-port.yaml | 3 --- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 2 +- sound/soc/generic/simple-card-utils.c | 13 ++++++------- 3 files changed, 7 insertions(+), 11 deletions(-) -- 2.7.4

4 36

[PATCH alsa-ucm-conf 1/2] codecs/rt5672: Move +12dB input boost from 'STO1 ADC Boost Gain' to 'ADC Capture Volume'
by Hans de Goede 13 Mar '21

13 Mar '21

Both the 'STO1 ADC Boost Gain' and the 'ADC Capture Volume' controls are digital and the 'ADC Capture Volume' control goes up to +30dB. When adding hw-volume-control support adding the +12dB boost gain on to of the max +30dB of 'ADC Capture Volume' is a bit too much. So move the +12dB from the 'STO1 ADC Boost Gain' control to the 'ADC Capture Volume' control. This way we keep the same setting for userspace which does not support hw-volume-control, while getting a better range when we start using 'ADC Capture Volume' as hw-volume-control. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/rt5672/EnableSeq.conf | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/rt5672/EnableSeq.conf b/ucm2/codecs/rt5672/EnableSeq.conf index ddfb1c7..e373275 100644 --- a/ucm2/codecs/rt5672/EnableSeq.conf +++ b/ucm2/codecs/rt5672/EnableSeq.conf @@ -26,14 +26,13 @@ EnableSequence [ # cset "name='LOUT MIX OUTMIX L Switch' on" # cset "name='LOUT MIX OUTMIX R Switch' on" - # 47=0dB, 0.375 dB/step - cset "name='ADC Capture Volume' 47" + # Both the DMICs and the headset mic (even with the 20dB boost below) + # are quite soft, set the default capture volume to +12dB + # 47=0dB, 0.375dB/step, 79=+12dB + cset "name='ADC Capture Volume' 79" cset "name='ADC Capture Switch' on" - - # Both the DMICs and the headset mic (even with the 20 dB boost below) - # are quite soft, set the ADC Boost gain (range 0-3) to 1 = 12 dB the - # next step is 24 dB which is a bit too much - cset "name='STO1 ADC Boost Gain Volume' 1" + # 0=0dB + cset "name='STO1 ADC Boost Gain Volume' 0" # Headset mic is quite soft, boost it a bit, 1 = 20dB which is the first # available boost step -- 2.30.1

2 3