October 2020 - 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

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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.

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[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

ASoc: soc_core.c stream direction from snd_soc_dai
by Matt Flax 26 Feb '21

26 Feb '21

Hi there, A large number of audio codecs allow different formats for playback and capture. This becomes very useful when there are different latencies between playback and capture hardware data lines. For example digital isolation chips typically have a 1 bit delay in propagation as the bit clock rate gets faster for higher sample rates. By setting the capture and playback formats to differ by one or two bit clock cycles, the delay problem is solved. There doesn't seem to be a simple way to detect stream direction in the codec driver's set_fmt function. The snd_soc_runtime_set_dai_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-core.c#L1480 calls the snd_soc_dai_set_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-dai.c#L101 which calls the set_fmt function : https://github.com/torvalds/linux/blob/master/include/sound/soc-dai.h#L189 The snd_soc_dai_ops set_fmt function is defined as : int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); Is there a simple way to find the stream direction from a snd_soc_dai ? If the stream direction can be detected then the playback and capture formats can be set independently for the codec. It there a different way to set the playback and capture formats for the codec independently at runtime, depending on the sample rate ? Matt

5 8

[PATCH] Add basic capture support for Pioneer DJ DJM-250MK2
by František Kučera 20 Jan '21

20 Jan '21

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows recording. However, DVS is not possible yet (see the comment in code). Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 62 ++++++++++++++++++++++++++++++++++------ sound/usb/quirks.c | 1 + 3 files changed, 56 insertions(+), 8 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..f5fa11deea4d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -374,6 +374,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ ep = 0x82; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f4fb002e3ef4..c6b2f0b8d3e2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3549,14 +3549,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * Pioneer DJ DJM-250MK2 - * PCM is 8 channels out @ 48 fixed (endpoints 0x01). - * The output from computer to the mixer is usable. - * - * The input (phono or line to computer) is not working. - * It should be at endpoint 0x82 and probably also 8 channels, - * but it seems that it works only with Pioneer proprietary software. - * Even on officially supported OS, the Audacity was unable to record - * and Mixxx to recognize the control vinyls. + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + * + * Both playback and recording is working, even simultaneously. + * + * Playback channels could be mapped to: + * - CH1 + * - CH2 + * - AUX + * + * Recording channels could be mapped to: + * - Post CH1 Fader + * - Post CH2 Fader + * - Cross Fader A + * - Cross Fader B + * - MIC + * - AUX + * - REC OUT + * + * There is remaining problem with recording directly from PHONO/LINE. + * If we map a channel to: + * - CH1 Control Tone PHONO + * - CH1 Control Tone LINE + * - CH2 Control Tone PHONO + * - CH2 Control Tone LINE + * it is silent. + * There is no signal even on other operating systems with official drivers. + * The signal appears only when a supported application is started. + * This needs to be investigated yet... + * (there is quite a lot communication on the USB in both directions) + * + * In current version this mixer could be used for playback + * and for recording from vinyls (through Post CH* Fader) + * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { @@ -3580,6 +3606,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .rate_max = 48000, .nr_rates = 1, .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..6a0be7e7f4fe 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1493,6 +1493,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, set_format_emu_quirk(subs, fmt); break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ -- 2.20.1

5 9

[PATCH v11 0/7] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 14 Jan '21

14 Jan '21

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 https://lore.kernel.org/patchwork/project/lkml/list/?series=464856 changes since V10: -- Moved hdmi regmap functions from lpass-hdmi.c to lpass-cpu.c -- Moved QCOM_REGMAP_FIELD_ALLOC macro from lpass-hdmi.c to lpass.h changes since V9: -- Removed unused structures lpass_hdmi.h changes since V8: -- Removed redundant structure wrapper for reg map field memebrs -- Updated lpass_hdmi_regmap_volatile API with appropriate registers as true and others as false. changes since V7: -- Fixed typo errors -- Created Separate patch for buffer size change changes since V6: -- Removed compile time define flag, which used for enabling HDMI code, based on corresponding config param is included. -- Updated reg map alloc API with reg map bulk API. -- Removed unnecessary line splits changes since V5: -- Removed unused struct regmap *map in lpass_platform_alloc_hdmidmactl_fields. -- DMA alloc and free API signature change in lpass-apq8016.c, lpass-ipq806x.c -- Keeping API "irqreturn_t lpass_platform_hdmiif_irq" under ifdef macro Changes Since v4: -- Updated with single compatible node for both I2S and HDMI. Changes Since v3: -- Removed id in lpass variant structure and used snd_soc_dai_driver id. Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. Changes Since v1: -- Commit messages are updated -- Addressed Rob Herring review comments V Sujith Kumar Reddy (7): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi Asoc:qcom:lpass-cpu:Update dts property read API Asoc: qcom: lpass:Update lpaif_dmactl members order ASoC: qcom: Add support for lpass hdmi driver Asoc: qcom: lpass-platform : Increase buffer size ASoC: qcom: sc7180: Add support for audio over DP .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 74 ++-- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 4 +- sound/soc/qcom/lpass-cpu.c | 249 ++++++++++++- sound/soc/qcom/lpass-hdmi.c | 258 ++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 102 ++++++ sound/soc/qcom/lpass-ipq806x.c | 4 +- sound/soc/qcom/lpass-lpaif-reg.h | 49 ++- sound/soc/qcom/lpass-platform.c | 395 +++++++++++++++++---- sound/soc/qcom/lpass-sc7180.c | 116 +++++- sound/soc/qcom/lpass.h | 124 ++++++- 13 files changed, 1240 insertions(+), 143 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] ASoC: tlv320adcx140: Fix a typo in a comment
by Christophe JAILLET 01 Dec '20

01 Dec '20

It is likely that this header file is about the TLV320ADCX140. (0 and 4 swapped) While at it fix a missing "H" in a comment related to the include guard. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/tlv320adcx140.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index eedbc1d7221f..cafcbc0dd5fd 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// TLV320ADCX104 Sound driver +// TLV320ADCX140 Sound driver // Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #ifndef _TLV320ADCX140_H @@ -145,4 +145,4 @@ #define ADCX140_GPO_CFG_MAX 4 #define ADCX140_GPO_DRV_MAX 5 -#endif /* _TLV320ADCX140_ */ +#endif /* _TLV320ADCX140_H */ -- 2.25.1

2 2

[PATCH v2] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 25 Nov '20

25 Nov '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..5a0d6737de5e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,19 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining( + struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..a071485383ed 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -708,11 +708,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -720,11 +733,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining || + !stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -767,6 +794,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

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[PATCH v3 00/10] Auxiliary bus implementation and SOF multi-client support
by Dave Ertman 15 Nov '20

15 Nov '20

Brief history of Auxiliary Bus ============================== The auxiliary bus code was originally submitted upstream as virtual bus, and was submitted through the netdev tree.  This process generated up to v4.  This discussion can be found here: https://lore.kernel.org/netdev/20191111192219.30259-1-jeffrey.t.kirsher@int… At this point, GregKH requested that we take the review and revision process to an internal mailing list and garner the buy-in of a respected kernel contributor. The auxiliary bus (then known as virtual bus) was originally submitted along with implementation code for the ice driver and irdma drive, causing the complication of also having dependencies in the rdma tree. This new submission is utilizing an auxiliary bus consumer in only the sound driver tree to create the initial implementation and a single user. Since implementation work has started on this patch set, there have been multiple inquiries about the time frame of its completion. It appears that there will be numerous consumers of this functionality. The process of internal review and implementation using the sound drivers generated 19 internal versions.  The changes, including the name change from virtual bus to auxiliary bus, from these versions can be summarized as the following: - Fixed compilation and checkpatch errors - Improved documentation to address the motivation for virtual bus. - Renamed virtual bus to auxiliary bus - increased maximum device name size - Correct order in Kconfig and Makefile - removed the mid-layer adev->release layer for device unregister - pushed adev->id management to parent driver - all error paths out of ancillary_device_register return error code - all error paths out of ancillary_device_register use put_device - added adev->name element - modname in register cannot be NULL - added KBUILD_MODNAME as prefix for match_name - push adev->id responsibility to registering driver - uevent now parses adev->dev name - match_id function now parses adev->dev name - changed drivers probe function to also take an ancillary_device_id param - split ancillary_device_register into device_initialize and device_add - adjusted what is done in device_initialize and device_add - change adev to ancildev and adrv to ancildrv - change adev to ancildev in documentation ========================== Introduces the auxiliary bus implementation along with the example usage in the Sound Open Firmware(SOF) audio driver. In some subsystems, the functionality of the core device (PCI/ACPI/other) may be too complex for a single device to be managed as a monolithic block or a part of the functionality might need to be exposed to a different subsystem. Splitting the functionality into smaller orthogonal devices makes it easier to manage data, power management and domain-specific communication with the hardware. Also, common auxiliary_device functionality across primary devices can be handled by a common auxiliary_device. A key requirement for such a split is that there is no dependency on a physical bus, device, register accesses or regmap support. These individual devices split from the core cannot live on the platform bus as they are not physical devices that are controlled by DT/ACPI. The same argument applies for not using MFD in this scenario as it relies on individual function devices being physical devices that are DT enumerated. An example for this kind of requirement is the audio subsystem where a single IP handles multiple entities such as HDMI, Soundwire, local devices such as mics/speakers etc. The split for the core's functionality can be arbitrary or be defined by the DSP firmware topology and include hooks for test/debug. This allows for the audio core device to be minimal and tightly coupled with handling the hardware-specific logic and communication. The auxiliary bus is intended to be minimal, generic and avoid domain-specific assumptions. Each auxiliary bus device represents a part of its parent functionality. The generic behavior can be extended and specialized as needed by encapsulating an auxiliary bus device within other domain-specific structures and the use of .ops callbacks. The SOF driver adopts the auxiliary bus for implementing the multi-client support. A client in the context of the SOF driver represents a part of the core device's functionality. It is not a physical device but rather an auxiliary device that needs to communicate with the DSP via IPCs. With multi-client support,the sound card can be separated into multiple orthogonal auxiliary devices for local devices (mic/speakers etc), HDMI, sensing, probes, debug etc. In this series, we demonstrate the usage of the auxiliary bus with the help of the IPC test client which is used for testing the serialization of IPCs when multiple clients talk to the DSP at the same time. v3 changes: rename to auxiliary bus move .c file to drivers/base/ split auxdev unregister flow into uninitialize and delete steps update kernel-doc on register functions for new unregister model update documentation with new name and unregister flow remove check for release in auxiliary bus, allow core to catch Change driver register so only probe and id_table mandatory Fix matching logic in auxillary_match_id to account for longer id->name utilize auxiliary_bus callbacks for probe, remove and shutdown add auxiliary_find_device function add code to ensure unique auxdrv name shorten initialize/uninitialize in function names to init/uninit simplify looping logic in match_id in drives/base/Kconfig changed from tristate to bool Modified signature SOF client register/unregister API Modified PM runtime enabling sequence in the SOF ipc test aux driver Removed driver.name from the aux driver Added Probes client aux driver and device registration in the SOF driver. This allows for enabling the probes functionality in the SOF firmware for audio data extraction from specific points in the audio pipeline. Without auxiliary bus, the implementation requires modifying existing 15+ machine drivers to add the probes DAI links to the sounds cards. Using the auxiliary bus allows for splitting the probes implementation from the SOF core making it easy to maintain. v2 changes: defined pr_fmt for kernel messages replaced WARN_ON calls in registration with pr_err calls adding kernel-doc function comments for device_initialize and device_add fix typo in documentation removed inaccurate line in documentation fixed formatting in drivers/bus/Makefile changed unwind path for sof_client_dev_alloc() improved comments for client list and mem freeing during client unreg removed debugfs entries in sof_ipc_test_client_drv during remove changed the signature of sof_debug_ipc_flood_test() fix a looping error in ancillary_match_id updated error value in sof_client_dev_register() mutex held while traversing client list when unregistering clients updated includes in sof-client.h Dave Ertman (1): Add auxiliary bus support Ranjani Sridharan (9): ASoC: SOF: Introduce descriptors for SOF client ASoC: SOF: Create client driver for IPC test ASoC: SOF: ops: Add ops for client registration ASoC: SOF: Intel: Define ops for client registration ASoC: SOF: Intel: Remove IPC flood test support in SOF core ASoC: SOF: sof-client: Add client APIs to access probes ops ASoC: SOF: compress: move and export sof_probe_compr_ops ASoC: SOF: Add new client driver for probes support ASoC: SOF: Intel: CNL: register probes client Documentation/driver-api/auxiliary_bus.rst | 228 ++++++++++ Documentation/driver-api/index.rst | 1 + drivers/base/Kconfig | 3 + drivers/base/Makefile | 1 + drivers/base/auxiliary.c | 267 ++++++++++++ include/linux/auxiliary_bus.h | 78 ++++ include/linux/mod_devicetable.h | 8 + scripts/mod/devicetable-offsets.c | 3 + scripts/mod/file2alias.c | 8 + sound/soc/sof/Kconfig | 47 ++- sound/soc/sof/Makefile | 10 +- sound/soc/sof/compress.c | 60 +-- sound/soc/sof/compress.h | 1 + sound/soc/sof/core.c | 18 +- sound/soc/sof/debug.c | 457 --------------------- sound/soc/sof/intel/Kconfig | 9 + sound/soc/sof/intel/Makefile | 3 + sound/soc/sof/intel/apl.c | 16 + sound/soc/sof/intel/bdw.c | 16 + sound/soc/sof/intel/byt.c | 20 + sound/soc/sof/intel/cnl.c | 32 ++ sound/soc/sof/intel/hda-dai.c | 27 -- sound/soc/sof/intel/hda.h | 6 - sound/soc/sof/intel/intel-client.c | 40 ++ sound/soc/sof/intel/intel-client.h | 26 ++ sound/soc/sof/ops.h | 14 + sound/soc/sof/pcm.c | 11 - sound/soc/sof/probe.c | 124 +++--- sound/soc/sof/probe.h | 41 +- sound/soc/sof/sof-client.c | 170 ++++++++ sound/soc/sof/sof-client.h | 91 ++++ sound/soc/sof/sof-ipc-test-client.c | 321 +++++++++++++++ sound/soc/sof/sof-priv.h | 23 +- sound/soc/sof/sof-probes-client.c | 414 +++++++++++++++++++ 34 files changed, 1968 insertions(+), 626 deletions(-) create mode 100644 Documentation/driver-api/auxiliary_bus.rst create mode 100644 drivers/base/auxiliary.c create mode 100644 include/linux/auxiliary_bus.h create mode 100644 sound/soc/sof/intel/intel-client.c create mode 100644 sound/soc/sof/intel/intel-client.h create mode 100644 sound/soc/sof/sof-client.c create mode 100644 sound/soc/sof/sof-client.h create mode 100644 sound/soc/sof/sof-ipc-test-client.c create mode 100644 sound/soc/sof/sof-probes-client.c -- 2.26.2

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[PATCH v1 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 10 Nov '20

10 Nov '20

Note: - This patch depends on this patch series ASoC: qcom: dt-bindings: Add sc7180 machine bindings https://patchwork.kernel.org/patch/11773221/ ASoC: qcom: sc7180: Add machine driver for sound card registration https://patchwork.kernel.org/patch/11773223/ - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> Thanks for the review! Ajye Huang (2): ASoC: qcom: dt-bindings: Modify sc7180 machine bindings ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/qcom,sc7180.yaml | 64 ++++++++++++++++++- sound/soc/qcom/sc7180.c | 60 +++++++++++++++++ 2 files changed, 123 insertions(+), 1 deletion(-) -- 2.25.1

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