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[PATCH] ALSA: usb-audio: Work around for XRUN with low latency playback
by Takashi Iwai 27 Aug '21

27 Aug '21

The recent change for low latency playback works in most of test cases but it turned out still to hit errors on some use cases, most notably with JACK with small buffer sizes. This is because USB-audio driver fills up and submits full URBs at the beginning, while the URBs would return immediately and try to fill more -- that can easily trigger XRUN. It was more or less expected, but in the small buffer size, the problem became pretty obvious. Fixing this behavior properly would require the change of the fundamental driver design, so it's no trivial task, unfortunately. Instead, here we work around the problem just by switching back to the old method when the given configuration is too fragile with the low latency stream handling. As a threshold, we calculate the total buffer bytes in all plus one URBs, and check whether it's beyond the PCM buffer bytes. The one extra URB is needed because XRUN happens at the next submission after the first round. Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 4 ++++ sound/usb/pcm.c | 13 +++++++++++-- 3 files changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 6c0a052a28f9..5b19901f305a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,7 @@ struct snd_usb_endpoint { struct list_head ready_playback_urbs; /* playback URB FIFO for implicit fb */ unsigned int nurbs; /* # urbs */ + unsigned int nominal_queue_size; /* total buffer sizes in URBs */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ char *syncbuf; /* sync buffer for all sync URBs */ @@ -187,6 +188,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + bool early_playback_start; /* early start needed for playback? */ struct media_ctl *media_ctl; }; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8b1bec51c806..bf26c04cf471 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1126,6 +1126,10 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep) INIT_LIST_HEAD(&u->ready_list); } + /* total buffer bytes of all URBs plus the next queue; + * referred in pcm.c + */ + ep->nominal_queue_size = maxsize * urb_packs * (ep->nurbs + 1); return 0; out_of_memory: diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4e5031a68064..f5cbf61ac366 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -614,6 +614,14 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->period_elapsed_pending = 0; runtime->delay = 0; + /* check whether early start is needed for playback stream */ + subs->early_playback_start = + subs->direction == SNDRV_PCM_STREAM_PLAYBACK && + subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes; + + if (subs->early_playback_start) + ret = start_endpoints(subs); + unlock: snd_usb_unlock_shutdown(chip); return ret; @@ -1394,7 +1402,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->trigger_tstamp_pending_update = false; } - if (period_elapsed && !subs->running) { + if (period_elapsed && !subs->running && !subs->early_playback_start) { subs->period_elapsed_pending = 1; period_elapsed = 0; } @@ -1448,7 +1456,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea prepare_playback_urb, retire_playback_urb, subs); - if (cmd == SNDRV_PCM_TRIGGER_START) { + if (!subs->early_playback_start && + cmd == SNDRV_PCM_TRIGGER_START) { err = start_endpoints(subs); if (err < 0) { snd_usb_endpoint_set_callback(subs->data_endpoint, -- 2.26.2

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Adjust appl_ptr for multi plugin on rewind/forward operations
by GitHub pull_request - edited 27 Aug '21

27 Aug '21

alsa-project/alsa-lib pull request #161 was edited from sava-j: * When doing rewind or forward operations, application pointer needs to be properly adjusted, in the same way when doing commit/write operations. * Introduce helper functions for manipulating the application pointer. Request URL : https://github.com/alsa-project/alsa-lib/pull/161 Patch URL : https://github.com/alsa-project/alsa-lib/pull/161.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: pcm: fix divide error in snd_pcm_lib_ioctl
by Zubin Mithra 27 Aug '21

27 Aug '21

Syzkaller reported a divide error in snd_pcm_lib_ioctl. fifo_size is of type snd_pcm_uframes_t(unsigned long). If frame_size is 0x100000000, the error occurs. Fixes: a9960e6a293e ("ALSA: pcm: fix fifo_size frame calculation") Signed-off-by: Zubin Mithra <zsm(a)chromium.org> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d5883432085..e41b4e01aa37 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1746,7 +1746,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, channels = params_channels(params); frame_size = snd_pcm_format_size(format, channels); if (frame_size > 0) - params->fifo_size /= (unsigned)frame_size; + params->fifo_size /= (unsigned long)frame_size; } return 0; } -- 2.33.0.259.gc128427fd7-goog

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[PATCH 0/6] ASoC: audio-graph: cleanups
by Kuninori Morimoto 27 Aug '21

27 Aug '21

Hi Mark These patches cleanups audio-graph. This is part of prepare for new audio-graph-card2. Kuninori Morimoto (6): ASoC: audio-graph: move audio_graph_card_probe() to simple-card-utils.c ASoC: audio-graph: move audio_graph_remove() to simple-card-utils.c ASoC: audio-graph: check ports if exists ASoC: audio-graph: remove "audio-graph-card," preix support ASoC: audio-graph: remove unused "node" from graph_parse_mclk_fs() ASoC: audio-graph: remove Platform support include/sound/graph_card.h | 4 -- include/sound/simple_card_utils.h | 3 ++ sound/soc/generic/audio-graph-card.c | 52 ++++-------------------- sound/soc/generic/simple-card-utils.c | 25 ++++++++++++ sound/soc/generic/simple-card.c | 7 ---- sound/soc/tegra/tegra_audio_graph_card.c | 4 +- 6 files changed, 38 insertions(+), 57 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH] seq: event_data_ctl: fix getter API for value parameter
by GitHub pull_request - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-gobject pull request #61 was opened from takaswie: ALSASeq.EventDataCtl.get_value() is programmed carelessly to return the value in parameter field instead of value field. This commit fixes the bug. Request URL : https://github.com/alsa-project/alsa-gobject/pull/61 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/61.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH] ASoC: dwc: Get IRQ optionally
by Robin Murphy 27 Aug '21

27 Aug '21

The IRQ is explicitly optional, so use platform_get_irq_optional() and avoid platform_get_irq() logging a spurious error when trying to use the thing in DMA mode. Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 8ebf76e04702..33ce257ae198 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -642,7 +642,7 @@ static int dw_i2s_probe(struct platform_device *pdev) dev->dev = &pdev->dev; - irq = platform_get_irq(pdev, 0); + irq = platform_get_irq_optional(pdev, 0); if (irq >= 0) { ret = devm_request_irq(&pdev->dev, irq, i2s_irq_handler, 0, pdev->name, dev); -- 2.25.1

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[PATCH 1/2] ASoC: fsl_rpmsg: add soc specific data structure
by Shengjiu Wang 27 Aug '21

27 Aug '21

Each platform has different supported rates and formats, so add soc specific data for each platform. This soc specific data is attached with compatible string. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_rpmsg.c | 47 +++++++++++++++++++++++++++++++++++---- sound/soc/fsl/fsl_rpmsg.h | 12 ++++++++++ 2 files changed, 55 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index d60f4dac6c1b..bda6cc96071d 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -138,11 +138,42 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", }; +static const struct fsl_rpmsg_soc_data imx7ulp_data = { + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mm_data = { + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U8 | + SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mn_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mp_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + static const struct of_device_id fsl_rpmsg_ids[] = { - { .compatible = "fsl,imx7ulp-rpmsg-audio"}, - { .compatible = "fsl,imx8mm-rpmsg-audio"}, - { .compatible = "fsl,imx8mn-rpmsg-audio"}, - { .compatible = "fsl,imx8mp-rpmsg-audio"}, + { .compatible = "fsl,imx7ulp-rpmsg-audio", .data = &imx7ulp_data}, + { .compatible = "fsl,imx8mm-rpmsg-audio", .data = &imx8mm_data}, + { .compatible = "fsl,imx8mn-rpmsg-audio", .data = &imx8mn_data}, + { .compatible = "fsl,imx8mp-rpmsg-audio", .data = &imx8mp_data}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); @@ -157,6 +188,14 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) if (!rpmsg) return -ENOMEM; + rpmsg->soc_data = of_device_get_match_data(&pdev->dev); + if (rpmsg->soc_data) { + fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; + fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + } + if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; rpmsg->buffer_size = LPA_LARGE_BUFFER_SIZE; diff --git a/sound/soc/fsl/fsl_rpmsg.h b/sound/soc/fsl/fsl_rpmsg.h index 4f5b49eb18d8..b04086fbf828 100644 --- a/sound/soc/fsl/fsl_rpmsg.h +++ b/sound/soc/fsl/fsl_rpmsg.h @@ -6,6 +6,16 @@ #ifndef __FSL_RPMSG_H #define __FSL_RPMSG_H +/* + * struct fsl_rpmsg_soc_data + * @rates: supported rates + * @formats: supported formats + */ +struct fsl_rpmsg_soc_data { + int rates; + u64 formats; +}; + /* * struct fsl_rpmsg - rpmsg private data * @@ -15,6 +25,7 @@ * @pll8k: parent clock for multiple of 8kHz frequency * @pll11k: parent clock for multiple of 11kHz frequency * @card_pdev: Platform_device pointer to register a sound card + * @soc_data: soc specific data * @mclk_streams: Active streams that are using baudclk * @force_lpa: force enable low power audio routine if condition satisfy * @enable_lpa: enable low power audio routine according to dts setting @@ -27,6 +38,7 @@ struct fsl_rpmsg { struct clk *pll8k; struct clk *pll11k; struct platform_device *card_pdev; + const struct fsl_rpmsg_soc_data *soc_data; unsigned int mclk_streams; int force_lpa; int enable_lpa; -- 2.17.1

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No cards found in 1.2.5
by GitHub issues - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-ucm-conf issue #111 was opened from Tooniis: On 1.2.4: ``` $ alsaucm listcards 0: HDA-Intel 1: SOF 2: bdw-rt5677 3: broadwell-rt286 4: broxton-rt298 5: bytcht-cx2072x 6: bytcht-es8316 7: bytcr-rt5640 8: bytcr-rt5651 9: cht-bsw-rt5672 10: chtmax98090 11: chtnau8824 chtnau8824 internal card 12: chtrt5645 Intel SoC Audio Device 13: chtrt5650 Intel SoC Audio Device 14: hda-dsp 15: kblrt5660 16: skylake-rt286 17: sof-hda-dsp 18: sof-soundwire ``` On 1.2.5: ``` $ alsaucm listcards list is empty ``` My card's config (snd-soc-apq8096, using DB820c config) isn't being loaded on 1.2.5 as a result and I get no sound. Note that in both cases above, alsa-ucm-conf and alsa-lib are of the same version; as mixing versions causes some other errors (lack of conf.virt.d on alsa-lib 1.2.5 alsa-ucm-conf 1.2.4, and invalid syntax 4 on alsa-lib 1.2.4 with alsa-ucm-conf 1.2.5) Using Arch Linux ARM. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/111 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v3 0/14] Patches to update for rockchip i2s
by Sugar Zhang 27 Aug '21

27 Aug '21

These patches fixup or update for rockchip i2s. Changes in v3: - Drop property 'rockchip,playback-only', 'rockchip,capture-only'. Implement it by 'dma-names' of DT instead. Changes in v2: - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - drop change-id Sugar Zhang (12): ASoC: rockchip: i2s: Add support for set bclk ratio ASoC: rockchip: i2s: Fixup clk div error ASoC: rockchip: i2s: Improve dma data transfer efficiency ASoC: rockchip: i2s: Fix regmap_ops hang ASoC: rockchip: i2s: Fix concurrency between tx/rx ASoC: rockchip: i2s: Reset the controller if soft reset failed ASoC: dt-bindings: rockchip: Document reset property for i2s ASoC: rockchip: i2s: Make playback/capture optional ASoC: rockchip: i2s: Add compatible for more SoCs ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs ASoC: rockchip: i2s: Add support for frame inversion ASoC: dt-bindings: rockchip: i2s: Document property TRCM Xiaotan Luo (1): ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B Xing Zheng (1): ASoC: rockchip: i2s: Add support for TRCM property .../devicetree/bindings/sound/rockchip-i2s.yaml | 19 ++ sound/soc/rockchip/rockchip_i2s.c | 278 +++++++++++++++------ sound/soc/rockchip/rockchip_i2s.h | 10 +- 3 files changed, 224 insertions(+), 83 deletions(-) -- 2.7.4

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[PATCH v1 0/7] Patches to update for rockchip pdm
by Sugar Zhang 27 Aug '21

27 Aug '21

These patches fixup or update for rockchip pdm. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 --------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 114 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 ++++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 226 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

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