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[PATCH] ASoC: Intel: sof_rt5682: Add support for max98360a speaker amp
by Brent Lu 02 Oct '21

02 Oct '21

From: Malik_Hsu <malik_hsu(a)wistron.corp-partner.google.com> Add a board config adl_mx98360a_rt5682 to support alc5682 headset codec and max98360a speaker amplifier. Follow Intel BT offload design by connecting alc5682 to SSP0 and max98360a to SSP1. Signed-off-by: Malik_Hsu <malik_hsu(a)wistron.corp-partner.google.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 11 +++++++++++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 13 +++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ad42d4c7ade5..613662eedd0d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1076,6 +1076,17 @@ static const struct platform_device_id board_ids[] = { SOF_MAX98360A_SPEAKER_AMP_PRESENT | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "adl_mx98360_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index e4ff280eac23..f5b21a95d222 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98360a_amp = { + .num_codecs = 1, + .codecs = {"MX98360A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_mx98360_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

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[PATCH v3 00/12] Add support for on demand pipeline setup/destroy
by Peter Ujfalusi 02 Oct '21

02 Oct '21

Hi, The previous, v2 of this series was sent by Daniel Baluta: https://lore.kernel.org/alsa-devel/20210917143659.401102-1-daniel.baluta@os… We have agreed that it might be better that someone from Intel is going to take it from here as we already have the infrastructure up to test and verify the dynamic pipelines support. Changes since v2 (sent by Daniel Baluta): - patch 10: Fix NULL point dereference in hda_dai_update_config() - I have kept Daniel's SoB for the series. Changes since v1: - Signed-off-by tag added by Daniel This series implements initial support for dynamic pipelines to setup/teardown pipeline as needed when a PCM is open/closed. Initially dynamic pipelines are only supported with single core setup which will be expanded with a follow-up series. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 The feature has been merged on 1st of April to sof-dev, all issues found since has been fixed and squashed to this upstream series. Regards, Peter --- Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 174 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1032 insertions(+), 538 deletions(-) -- 2.33.0

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[PATCH 1/2] ASoC: rt5682s: Enable ASRC auto-disable to fix pop during jack plug-in while playback
by derek.fang＠realtek.com 02 Oct '21

02 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> When codec's ASRC was enabled, the DA and AD filters use the reference clk which tracks the I2S clks. And if the I2S clks' timing of open and close are not expected, this patch allows the filters to switch to use sysclk if ASRC's ref clks disappeared and could fix the below possible issues: 1. Avoid DA filter to keep surplus samples. 2. Avoid that AD filter works failed during dapm's power on. For example, if I2S clks were closed before dacdat during playback off due to jack unplug, it causes ref clks disappeared and DA filter remained some samples which will produce pop noise on the next HP playback. ASRC auto-disable could clear the samples during the playback off. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 3 +++ sound/soc/codecs/rt5682s.h | 13 +++++++++++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 122adeb7beb1..5ae54a51ff9e 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1061,6 +1061,9 @@ int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); } + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_11, + RT5682S_ASRCIN_AUTO_CLKOUT_MASK, RT5682S_ASRCIN_AUTO_CLKOUT_EN); + return 0; } EXPORT_SYMBOL_GPL(rt5682s_sel_asrc_clk_src); diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 55f1febb81e9..59ba4ea3b062 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -994,6 +994,19 @@ #define RT5682S_ASRCIN_FTK_M2_MASK (0x7 << 4) #define RT5682S_ASRCIN_FTK_M2_SFT 4 +/* ASRC Control 11 (0x008c) */ +#define RT5682S_ASRCIN_AUTO_CLKOUT_MASK (0x1 << 5) +#define RT5682S_ASRCIN_AUTO_CLKOUT_EN (0x1 << 5) +#define RT5682S_ASRCIN_AUTO_CLKOUT_DIS (0x0 << 5) +#define RT5682S_ASRCIN_AUTO_RST_MASK (0x1 << 4) +#define RT5682S_ASRCIN_AUTO_RST_EN (0x1 << 4) +#define RT5682S_ASRCIN_AUTO_RST_DIS (0x0 << 4) +#define RT5682S_SEL_LRCK_DET_MASK (0x3) +#define RT5682S_SEL_LRCK_DET_DIV8 (0x3) +#define RT5682S_SEL_LRCK_DET_DIV4 (0x2) +#define RT5682S_SEL_LRCK_DET_DIV2 (0x1) +#define RT5682S_SEL_LRCK_DET_DIV1 (0x0) + /* Depop Mode Control 1 (0x008e) */ #define RT5682S_OUT_HP_L_EN (0x1 << 6) #define RT5682S_OUT_HP_R_EN (0x1 << 5) -- 2.17.1

2 2

[PATCH] ASoC: tegra: Constify static snd_soc_ops
by Rikard Falkeborn 02 Oct '21

02 Oct '21

The struct tegra_machine_snd_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/tegra/tegra_asoc_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index 735909310a26..3cbe6ef1cf9f 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -313,7 +313,7 @@ static int tegra_machine_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_machine_snd_ops = { +static const struct snd_soc_ops tegra_machine_snd_ops = { .hw_params = tegra_machine_hw_params, }; -- 2.33.0

2 1

[PATCH] ASoC: qcom: apq8096: Constify static snd_soc_ops
by Rikard Falkeborn 02 Oct '21

02 Oct '21

The struct iapq8096_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/qcom/apq8096.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1a69baefc5ce..c7b7d0864d1a 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -60,7 +60,7 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops apq8096_ops = { +static const struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, }; -- 2.33.0

2 1

[PATCH] ASoC: mediatek: mt8195: move of_node_put to remove function
by Trevor Wu 02 Oct '21

02 Oct '21

platforms->of_node and codes->of_node are assigned in probe function, and of_node_put is called at the end of probe function, because of_node seems to be not used after probe functon In this patch, of_node_put is moved to platform remove function in case of_node is used at any occasions after probe function in the future. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 69 +++++++++++-------- 1 file changed, 42 insertions(+), 27 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index cb15467e5fc5..95abaadcd842 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -26,6 +26,9 @@ #define RT5682_DEV0_NAME "rt5682.2-001a" struct mt8195_mt6359_rt1019_rt5682_priv { + struct device_node *platform_node; + struct device_node *hdmi_node; + struct device_node *dp_node; struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; @@ -994,33 +997,36 @@ static struct snd_soc_card mt8195_mt6359_rt1019_rt5682_soc_card = { static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8195_mt6359_rt1019_rt5682_soc_card; - struct device_node *platform_node; - struct device_node *dp_node; - struct device_node *hdmi_node; struct snd_soc_dai_link *dai_link; - struct mt8195_mt6359_rt1019_rt5682_priv *priv = NULL; + struct mt8195_mt6359_rt1019_rt5682_priv *priv; int ret, i; card->dev = &pdev->dev; - platform_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,platform", 0); - if (!platform_node) { + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!priv->platform_node) { dev_dbg(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { if (!dai_link->platforms->name) - dai_link->platforms->of_node = platform_node; + dai_link->platforms->of_node = priv->platform_node; if (strcmp(dai_link->name, "DPTX_BE") == 0) { - dp_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,dptx-codec", 0); - if (!dp_node) { + priv->dp_node = + of_parse_phandle(pdev->dev.of_node, + "mediatek,dptx-codec", 0); + + if (!priv->dp_node) { dev_dbg(&pdev->dev, "No property 'dptx-codec'\n"); } else { - dai_link->codecs->of_node = dp_node; + dai_link->codecs->of_node = priv->dp_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_dptx_codec_init; @@ -1028,12 +1034,13 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - hdmi_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,hdmi-codec", 0); - if (!hdmi_node) { + priv->hdmi_node = + of_parse_phandle(pdev->dev.of_node, + "mediatek,hdmi-codec", 0); + if (!priv->hdmi_node) { dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n"); } else { - dai_link->codecs->of_node = hdmi_node; + dai_link->codecs->of_node = priv->hdmi_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_hdmi_codec_init; @@ -1041,26 +1048,33 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } } - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - ret = -ENOMEM; - goto out; - } - snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret) { dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(priv->hdmi_node); + of_node_put(priv->dp_node); + of_node_put(priv->platform_node); + } -out: - of_node_put(hdmi_node); - of_node_put(dp_node); - of_node_put(platform_node); return ret; } +static int mt8195_mt6359_rt1019_rt5682_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct mt8195_mt6359_rt1019_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + + of_node_put(priv->hdmi_node); + of_node_put(priv->dp_node); + of_node_put(priv->platform_node); + + return 0; +} + #ifdef CONFIG_OF static const struct of_device_id mt8195_mt6359_rt1019_rt5682_dt_match[] = { {.compatible = "mediatek,mt8195_mt6359_rt1019_rt5682",}, @@ -1082,6 +1096,7 @@ static struct platform_driver mt8195_mt6359_rt1019_rt5682_driver = { .pm = &mt8195_mt6359_rt1019_rt5682_pm_ops, }, .probe = mt8195_mt6359_rt1019_rt5682_dev_probe, + .remove = mt8195_mt6359_rt1019_rt5682_dev_remove, }; module_platform_driver(mt8195_mt6359_rt1019_rt5682_driver); -- 2.18.0

2 1

[PATCH] ASoC: SOF: Intel: hda-stream: limit PROCEN workaround
by Peter Ujfalusi 01 Oct '21

01 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The work-around enabled in hda-stream.c is only required on earlier versions of SOCs/PCH (Skylake, KabyLake, ApolloLake, GeminiLake). Before setting the format on the host DMA, it is required to couple the host and link DMA - which as a consequence shall use the same format. This patch introduces a quirk field in the platform descriptor and makes the work-around conditional. Newer platforms have no limitations on the use of host and link DMA, which can use different formats. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/hda-stream.c | 18 ++++++++++++------ sound/soc/sof/intel/shim.h | 4 ++++ 3 files changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index e6a1f6532547..16b7d8b1adff 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -143,5 +143,6 @@ const struct sof_intel_dsp_desc apl_chip_info = { .rom_init_timeout = 150, .ssp_count = APL_SSP_COUNT, .ssp_base_offset = APL_SSP_BASE_OFFSET, + .quirks = SOF_INTEL_PROCEN_FMT_QUIRK, }; EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..47a465b06da5 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -405,6 +405,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params) { + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *hstream = &stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); @@ -538,6 +539,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, /* * Recommended hardware programming sequence for HDAudio DMA format + * on earlier platforms - this is not needed on newer platforms * * 1. Put DMA into coupled mode by clearing PPCTL.PROCEN bit * for corresponding stream index before the time of writing @@ -547,9 +549,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, * enable decoupled mode */ - /* couple host and link DMA, disable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, 0); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* couple host and link DMA, disable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, 0); + } /* program stream format */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -557,9 +561,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, SOF_HDA_ADSP_REG_CL_SD_FORMAT, 0xffff, hstream->format_val); - /* decouple host and link DMA, enable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, mask); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* decouple host and link DMA, enable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, mask); + } /* program last valid index */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index e9f7d4d7fcce..8e2a0f4c0f66 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -151,6 +151,9 @@ #define PCI_PMCS 0x84 #define PCI_PMCS_PS_MASK 0x3 +/* Intel quirks */ +#define SOF_INTEL_PROCEN_FMT_QUIRK BIT(0) + /* DSP hardware descriptor */ struct sof_intel_dsp_desc { int cores_num; @@ -166,6 +169,7 @@ struct sof_intel_dsp_desc { int ssp_base_offset; /* base address of the SSPs */ u32 sdw_shim_base; u32 sdw_alh_base; + u32 quirks; bool (*check_sdw_irq)(struct snd_sof_dev *sdev); }; -- 2.33.0

3 4

[PATCH][next] ALSA: usb-audio: Fix sum of uninitialized variable sample_accum
by Colin King 01 Oct '21

01 Oct '21

From: Colin Ian King <colin.king(a)canonical.com> Variable sample_accum is not being intialized and then has ep->sample_rem added to it, leading to a bogus value. One solution is to initialize it to zero at declaration time, but it is probably best to just assign it to ep->sample_rem on first use. Addresses-Coveriry: ("Uninitialized scalar variable") Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 42c0d2db8ba8..c6a33732db3f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -182,7 +182,7 @@ static int next_packet_size(struct snd_usb_endpoint *ep, unsigned int avail) if (ep->fill_max) return ep->maxframesize; - sample_accum += ep->sample_rem; + sample_accum = ep->sample_rem; if (sample_accum >= ep->pps) { sample_accum -= ep->pps; ret = ep->packsize[1]; -- 2.32.0

3 2

[PATCH] ALSA: usb-audio: Fix packet size calculation regression
by Takashi Iwai 01 Oct '21

01 Oct '21

The commit d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") introduced the available frame size check, but the conversion forgot to initialize the temporary variable properly, and it resulted in a bogus calculation. This patch fixes it. Fixes: d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") Reported-by: Colin Ian King <colin.king(a)canonical.com> Link: https://lore.kernel.org/r/20211001104417.14291-1-colin.king@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 42c0d2db8ba8..743b8287cfcd 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -182,7 +182,7 @@ static int next_packet_size(struct snd_usb_endpoint *ep, unsigned int avail) if (ep->fill_max) return ep->maxframesize; - sample_accum += ep->sample_rem; + sample_accum = ep->sample_accum + ep->sample_rem; if (sample_accum >= ep->pps) { sample_accum -= ep->pps; ret = ep->packsize[1]; -- 2.26.2

1 0

ALSA kernel projects - for academic purposes
by Muni Sekhar 01 Oct '21

01 Oct '21

Hi all, What small projects would you suggest to a novice with the ALSA kernel. The aim is to develop a familiarity with the ALSA kernel source code, and also to submit it for academic purposes. -- Thanks, Sekhar

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