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

[PATCH] ALSA: firewire-motu: fix truncated bytes in message tracepoints
by Takashi Sakamoto 21 Sep '21

21 Sep '21

In MOTU protocol v2/v3, first two data chunks across 2nd and 3rd data channels includes message bytes from device. The total size of message is 48 bits per data block. The 'data_block_message' tracepoints event produced by ALSA firewire-motu driver exposes the sequence of messages to userspace in 64 bit storage, however lower 32 bits are actually available since current implementation truncates 16 bits in upper of the message as a result of bit shift operation within 32 bit storage. This commit fixes the bug by perform the bit shift in 64 bit storage. Fixes: c6b0b9e65f09 ("ALSA: firewire-motu: add tracepoints for messages for unique protocol") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 5388b85fb60e..a18c2c033e83 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, /* This is just for v2/v3 protocol. */ for (i = 0; i < data_blocks; ++i) { - *frames = (be32_to_cpu(buffer[1]) << 16) | - (be32_to_cpu(buffer[2]) >> 16); + *frames = be32_to_cpu(buffer[1]); + *frames <<= 16; + *frames |= be32_to_cpu(buffer[2]) >> 16; + ++frames; buffer += data_block_quadlets; - frames++; } } -- 2.30.2

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[GIT PULL] ASoC fixes for v5.15-rc2
by Mark Brown 21 Sep '21

21 Sep '21

The following changes since commit fcb958ee8e832e9cdf43408535207e15f14af755: ASoC: rockchip: i2s: Fix concurrency between tx/rx (2021-09-03 16:19:01 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc2 for you to fetch changes up to cfacfefd382af3b42905108b54f02820dca225c4: ASoC: SOF: trace: Omit error print when waking up trace sleepers (2021-09-17 13:16:36 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A crop of mostly device specific fixes that have been applied since the merge window, nothing particularly standout here. ---------------------------------------------------------------- Arnaud Pouliquen (1): MAINTAINERS: fix update references to stm32 audio bindings Marc Herbert (1): ASoC: SOF: loader: release_firmware() on load failure to avoid batching Mark Brown (1): Merge series "ASoC: fsl: register platform component before registering cpu dai" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Peter Ujfalusi (4): ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication ASoC: SOF: trace: Omit error print when waking up trace sleepers Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic Shengjiu Wang (5): ASoC: fsl_sai: register platform component before registering cpu dai ASoC: fsl_esai: register platform component before registering cpu dai ASoC: fsl_micfil: register platform component before registering cpu dai ASoC: fsl_spdif: register platform component before registering cpu dai ASoC: fsl_xcvr: register platform component before registering cpu dai Trevor Wu (3): ASoC: mediatek: add required config dependency ASoC: mediatek: common: handle NULL case in suspend/resume function ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX Yong Zhi (1): ASoC: SOF: Fix DSP oops stack dump output contents MAINTAINERS | 3 ++- sound/soc/fsl/fsl_esai.c | 16 ++++++++++------ sound/soc/fsl/fsl_micfil.c | 15 ++++++++++----- sound/soc/fsl/fsl_sai.c | 14 +++++++++----- sound/soc/fsl/fsl_spdif.c | 14 +++++++++----- sound/soc/fsl/fsl_xcvr.c | 15 ++++++++++----- sound/soc/intel/boards/sof_sdw.c | 5 +++++ sound/soc/mediatek/Kconfig | 3 +++ sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- .../soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 7 +++---- sound/soc/sof/core.c | 4 +--- sound/soc/sof/imx/imx8.c | 9 ++++++++- sound/soc/sof/imx/imx8m.c | 9 ++++++++- sound/soc/sof/loader.c | 8 +++++--- sound/soc/sof/trace.c | 1 - sound/soc/sof/xtensa/core.c | 4 ++-- 16 files changed, 96 insertions(+), 50 deletions(-)

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Re: [PATCH] ALSA: usx2y: Prefer struct_size over open coded arithmetic
by Takashi Iwai 21 Sep '21

21 Sep '21

On Sun, 19 Sep 2021 15:37:27 +0200, Len Baker wrote: > > As noted in the "Deprecated Interfaces, Language Features, Attributes, > and Conventions" documentation [1], size calculations (especially > multiplication) should not be performed in memory allocator (or similar) > function arguments due to the risk of them overflowing. This could lead > to values wrapping around and a smaller allocation being made than the > caller was expecting. Using those allocations could lead to linear > overflows of heap memory and other misbehaviors. > > In this case this is not actually dynamic size: all the operands > involved in the calculation are constant values. However it is better to > refactor this anyway, just to keep the open-coded math idiom out of > code. > > So, use the struct_size() helper to do the arithmetic instead of the > argument "size + size * count" in the kzalloc() function. > > Also, take the opportunity to refactor the declaration variables to make > it more easy to read. > > [1] https://www.kernel.org/doc/html/latest/process/deprecated.html#open-coded-a… > > Signed-off-by: Len Baker <len.baker(a)gmx.com> Thanks, applied. Takashi

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[PATCH] ASoC: fsl_spdif: Add support for i.MX8ULP
by Shengjiu Wang 21 Sep '21

21 Sep '21

On i.MX8ULP the spdif works with EDMA, so add compatible string and soc specific data for i.MX8ULP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8ffb1a6048d6..61ba08322f23 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -186,6 +186,16 @@ static struct fsl_spdif_soc_data fsl_spdif_imx8mm = { .tx_formats = FSL_SPDIF_FORMATS_PLAYBACK, }; +static struct fsl_spdif_soc_data fsl_spdif_imx8ulp = { + .imx = true, + .shared_root_clock = true, + .raw_capture_mode = false, + .interrupts = 1, + .tx_burst = 2, /* Applied for EDMA */ + .rx_burst = 2, /* Applied for EDMA */ + .tx_formats = SNDRV_PCM_FMTBIT_S24_LE, /* Applied for EDMA */ +}; + /* Check if clk is a root clock that does not share clock source with others */ static inline bool fsl_spdif_can_set_clk_rate(struct fsl_spdif_priv *spdif, int clk) { @@ -1556,6 +1566,7 @@ static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx6sx-spdif", .data = &fsl_spdif_imx6sx, }, { .compatible = "fsl,imx8qm-spdif", .data = &fsl_spdif_imx8qm, }, { .compatible = "fsl,imx8mm-spdif", .data = &fsl_spdif_imx8mm, }, + { .compatible = "fsl,imx8ulp-spdif", .data = &fsl_spdif_imx8ulp, }, {} }; MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); -- 2.17.1

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[PATCH] ASoC: fsl_xcvr: Fix channel swap issue with ARC
by Shengjiu Wang 21 Sep '21

21 Sep '21

With pause and resume test for ARC, there is occasionally channel swap issue. The reason is that currently driver set the DPATH out of reset first, then start the DMA, the first data got from FIFO may not be the Left channel. Moving DPATH out of reset operation after the dma enablement to fix this issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 31c5ee641fe7..6e67033b6cde 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -487,8 +487,9 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return ret; } - /* clear DPATH RESET */ + /* set DPATH RESET */ m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); + v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -590,10 +591,6 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, val |= FSL_XCVR_EXT_CTRL_CMDC_RESET(tx); } - /* set DPATH RESET */ - mask |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - val |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, mask, val); if (ret < 0) { dev_err(dai->dev, "Err setting DPATH RESET: %d\n", ret); @@ -643,6 +640,16 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(dai->dev, "Failed to enable DMA: %d\n", ret); return ret; } + + /* clear DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear DPATH RESET: %d\n", ret); + return ret; + } + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: -- 2.17.1

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[PATCH] ASoC: qdsp6: q6afe-dai: Fix spelling mistake "Fronend" -> "Frontend"
by Colin King 21 Sep '21

21 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the module description. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index ac8f7324e94b..8b664cbf6fa6 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1747,5 +1747,5 @@ static struct platform_driver q6afe_dai_platform_driver = { }; module_platform_driver(q6afe_dai_platform_driver); -MODULE_DESCRIPTION("Q6 Audio Fronend dai driver"); +MODULE_DESCRIPTION("Q6 Audio Frontend dai driver"); MODULE_LICENSE("GPL v2"); -- 2.32.0

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[PATCH] ASoC: fsl: Constify static snd_soc_ops
by Rikard Falkeborn 21 Sep '21

21 Sep '21

These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-card.c | 4 ++-- sound/soc/fsl/imx-hdmi.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index a364e2415de0..0d637929bfef 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -132,12 +132,12 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops imx_audmix_fe_ops = { +static const struct snd_soc_ops imx_audmix_fe_ops = { .startup = imx_audmix_fe_startup, .hw_params = imx_audmix_fe_hw_params, }; -static struct snd_soc_ops imx_audmix_be_ops = { +static const struct snd_soc_ops imx_audmix_be_ops = { .hw_params = imx_audmix_be_hw_params, }; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 58fd0639a069..05dff2dc1d19 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -442,12 +442,12 @@ static int imx_aif_startup(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops imx_aif_ops = { +static const struct snd_soc_ops imx_aif_ops = { .hw_params = imx_aif_hw_params, .startup = imx_aif_startup, }; -static struct snd_soc_ops imx_aif_ops_be = { +static const struct snd_soc_ops imx_aif_ops_be = { .hw_params = imx_aif_hw_params, }; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 34a0dceae621..2b663c39edb2 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -59,7 +59,7 @@ static int imx_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops imx_hdmi_ops = { +static const struct snd_soc_ops imx_hdmi_ops = { .hw_params = imx_hdmi_hw_params, }; -- 2.33.0

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[PATCH v3 0/3] Convert name-prefix doc to json-schema
by Sameer Pujar 21 Sep '21

21 Sep '21

Following are the changes: - Add json-schema for 'sound-name-prefix' documentation under 'name-perfix.yaml' - Use schema references wherever needed. - Remove txt based doc Changelog ========= v2 -> v3 -------- * Removed examples in patch 1/3 as suggested by Rob * Added reviewed tag from Jerome for the series * Added reviewed tag from Rob for patch 2/3 v1 -> v2 -------- * Provide top reference to name-prefix.yaml as suggested by Rob for patch 2/3 * Dropped couple of unreachable email ids from Cc list in commit message of patch 2/3 * No changes in remaining patches Sameer Pujar (3): ASoC: Add json-schema documentation for sound-name-prefix ASoC: Use schema reference for sound-name-prefix ASoC: Remove name-prefix.txt .../devicetree/bindings/sound/name-prefix.txt | 24 ---------------------- .../devicetree/bindings/sound/name-prefix.yaml | 21 +++++++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 9 +++----- .../bindings/sound/nvidia,tegra210-dmic.yaml | 9 +++----- .../bindings/sound/nvidia,tegra210-i2s.yaml | 9 +++----- .../devicetree/bindings/sound/nxp,tfa989x.yaml | 9 +++----- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- .../bindings/sound/simple-audio-mux.yaml | 9 +++----- 8 files changed, 37 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.yaml -- 2.7.4

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[PATCH 0/7] Update Lpass digital codec macro drivers
by Srinivasa Rao Mandadapu 21 Sep '21

21 Sep '21

This patch set is to add support for lpass sc7280 based targets. Upadate compatible name and change of bulk clock voting to individual clock voting in digital codec va, rx, tx macro drivers. Srinivasa Rao Mandadapu (7): ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Change mic control registers to volatile ASoC: codecs: lpass-va-macro: Change bulk voting to individual clock voting ASoC: codecs: lpass-rx-macro: Change bulk voting to individual clock voting ASoC: codecs: lpass-tx-macro: Change bulk voting to individual clock voting ASoC: codecs: lpass-va-macro: set mclk clock rate correctly .../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +- sound/soc/codecs/lpass-rx-macro.c | 69 ++++++++++++------ sound/soc/codecs/lpass-tx-macro.c | 81 ++++++++++++++++------ sound/soc/codecs/lpass-va-macro.c | 49 ++++++++----- sound/soc/codecs/lpass-wsa-macro.c | 1 + 8 files changed, 153 insertions(+), 63 deletions(-) -- 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] ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
by Jaroslav Kysela 20 Sep '21

20 Sep '21

The new framing mode causes the user space regression, because the alsa-lib code does not initialize the reserved space in the params structure when the device is opened. This change adds SNDRV_RAWMIDI_IOCTL_USER_PVERSION like we do for the PCM interface for the protocol acknowledgment. Cc: David Henningsson <coding(a)diwic.se> Cc: <stable(a)vger.kernel.org> Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/rawmidi.c | 9 +++++++++ 3 files changed, 11 insertions(+) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 989e1517332d..7a08ed2acd60 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -98,6 +98,7 @@ struct snd_rawmidi_file { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; + unsigned int user_pversion; /* supported protocol version */ }; struct snd_rawmidi_str { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1d84ec9db93b..f906e50a7919 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -784,6 +784,7 @@ struct snd_rawmidi_status { #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('A', 0x02, int) #define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) #define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) #define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e..6f30231bdb88 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) -- 2.31.1

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