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

[PATCH 0/2] ASoC: codecs: lpass: add support for v2.6 rx macro
by srinivas.kandagatla＠linaro.org 17 Jun '24

17 Jun '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support to reading codec version and also adds support for v2.6 codec version in rx macro. LPASS 2.6 has changes in some of the rx block which are required to get headset functional correctly. Tested this on X13s and x1e80100 crd. Thanks, Srini Srinivas Kandagatla (2): ASoC: codecs: lpass-macro: add helpers to get codec version ASoC: codec: lpass-rx-macro: add suppor for 2.6 codec version sound/soc/codecs/lpass-macro-common.c | 14 + sound/soc/codecs/lpass-macro-common.h | 35 ++ sound/soc/codecs/lpass-rx-macro.c | 565 +++++++++++++++++++------- sound/soc/codecs/lpass-va-macro.c | 29 ++ 4 files changed, 488 insertions(+), 155 deletions(-) -- 2.25.1

3 5

[PATCH 1/2] ASoC: soc-dai.h: Constify DAI ops auto_selectable_formats
by Krzysztof Kozlowski 17 Jun '24

17 Jun '24

The core ASoC code does not modify contents of the 'auto_selectable_formats' array passed in 'struct snd_soc_dai_ops', so make it const for code safety. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 15ef268c9845..279223c4ef5e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -361,7 +361,7 @@ struct snd_soc_dai_ops { * see * snd_soc_dai_get_fmt() */ - u64 *auto_selectable_formats; + const u64 *auto_selectable_formats; int num_auto_selectable_formats; /* probe ordering - for components with runtime dependencies */ -- 2.43.0

2 4

[PATCH 0/2] ASoC: Add ak4619 codec support
by Kuninori Morimoto 17 Jun '24

17 Jun '24

Hi Mark This patch-set adds ak4619 driver. It was created by Khanh, and I updated/adjusted to upstream. It was tested on Renesas V4M GrayH Khanh Le (1): ASoC: Add ak4619 codec support Kuninori Morimoto (1): ASoC: dt-bindings: ak4619: Add initial DT binding .../devicetree/bindings/sound/ak4619.yaml | 48 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4619.c | 903 ++++++++++++++++++ 4 files changed, 958 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4619.yaml create mode 100644 sound/soc/codecs/ak4619.c -- 2.43.0

2 4

[PATCH v2 0/5] ASoC: simple-card / audio-graph: add link-trigger-order
by Kuninori Morimoto 17 Jun '24

17 Jun '24

Hi Mark, Rob, Krzysztof Cc Kochetkov This patch-set adds link-trigger-order to Simple-Card / Audio-Graph-Card. Link: https://lore.kernel.org/r/87v82ls82e.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - merge header into Doc - update git-log Kuninori Morimoto (5): ASoC: audio-graph-port: add link-trigger-order ASoC: simple-card-utils: add link-trigger-order support ASoC: simple-audio-card: add link-trigger-order support ASoC: audio-graph-card: add link-trigger-order support ASoC: audio-graph-card2: add link-trigger-order support .../bindings/sound/audio-graph-port.yaml | 9 +++ include/dt-bindings/sound/audio-graph.h | 26 +++++++ include/sound/simple_card_utils.h | 4 ++ sound/soc/generic/audio-graph-card.c | 13 ++++ sound/soc/generic/audio-graph-card2.c | 13 ++++ sound/soc/generic/simple-card-utils.c | 71 +++++++++++++++++++ sound/soc/generic/simple-card.c | 10 +++ 7 files changed, 146 insertions(+) create mode 100644 include/dt-bindings/sound/audio-graph.h -- 2.43.0

1 5

[PATCH v5 resend 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 17 Jun '24

17 Jun '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer 2 weeks past. This is resend of v5 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

1 4

[PATCH] ASoC: fsl: imx-pcm-fiq: add missing MODULE_DESCRIPTION() macro
by Jeff Johnson 16 Jun '24

16 Jun '24

With ARCH=arm, make allmodconfig && make W=1 C=1 reports: WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/fsl/imx-pcm-fiq.o Add the missing invocation of the MODULE_DESCRIPTION() macro. Signed-off-by: Jeff Johnson <quic_jjohnson(a)quicinc.com> --- sound/soc/fsl/imx-pcm-fiq.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 0d124002678e..5ea6dd4c89a1 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -319,4 +319,5 @@ void imx_pcm_fiq_exit(struct platform_device *pdev) } EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); +MODULE_DESCRIPTION("Freescle i.MX PCM FIQ handler"); MODULE_LICENSE("GPL"); --- base-commit: 83a7eefedc9b56fe7bfeff13b6c7356688ffa670 change-id: 20240615-md-arm-sound-soc-fsl-c598fb353e69

2 2

[PATCH] slimbus: Fix struct and documentation alignment in stream.c
by Amit Vadhavana 16 Jun '24

16 Jun '24

The placement of the `segdist_codes` array documentation was corrected to conform with kernel documentation guidelines. The `@segdist_codes` was placed incorrectly within the struct `segdist_code` documentation block, which led to a potential misinterpretation of the code structure. The `segdist_codes` array documentation was moved outside the struct block, and a separate comment block was provided for it. This change ensures that clarity and proper alignment with kernel documentation standards are maintained. A kernel-doc warning was addressed: ./drivers/slimbus/stream.c:49: warning: Excess struct member 'segdist_codes' description in 'segdist_code' Signed-off-by: Amit Vadhavana <av2082000(a)gmail.com> --- drivers/slimbus/stream.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..62661211a409 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -18,7 +18,6 @@ * and the first slot of the next consecutive Segment. * @segdist_code: Segment Distribution Code SD[11:0] * @seg_offset_mask: Segment offset mask in SD[11:0] - * @segdist_codes: List of all possible Segmet Distribution codes. */ static const struct segdist_code { int ratem; @@ -26,7 +25,10 @@ static const struct segdist_code { int segdist_code; u32 seg_offset_mask; -} segdist_codes[] = { +}; + +/* segdist_codes - List of all possible Segment Distribution codes. */ +static const struct segdist_code segdist_codes[] = { {1, 1536, 0x200, 0xdff}, {2, 768, 0x100, 0xcff}, {4, 384, 0x080, 0xc7f}, -- 2.25.1

2 2

[PATCH v4 0/3] ASoC: codecs: lpass: add support for v2.5 rx macro
by Srinivas Kandagatla 15 Jun '24

15 Jun '24

This patchset adds support to reading codec version and also adds support for v2.5 codec version in rx macro. LPASS 2.5 and up versions have changes in some of the rx blocks which are required to get headset functional correctly. Tested this on SM8450, X13s and x1e80100 crd. This changes also fixes issue with sm8450, sm8550, sm8660 and x1e80100. @Neil Armstrong can you pl test it on sm8650 @Krzysztof Kozlowski can you pl test it on sm8550 Thanks, Srini Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes in v3: - moved dev_info to dev_dbg to print codec version. - added explicit cases for codec version checks instead of using range - changed err handling. - Link to v1: https://lore.kernel.org/r/20240612-lpass-codec-v25-v1-0-9f40611a1370@linaro… --- Srinivas Kandagatla (3): ASoC: codecs: lpass-macro: add helpers to get codec version ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions ASoC: codec: lpass-rx-macro: add support for 2.5 codec version sound/soc/codecs/lpass-macro-common.c | 23 ++ sound/soc/codecs/lpass-macro-common.h | 35 ++ sound/soc/codecs/lpass-rx-macro.c | 595 +++++++++++++++++++++++++--------- sound/soc/codecs/lpass-va-macro.c | 28 ++ 4 files changed, 528 insertions(+), 153 deletions(-) --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240612-lpass-codec-v25-4e960abd661f Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

3 7

[PATCH v2] ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
by Richard Fitzgerald 15 Jun '24

15 Jun '24

If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer sources should be initialized to disconnected. There aren't currently any available products using the ASP so this doesn't affect any existing systems. The cs35l56 does not have any fixed default for the mixer source registers. When the cs35l56 boots, its firmware patches these registers to setup a system-specific routing; this is so that Windows can use generic SDCA drivers instead of needing knowledge of chip-specific registers. The setup varies between end-products, which each have customized firmware, and so the default register state varies between end-products. It can also change if the firmware on an end-product is upgraded - for example if a change was needed to the routing for Windows use-cases. It must be emphasized that the settings applied by the firmware are not internal magic tuning; they are statically implementing use-case setup that on Linux would be done via ALSA controls. The driver is currently syncing the mixer controls with whatever initial state the firmware wrote to the registers, so that they report the actual audio routing. But if the ASP DAI is hooked up this can create a powered-up DAPM graph without anything intentionally setting up a path. This can lead to parts of the audio system powering up unexpectedly. For example when cs35l56 is connected to cs42l43 using a codec-codec link, this can create a complete DAPM graph which then powers-up cs42l43. But the cs42l43 can only be clocked from its SoundWire bus so this causes a bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up without a clock. If the host is taking ownership of the ASP (either directly or as a codec-to-codec link) there is no need to keep the mixer settings that the firmware wrote. The driver has ALSA controls for setting these using standard Linux mechanisms. So if the machine driver hooks up the ASP the ASP mixers are initialized to "None" (no input). This prevents unintended DAPM-graph power-ups, and means the initial state of the mixers is always going to be None. Since the initial state of the mixers can vary from system to system and potentially between firmware upgrades, no use-case manager can currently assume that cs35l56 has a known initial state. The firmware could just as easily default them to "None" as to any input source. So defaulting them to "None" in the driver is not increasing the entropy of the system. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e89027cd40d1..27869e14e9c8 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -215,6 +215,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = { REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018), + REG_SEQ0(CS35L56_ASP1TX1_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX2_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX3_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX4_INPUT, 0x00000000), }; /* -- 2.39.2

2 1

[PATCH] ASoC: q6apm-lpass-dai: close graph on prepare errors
by Srinivas Kandagatla 15 Jun '24

15 Jun '24

There is an issue around with error handling and graph management with the exising code, none of the error paths close the graph, which result in leaving the loaded graph in dsp, however the driver thinks otherwise. This can have a nasty side effect specially when we try to load the same graph to dsp, dsp returns error which leaves the board with no sound and requires restart. Fix this by properly closing the graph when we hit errors between open and close. Fixes: 30ad723b93ad ("ASoC: qdsp6: audioreach: add q6apm lpass dai support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index 68a38f63a2db..66b911b49e3f 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -141,14 +141,17 @@ static void q6apm_lpass_dai_shutdown(struct snd_pcm_substream *substream, struct struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev); int rc; - if (!dai_data->is_port_started[dai->id]) - return; - rc = q6apm_graph_stop(dai_data->graph[dai->id]); - if (rc < 0) - dev_err(dai->dev, "fail to close APM port (%d)\n", rc); + if (dai_data->is_port_started[dai->id]) { + rc = q6apm_graph_stop(dai_data->graph[dai->id]); + dai_data->is_port_started[dai->id] = false; + if (rc < 0) + dev_err(dai->dev, "fail to close APM port (%d)\n", rc); + } - q6apm_graph_close(dai_data->graph[dai->id]); - dai_data->is_port_started[dai->id] = false; + if (dai_data->graph[dai->id]) { + q6apm_graph_close(dai_data->graph[dai->id]); + dai_data->graph[dai->id] = NULL; + } } static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -163,8 +166,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s q6apm_graph_stop(dai_data->graph[dai->id]); dai_data->is_port_started[dai->id] = false; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { q6apm_graph_close(dai_data->graph[dai->id]); + dai_data->graph[dai->id] = NULL; + } } /** @@ -183,26 +188,29 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s cfg->direction = substream->stream; rc = q6apm_graph_media_format_pcm(dai_data->graph[dai->id], cfg); - if (rc) { dev_err(dai->dev, "Failed to set media format %d\n", rc); - return rc; + goto err; } rc = q6apm_graph_prepare(dai_data->graph[dai->id]); if (rc) { dev_err(dai->dev, "Failed to prepare Graph %d\n", rc); - return rc; + goto err; } rc = q6apm_graph_start(dai_data->graph[dai->id]); if (rc < 0) { dev_err(dai->dev, "fail to start APM port %x\n", dai->id); - return rc; + goto err; } dai_data->is_port_started[dai->id] = true; return 0; +err: + q6apm_graph_close(dai_data->graph[dai->id]); + dai_data->graph[dai->id] = NULL; + return rc; } static int q6apm_lpass_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240613-q6apm-fixes-6a9c84852713 Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

3 3