June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/4] soundwire: add SoundWire 1.2 spec support
by Bard Liao 30 Jun '20

30 Jun '20

This series adds basic support for SoundWire 1.2. Pierre-Louis Bossart (4): soundwire: add definitions for 1.2 spec soundwire: bus_type: convert open-coded while() to for() loop soundwire: extend SDW_SLAVE_ENTRY soundwire: bus: initialize bus clock base and scale registers drivers/soundwire/bus.c | 107 ++++++++++++++++++++++ drivers/soundwire/bus_type.c | 19 ++-- include/linux/mod_devicetable.h | 2 + include/linux/soundwire/sdw.h | 11 ++- include/linux/soundwire/sdw_registers.h | 117 +++++++++++++++++++++++- scripts/mod/devicetable-offsets.c | 2 + scripts/mod/file2alias.c | 6 +- 7 files changed, 250 insertions(+), 14 deletions(-) -- 2.17.1

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Re: XHCI vs PCM2903B/PCM2904 part 2
by Mathias Nyman 30 Jun '20

30 Jun '20

On 30.6.2020 16.08, Rik van Riel wrote: > I misread the code, it's not a bitfield, so state 1 means an endpoint marked with running state. The next urb is never getting a response, though. > > However, the xhci spec says an endpoint is halted upon a babble error. I was looking at the same, so according to specs this state shouldn't be possible. > > The code right above the babble handling case adds halted into the endpoint state itself. Does the code handling the babble error need to do something similar to trigger cleanup elsewhere? It's a flag to prevent ringing the doorbell for a halted endpoint. Anyway, reset endpoint is meant to recover an endpoint in a halted state. Resetting non-halted endpoints will just lead to a context state error, and besides, isoc endpoints shouldn't halt. Anyways, I haven't got any better idea at the moment. You can try and see what a forced reset does with: diff --git a/drivers/usb/host/xhci-ring.c b/drivers/usb/host/xhci-ring.c index 2c255d0620b0..d79aca0df6d4 100644 --- a/drivers/usb/host/xhci-ring.c +++ b/drivers/usb/host/xhci-ring.c @@ -1900,8 +1900,7 @@ static int xhci_requires_manual_halt_cleanup(struct xhci_hcd *xhci, * endpoint anyway. Check if a babble halted the * endpoint. */ - if (GET_EP_CTX_STATE(ep_ctx) == EP_STATE_HALTED) - return 1; + return 1; return 0; } Traces also showed thet endpoint doorbell was rang after th babble error, so we know that didn't help restarting the endpoint. -Mathias

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[PATCH 1/2] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 30 Jun '20

30 Jun '20

Just convert Rockchip rk3328 audio codec binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 --------- .../bindings/sound/rockchip,rk3328-codec.yaml | 61 +++++++++++++++++++ 2 files changed, 61 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..08a74e28e483 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + + clock-names: + items: + - const: pclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + +examples: + - | + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = 100; + }; -- 2.27.0

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[PATCH] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 30 Jun '20

30 Jun '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 58 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 56 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..fcafc8ecb131 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,59 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + bool clk_sel[2]; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + if (i == ASRC_CLK_MAP_LEN) { + select_clk[j] = OUTCLK_ASRCK1_CLK; + clk_sel[j] = false; + } else { + select_clk[j] = i; + clk_sel[j] = true; + } + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (!clk_sel[IN] || !clk_sel[OUT]) + select_clk[IN] = INCLK_NONE; + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +653,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +666,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

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ucm2: Fixes for moving DragonBoard UCM files.
by GitHub pull_request - opened 30 Jun '20

30 Jun '20

alsa-project/alsa-ucm-conf pull request #37 was opened from Srinivas-Kandagatla: Hi Jaroslav, Nico just noticed that you have moved and renamed all the DragonBoard ucm2 files. We have below few issues with that patches: 1> DB820c belongs to APQ8096 SoC, so it should be moved to ucm2/apq8096 folder rather than apq8016. 2> DB410c belongs to APQ8016 SoC, so it should be under ucm2/apq8016 rather than apq8016-sbc. 3> All the board specific ucm conf file names with actual board file are now replaced with SoC names, like DB845c is replace with sdm845. Any particular reason for doing this? Are you suggesting us to use some matching on card long names to include correct board files from top conf file? Can you provide us some pointers on how board specific config files should endup looking like? This PR fixes (1), and (2). Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/37 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/37.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] asoc: Update supported rate and format for dummy dai
by Rohit kumar 29 Jun '20

29 Jun '20

Add support for 384KHz sample rate and S24_3LE bitwidth for dummy dai. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-utils.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 922eac9..364b248 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -86,12 +86,13 @@ static const struct snd_soc_component_driver dummy_codec = { .non_legacy_dai_naming = 1, }; -#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_RATES SNDRV_PCM_RATE_8000_384000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_U16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ SNDRV_PCM_FMTBIT_U24_LE | \ SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ -- 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 -next] ASoC: Documentation: fix reference to renamed source file
by Randy Dunlap 29 Jun '20

29 Jun '20

From: Randy Dunlap <rdunlap(a)infradead.org> sound/soc/soc-io.c was merged into sound/soc/soc-component.c, so fixup the Documentation to use the updated file name. Error: Cannot open file ../sound/soc/soc-io.c WARNING: kernel-doc '../scripts/kernel-doc -rst -enable-lineno ../sound/soc/soc-io.c' failed with return code 1 Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/kernel-api/alsa-driver-api.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20200626.orig/Documentation/sound/kernel-api/alsa-driver-api.rst +++ linux-next-20200626/Documentation/sound/kernel-api/alsa-driver-api.rst @@ -99,7 +99,7 @@ ASoC Core API .. kernel-doc:: include/sound/soc.h .. kernel-doc:: sound/soc/soc-core.c .. kernel-doc:: sound/soc/soc-devres.c -.. kernel-doc:: sound/soc/soc-io.c +.. kernel-doc:: sound/soc/soc-component.c .. kernel-doc:: sound/soc/soc-pcm.c .. kernel-doc:: sound/soc/soc-ops.c .. kernel-doc:: sound/soc/soc-compress.c

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[PATCH] ASoC: qcom: Drop HAS_DMA dependency to fix link failure
by Geert Uytterhoeven 29 Jun '20

29 Jun '20

When building on allyesconfig kernel for a NO_DMA=y platform (e.g. Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n, leading to a link failure: sound/soc/qcom/common.o: In function `qcom_snd_parse_of': common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port' While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845 don't, so the following warning is seen: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y] Until recently, this warning was harmless (from a compile-testing point-of-view), but the new user of q6afe_is_rx_port() turned this into a hard failure. As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is safe to increase compile testing coverage. Hence fix the link failure by dropping the HAS_DMA dependency of SND_SOC_QDSP6. Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks") Fixes: 6b1687bf76ef84cb ("ASoC: qcom: add sdm845 sound card support") Fixes: a6f933f63f2ffdb2 ("ASoC: qcom: apq8096: Add db820c machine driver") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d87b5..92f51d0e9fe2bf02 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" - depends on QCOM_APR && HAS_DMA + depends on QCOM_APR select SND_SOC_QDSP6_COMMON select SND_SOC_QDSP6_CORE select SND_SOC_QDSP6_AFE -- 2.17.1

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[PATCH v2 0/3] ASoC: rt1015: fix compilation errors and warnings
by Tzung-Bi Shih 29 Jun '20

29 Jun '20

The series fixes compilation errors and warnings discovered in the thread https://patchwork.kernel.org/patch/11622319/. The original patch: (1) adds acpi.h for fixing implicit declaration of function 'ACPI_PTR' (2) sorts header inclusions in alphabetical The compilation errors and warnings are likely introduced by (2). The 1st and 2nd patches fix the newly discovered errors and warnings. The 3rd patch is the same as the original patch. Tzung-Bi Shih (3): ASoC: core: move definition of enum snd_soc_bias_level ASoC: dapm: declare missing structure prototypes ASoC: rt1015: add missing header inclusion include/sound/soc-dapm.h | 20 ++++++++++++++++++++ include/sound/soc.h | 18 ------------------ sound/soc/codecs/rt1015.c | 17 +++++++++-------- 3 files changed, 29 insertions(+), 26 deletions(-) -- 2.27.0.212.ge8ba1cc988-goog

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[PATCH v4 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 29 Jun '20

29 Jun '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v4: - Add review tag by Charles - fix the lock deadlock pointed by Charles Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 10 ++- sound/core/compress_offload.c | 4 + 3 files changed, 96 insertions(+), 1 deletion(-) -- 2.26.2

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