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

[PATCH] ASoC: msm8916-wcd-analog: fix register Interrupt offset
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

For some reason interrupt set and clear register offsets are not set correctly. This patch corrects them! Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4428c62e25cf..3ddd822240e3 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -19,8 +19,8 @@ #define CDC_D_REVISION1 (0xf000) #define CDC_D_PERPH_SUBTYPE (0xf005) -#define CDC_D_INT_EN_SET (0x015) -#define CDC_D_INT_EN_CLR (0x016) +#define CDC_D_INT_EN_SET (0xf015) +#define CDC_D_INT_EN_CLR (0xf016) #define MBHC_SWITCH_INT BIT(7) #define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6) #define MBHC_BUTTON_PRESS_DET BIT(5) -- 2.21.0

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[PATCH] [v2] ASoC: intel: Fix memleak in sst_media_open
by Dinghao Liu 18 Aug '20

18 Aug '20

When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c1a68b ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- Changelog: v2: - Add a new label 'out_power_up' to unify code style. --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 49b9f18472bc..b1cac7abdc0a 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -331,7 +331,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, ret_val = power_up_sst(stream); if (ret_val < 0) - return ret_val; + goto out_power_up; /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -340,8 +340,9 @@ static int sst_media_open(struct snd_pcm_substream *substream, return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); out_ops: - kfree(stream); mutex_unlock(&sst_lock); +out_power_up: + kfree(stream); return ret_val; } -- 2.17.1

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[PATCH] ASoC: codecs: delete repeated words in comments
by Randy Dunlap 18 Aug '20

18 Aug '20

Drop the repeated words {start, it, the} in comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) --- linux-next-20200807.orig/sound/soc/codecs/nau8825.c +++ linux-next-20200807/sound/soc/codecs/nau8825.c @@ -251,7 +251,7 @@ static const unsigned short logtable[256 * * Acquires the semaphore without jiffies. Try to acquire the semaphore * atomically. Returns 0 if the semaphore has been acquired successfully - * or 1 if it it cannot be acquired. + * or 1 if it cannot be acquired. */ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) { --- linux-next-20200807.orig/sound/soc/codecs/tas5086.c +++ linux-next-20200807/sound/soc/codecs/tas5086.c @@ -487,7 +487,7 @@ static int tas5086_init(struct device *d /* * If any of the channels is configured to start in Mid-Z mode, * configure 'part 1' of the PWM starts to use Mid-Z, and tell - * all configured mid-z channels to start start under 'part 1'. + * all configured mid-z channels to start under 'part 1'. */ if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, --- linux-next-20200807.orig/sound/soc/codecs/wm9713.c +++ linux-next-20200807/sound/soc/codecs/wm9713.c @@ -755,7 +755,7 @@ static void pll_factors(struct snd_soc_c u64 Kpart; unsigned int K, Ndiv, Nmod, target; - /* The the PLL output is always 98.304MHz. */ + /* The PLL output is always 98.304MHz. */ target = 98304000; /* If the input frequency is over 14.4MHz then scale it down. */

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[PATCH 00/21] ASoC: Intel: fix cppcheck warnings
by Pierre-Louis Bossart 18 Aug '20

18 Aug '20

This patchset tries to reduce the number of warnings on those drivers, so that cppcheck can become a viable tool to detect issues (currently hundreds of reports). Most of the problems are related to unnecessary/redundant variable assignments, prototypes and one nice logical mistake resulting in an always-true condition. Pierre-Louis Bossart (21): ASoC: Intel: Atom: sst-atom-controls: remove redundant assignments ASoC: Intel: Atom: compress: remove redundant assignment ASoC: Intel: Atom: platform-pcm: remove redundant assignment ASoC: Intel: Atom: sst: remove useless NULL assignment ASoC: Intel: Atom: remove redundant initialization ASoC: Intel: Atom: sst_pvt: remove redundant initialization ASoC: Intel: Atom: platform-pcm: fix redundant return ASoC: Intel: Atom: remove useless assignment ASoC: Intel: Atom: sst_loader: remove always-true condition ASoC: Intel: Atom: sst_pvt: simplify return handling ASoC: Intel: Atom: (cosmetic) align parameters ASoC: Intel: Baytrail: (cosmetic) align function parameters ASoC: Intel: common: (cosmetic) align function parameters ASoC: Intel: haswell: (cosmetic) align function parameters ASoC: Intel: haswell-ipc: remove redundant assignments ASoC: Intel: Skylake: skl-nhlt: remove redundant initialization ASoC: Intel: Skylake: cldma: remove redundant initialization ASoC: Intel: Skylake: sst-utils: remove redundant assignment ASoC: Intel: Skylake: skl-topology: remove redundant assignments ASoC: Intel: Skylake: skl-topology: remove redundant assignment ASoC: Intel: Skylake: (cosmetic) align function parameters sound/soc/intel/atom/sst-atom-controls.c | 4 +-- .../intel/atom/sst-mfld-platform-compress.c | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +-- sound/soc/intel/atom/sst-mfld-platform.h | 4 +-- sound/soc/intel/atom/sst/sst.c | 5 ++- sound/soc/intel/atom/sst/sst.h | 34 +++++++++---------- sound/soc/intel/atom/sst/sst_loader.c | 3 +- sound/soc/intel/atom/sst/sst_pvt.c | 4 +-- sound/soc/intel/baytrail/sst-baytrail-ipc.h | 3 +- sound/soc/intel/common/sst-dsp-priv.h | 2 +- sound/soc/intel/common/sst-dsp.h | 15 ++++---- sound/soc/intel/haswell/sst-haswell-ipc.c | 4 +-- sound/soc/intel/haswell/sst-haswell-ipc.h | 2 +- sound/soc/intel/skylake/cnl-sst-dsp.h | 4 +-- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/intel/skylake/skl-sst-cldma.c | 2 +- sound/soc/intel/skylake/skl-sst-ipc.h | 16 ++++----- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 8 ++--- sound/soc/intel/skylake/skl-topology.h | 8 ++--- sound/soc/intel/skylake/skl.h | 2 +- 21 files changed, 63 insertions(+), 68 deletions(-) -- 2.25.1

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[PATCH 1/2] ASoC: dt-bindings: ak4458: Add power supply property
by Shengjiu Wang 18 Aug '20

18 Aug '20

AVDD-supply is for Analog power supply DVDD-supply is for Digital power supply Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak4458.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index e5820235e0d5..8f6c84f21468 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -10,6 +10,8 @@ Required properties: Optional properties: - reset-gpios: A GPIO specifier for the power down & reset pin - mute-gpios: A GPIO specifier for the soft mute pin +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply Example: -- 2.27.0

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[PATCH] ASoC: various vendors: delete repeated words in comments
by Randy Dunlap 18 Aug '20

18 Aug '20

Drop the repeated words {related, we, is, the} in comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/intel/skylake/skl-sst.c | 2 +- sound/soc/meson/axg-tdm-formatter.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) --- linux-next-20200807.orig/sound/soc/fsl/fsl_dma.c +++ linux-next-20200807/sound/soc/fsl/fsl_dma.c @@ -154,7 +154,7 @@ static void fsl_dma_abort_stream(struct /** * fsl_dma_update_pointers - update LD pointers to point to the next period * - * As each period is completed, this function changes the the link + * As each period is completed, this function changes the link * descriptor pointers for that period to point to the next period. */ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) --- linux-next-20200807.orig/sound/soc/intel/skylake/skl-sst.c +++ linux-next-20200807/sound/soc/intel/skylake/skl-sst.c @@ -354,7 +354,7 @@ static int skl_transfer_module(struct ss /* * if bytes_left > 0 then wait for BDL complete interrupt and * copy the next chunk till bytes_left is 0. if bytes_left is - * is zero, then wait for load module IPC reply + * zero, then wait for load module IPC reply */ while (bytes_left > 0) { curr_pos = size - bytes_left; --- linux-next-20200807.orig/sound/soc/meson/axg-tdm-formatter.c +++ linux-next-20200807/sound/soc/meson/axg-tdm-formatter.c @@ -398,7 +398,7 @@ void axg_tdm_stream_free(struct axg_tdm_ /* * If the list is not empty, it would mean that one of the formatter * widget is still powered and attached to the interface while we - * we are removing the TDM DAI. It should not be possible + * are removing the TDM DAI. It should not be possible */ WARN_ON(!list_empty(&ts->formatter_list)); mutex_destroy(&ts->lock); --- linux-next-20200807.orig/sound/soc/sprd/sprd-pcm-compress.c +++ linux-next-20200807/sound/soc/sprd/sprd-pcm-compress.c @@ -559,7 +559,7 @@ static int sprd_platform_compr_copy(stru } else { /* * If the data count is larger than the available spaces - * of the the stage 0 IRAM buffer, we should copy one + * of the stage 0 IRAM buffer, we should copy one * partial data to the stage 0 IRAM buffer, and copy * the left to the stage 1 DDR buffer. */ --- linux-next-20200807.orig/sound/soc/sunxi/sun4i-codec.c +++ linux-next-20200807/sound/soc/sunxi/sun4i-codec.c @@ -335,7 +335,7 @@ static int sun4i_codec_prepare_capture(s /* * FIXME: Undocumented in the datasheet, but - * Allwinner's code mentions that it is related + * Allwinner's code mentions that it is * related to microphone gain */ if (of_device_is_compatible(scodec->dev->of_node, --- linux-next-20200807.orig/sound/soc/ti/davinci-mcasp.c +++ linux-next-20200807/sound/soc/ti/davinci-mcasp.c @@ -633,7 +633,7 @@ static int __davinci_mcasp_set_clkdiv(st * right channels), so it has to be divided by number * of tdm-slots (for I2S - divided by 2). * Instead of storing this ratio, we calculate a new - * tdm_slot width by dividing the the ratio by the + * tdm_slot width by dividing the ratio by the * number of configured tdm slots. */ mcasp->slot_width = div / mcasp->tdm_slots;

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[PATCH v2] ASoC: fsl-asoc-card: Get "extal" clock rate by clk_get_rate
by Shengjiu Wang 18 Aug '20

18 Aug '20

On some platform(.e.g. i.MX8QM MEK), the "extal" clock is different with the mclk of codec, then the clock rate is also different. So it is better to get clock rate of "extal" rate by clk_get_rate, don't reuse the clock rate of mclk. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add defer probe handler sound/soc/fsl/fsl-asoc-card.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 52adedc03245..32f8f756e6bb 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -696,6 +696,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } } else if (of_node_name_eq(cpu_np, "esai")) { + struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal"); + + if (!IS_ERR(esai_clk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk); + priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk); + clk_put(esai_clk); + } else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto asrc_fail; + } + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; } else if (of_node_name_eq(cpu_np, "sai")) { -- 2.27.0

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[PATCH] ASoC: SOF: delete repeated words in comments
by Randy Dunlap 18 Aug '20

18 Aug '20

Drop the repeated words {that, the} in comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/sof-audio.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- linux-next-20200807.orig/sound/soc/sof/intel/cnl.c +++ linux-next-20200807/sound/soc/sof/intel/cnl.c @@ -202,7 +202,7 @@ static int cnl_ipc_send_msg(struct snd_s * IPCs are sent at a high-rate. mod_delayed_work() * modifies the timer if the work is pending. * Also, a new delayed work should not be queued after the - * the CTX_SAVE IPC, which is sent before the DSP enters D3. + * CTX_SAVE IPC, which is sent before the DSP enters D3. */ if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) mod_delayed_work(system_wq, &hdev->d0i3_work, --- linux-next-20200807.orig/sound/soc/sof/sof-audio.c +++ linux-next-20200807/sound/soc/sof/sof-audio.c @@ -29,7 +29,7 @@ bool snd_sof_dsp_only_d0i3_compatible_st continue; /* - * substream->runtime being not NULL indicates that + * substream->runtime being not NULL indicates * that the stream is open. No need to check the * stream state. */

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[PATCH 1/3] ASoC: core: Add common helper to parse aux devs from device tree
by Stephan Gerhold 18 Aug '20

18 Aug '20

simple-card.c and meson-card-utils.c use pretty much the same helper function to parse auxiliary devices from the device tree. Make it easier for other drivers to parse these from the device tree as well by adding a shared helper function to soc-core.c. snd_soc_of_parse_aux_devs() is pretty much a copy of meson_card_add_aux_devices() from meson-card-utils.c with two minor changes: - Make property name configurable as parameter - Change dev_err() message slightly for consistency with other error messages in soc-core.c Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- I have another patch set that I will submit separately which makes use of this function to parse aux devs from the device tree within qcom_snd_parse_of(). This is preparation for that patch set. --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e3919ffb00c..a0918d159fd3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1331,6 +1331,7 @@ void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..bf46f410c8c6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2827,6 +2827,37 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, propname, NULL); + if (num == -ENOENT) { + return 0; + } else if (num < 0) { + dev_err(card->dev, "ASOC: Property '%s' could not be read: %d\n", + propname, num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for_each_card_pre_auxs(card, i, aux) { + aux->dlc.of_node = of_parse_phandle(node, propname, i); + if (!aux->dlc.of_node) + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_aux_devs); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, -- 2.27.0

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[PATCH 1/2] dt-bindings: tas2562: Remove tas2562 text file
by Dan Murphy 18 Aug '20

18 Aug '20

Remove the tas2562 text file as the tas2562.yaml is now available. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 37 ------------------- 1 file changed, 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index dc6d7362ded7..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. This slot must always be - greater then ti,imon-slot-no. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down-gpio: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down-gpio = <&gpio1 15 0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <1>; -}; - -- 2.28.0

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