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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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[PATCH] ASoC: qcom: add a dedicated menuconfig
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

Currently list of Qualcomm drivers is growing, so put them in to a proper menu so that it does not mix up with other ASOC configs in menuconfig. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/Kconfig | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5d6b2466a2f2..be6b8d0e2f70 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -1,11 +1,13 @@ # SPDX-License-Identifier: GPL-2.0-only -config SND_SOC_QCOM +menuconfig SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help Say Y or M if you want to add support to use audio devices in Qualcomm Technologies SOC-based platforms. +if SND_SOC_QCOM + config SND_SOC_LPASS_CPU tristate select REGMAP_MMIO @@ -26,7 +28,6 @@ config SND_SOC_LPASS_APQ8016 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -35,7 +36,6 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 select SND_SOC_QCOM_COMMON help @@ -110,3 +110,5 @@ config SND_SOC_SDM845 To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. Say Y if you want to use audio device on this SoCs. + +endif #SND_SOC_QCOM -- 2.21.0

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[PATCH v6 00/12] ASoC: qcom: Add support for SC7180 lpass variant
by Rohit kumar 18 Aug '20

18 Aug '20

This patch chain add audio support for SC7180 soc by doing the required modification in existing common lpass-cpu/lpass-platform driver. This also fixes some concurrency issue. This patch series is already tested by Srinivas on Dragon Board 410c. Changes since v5: - Fixed remove api in lpass-sc7180.c - Addressed comments by Rob in yaml Documentation. Ajit Pandey (4): ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-platform: Replace card->dev with component->dev include: dt-bindings: sound: Add sc7180-lpass bindings header ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio Rohit kumar (8): ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function ASoC: qcom: lpass-platform: fix memory leak ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers ASoC: qcom: lpass-cpu: fix concurrency issue dt-bindings: sound: lpass-cpu: Add sc7180 lpass cpu node ASoC: qcom: lpass-cpu: Use platform_get_resource ASoC: qcom: lpass-platform: Use platform_get_irq dt-bindings: sound: lpass-cpu: Move to yaml format .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 -------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 189 ++++++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 86 ++++++-- sound/soc/qcom/lpass-cpu.c | 204 ++++++++++--------- sound/soc/qcom/lpass-ipq806x.c | 67 +++++++ sound/soc/qcom/lpass-lpaif-reg.h | 157 ++++++++------- sound/soc/qcom/lpass-platform.c | 155 +++++++++++---- sound/soc/qcom/lpass-sc7180.c | 216 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 63 +++++- 12 files changed, 934 insertions(+), 299 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- 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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