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

[PATCH v2 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 15 Sep '21

15 Sep '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. These patches depends on the dt-bindings header patch -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Chnages Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 169 ++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 343 +++++++++++++++++++++ 4 files changed, 526 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.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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[PATCH] ASoC: atmel: Convert to new style DAI format definitions
by Mark Brown 15 Sep '21

15 Sep '21

Convert the Atmel drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 4 ++-- sound/soc/atmel/atmel_ssc_dai.c | 10 +++++----- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++++---- sound/soc/atmel/mikroe-proto.c | 4 ++-- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- 7 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6b3d9c05eaf2..e119abbca099 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -343,7 +343,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* codec is slave, so cpu is master */ mr |= ATMEL_I2SC_MR_MODE_MASTER; ret = atmel_i2s_get_gck_param(dev, params_rate(params)); @@ -351,7 +351,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return ret; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* codec is master, so cpu is slave */ mr |= ATMEL_I2SC_MR_MODE_SLAVE; dev->gck_param = NULL; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6a63e8797a0b..4846a8958337 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -210,7 +210,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, return frame_size; switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) && ssc->clk_from_rk_pin) /* Receiver Frame Synchro (i.e. capture) @@ -220,7 +220,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, mck_div = 3; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) && !ssc->clk_from_rk_pin) /* Transmit Frame Synchro (i.e. playback) @@ -233,7 +233,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: r.num = ssc_p->mck_rate / mck_div / frame_size; ret = snd_interval_ratnum(i, 1, &r, &num, &den); @@ -243,8 +243,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBP_CFP: t.min = 8000; t.max = ssc_p->mck_rate / mck_div / frame_size; t.openmin = t.openmax = 0; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 9e237580afa9..9c974c4e187d 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -66,7 +66,7 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { .stream_name = "WM8904 PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &atmel_asoc_wm8904_ops, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 8988f024a732..67550e3d9ee3 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -350,7 +350,7 @@ static int mchp_i2s_mcc_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; /* We can't generate only FSYNC */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFS) + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBP_CFC) return -EINVAL; /* We can only reconfigure the IP when it's stopped */ @@ -547,19 +547,19 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* cpu is BCLK and LRC master */ mra |= MCHP_I2SMCC_MRA_MODE_MASTER; if (dev->sysclk) mra |= MCHP_I2SMCC_MRA_IMCKMODE_GEN; set_divs = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: /* cpu is BCLK master */ mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; set_divs = 1; fallthrough; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* cpu is slave */ mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; if (dev->sysclk) diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index 0be7b4221c14..f9331f7e80fe 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -129,9 +129,9 @@ static int snd_proto_probe(struct platform_device *pdev) } if (bitclkmaster) { if (codec_np == bitclkmaster) - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; else - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; } else { dai_fmt |= snd_soc_daifmt_parse_clock_provider_as_flag(np, NULL); } diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index ed1f69b57024..915da92e1ec8 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -126,7 +126,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .stream_name = "WM8731 PCM", .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 50c3dc6936f9..1b3a31296c9b 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link tse850_dailink = { .stream_name = "TSE-850-PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFS, + | SND_SOC_DAIFMT_CBP_CFC, SND_SOC_DAILINK_REG(pcm), }; -- 2.20.1

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[PATCH] ASoC: amd: Convert to new style DAI format definitions
by Mark Brown 15 Sep '21

15 Sep '21

Convert the AMD machine drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 20 ++++++++++---------- sound/soc/amd/acp-rt5645.c | 4 ++-- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +++--- 3 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b2065f3fe42c..3bf86c2424ae 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -522,7 +522,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_da7219_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -533,7 +533,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, @@ -543,7 +543,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_max_play_ops, @@ -554,7 +554,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, @@ -565,7 +565,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, @@ -578,7 +578,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_rt5682_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -589,7 +589,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, @@ -599,7 +599,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, @@ -610,7 +610,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, @@ -621,7 +621,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index d6ba94677ac2..58b75cf399fc 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -111,7 +111,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-play", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_init, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware1, codec, platform), @@ -120,7 +120,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-cap", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware2, codec, platform), }, diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e561464f7d60..7af2bfdbf6d7 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -51,7 +51,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) /* set rt5682 dai fmt */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); + | SND_SOC_DAIFMT_CBP_CFP); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set rt5682 dai fmt: %d\n", ret); @@ -302,7 +302,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = acp3x_5682_init, .dpcm_playback = 1, .dpcm_capture = 1, @@ -313,7 +313,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .dpcm_playback = 1, .ops = &acp3x_max_play_ops, .cpus = acp3x_bt, -- 2.20.1

1 1

[PATCH 00/12] ASoC: SOF: Abstract away the 'bar' usage (mostly)
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Hi, The direct use of 'bar' as address regions in SOF can limit portability and makes assumption that all platforms are using memory mapped regions or BARs to communicate with the DSP. The core API should not use 'bar' in it's parameter lists or make assumption that the platform needs to comply with this system integration. Use the region type as an ID in core and translate that in platform dependent code to a way to access registers, mailbox or DSP memory. Regards, Peter --- Peter Ujfalusi (12): ASoC: SOF: Intel: bdw: Set the mailbox offset directly in bdw_probe ASoC: SOF: ipc: Remove snd_sof_dsp_mailbox_init() ASoC: SOF: imx: Do not initialize the snd_sof_dsp_ops.read64 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: No need to export snd_sof_fw_parse_ext_data() ASoC: SOF: core: Do not use 'bar' as parameter for block_read/write ASoC: SOF: debug: Add generic API and ops for DSP regions ASoC: SOF: imx: Provide debugfs_add_region_item ops for core ASoC: SOF: Intel: Provide debugfs_add_region_item ops for core ASoC: SOF: loader: Use the generic ops for region debugfs handling ASoC: SOF: debug: No need to export the snd_sof_debugfs_io_item() sound/soc/sof/debug.c | 24 ++++-- sound/soc/sof/imx/imx8.c | 17 +++-- sound/soc/sof/imx/imx8m.c | 13 +++- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/bdw.c | 5 +- sound/soc/sof/intel/byt.c | 2 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/pci-tng.c | 1 + sound/soc/sof/intel/tgl.c | 1 + sound/soc/sof/ipc.c | 34 +++------ sound/soc/sof/loader.c | 137 +++++++++++++++------------------- sound/soc/sof/ops.h | 25 +++++-- sound/soc/sof/sof-priv.h | 55 +++++++------- sound/soc/sof/utils.c | 28 +++++-- 15 files changed, 187 insertions(+), 158 deletions(-) -- 2.33.0

2 15

[PATCH] ASoC: mediatek: mt6359: Fix unexpected error in bind/unbind flow
by Trevor Wu 15 Sep '21

15 Sep '21

mt6359-sound is a MFD driver. Because its regmap is retrieved from its parent, it shouldn't be freed in mt6359-sound driver. snd_soc_component_exit_regmap() will do regmap_exit(), this results in unexpected results if sound card unregister flow is invoked when users try to bind/unbind audio codec. Remove the usage of snd_soc_component_exit_regmap(). Instead, set component->regmap = NULL in the component remove function. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/codecs/mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 2d6a4a29b850..f8532aa7e4aa 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2697,7 +2697,7 @@ static int mt6359_codec_probe(struct snd_soc_component *cmpnt) static void mt6359_codec_remove(struct snd_soc_component *cmpnt) { - snd_soc_component_exit_regmap(cmpnt); + cmpnt->regmap = NULL; } static const DECLARE_TLV_DB_SCALE(hp_playback_tlv, -2200, 100, 0); -- 2.18.0

2 1

[PATCH 0/4] Support ALC5682I-VS codec
by Brent Lu 15 Sep '21

15 Sep '21

Support the ALC5682I-VS codec in Intel's rt5682 machine driver with three board configurations. Brent Lu (4): ASoC: Intel: sof_rt5682: support ALC5682I-VS codec ASoC: intel: sof_rt5682: support jsl_rt5682s_rt1015p board ASoC: intel: sof_rt5682: support jsl_rt5682s_rt1015 board ASoC: intel: sof_rt5682: support jsl_rt5682s_mx98360a board sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_rt5682.c | 105 +++++++++++++++--- .../intel/common/soc-acpi-intel-jsl-match.c | 24 ++++ 3 files changed, 112 insertions(+), 18 deletions(-) -- 2.25.1

3 6

[PATCH 0/3] ASoC: SOF: Intel: hda: Cleanups for local function uses
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Hi, Small cleanups regarding to HDA function locations and dropping of definitions of not implemented functions. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda-dsp: Declare locally used functions as static ASoC: SOF: Intel: hda: Remove boot_firmware skl and iccmax_icl declarations ASoC: SOF: Intel: hda: Relocate inline definitions from hda.h to hda.c for sdw sound/soc/sof/intel/hda-dsp.c | 61 +++++++++++++++++------------------ sound/soc/sof/intel/hda.c | 33 ++++++++++++++++++- sound/soc/sof/intel/hda.h | 41 ----------------------- 3 files changed, 62 insertions(+), 73 deletions(-) -- 2.33.0

2 4

[PATCH] ASoC: mediatek: mt8195: make array adda_dai_list static const
by Colin King 15 Sep '21

15 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array adda_dai_list on the stack but instead it static const. Also makes the object code smaller by 33 bytes: Before: text data bss dec hex filename 28271 11640 0 39911 9be7 mt8195/mt8195-dai-adda.o After: text data bss dec hex filename 28142 11736 0 39878 9bc6 mt8195/mt8195-dai-adda.o (gcc version 11.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c index 878dec0b69ed..f04bd1781356 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c @@ -788,9 +788,11 @@ static int init_adda_priv_data(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; struct mtk_dai_adda_priv *adda_priv; - int adda_dai_list[] = { MT8195_AFE_IO_DL_SRC, - MT8195_AFE_IO_UL_SRC1, - MT8195_AFE_IO_UL_SRC2}; + static const int adda_dai_list[] = { + MT8195_AFE_IO_DL_SRC, + MT8195_AFE_IO_UL_SRC1, + MT8195_AFE_IO_UL_SRC2 + }; int i; for (i = 0; i < ARRAY_SIZE(adda_dai_list); i++) { -- 2.32.0

2 1

[PATCH 1/6] ASoC: cs35l41: Fix use of an uninitialised variable
by Charles Keepax 15 Sep '21

15 Sep '21

The loop checking PDN_DONE doesn't check the return value from regmap_read, nor does it initialise val. This means if regmap_read fails val will be checked for the PDN_DONE bit whilst being uninitialised. Fix this up by switching to regmap_read_poll_timeout which tidies up the code and avoids the uninitialised variable. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index dbec54a28a9e3..d2a11cc336833 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -580,7 +580,6 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret = 0; bool pdn; - int i; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -599,19 +598,11 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, CS35L41_GLOBAL_EN_MASK, 0); pdn = false; - for (i = 0; i < 100; i++) { - regmap_read(cs35l41->regmap, - CS35L41_IRQ1_STATUS1, - &val); - if (val & CS35L41_PDN_DONE_MASK) { - pdn = true; - break; - } - usleep_range(1000, 1100); - } - - if (!pdn) - dev_warn(cs35l41->dev, "PDN failed\n"); + ret = regmap_read_poll_timeout(cs35l41->regmap, CS35L41_IRQ1_STATUS1, + val, val & CS35L41_PDN_DONE_MASK, + 1000, 100000); + if (ret) + dev_warn(cs35l41->dev, "PDN failed: %d\n", ret); regmap_write(cs35l41->regmap, CS35L41_IRQ1_STATUS1, CS35L41_PDN_DONE_MASK); -- 2.11.0

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[PATCH 0/2] ASoC: SOF: Remove unused members from struct sof_dev_desc
by Peter Ujfalusi 15 Sep '21

15 Sep '21

Hi, dma_engine, dma_size and resindex_dma_base is unused from sof_dev_desc, drop them. resindex_dma_base is initialized to -1 for Intel platforms, but it is not used. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: intel: Do no initialize resindex_dma_base ASoC: SOF: Drop resindex_dma_base, dma_engine, dma_size from sof_dev_desc include/sound/sof.h | 5 ----- sound/soc/sof/intel/pci-apl.c | 2 -- sound/soc/sof/intel/pci-cnl.c | 3 --- sound/soc/sof/intel/pci-icl.c | 2 -- sound/soc/sof/intel/pci-tgl.c | 5 ----- sound/soc/sof/intel/pci-tng.c | 1 - 6 files changed, 18 deletions(-) -- 2.33.0

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