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author | Mark Brown <broonie@kernel.org> | 2021-03-04 01:52:07 +0100 |
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committer | Mark Brown <broonie@kernel.org> | 2021-03-10 14:13:24 +0100 |
commit | 98537d5c925711f65b4a88eef571d2cf0d2c51ac (patch) | |
tree | 0e701450a5caaef4ceda707f83e05010711bb5d3 | |
parent | Merge series "ASoC: realtek: fix cppcheck warnings" from Pierre-Louis Bossart... (diff) | |
parent | ASoC: qcom: q6dsp: fix kernel-doc warning (diff) | |
download | linux-98537d5c925711f65b4a88eef571d2cf0d2c51ac.tar.xz linux-98537d5c925711f65b4a88eef571d2cf0d2c51ac.zip |
Merge series "ASoC: remove more make W=1 warnings" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
These warnings get in the way of automation/CI, let's remove them.
Pierre-Louis Bossart (9):
ASoC: cs4270: fix kernel-doc
ASoC: jz4760: fix set but not used warning
ASoC: rt5631: fix kernel-doc warning
ASoC: sigmadsp-regmap: fix kernel-doc warning
ASoC: amd: renoir: remove invalid kernel-doc comment
ASoC: fsl: fsl_ssi: fix kernel-doc warning
ASoC: fsl: fsl_easrc: fix kernel-doc warning
ASoC: Intel: bytcr_wm5102: remove unused static variable
ASoC: qcom: q6dsp: fix kernel-doc warning
sound/soc/amd/renoir/rn-pci-acp3x.c | 2 +-
sound/soc/codecs/cs4270.c | 1 +
sound/soc/codecs/jz4760.c | 4 ++--
sound/soc/codecs/rt5631.c | 2 +-
sound/soc/codecs/sigmadsp-regmap.c | 2 +-
sound/soc/fsl/fsl_easrc.c | 2 +-
sound/soc/fsl/fsl_ssi.c | 2 +-
sound/soc/intel/boards/bytcr_wm5102.c | 8 --------
sound/soc/qcom/qdsp6/q6afe.c | 2 +-
9 files changed, 9 insertions(+), 16 deletions(-)
--
2.25.1
-rw-r--r-- | sound/soc/amd/renoir/rn-pci-acp3x.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/cs4270.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/jz4760.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/rt5631.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/sigmadsp-regmap.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_easrc.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 2 | ||||
-rw-r--r-- | sound/soc/intel/boards/bytcr_wm5102.c | 8 | ||||
-rw-r--r-- | sound/soc/qcom/qdsp6/q6afe.c | 2 |
9 files changed, 9 insertions, 16 deletions
diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 050a61fe9693..19438da5dfa5 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -20,7 +20,7 @@ static int acp_power_gating; module_param(acp_power_gating, int, 0644); MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating"); -/** +/* * dmic_acpi_check = -1 - Use ACPI/DMI method to detect the DMIC hardware presence at runtime * = 0 - Skip the DMIC device creation and return probe failure * = 1 - Force DMIC support diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ddd95c8269ed..2d239e983a83 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -400,6 +400,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute + * @direction: (ignored) * * This function toggles the mute bits in the MUTE register. The CS4270's * mute capability is intended for external muting circuitry, so if the diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index 5ae0e312bcfc..d96a4f6c9183 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -198,7 +198,7 @@ static int jz4760_codec_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec); - int ret; + int ret = 0; /* * SYSCLK output from the codec to the AIC is required to keep the @@ -207,7 +207,7 @@ static int jz4760_codec_startup(struct snd_pcm_substream *substream, */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ret = snd_soc_dapm_force_enable_pin(dapm, "SYSCLK"); - return 0; + return ret; } static void jz4760_codec_shutdown(struct snd_pcm_substream *substream, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index afc1305a7fa5..3000bc128b5b 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -436,7 +436,7 @@ static void onebit_depop_mute_stage(struct snd_soc_component *component, int ena } /** - * onebit_depop_power_stage - step by step depop sequence in power stage. + * depop_seq_power_stage - step by step depop sequence in power stage. * @component: ASoC component * @enable: power on/off * diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c index bf1c4086da9f..ba9a6795e470 100644 --- a/sound/soc/codecs/sigmadsp-regmap.c +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -26,7 +26,7 @@ static int sigmadsp_read_regmap(void *control_data, } /** - * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance + * devm_sigmadsp_init_regmap() - Initialize SigmaDSP instance * @dev: The parent device * @regmap: Regmap instance to use * @ops: The sigmadsp_ops to use for this instance diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index e823c9c13764..600e0d670ca6 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -380,7 +380,7 @@ static int fsl_easrc_resampler_config(struct fsl_asrc *easrc) } /** - * Scale filter coefficients (64 bits float) + * fsl_easrc_normalize_filter - Scale filter coefficients (64 bits float) * For input float32 normalized range (1.0,-1.0) -> output int[16,24,32]: * scale it by multiplying filter coefficients by 2^31 * For input int[16, 24, 32] -> output float32 diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4e2ce47a7c6e..2b57b60431bb 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -363,7 +363,7 @@ static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi *ssi) } /** - * fsl_ssi_irq - Interrupt handler to gather states + * fsl_ssi_isr - Interrupt handler to gather states * @irq: irq number * @dev_id: context */ diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index f38850eb2eaf..fd584e380340 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -213,14 +213,6 @@ static int byt_wm5102_init(struct snd_soc_pcm_runtime *runtime) return 0; } -static const struct snd_soc_pcm_stream byt_wm5102_dai_params = { - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, -}; - static int byt_wm5102_codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 6f700c311337..38f6042fe893 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1448,7 +1448,7 @@ int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg) EXPORT_SYMBOL_GPL(q6afe_i2s_port_prepare); /** - * q6afe_dam_port_prepare() - Prepare dma afe port. + * q6afe_cdc_dma_port_prepare() - Prepare dma afe port. * * @port: Instance of afe port * @cfg: DMA configuration for the afe port |