March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda/tegra: avoid build error without CONFIG_PM
by Arnd Bergmann 13 Mar '19

13 Mar '19

The #ifdef protection around the PM functions is wrong, leading to a failed reference in some configurations: sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend': sound/pci/hda/hda_tegra.c:273:2: error: implicit declaration of function 'hda_tegra_disable_clocks'; did you mean 'hda_tegra_enable_clocks'? [-Werror=implicit-function-declaration] Better remove the #ifdefs entirely and rely on the compiler silently dropping unused functions marked __maybe_unused. Fixes: 707e0759f2f4 ("ALSA: hda/tegra: implement runtime suspend/resume") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/hda_tegra.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index dbd8da5685cb..3d68f9ef7694 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -230,7 +229,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) /* * power management */ -static int hda_tegra_suspend(struct device *dev) +static int __maybe_unused hda_tegra_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); int rc; @@ -243,7 +242,7 @@ static int hda_tegra_suspend(struct device *dev) return 0; } -static int hda_tegra_resume(struct device *dev) +static int __maybe_unused hda_tegra_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); int rc; @@ -255,10 +254,8 @@ static int hda_tegra_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM -static int hda_tegra_runtime_suspend(struct device *dev) +static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; @@ -275,7 +272,7 @@ static int hda_tegra_runtime_suspend(struct device *dev) return 0; } -static int hda_tegra_runtime_resume(struct device *dev) +static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; @@ -292,7 +289,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) -- 2.20.0

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Re: [alsa-devel] [PATCH] ALSA: usx2y: Fix potential NULL pointer dereference
by Takashi Iwai 13 Mar '19

13 Mar '19

On Tue, 05 Mar 2019 00:00:02 +0100, Aditya Pakki wrote: > > usb_alloc_urb can fail due to kmalloc failure and push the error > upstream. Further this can cause a NULL pointer dereference in > init_pipe_urbs. This patch avoids such a scenario. > > Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH 0/1] sound/hda/hdac_stream: Avoid NULL pointer dereference
by Mariusz Ceier 13 Mar '19

13 Mar '19

Seems like commit 9b6f7e7a296e17990aae298c809b001e99ddd151 introduced NULL pointer dereference for ca0132 codec. When ca0132 loads firmware, snd_hdac_stream_start is called with azx_dev->substream being NULL. This patch calls snd_hdac_get_stream_stripe_ctl only when azx_dev->substream is not NULL. Even if I'm not sure if this is correct, since it might be that ca0132 codec does something wrong, with this change NULL pointer dereference doesn't happen and ca0132 works again on my system with Recon3Di. Mariusz Ceier (1): sound/hda/hdac_stream: Avoid NULL pointer dereference sound/hda/hdac_stream.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) -- 2.21.0

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[alsa-devel] [RFC PATCH 1/4] alsa: make hw_params negotiation infrastructure 'bclk_ratio aware'
by Sven Van Asbroeck 13 Mar '19

13 Mar '19

Negotiation seems to work ok, but bclk_ratio is exposed to userspace via snd_pcm_hw_params, which is not acceptable. Constrain bclk_ratio by: - cpu dai capabilities && rules - codec dai capabilities && rules - minimum bclk_ratio is sample_width * channels In hw_params_choose(), pick the smallest supported bclk_ratio, which should correspond to the most efficient solution. If cpu and codec dais do not specify or constrain supported bclk_ratios, alsa will pick sample_width * channels. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- include/sound/pcm.h | 11 +++++++++++ include/sound/soc.h | 2 ++ include/uapi/sound/asound.h | 5 +++-- sound/core/pcm_native.c | 34 +++++++++++++++++++++++++++++++++- sound/soc/soc-pcm.c | 8 ++++++++ 5 files changed, 57 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index d6bd3caf6878..71ac7e8de23d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -56,6 +56,8 @@ struct snd_pcm_hardware { unsigned int periods_min; /* min # of periods */ unsigned int periods_max; /* max # of periods */ size_t fifo_size; /* fifo size in bytes */ + unsigned int bclk_ratio_min; /* min bclk ratio for wire format */ + unsigned int bclk_ratio_max; /* max bclk ratio for wire format */ }; struct snd_pcm_substream; @@ -980,6 +982,15 @@ static inline unsigned int params_buffer_bytes(const struct snd_pcm_hw_params *p return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min; } +/** + * params_bclk_ratio - Get the bclk_ratio from the hw params + * @p: hw params + */ +static inline unsigned int params_bclk_ratio(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BCLK_RATIO)->min; +} + int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); int snd_interval_list(struct snd_interval *i, unsigned int count, const unsigned int *list, unsigned int mask); diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f111b0d2..96d669423688 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -732,6 +732,8 @@ struct snd_soc_pcm_stream { unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ unsigned int sig_bits; /* number of bits of content */ + unsigned int bclk_ratio_min; /* min bclk ratio for wire format */ + unsigned int bclk_ratio_max; /* max bclk ratio for wire format */ }; /* SoC audio ops */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 404d4b9ffe76..c3ea94eaaa77 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -371,8 +371,9 @@ typedef int snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_BUFFER_SIZE 17 /* Size of buffer in frames */ #define SNDRV_PCM_HW_PARAM_BUFFER_BYTES 18 /* Size of buffer in bytes */ #define SNDRV_PCM_HW_PARAM_TICK_TIME 19 /* Approx tick duration in us */ +#define SNDRV_PCM_HW_PARAM_BCLK_RATIO 20 /* bclk_ratio for wire format */ #define SNDRV_PCM_HW_PARAM_FIRST_INTERVAL SNDRV_PCM_HW_PARAM_SAMPLE_BITS -#define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME +#define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_BCLK_RATIO #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ #define SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER (1<<1) /* export buffer */ @@ -399,7 +400,7 @@ struct snd_pcm_hw_params { struct snd_mask mres[5]; /* reserved masks */ struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; - struct snd_interval ires[9]; /* reserved intervals */ + struct snd_interval ires[8]; /* reserved intervals */ unsigned int rmask; /* W: requested masks */ unsigned int cmask; /* R: changed masks */ unsigned int info; /* R: Info flags for returned setup */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 818dff1de545..23dbe43a6691 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -516,6 +516,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int err; + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_BCLK_RATIO); params->info = 0; params->fifo_size = 0; @@ -525,6 +526,12 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, params->rate_num = 0; params->rate_den = 0; } + /* + * if left zero (not empty), assume userspace is oblivious, and + * completely flexible + */ + if (snd_interval_single(r) && snd_interval_min(r) == 0) + snd_interval_any(r); err = constrain_mask_params(substream, params); if (err < 0) @@ -610,7 +617,8 @@ static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, - * min rate, min period time, max buffer size, min tick time + * min rate, min period time, max buffer size, min tick time, + * min bclk_ratio * * Return: Zero if successful, or a negative error code on failure. */ @@ -626,6 +634,7 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_TICK_TIME, + SNDRV_PCM_HW_PARAM_BCLK_RATIO, -1 }; const int *v; @@ -2100,6 +2109,18 @@ static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, return snd_mask_refine(mask, &m); } +static int snd_pcm_hw_rule_bclk_ratio(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval i; + struct snd_interval *ratios = hw_param_interval(params, SNDRV_PCM_HW_PARAM_BCLK_RATIO); + snd_interval_any(&i); + i.openmax = 1; + i.min = params_channels(params) * params_width(params); + i.integer = 1; + return snd_interval_refine(ratios, &i); +} + static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -2180,12 +2201,18 @@ int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)); snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)); snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS)); + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BCLK_RATIO)); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, snd_pcm_hw_rule_format, NULL, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BCLK_RATIO, + snd_pcm_hw_rule_bclk_ratio, NULL, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_sample_bits, NULL, SNDRV_PCM_HW_PARAM_FORMAT, @@ -2341,6 +2368,11 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BCLK_RATIO, + hw->bclk_ratio_min, hw->bclk_ratio_max); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_buffer_bytes_max, substream, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 03f36e534050..747026595634 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -381,6 +381,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) struct snd_soc_pcm_stream *cpu_stream; unsigned int chan_min = 0, chan_max = UINT_MAX; unsigned int rate_min = 0, rate_max = UINT_MAX; + unsigned int bclk_ratio_min = 0, bclk_ratio_max = UINT_MAX; unsigned int rates = UINT_MAX; u64 formats = ULLONG_MAX; int i; @@ -413,6 +414,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) codec_stream = &codec_dai_drv->capture; chan_min = max(chan_min, codec_stream->channels_min); chan_max = min(chan_max, codec_stream->channels_max); + bclk_ratio_min = max(bclk_ratio_min, codec_stream->bclk_ratio_min); + bclk_ratio_max = min_not_zero(bclk_ratio_max, codec_stream->bclk_ratio_max); rate_min = max(rate_min, codec_stream->rate_min); rate_max = min_not_zero(rate_max, codec_stream->rate_max); formats &= codec_stream->formats; @@ -443,6 +446,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); hw->rate_max = min_not_zero(hw->rate_max, rate_max); + + hw->bclk_ratio_min = max(hw->bclk_ratio_min, cpu_stream->bclk_ratio_min); + hw->bclk_ratio_min = max(hw->bclk_ratio_min, bclk_ratio_min); + hw->bclk_ratio_max = min_not_zero(hw->bclk_ratio_max, cpu_stream->bclk_ratio_max); + hw->bclk_ratio_max = min_not_zero(hw->bclk_ratio_max, bclk_ratio_max); } static int soc_pcm_components_close(struct snd_pcm_substream *substream, -- 2.17.1

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[alsa-devel] [PATCH v2 0/5] use extended audio dmac register
by Jiada Wang 13 Mar '19

13 Mar '19

According to user reference manual for R-CAR H3, M3-W and M3-N SoCs, in order to access busif4 ~ busif7, extended audio dmac registers (PDMASAREn, PDMADAREn, PDMACHCREn) need to be used, rather than basic audio dmac registers (PDMASARn, PDMADARn, PDMACHCRn) This patch set updates H3 (= r8a7795), M3-W (= r8a7796) and M3-N (=r8a77965) to use extended audio dmac registers R-CAR E3 only has extended register set and is already using it, rename register set name to indicate this. --- v2: update rsnd driver to use extended register set when available rename R-CAR E3 audio dmac regisetr v1: initial version Jiada Wang (5): arm64: dts: renesas: r8a7795: add extended audio dmac register arm64: dts: renesas: r8a7796: add extended audio dmac register arm64: dts: renesas: r8a77965: add extended audio dmac register arm64: dts: renesas: r8a77990: rename audio dmac register ASoC: rsnd: dma: use extended audio dmac registers when available arch/arm64/boot/dts/renesas/r8a7795.dtsi | 5 +++-- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 5 +++-- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 5 +++-- arch/arm64/boot/dts/renesas/r8a77990.dtsi | 4 ++-- sound/soc/sh/rcar/dma.c | 5 ++++- 5 files changed, 15 insertions(+), 9 deletions(-) -- 2.19.2

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[alsa-devel] [PATCH 1/2] ASoC: audio-graph-card: test memory allocation
by Pierre-Louis Bossart 12 Mar '19

12 Mar '19

devm_kcalloc() return value needs to be checked. Fixes: 1e4771a62fd7a ('ASoC: audio-graph-card: add link_info') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351330e8..ba65b48c10a7 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -670,7 +670,7 @@ static int graph_probe(struct platform_device *pdev) dai_link = devm_kcalloc(dev, li.link, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, li.dais, sizeof(*dais), GFP_KERNEL); cconf = devm_kcalloc(dev, li.conf, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) + if (!dai_props || !dai_link || !dais || !cconf) return -ENOMEM; /* -- 2.17.1

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[alsa-devel] [PATCH] media: Drop superfluous PCM preallocation error checks
by Takashi Iwai 12 Mar '19

12 Mar '19

snd_pcm_lib_preallocate_pages() and co always succeed, so the error check is simply redundant. Drop it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a piece I overlooked in the previous patchset: https://www.spinics.net/lists/alsa-devel/msg87223.html Media people, please give ACK if this is OK. Then I'll merge it together with other relevant patches. Thanks! drivers/media/pci/solo6x10/solo6x10-g723.c | 4 +--- drivers/media/pci/tw686x/tw686x-audio.c | 3 ++- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/drivers/media/pci/solo6x10/solo6x10-g723.c b/drivers/media/pci/solo6x10/solo6x10-g723.c index 2cc05a9d57ac..a16242a9206f 100644 --- a/drivers/media/pci/solo6x10/solo6x10-g723.c +++ b/drivers/media/pci/solo6x10/solo6x10-g723.c @@ -360,13 +360,11 @@ static int solo_snd_pcm_init(struct solo_dev *solo_dev) ss; ss = ss->next, i++) sprintf(ss->name, "Camera #%d Audio", i); - ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), G723_PERIOD_BYTES * PERIODS, G723_PERIOD_BYTES * PERIODS); - if (ret < 0) - return ret; solo_dev->snd_pcm = pcm; diff --git a/drivers/media/pci/tw686x/tw686x-audio.c b/drivers/media/pci/tw686x/tw686x-audio.c index a28329698e20..fb0e7573b5ae 100644 --- a/drivers/media/pci/tw686x/tw686x-audio.c +++ b/drivers/media/pci/tw686x/tw686x-audio.c @@ -301,11 +301,12 @@ static int tw686x_snd_pcm_init(struct tw686x_dev *dev) ss; ss = ss->next, i++) snprintf(ss->name, sizeof(ss->name), "vch%u audio", i); - return snd_pcm_lib_preallocate_pages_for_all(pcm, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(dev->pci_dev), TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX, TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX); + return 0; } static void tw686x_audio_dma_free(struct tw686x_dev *dev, -- 2.16.4

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Fix DAPM routes for capture stream
by Sylwester Nawrocki 11 Mar '19

11 Mar '19

This patch sets missing stream_name of capture part of the DAI driver so we can define DAPM routing properly also for the capture stream. While at it "Playback" suffix is added to the playback stream names to clearly identify playback/capture. Together with related dts patch this fixes NULL pointer dereference when opening ALSA device for recording on Odroid XU3. Fixes: 64aba9bca5bd ("ASoC: samsung: i2s: Add widgets and routes for DPCM support") Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 4231001226f4..ab471d550d17 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1130,11 +1130,11 @@ static const struct snd_soc_dapm_widget samsung_i2s_widgets[] = { }; static const struct snd_soc_dapm_route samsung_i2s_dapm_routes[] = { - { "Playback Mixer", NULL, "Primary" }, - { "Playback Mixer", NULL, "Secondary" }, + { "Playback Mixer", NULL, "Primary Playback" }, + { "Playback Mixer", NULL, "Secondary Playback" }, { "Mixer DAI TX", NULL, "Playback Mixer" }, - { "Playback Mixer", NULL, "Mixer DAI RX" }, + { "Primary Capture", NULL, "Mixer DAI RX" }, }; static const struct snd_soc_component_driver samsung_i2s_component = { @@ -1155,7 +1155,8 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, int num_dais) { static const char *dai_names[] = { "samsung-i2s", "samsung-i2s-sec" }; - static const char *stream_names[] = { "Primary", "Secondary" }; + static const char *stream_names[] = { "Primary Playback", + "Secondary Playback" }; struct snd_soc_dai_driver *dai_drv; struct i2s_dai *dai; int i; @@ -1201,6 +1202,7 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, dai_drv->capture.channels_max = 2; dai_drv->capture.rates = i2s_dai_data->pcm_rates; dai_drv->capture.formats = SAMSUNG_I2S_FMTS; + dai_drv->capture.stream_name = "Primary Capture"; return 0; } -- 2.17.1

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[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 11 Mar '19

11 Mar '19

The patch ASoC: rsnd: src: fix compiler warnings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 825b22e5961a47ba7387d3ee6729956d6bb3b495 Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 11:34:53 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index db81e066b92e..79716725f67b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -135,7 +135,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -144,7 +144,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -153,7 +153,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -162,7 +162,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -171,7 +171,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -180,7 +180,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

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[alsa-devel] [PATCH 0/3] ASoC: hda: capture-related fixes
by Rander Wang 11 Mar '19

11 Mar '19

There are several sources of errors fixed in this series, such as mismatch between TX and RX mask used to store the stream_tag, the use of hw_param fixups not properly handled in the hdac_hda codec and registers set on capture when they should only be used on playback. The first patch fixes a issue caused by tx|rx_mask for TDM. The Second one fixes the mismatch between BE dai format and hda codec format. The last one removes the register of stream id setting on capture. this series was tested successfully with the Intel Skylake driver on a Skylake Dell XPS13 (same as Linus), KabyLake NUC, Whiskylake laptop and Apollolake LeafHill board. The two soc-pcm and hdac_hda fixes are needed by the SOF driver as well, but SOF-specific patches will be provided in the next batch and are not included here Rander Wang (3): ASoC:soc-pcm:fix a codec fixup issue in TDM case ASoC:hdac_hda:use correct format to setup hda codec ASoC:intel:skl:fix a simultaneous playback & capture issue on hda platform sound/soc/codecs/hdac_hda.c | 53 ++++++++++++++++++++++++++++----------- sound/soc/codecs/hdac_hda.h | 1 + sound/soc/intel/skylake/skl-pcm.c | 19 ++++++++++---- sound/soc/soc-pcm.c | 7 ++++-- 4 files changed, 59 insertions(+), 21 deletions(-) -- 2.14.1

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