[PATCH v2 00/25] ASoC: use helper function and cleanup
Hi Mark
These are v2 patch-set to use helper function on ASoC.
I'm posting about DT schema fixup, but it seems take longer time. This patch-set is 100% independent from it, so I will post it.
struct snd_soc_dai need to have info for playback/capture, but it is using "playback/capture_xxx" or "tx/tx_xxx" or array. This kind of random definition is very difficult to read.
This patch-set add helper functions and each driver use it. And cleanup the definition.
v1 -> v2 - added Reviewed-by on each patches if it got - fixup soc-dai.c helper bug
Link: https://lore.kernel.org/r/87a62dlmmn.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (25): ASoC: soc-dai.h: add missing snd_soc_dai_set_widget() ASoC: soc-dai.h: add snd_soc_dai_dma_data_set/get() for low level ASoC: soc-dai.h: add snd_soc_dai_tdm_mask_set/get() helper ASoC: soc-dai.h: add snd_soc_dai_get/set_widget_playback/capture() helper ASoC: soc.h: add snd_soc_card_is_instantiated() helper ASoC: hdmi-codec: use helper function ASoC: hda: use helper function ASoC: max: use helper function ASoC: rt: use helper function ASoC: cirrus: use helper function ASoC: spear: use helper function ASoC: sdw-mockup: use helper function ASoC: intel: use helper function ASoC: mediatek: use helper function ASoC: meson: use helper function ASoC: rockchip: use helper function ASoC: sof: use helper function ASoC: tegra: use helper function ASoC: ti: use helper function ASoC: soc-core.c: use helper function ASoC: soc-dapm.c: use helper function ASoC: soc-dai.c: use helper function ASoC: soc-pcm.c: use helper function ASoC: soc-topology.c: use helper function ASoC: soc-dai.h: cleanup Playback/Capture data for snd_soc_dai
include/sound/soc-dai.h | 90 ++++++++------- include/sound/soc.h | 6 + sound/soc/cirrus/ep93xx-ac97.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 4 +- sound/soc/codecs/hda.c | 7 +- sound/soc/codecs/hdmi-codec.c | 11 +- sound/soc/codecs/max98373-sdw.c | 5 +- sound/soc/codecs/rt1308-sdw.c | 5 +- sound/soc/codecs/rt1316-sdw.c | 5 +- sound/soc/codecs/rt1318-sdw.c | 5 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5665.c | 4 +- sound/soc/codecs/rt5668.c | 4 +- sound/soc/codecs/rt5682-sdw.c | 5 +- sound/soc/codecs/rt5682.c | 4 +- sound/soc/codecs/rt5682s.c | 4 +- sound/soc/codecs/rt700.c | 7 +- sound/soc/codecs/rt711-sdca.c | 7 +- sound/soc/codecs/rt711.c | 7 +- sound/soc/codecs/rt715-sdca.c | 5 +- sound/soc/codecs/rt715.c | 5 +- sound/soc/codecs/sdw-mockup.c | 5 +- sound/soc/intel/atom/sst-atom-controls.c | 9 +- sound/soc/intel/avs/boards/nau8825.c | 5 +- sound/soc/intel/avs/pcm.c | 15 +-- sound/soc/intel/skylake/skl-topology.c | 19 +--- .../soc/mediatek/common/mtk-dsp-sof-common.c | 18 ++- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 8 +- sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 7 +- sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 7 +- sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 7 +- sound/soc/meson/aiu-fifo-i2s.c | 4 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.c | 21 ++-- sound/soc/meson/axg-tdm-interface.c | 34 +++--- sound/soc/meson/axg-tdmin.c | 2 +- sound/soc/meson/axg-tdmout.c | 2 +- sound/soc/meson/meson-codec-glue.c | 8 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dai.c | 13 ++- sound/soc/soc-dapm.c | 26 ++--- sound/soc/soc-pcm.c | 12 +- sound/soc/soc-topology.c | 4 +- sound/soc/sof/topology.c | 104 +++++++----------- sound/soc/spear/spdif_out.c | 3 +- sound/soc/tegra/tegra20_ac97.c | 4 +- sound/soc/tegra/tegra20_i2s.c | 4 +- sound/soc/tegra/tegra20_spdif.c | 3 +- sound/soc/tegra/tegra210_admaif.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/ti/davinci-i2s.c | 5 +- sound/soc/ti/davinci-mcasp.c | 5 +- sound/soc/ti/davinci-vcif.c | 5 +- 57 files changed, 274 insertions(+), 315 deletions(-)
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has snd_soc_dai_get_widget() (= _get_) but doesn't have _set_ function. This patch adds it.
This patch also cleanup unnecessary line break for _get_ function.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- include/sound/soc-dai.h | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ea7509672086..ea704d92deaa 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -478,13 +478,21 @@ snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) }
static inline -struct snd_soc_dapm_widget *snd_soc_dai_get_widget( - struct snd_soc_dai *dai, int stream) +struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream) { return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? dai->playback_widget : dai->capture_widget; }
+static inline +void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_widget = widget; + else + dai->capture_widget = widget; +} + static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) {
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has snd_soc_dai_set/get_dma_data() which is assuming struct snd_pcm_substream to get Playback/Capture direction.
But, many drivers want to use it not through snd_pcm_substream. This patch adds more low level snd_soc_dai_dma_data_set/get() for it, and previous functions will be macro for it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- include/sound/soc-dai.h | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ea704d92deaa..f27b33dd97bd 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -493,18 +493,21 @@ void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_ dai->capture_widget = widget; }
-static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, - const struct snd_pcm_substream *ss) +#define snd_soc_dai_dma_data_get_playback(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_PLAYBACK) +#define snd_soc_dai_dma_data_get_capture(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_CAPTURE) +#define snd_soc_dai_get_dma_data(dai, ss) snd_soc_dai_dma_data_get(dai, ss->stream) +static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int stream) { - return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? dai->playback_dma_data : dai->capture_dma_data; }
-static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, - const struct snd_pcm_substream *ss, - void *data) +#define snd_soc_dai_dma_data_set_playback(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_PLAYBACK, data) +#define snd_soc_dai_dma_data_set_capture(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_CAPTURE, data) +#define snd_soc_dai_set_dma_data(dai, ss, data) snd_soc_dai_dma_data_set(dai, ss->stream, data) +static inline void snd_soc_dai_dma_data_set(struct snd_soc_dai *dai, int stream, void *data) { - if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) dai->playback_dma_data = data; else dai->capture_dma_data = data;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has tx/rx_mask, and is directly accessing to them, but accessing to it via function is nice idea. This patch adds snd_soc_dai_tdm_mask_set/get() for it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- include/sound/soc-dai.h | 17 +++++++++++++++++ 1 file changed, 17 insertions(+)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index f27b33dd97bd..9b18e230e5b3 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -520,6 +520,23 @@ static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, dai->capture_dma_data = capture; }
+static inline unsigned int snd_soc_dai_tdm_mask_get(struct snd_soc_dai *dai, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return dai->tx_mask; + else + return dai->rx_mask; +} + +static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream, + unsigned int tdm_mask) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->tx_mask = tdm_mask; + else + dai->rx_mask = tdm_mask; +} + static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) {
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
snd_soc_dai_get_widget() requests SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. This patch adds helper for it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- include/sound/soc-dai.h | 4 ++++ 1 file changed, 4 insertions(+)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 9b18e230e5b3..197dc1629708 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -477,6 +477,8 @@ snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) &dai->driver->playback : &dai->driver->capture; }
+#define snd_soc_dai_get_widget_playback(dai) snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_PLAYBACK) +#define snd_soc_dai_get_widget_capture(dai) snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_CAPTURE) static inline struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream) { @@ -484,6 +486,8 @@ struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int dai->playback_widget : dai->capture_widget; }
+#define snd_soc_dai_set_widget_playback(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_PLAYBACK, widget) +#define snd_soc_dai_set_widget_capture(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_CAPTURE, widget) static inline void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget) {
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
ASoC framework/driver checks whether card was instantiated every where. Then, it should check card pointer too in such case. This patch adds snd_soc_card_is_instantiated() for it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- include/sound/soc.h | 6 ++++++ 1 file changed, 6 insertions(+)
diff --git a/include/sound/soc.h b/include/sound/soc.h index 37bbfc8b45cb..e58b43b5da7c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1052,6 +1052,12 @@ struct snd_soc_card { #define for_each_card_widgets_safe(card, w, _w) \ list_for_each_entry_safe(w, _w, &card->widgets, list)
+ +static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) +{ + return card && card->instantiated; +} + /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { struct device *dev;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/hdmi-codec.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-)
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 74cbbe16f9ae..01e8ffda2a4b 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -518,7 +518,7 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); - struct hdmi_codec_daifmt *cf = dai->playback_dma_data; + struct hdmi_codec_daifmt *cf = snd_soc_dai_dma_data_get_playback(dai); struct hdmi_codec_params hp = { .iec = { .status = { 0 }, @@ -562,7 +562,7 @@ static int hdmi_codec_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); - struct hdmi_codec_daifmt *cf = dai->playback_dma_data; + struct hdmi_codec_daifmt *cf = snd_soc_dai_dma_data_get_playback(dai); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int channels = runtime->channels; unsigned int width = snd_pcm_format_width(runtime->format); @@ -597,7 +597,7 @@ static int hdmi_codec_prepare(struct snd_pcm_substream *substream, static int hdmi_codec_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - struct hdmi_codec_daifmt *cf = dai->playback_dma_data; + struct hdmi_codec_daifmt *cf = snd_soc_dai_dma_data_get_playback(dai);
/* Reset daifmt */ memset(cf, 0, sizeof(*cf)); @@ -834,7 +834,8 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) if (!daifmt) return -ENOMEM;
- dai->playback_dma_data = daifmt; + snd_soc_dai_dma_data_set_playback(dai, daifmt); + return 0; }
@@ -891,7 +892,7 @@ static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) if (ret) return ret;
- cf = dai->playback_dma_data; + cf = snd_soc_dai_dma_data_get_playback(dai); cf->fmt = HDMI_SPDIF;
return 0;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/hda.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index 4b8ec6f77337..d57b043d6bfe 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -126,12 +126,15 @@ static void hda_codec_unregister_dais(struct hda_codec *codec, struct hda_pcm *pcm;
for_each_component_dais_safe(component, dai, save) { + int stream; + list_for_each_entry(pcm, &codec->pcm_list_head, list) { if (strcmp(dai->driver->name, pcm->name)) continue;
- snd_soc_dapm_free_widget(dai->playback_widget); - snd_soc_dapm_free_widget(dai->capture_widget); + for_each_pcm_streams(stream) + snd_soc_dapm_free_widget(snd_soc_dai_get_widget(dai, stream)); + snd_soc_unregister_dai(dai); break; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/max98373-sdw.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 3cd1be743d9e..c9a2d4dabd3c 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -689,10 +689,7 @@ static int max98373_set_sdw_stream(struct snd_soc_dai *dai, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/rt1308-sdw.c | 5 +---- sound/soc/codecs/rt1316-sdw.c | 5 +---- sound/soc/codecs/rt1318-sdw.c | 5 +---- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5665.c | 4 ++-- sound/soc/codecs/rt5668.c | 4 ++-- sound/soc/codecs/rt5682-sdw.c | 5 +---- sound/soc/codecs/rt5682.c | 4 ++-- sound/soc/codecs/rt5682s.c | 4 ++-- sound/soc/codecs/rt700.c | 7 ++----- sound/soc/codecs/rt711-sdca.c | 7 ++----- sound/soc/codecs/rt711.c | 7 ++----- sound/soc/codecs/rt715-sdca.c | 5 +---- sound/soc/codecs/rt715.c | 5 +---- 14 files changed, 21 insertions(+), 48 deletions(-)
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index ca2790d63b71..45544b530d3d 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -508,10 +508,7 @@ static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index e6294cc7a995..6dbb59f51426 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -507,10 +507,7 @@ static int rt1316_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c index f85f5ab2c6d0..784147a1004d 100644 --- a/sound/soc/codecs/rt1318-sdw.c +++ b/sound/soc/codecs/rt1318-sdw.c @@ -575,10 +575,7 @@ static int rt1318_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + dai->stream[direction].dma_data = stream;
return 0; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 620ecbfa4a7a..7c7cbb6362ea 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3157,7 +3157,7 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse snd_soc_dapm_force_enable_pin(dapm, "LDO2"); snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); - if (!dapm->card->instantiated) { + if (!snd_soc_card_is_instantiated(dapm->card)) { /* Power up necessary bits for JD if dapm is not ready yet */ regmap_update_bits(rt5645->regmap, RT5645_PWR_ANLG1, diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 6e66cc218fa8..17afaef85c77 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1298,7 +1298,7 @@ static void rt5665_jack_detect_handler(struct work_struct *work) usleep_range(10000, 15000); }
- while (!rt5665->component->card->instantiated) { + while (!snd_soc_card_is_instantiated(rt5665->component->card)) { pr_debug("%s\n", __func__); usleep_range(10000, 15000); } @@ -4748,7 +4748,7 @@ static void rt5665_calibrate_handler(struct work_struct *work) struct rt5665_priv *rt5665 = container_of(work, struct rt5665_priv, calibrate_work.work);
- while (!rt5665->component->card->instantiated) { + while (!snd_soc_card_is_instantiated(rt5665->component->card)) { pr_debug("%s\n", __func__); usleep_range(10000, 15000); } diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index beb0951ff680..ecf3b0527dbe 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,8 +1022,8 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type;
- if (!rt5668->component || !rt5668->component->card || - !rt5668->component->card->instantiated) { + if (!rt5668->component || + !snd_soc_card_is_instantiated(rt5668->component->card)) { /* card not yet ready, try later */ mod_delayed_work(system_power_efficient_wq, &rt5668->jack_detect_work, msecs_to_jiffies(15)); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index d8a573dcb771..5f80a5d59b65 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -107,10 +107,7 @@ static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7e3eb65afe16..f6c798b65c08 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1094,8 +1094,8 @@ void rt5682_jack_detect_handler(struct work_struct *work) struct snd_soc_dapm_context *dapm; int val, btn_type;
- if (!rt5682->component || !rt5682->component->card || - !rt5682->component->card->instantiated) { + if (!rt5682->component || + !snd_soc_card_is_instantiated(rt5682->component->card)) { /* card not yet ready, try later */ mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(15)); diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index f5e5dbc3b0f0..77dc62219bb8 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -834,8 +834,8 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) struct snd_soc_dapm_context *dapm; int val, btn_type;
- if (!rt5682s->component || !rt5682s->component->card || - !rt5682s->component->card->instantiated) { + if (!rt5682s->component || + !snd_soc_card_is_instantiated(rt5682s->component->card)) { /* card not yet ready, try later */ mod_delayed_work(system_power_efficient_wq, &rt5682s->jack_detect_work, msecs_to_jiffies(15)); diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 6534c9b51442..659ce26e9f3b 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -163,7 +163,7 @@ static void rt700_jack_detect_handler(struct work_struct *work) if (!rt700->hs_jack) return;
- if (!rt700->component->card || !rt700->component->card->instantiated) + if (!snd_soc_card_is_instantiated(rt700->component->card)) return;
reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; @@ -887,10 +887,7 @@ static int rt700_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 7cdf184d380b..c65abe812a4c 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -295,7 +295,7 @@ static void rt711_sdca_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return;
- if (!rt711->component->card || !rt711->component->card->instantiated) + if (!snd_soc_card_is_instantiated(rt711->component->card)) return;
/* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ @@ -1249,10 +1249,7 @@ static int rt711_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 78e1da9b0738..862f50950565 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -243,7 +243,7 @@ static void rt711_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return;
- if (!rt711->component->card || !rt711->component->card->instantiated) + if (!snd_soc_card_is_instantiated(rt711->component->card)) return;
if (pm_runtime_status_suspended(rt711->slave->dev.parent)) { @@ -976,10 +976,7 @@ static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 1fca7a3f46ea..920510365fd7 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -793,10 +793,7 @@ static int rt715_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; } diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 1a2036ccfbac..c6dd9df7be45 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -777,10 +777,7 @@ static int rt715_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; }
On Tue, Jan 24, 2023 at 02:10:20AM +0000, Kuninori Morimoto wrote:
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
This breaks an x86 allmodconfig build (missing use of the helper):
/build/stage/linux/sound/soc/codecs/rt1318-sdw.c: In function ‘rt1318_set_sdw_stream’: /build/stage/linux/sound/soc/codecs/rt1318-sdw.c:578:12: error: ‘struct snd_soc_dai’ has no member named ‘stream’ 578 | dai->stream[direction].dma_data = stream; | ^~ make[5]: *** [/build/stage/linux/scripts/Makefile.build:252: sound/soc/codecs/rt1318-sdw.o] Error 1
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/cirrus/ep93xx-ac97.c | 4 ++-- sound/soc/cirrus/ep93xx-i2s.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 37593abe6053..cec067c91a0f 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -323,8 +323,8 @@ static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai) info->dma_params_tx.filter_data = &ep93xx_ac97_pcm_out; info->dma_params_rx.filter_data = &ep93xx_ac97_pcm_in;
- dai->playback_dma_data = &info->dma_params_tx; - dai->capture_dma_data = &info->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &info->dma_params_tx, + &info->dma_params_rx);
return 0; } diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 982151330c89..8265173a7932 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -202,8 +202,8 @@ static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) info->dma_params_rx.filter_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE];
- dai->playback_dma_data = &info->dma_params_tx; - dai->capture_dma_data = &info->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &info->dma_params_tx, + &info->dma_params_rx);
return 0; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/spear/spdif_out.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index fb107c5790ad..469373d1bb41 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -244,7 +244,8 @@ static int spdif_soc_dai_probe(struct snd_soc_dai *dai) struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
host->dma_params_tx.filter_data = &host->dma_params; - dai->playback_dma_data = &host->dma_params_tx; + + snd_soc_dai_dma_data_set_playback(dai, &host->dma_params_tx);
return snd_soc_add_dai_controls(dai, spdif_out_controls, ARRAY_SIZE(spdif_out_controls));
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/sdw-mockup.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/codecs/sdw-mockup.c b/sound/soc/codecs/sdw-mockup.c index af52f2728854..62b02d764f09 100644 --- a/sound/soc/codecs/sdw-mockup.c +++ b/sound/soc/codecs/sdw-mockup.c @@ -57,10 +57,7 @@ static int sdw_mockup_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, stream->sdw_stream = sdw_stream;
/* Use tx_mask or rx_mask to configure stream tag and set dma_data */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = stream; - else - dai->capture_dma_data = stream; + snd_soc_dai_dma_data_set(dai, direction, stream);
return 0; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/intel/atom/sst-atom-controls.c | 9 ++------- sound/soc/intel/avs/boards/nau8825.c | 5 +++-- sound/soc/intel/avs/pcm.c | 15 ++++++++------- sound/soc/intel/skylake/skl-topology.c | 19 ++++--------------- 4 files changed, 17 insertions(+), 31 deletions(-)
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index fd59b35a62ba..38116c758717 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1327,15 +1327,13 @@ static bool is_sst_dapm_widget(struct snd_soc_dapm_widget *w) int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) { struct sst_data *drv = snd_soc_dai_get_drvdata(dai); - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); struct snd_soc_dapm_path *p;
dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream); + dev_dbg(dai->dev, "Stream name=%s\n", w->name);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - dev_dbg(dai->dev, "Stream name=%s\n", - dai->playback_widget->name); - w = dai->playback_widget; snd_soc_dapm_widget_for_each_sink_path(w, p) { if (p->connected && !p->connected(w, p->sink)) continue; @@ -1352,9 +1350,6 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) } } } else { - dev_dbg(dai->dev, "Stream name=%s\n", - dai->capture_widget->name); - w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { if (p->connected && !p->connected(w, p->source)) continue; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 6731d8a49076..b31fa931ba8b 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -258,14 +258,15 @@ static int avs_card_resume_post(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + int stream = SNDRV_PCM_STREAM_PLAYBACK;
if (!codec_dai) { dev_err(card->dev, "Codec dai not found\n"); return -EINVAL; }
- if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && - codec_dai->playback_widget->active) + if (snd_soc_dai_stream_active(codec_dai, stream) && + snd_soc_dai_get_widget(codec_dai, stream)->active) snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN);
return snd_soc_component_set_jack(codec_dai->component, jack, NULL); diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index c4376c9c35ef..211edd51a896 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -35,15 +35,13 @@ struct avs_dma_data { static struct avs_tplg_path_template * avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction) { - struct snd_soc_dapm_widget *dw; + struct snd_soc_dapm_widget *dw = snd_soc_dai_get_widget(dai, direction); struct snd_soc_dapm_path *dp; enum snd_soc_dapm_direction dir;
if (direction == SNDRV_PCM_STREAM_CAPTURE) { - dw = dai->capture_widget; dir = is_fe ? SND_SOC_DAPM_DIR_OUT : SND_SOC_DAPM_DIR_IN; } else { - dw = dai->playback_widget; dir = is_fe ? SND_SOC_DAPM_DIR_IN : SND_SOC_DAPM_DIR_OUT; }
@@ -929,7 +927,7 @@ static int avs_component_pm_op(struct snd_soc_component *component, bool be, int ret;
for_each_component_dais(component, dai) { - data = dai->playback_dma_data; + data = snd_soc_dai_dma_data_get_playback(dai); if (data) { rtd = asoc_substream_to_rtd(data->substream); if (rtd->dai_link->no_pcm == be && !rtd->dai_link->ignore_suspend) { @@ -942,7 +940,7 @@ static int avs_component_pm_op(struct snd_soc_component *component, bool be, } }
- data = dai->capture_dma_data; + data = snd_soc_dai_dma_data_get_capture(dai); if (data) { rtd = asoc_substream_to_rtd(data->substream); if (rtd->dai_link->no_pcm == be && !rtd->dai_link->ignore_suspend) { @@ -1291,11 +1289,14 @@ static void avs_component_hda_unregister_dais(struct snd_soc_component *componen sprintf(name, "%s-cpu", dev_name(&codec->core.dev));
for_each_component_dais_safe(component, dai, save) { + int stream; + if (!strstr(dai->driver->name, name)) continue;
- snd_soc_dapm_free_widget(dai->playback_widget); - snd_soc_dapm_free_widget(dai->capture_widget); + for_each_pcm_streams(stream) + snd_soc_dapm_free_widget(snd_soc_dai_get_widget(dai, stream)); + snd_soc_unregister_dai(dai); } } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index b20643b83401..96cfebded072 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1663,11 +1663,10 @@ int skl_tplg_update_pipe_params(struct device *dev, struct skl_module_cfg * skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); struct snd_soc_dapm_path *p = NULL;
if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - w = dai->playback_widget; snd_soc_dapm_widget_for_each_sink_path(w, p) { if (p->connect && p->sink->power && !is_skl_dsp_widget_type(p->sink, dai->dev)) @@ -1680,7 +1679,6 @@ skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream) } } } else { - w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { if (p->connect && p->source->power && !is_skl_dsp_widget_type(p->source, dai->dev)) @@ -1744,14 +1742,12 @@ static struct skl_module_cfg *skl_get_mconfig_cap_cpr( struct skl_module_cfg * skl_tplg_be_get_cpr_module(struct snd_soc_dai *dai, int stream) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); struct skl_module_cfg *mconfig;
if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - w = dai->playback_widget; mconfig = skl_get_mconfig_pb_cpr(dai, w); } else { - w = dai->capture_widget; mconfig = skl_get_mconfig_cap_cpr(dai, w); } return mconfig; @@ -1905,20 +1901,13 @@ static int skl_tplg_be_set_sink_pipe_params(struct snd_soc_dai *dai, int skl_tplg_be_update_params(struct snd_soc_dai *dai, struct skl_pipe_params *params) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, params->stream);
if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { - w = dai->playback_widget; - return skl_tplg_be_set_src_pipe_params(dai, w, params); - } else { - w = dai->capture_widget; - return skl_tplg_be_set_sink_pipe_params(dai, w, params); } - - return 0; }
static const struct snd_soc_tplg_widget_events skl_tplg_widget_ops[] = { @@ -2978,7 +2967,7 @@ void skl_cleanup_resources(struct skl_dev *skl) return;
card = soc_component->card; - if (!card || !card->instantiated) + if (!snd_soc_card_is_instantiated(card)) return;
list_for_each_entry(w, &card->widgets, list) {
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 18 +++++++----------- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 8 +++++--- sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 7 ++++--- sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 7 ++++--- sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 8 +++++--- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 7 +++++-- 6 files changed, 30 insertions(+), 25 deletions(-)
diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c index 8b1b623207be..6fef16306f74 100644 --- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c @@ -32,7 +32,7 @@ int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, continue;
for_each_rtd_cpu_dais(runtime, j, cpu_dai) { - if (cpu_dai->stream_active[conn->stream_dir] > 0) { + if (snd_soc_dai_stream_active(cpu_dai, conn->stream_dir) > 0) { sof_dai_link = runtime->dai_link; break; } @@ -111,21 +111,17 @@ int mtk_sof_card_late_probe(struct snd_soc_card *card) for_each_rtd_cpu_dais(sof_rtd, j, cpu_dai) { struct snd_soc_dapm_route route; struct snd_soc_dapm_path *p = NULL; - struct snd_soc_dapm_widget *play_widget = - cpu_dai->playback_widget; - struct snd_soc_dapm_widget *cap_widget = - cpu_dai->capture_widget; + struct snd_soc_dapm_widget *widget = snd_soc_dai_get_widget(cpu_dai, conn->stream_dir); + memset(&route, 0, sizeof(route)); - if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && - cap_widget) { - snd_soc_dapm_widget_for_each_sink_path(cap_widget, p) { + if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && widget) { + snd_soc_dapm_widget_for_each_sink_path(widget, p) { route.source = conn->sof_dma; route.sink = p->sink->name; snd_soc_dapm_add_routes(&card->dapm, &route, 1); } - } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && - play_widget) { - snd_soc_dapm_widget_for_each_source_path(play_widget, p) { + } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && widget) { + snd_soc_dapm_widget_for_each_source_path(widget, p) { route.source = p->source->name; route.sink = conn->sof_dma; snd_soc_dapm_add_routes(&card->dapm, &route, 1); diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 51f736f319e4..8a309b0734f7 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -183,6 +183,8 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct snd_soc_dapm_widget *p = snd_soc_dai_get_widget_playback(dai); + struct snd_soc_dapm_widget *c = snd_soc_dai_get_widget_capture(dai); unsigned int rate = params_rate(params); unsigned int rate_reg = mt6797_rate_transform(afe->dev, rate, dai->id); unsigned int pcm_con = 0; @@ -193,10 +195,10 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, substream->stream, rate, rate_reg, - dai->playback_widget->active, - dai->capture_widget->active); + p->active, + c->active);
- if (dai->playback_widget->active || dai->capture_widget->active) + if (p->active || c->active) return 0;
switch (dai->id) { diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-pcm.c b/sound/soc/mediatek/mt8183/mt8183-dai-pcm.c index 38ce0e36cdb4..4e25287fc0e4 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-pcm.c @@ -183,6 +183,8 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct snd_soc_dapm_widget *p = snd_soc_dai_get_widget_playback(dai); + struct snd_soc_dapm_widget *c = snd_soc_dai_get_widget_capture(dai); unsigned int rate = params_rate(params); unsigned int rate_reg = mt8183_rate_transform(afe->dev, rate, dai->id); unsigned int pcm_con = 0; @@ -193,10 +195,9 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, substream->stream, rate, rate_reg, - dai->playback_widget->active, - dai->capture_widget->active); + p->active, c->active);
- if (dai->playback_widget->active || dai->capture_widget->active) + if (p->active || c->active) return 0;
switch (dai->id) { diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c index 41221a66111c..a50aa294960b 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c @@ -218,6 +218,8 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, { struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct snd_soc_dapm_widget *p = snd_soc_dai_get_widget_playback(dai); + struct snd_soc_dapm_widget *c = snd_soc_dai_get_widget_capture(dai); int pcm_id = dai->id; struct mtk_afe_pcm_priv *pcm_priv = afe_priv->dai_priv[pcm_id]; unsigned int rate = params_rate(params); @@ -230,12 +232,11 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, unsigned int pcm_con = 0;
dev_dbg(afe->dev, "%s(), id %d, stream %d, widget active p %d, c %d\n", - __func__, dai->id, substream->stream, dai->playback_widget->active, - dai->capture_widget->active); + __func__, dai->id, substream->stream, p->active, c->active); dev_dbg(afe->dev, "%s(), rate %d, rate_reg %d, data_width %d, wlen_width %d\n", __func__, rate, rate_reg, data_width, wlen_width);
- if (dai->playback_widget->active || dai->capture_widget->active) + if (p->active || c->active) return 0;
switch (dai->id) { diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-pcm.c b/sound/soc/mediatek/mt8192/mt8192-dai-pcm.c index 239e3f5b53d3..2847a2e747be 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-pcm.c @@ -273,6 +273,8 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct snd_soc_dapm_widget *p = snd_soc_dai_get_widget_playback(dai); + struct snd_soc_dapm_widget *c = snd_soc_dai_get_widget_capture(dai); unsigned int rate = params_rate(params); unsigned int rate_reg = mt8192_rate_transform(afe->dev, rate, dai->id); unsigned int pcm_con = 0; @@ -283,10 +285,10 @@ static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, substream->stream, rate, rate_reg, - dai->playback_widget->active, - dai->capture_widget->active); + p->active, + c->active);
- if (dai->playback_widget->active || dai->capture_widget->active) + if (p->active || c->active) return 0;
switch (dai->id) { diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c index caceb0deb467..051433689ff5 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c @@ -213,11 +213,14 @@ static int mtk_dai_pcm_configure(struct snd_pcm_substream *substream, static int mtk_dai_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct snd_soc_dapm_widget *p = snd_soc_dai_get_widget_playback(dai); + struct snd_soc_dapm_widget *c = snd_soc_dai_get_widget_capture(dai); + dev_dbg(dai->dev, "%s(), id %d, stream %d, widget active p %d, c %d\n", __func__, dai->id, substream->stream, - dai->playback_widget->active, dai->capture_widget->active); + p->active, c->active);
- if (dai->playback_widget->active || dai->capture_widget->active) + if (p->active || c->active) return 0;
return mtk_dai_pcm_configure(substream, dai);
Il 24/01/23 03:12, Kuninori Morimoto ha scritto:
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Reviewed-by: AngeloGioacchino Del Regno angelogioacchino.delregno@collabora.com
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/meson/aiu-fifo-i2s.c | 4 ++-- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.c | 21 +++++++++--------- sound/soc/meson/axg-tdm-interface.c | 34 +++++++++++++++-------------- sound/soc/meson/axg-tdmin.c | 2 +- sound/soc/meson/axg-tdmout.c | 2 +- sound/soc/meson/meson-codec-glue.c | 8 +++---- 7 files changed, 38 insertions(+), 35 deletions(-)
diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c index 57e6e7160d2f..59e00a74b5f8 100644 --- a/sound/soc/meson/aiu-fifo-i2s.c +++ b/sound/soc/meson/aiu-fifo-i2s.c @@ -88,7 +88,7 @@ static int aiu_fifo_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); unsigned int val; int ret;
@@ -158,7 +158,7 @@ int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai) if (ret) return ret;
- fifo = dai->playback_dma_data; + fifo = snd_soc_dai_dma_data_get_playback(dai);
fifo->pcm = &fifo_i2s_pcm; fifo->mem_offset = AIU_MEM_I2S_START; diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c index 2fb30f89bf7a..ddbd2fc40185 100644 --- a/sound/soc/meson/aiu-fifo-spdif.c +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -173,7 +173,7 @@ int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai) if (ret) return ret;
- fifo = dai->playback_dma_data; + fifo = snd_soc_dai_dma_data_get_playback(dai);
fifo->pcm = &fifo_spdif_pcm; fifo->mem_offset = AIU_MEM_IEC958_START; diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index d67ff4cdabd5..543d41856c12 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -34,7 +34,7 @@ snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_dai *dai = aiu_fifo_dai(substream); - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int addr;
@@ -46,7 +46,7 @@ snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, static void aiu_fifo_enable(struct snd_soc_dai *dai, bool enable) { struct snd_soc_component *component = dai->component; - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); unsigned int en_mask = (AIU_MEM_CONTROL_FILL_EN | AIU_MEM_CONTROL_EMPTY_EN);
@@ -80,7 +80,7 @@ int aiu_fifo_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai);
snd_soc_component_update_bits(component, fifo->mem_offset + AIU_MEM_CONTROL, @@ -98,7 +98,7 @@ int aiu_fifo_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_component *component = dai->component; - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); dma_addr_t end;
/* Setup the fifo boundaries */ @@ -132,7 +132,7 @@ static irqreturn_t aiu_fifo_isr(int irq, void *dev_id) int aiu_fifo_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); int ret;
snd_soc_set_runtime_hwparams(substream, fifo->pcm); @@ -168,7 +168,7 @@ int aiu_fifo_startup(struct snd_pcm_substream *substream, void aiu_fifo_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai);
free_irq(fifo->irq, substream); clk_disable_unprepare(fifo->pclk); @@ -178,7 +178,7 @@ int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_card *card = rtd->card->snd_card; - struct aiu_fifo *fifo = dai->playback_dma_data; + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); size_t size = fifo->pcm->buffer_bytes_max; int ret;
@@ -200,15 +200,16 @@ int aiu_fifo_dai_probe(struct snd_soc_dai *dai) if (!fifo) return -ENOMEM;
- dai->playback_dma_data = fifo; + snd_soc_dai_dma_data_set_playback(dai, fifo);
return 0; }
int aiu_fifo_dai_remove(struct snd_soc_dai *dai) { - kfree(dai->playback_dma_data); + struct aiu_fifo *fifo = snd_soc_dai_dma_data_get_playback(dai); + + kfree(fifo);
return 0; } - diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index c040c83637e0..7624aafe9009 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -37,10 +37,8 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, unsigned int slot_width) { struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); - struct axg_tdm_stream *tx = (struct axg_tdm_stream *) - dai->playback_dma_data; - struct axg_tdm_stream *rx = (struct axg_tdm_stream *) - dai->capture_dma_data; + struct axg_tdm_stream *tx = snd_soc_dai_dma_data_get_playback(dai); + struct axg_tdm_stream *rx = snd_soc_dai_dma_data_get_capture(dai); unsigned int tx_slots, rx_slots; unsigned int fmt = 0;
@@ -362,11 +360,14 @@ static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream,
static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) { - if (dai->capture_dma_data) - axg_tdm_stream_free(dai->capture_dma_data); + int stream;
- if (dai->playback_dma_data) - axg_tdm_stream_free(dai->playback_dma_data); + for_each_pcm_streams(stream) { + struct axg_tdm_stream *ts = snd_soc_dai_dma_data_get(dai, stream); + + if (ts) + axg_tdm_stream_free(ts); + }
return 0; } @@ -374,19 +375,20 @@ static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) { struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int stream;
- if (dai->capture_widget) { - dai->capture_dma_data = axg_tdm_stream_alloc(iface); - if (!dai->capture_dma_data) - return -ENOMEM; - } + for_each_pcm_streams(stream) { + struct axg_tdm_stream *ts; + + if (!snd_soc_dai_get_widget(dai, stream)) + continue;
- if (dai->playback_widget) { - dai->playback_dma_data = axg_tdm_stream_alloc(iface); - if (!dai->playback_dma_data) { + ts = axg_tdm_stream_alloc(iface); + if (!ts) { axg_tdm_iface_remove_dai(dai); return -ENOMEM; } + snd_soc_dai_dma_data_set(dai, stream, ts); }
return 0; diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 49b613a1faf2..c8f6ea24ae78 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -83,7 +83,7 @@ axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w) if (!be) return NULL;
- return be->capture_dma_data; + return snd_soc_dai_dma_data_get_capture(be); }
static void axg_tdmin_enable(struct regmap *map) diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c index 22d519fc07b2..c4039e4f0847 100644 --- a/sound/soc/meson/axg-tdmout.c +++ b/sound/soc/meson/axg-tdmout.c @@ -81,7 +81,7 @@ axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w) if (!be) return NULL;
- return be->playback_dma_data; + return snd_soc_dai_dma_data_get_playback(be); }
static void axg_tdmout_enable(struct regmap *map) diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c index 80c5ed196961..5913486c43ab 100644 --- a/sound/soc/meson/meson-codec-glue.c +++ b/sound/soc/meson/meson-codec-glue.c @@ -39,13 +39,13 @@ meson_codec_glue_get_input(struct snd_soc_dapm_widget *w) static void meson_codec_glue_input_set_data(struct snd_soc_dai *dai, struct meson_codec_glue_input *data) { - dai->playback_dma_data = data; + snd_soc_dai_dma_data_set_playback(dai, data); }
struct meson_codec_glue_input * meson_codec_glue_input_get_data(struct snd_soc_dai *dai) { - return dai->playback_dma_data; + return snd_soc_dai_dma_data_get_playback(dai); } EXPORT_SYMBOL_GPL(meson_codec_glue_input_get_data);
@@ -99,8 +99,8 @@ int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct meson_codec_glue_input *in_data = - meson_codec_glue_output_get_input_data(dai->capture_widget); + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget_capture(dai); + struct meson_codec_glue_input *in_data = meson_codec_glue_output_get_input_data(w);
if (!in_data) return -ENODEV;
On Tue 24 Jan 2023 at 02:13, Kuninori Morimoto kuninori.morimoto.gx@renesas.com wrote:
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Reviewed-by: Jerome Brunet jbrunet@baylibre.com
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 ++-- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 08b90ec5cc80..166257c6ae14 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1070,9 +1070,9 @@ static int rockchip_i2s_tdm_dai_probe(struct snd_soc_dai *dai) struct rk_i2s_tdm_dev *i2s_tdm = snd_soc_dai_get_drvdata(dai);
if (i2s_tdm->has_capture) - dai->capture_dma_data = &i2s_tdm->capture_dma_data; + snd_soc_dai_dma_data_set_capture(dai, &i2s_tdm->capture_dma_data); if (i2s_tdm->has_playback) - dai->playback_dma_data = &i2s_tdm->playback_dma_data; + snd_soc_dai_dma_data_set_playback(dai, &i2s_tdm->playback_dma_data);
if (i2s_tdm->mclk_calibrate) snd_soc_add_dai_controls(dai, &rockchip_i2s_tdm_compensation_control, 1); diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 5b1e47bdc376..6ce92b1db790 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -373,7 +373,7 @@ static int rockchip_pdm_dai_probe(struct snd_soc_dai *dai) { struct rk_pdm_dev *pdm = to_info(dai);
- dai->capture_dma_data = &pdm->capture_dma_data; + snd_soc_dai_dma_data_set_capture(dai, &pdm->capture_dma_data);
return 0; } diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 5b4f00457587..2d937fcf357d 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -196,7 +196,7 @@ static int rk_spdif_dai_probe(struct snd_soc_dai *dai) { struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
- dai->playback_dma_data = &spdif->playback_dma_data; + snd_soc_dai_dma_data_set_playback(dai, &spdif->playback_dma_data);
return 0; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sof/topology.c | 104 ++++++++++++++------------------------- 1 file changed, 38 insertions(+), 66 deletions(-)
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 560771ba8fb9..e9a2d4a240b3 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1055,6 +1055,7 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, struct snd_soc_card *card = scomp->card; struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *cpu_dai; + int stream; int i;
if (!w->sname) { @@ -1062,62 +1063,41 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, return -EINVAL; }
+ if (w->id == snd_soc_dapm_dai_out) + stream = SNDRV_PCM_STREAM_CAPTURE; + if (w->id == snd_soc_dapm_dai_in) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + goto end; + list_for_each_entry(rtd, &card->rtd_list, list) { /* does stream match DAI link ? */ if (!rtd->dai_link->stream_name || strcmp(w->sname, rtd->dai_link->stream_name)) continue;
- switch (w->id) { - case snd_soc_dapm_dai_out: - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - /* - * Please create DAI widget in the right order - * to ensure BE will connect to the right DAI - * widget. - */ - if (!cpu_dai->capture_widget) { - cpu_dai->capture_widget = w; - break; - } - } - if (i == rtd->dai_link->num_cpus) { - dev_err(scomp->dev, "error: can't find BE for DAI %s\n", - w->name); - - return -EINVAL; - } - dai->name = rtd->dai_link->name; - dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", - w->name, rtd->dai_link->name); - break; - case snd_soc_dapm_dai_in: - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - /* - * Please create DAI widget in the right order - * to ensure BE will connect to the right DAI - * widget. - */ - if (!cpu_dai->playback_widget) { - cpu_dai->playback_widget = w; - break; - } + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + /* + * Please create DAI widget in the right order + * to ensure BE will connect to the right DAI + * widget. + */ + if (!snd_soc_dai_get_widget(cpu_dai, stream)) { + snd_soc_dai_set_widget(cpu_dai, stream, w); + break; } - if (i == rtd->dai_link->num_cpus) { - dev_err(scomp->dev, "error: can't find BE for DAI %s\n", - w->name); + } + if (i == rtd->dai_link->num_cpus) { + dev_err(scomp->dev, "error: can't find BE for DAI %s\n", w->name);
- return -EINVAL; - } - dai->name = rtd->dai_link->name; - dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", - w->name, rtd->dai_link->name); - break; - default: - break; + return -EINVAL; } - }
+ dai->name = rtd->dai_link->name; + dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", + w->name, rtd->dai_link->name); + } +end: /* check we have a connection */ if (!dai->name) { dev_err(scomp->dev, "error: can't connect DAI %s stream %s\n", @@ -1134,37 +1114,29 @@ static void sof_disconnect_dai_widget(struct snd_soc_component *scomp, struct snd_soc_card *card = scomp->card; struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *cpu_dai; - int i; + int i, stream;
if (!w->sname) return;
+ if (w->id == snd_soc_dapm_dai_out) + stream = SNDRV_PCM_STREAM_CAPTURE; + else if (w->id == snd_soc_dapm_dai_in) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + return; + list_for_each_entry(rtd, &card->rtd_list, list) { /* does stream match DAI link ? */ if (!rtd->dai_link->stream_name || strcmp(w->sname, rtd->dai_link->stream_name)) continue;
- switch (w->id) { - case snd_soc_dapm_dai_out: - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (cpu_dai->capture_widget == w) { - cpu_dai->capture_widget = NULL; - break; - } - } - break; - case snd_soc_dapm_dai_in: - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (cpu_dai->playback_widget == w) { - cpu_dai->playback_widget = NULL; - break; - } + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (snd_soc_dai_get_widget(cpu_dai, stream) == w) { + snd_soc_dai_set_widget(cpu_dai, stream, NULL); + break; } - break; - default: - break; - } } }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/tegra/tegra20_ac97.c | 4 ++-- sound/soc/tegra/tegra20_i2s.c | 4 ++-- sound/soc/tegra/tegra20_spdif.c | 3 +-- sound/soc/tegra/tegra210_admaif.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 5 files changed, 9 insertions(+), 10 deletions(-)
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 87facfbcdd11..d23d88a10899 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -212,8 +212,8 @@ static int tegra20_ac97_probe(struct snd_soc_dai *dai) { struct tegra20_ac97 *ac97 = snd_soc_dai_get_drvdata(dai);
- dai->capture_dma_data = &ac97->capture_dma_data; - dai->playback_dma_data = &ac97->playback_dma_data; + snd_soc_dai_init_dma_data(dai, &ac97->playback_dma_data, + &ac97->capture_dma_data);
return 0; } diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index fff0cd6588f5..d37a9f2603e8 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -256,8 +256,8 @@ static int tegra20_i2s_probe(struct snd_soc_dai *dai) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- dai->capture_dma_data = &i2s->capture_dma_data; - dai->playback_dma_data = &i2s->playback_dma_data; + snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, + &i2s->capture_dma_data);
return 0; } diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index ca7b222e07d0..86bef54dfdf2 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -236,8 +236,7 @@ static int tegra20_spdif_probe(struct snd_soc_dai *dai) { struct tegra20_spdif *spdif = dev_get_drvdata(dai->dev);
- dai->capture_dma_data = NULL; - dai->playback_dma_data = &spdif->playback_dma_data; + snd_soc_dai_init_dma_data(dai, &spdif->playback_dma_data, NULL);
return 0; } diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 1a2e868a6220..100a2b6e6063 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -544,8 +544,8 @@ static int tegra_admaif_dai_probe(struct snd_soc_dai *dai) { struct tegra_admaif *admaif = snd_soc_dai_get_drvdata(dai);
- dai->capture_dma_data = &admaif->capture_dma_data[dai->id]; - dai->playback_dma_data = &admaif->playback_dma_data[dai->id]; + snd_soc_dai_init_dma_data(dai, &admaif->playback_dma_data[dai->id], + &admaif->capture_dma_data[dai->id]);
return 0; } diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 10cd37096fb3..c26f960c6afd 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -297,8 +297,8 @@ static int tegra30_i2s_probe(struct snd_soc_dai *dai) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- dai->capture_dma_data = &i2s->capture_dma_data; - dai->playback_dma_data = &i2s->playback_dma_data; + snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, + &i2s->capture_dma_data);
return 0; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/ti/davinci-i2s.c | 5 +++-- sound/soc/ti/davinci-mcasp.c | 5 +++-- sound/soc/ti/davinci-vcif.c | 5 +++-- 3 files changed, 9 insertions(+), 6 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index e6e77a5f3c1e..3ccd0cfca008 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -614,9 +614,10 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + int stream;
- dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + for_each_pcm_streams(stream) + snd_soc_dai_dma_data_set(dai, stream, &dev->dma_data[stream]);
return 0; } diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index f5ac2ab77f5b..578254549d2d 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1699,9 +1699,10 @@ static void davinci_mcasp_init_iec958_status(struct davinci_mcasp *mcasp) static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + int stream;
- dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + for_each_pcm_streams(stream) + snd_soc_dai_dma_data_set(dai, stream, &mcasp->dma_data[stream]);
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) { davinci_mcasp_init_iec958_status(mcasp); diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c index 36fa97e2b9e2..fc16b3b8f71a 100644 --- a/sound/soc/ti/davinci-vcif.c +++ b/sound/soc/ti/davinci-vcif.c @@ -161,9 +161,10 @@ static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) { struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + int stream;
- dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + for_each_pcm_streams(stream) + snd_soc_dai_dma_data_set(dai, stream, &dev->dma_data[stream]);
return 0; }
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e613698824fd..71b022f7edfd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -553,7 +553,7 @@ int snd_soc_suspend(struct device *dev) int i;
/* If the card is not initialized yet there is nothing to do */ - if (!card->instantiated) + if (!snd_soc_card_is_instantiated(card)) return 0;
/* @@ -695,7 +695,7 @@ int snd_soc_resume(struct device *dev) struct snd_soc_component *component;
/* If the card is not initialized yet there is nothing to do */ - if (!card->instantiated) + if (!snd_soc_card_is_instantiated(card)) return 0;
/* activate pins from sleep state */ @@ -1915,7 +1915,7 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card)
static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) { - if (card->instantiated) { + if (snd_soc_card_is_instantiated(card)) { card->instantiated = false; snd_soc_flush_all_delayed_work(card);
@@ -2126,7 +2126,7 @@ int snd_soc_poweroff(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_component *component;
- if (!card->instantiated) + if (!snd_soc_card_is_instantiated(card)) return 0;
/*
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- sound/soc/soc-dapm.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f61c8633e7eb..5d9a671e50f1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -149,7 +149,7 @@ static int dapm_down_seq[] = {
static void dapm_assert_locked(struct snd_soc_dapm_context *dapm) { - if (dapm->card && dapm->card->instantiated) + if (snd_soc_card_is_instantiated(dapm->card)) lockdep_assert_held(&dapm->card->dapm_mutex); }
@@ -1297,7 +1297,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); LIST_HEAD(widgets); int paths; int ret; @@ -1305,12 +1305,10 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - w = dai->playback_widget; invalidate_paths_ep(w, SND_SOC_DAPM_DIR_OUT); paths = is_connected_output_ep(w, &widgets, custom_stop_condition); } else { - w = dai->capture_widget; invalidate_paths_ep(w, SND_SOC_DAPM_DIR_IN); paths = is_connected_input_ep(w, &widgets, custom_stop_condition); @@ -2614,7 +2612,7 @@ int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm) * Suppress early reports (eg, jacks syncing their state) to avoid * silly DAPM runs during card startup. */ - if (!dapm->card || !dapm->card->instantiated) + if (!snd_soc_card_is_instantiated(dapm->card)) return 0;
return dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP); @@ -2908,7 +2906,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(path->node[dir], "Route added"); }
- if (dapm->card->instantiated && path->connect) + if (snd_soc_card_is_instantiated(dapm->card) && path->connect) dapm_path_invalidate(path);
return 0; @@ -4229,7 +4227,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, return PTR_ERR(w);
w->priv = dai; - dai->playback_widget = w; + snd_soc_dai_set_widget_playback(dai, w); }
if (dai->driver->capture.stream_name) { @@ -4245,7 +4243,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, return PTR_ERR(w);
w->priv = dai; - dai->capture_widget = w; + snd_soc_dai_set_widget_capture(dai, w); }
return 0; @@ -4339,16 +4337,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, int stream;
if (dai_link->params) { - playback_cpu = cpu_dai->capture_widget; - capture_cpu = cpu_dai->playback_widget; + playback_cpu = snd_soc_dai_get_widget_capture(cpu_dai); + capture_cpu = snd_soc_dai_get_widget_playback(cpu_dai); } else { - playback_cpu = cpu_dai->playback_widget; - capture_cpu = cpu_dai->capture_widget; + playback_cpu = snd_soc_dai_get_widget_playback(cpu_dai); + capture_cpu = snd_soc_dai_get_widget_capture(cpu_dai); }
/* connect BE DAI playback if widgets are valid */ stream = SNDRV_PCM_STREAM_PLAYBACK; - codec = codec_dai->playback_widget; + codec = snd_soc_dai_get_widget(codec_dai, stream);
if (playback_cpu && codec) { if (dai_link->params && !rtd->c2c_widget[stream]) { @@ -4367,7 +4365,7 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, capture: /* connect BE DAI capture if widgets are valid */ stream = SNDRV_PCM_STREAM_CAPTURE; - codec = codec_dai->capture_widget; + codec = snd_soc_dai_get_widget(codec_dai, stream);
if (codec && capture_cpu) { if (dai_link->params && !rtd->c2c_widget[stream]) {
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Link: https://lore.kernel.org/all/6f047ec5-4055-761d-c1ea-c2d0b606e53a@linux.intel... Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- v1 -> v2 - use pointer for tx_mask/rx_mast to avoid overwrite
sound/soc/soc-dai.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 29a75fdf90e0..69f534f0d4bf 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -267,6 +267,11 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { int ret = -ENOTSUPP; + int stream; + unsigned int *tdm_mask[] = { + &tx_mask, + &rx_mask, + };
if (dai->driver->ops && dai->driver->ops->xlate_tdm_slot_mask) @@ -275,8 +280,8 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, else snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
- dai->tx_mask = tx_mask; - dai->rx_mask = rx_mask; + for_each_pcm_streams(stream) + snd_soc_dai_tdm_mask_set(dai, stream, *tdm_mask[stream]);
if (dai->driver->ops && dai->driver->ops->set_tdm_slot)
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- sound/soc/soc-pcm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 82bb46c7f5cc..8023cc1fe066 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1012,6 +1012,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
for_each_rtd_codec_dais(rtd, i, codec_dai) { struct snd_pcm_hw_params codec_params; + unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream);
/* * Skip CODECs which don't support the current stream type, @@ -1034,15 +1035,8 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, codec_params = *params;
/* fixup params based on TDM slot masks */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->tx_mask) - soc_pcm_codec_params_fixup(&codec_params, - codec_dai->tx_mask); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && - codec_dai->rx_mask) - soc_pcm_codec_params_fixup(&codec_params, - codec_dai->rx_mask); + if (tdm_mask) + soc_pcm_codec_params_fixup(&codec_params, tdm_mask);
ret = snd_soc_dai_hw_params(codec_dai, substream, &codec_params);
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ASoC has many helper function. This patch use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/soc-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c3be24b2fac5..6bb3efcee3b9 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1468,7 +1468,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg,
/* card dapm mutex is held by the core if we are loading topology * data during sound card init. */ - if (card->instantiated) + if (snd_soc_card_is_instantiated(card)) widget = snd_soc_dapm_new_control(dapm, &template); else widget = snd_soc_dapm_new_control_unlocked(dapm, &template); @@ -1556,7 +1556,7 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) /* Card might not have been registered at this point. * If so, just return success. */ - if (!card || !card->instantiated) { + if (!snd_soc_card_is_instantiated(card)) { dev_warn(tplg->dev, "ASoC: Parent card not yet available," " widget card binding deferred\n"); return 0;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current snd_soc_dai has data for Playback/Capture, but it is very random. Someone is array (A), someone is playback/capture (B), and someone is tx/rx (C);
struct snd_soc_dai { ... (A) unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1];
(B) struct snd_soc_dapm_widget *playback_widget; (B) struct snd_soc_dapm_widget *capture_widget;
(B) void *playback_dma_data; (B) void *capture_dma_data;
...
(C) unsigned int tx_mask; (C) unsigned int rx_mask; };
Because of it, the code was very complicated. This patch creates new data structure to merge these into one, and tidyup the code.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Reviewed-by: Charles Keepax ckeepax@opensource.cirrus.com --- include/sound/soc-dai.h | 68 ++++++++++++++++------------------------- sound/soc/soc-dai.c | 4 +-- 2 files changed, 28 insertions(+), 44 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 197dc1629708..e3906ecda740 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -423,6 +423,16 @@ struct snd_soc_dai_driver { int remove_order; };
+/* for Playback/Capture */ +struct snd_soc_dai_stream { + struct snd_soc_dapm_widget *widget; + + unsigned int active; /* usage count */ + unsigned int tdm_mask; /* CODEC TDM slot masks and params (for fixup) */ + + void *dma_data; /* DAI DMA data */ +}; + /* * Digital Audio Interface runtime data. * @@ -437,14 +447,7 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver;
/* DAI runtime info */ - unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */ - - struct snd_soc_dapm_widget *playback_widget; - struct snd_soc_dapm_widget *capture_widget; - - /* DAI DMA data */ - void *playback_dma_data; - void *capture_dma_data; + struct snd_soc_dai_stream stream[SNDRV_PCM_STREAM_LAST + 1];
/* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; @@ -454,10 +457,6 @@ struct snd_soc_dai { /* parent platform/codec */ struct snd_soc_component *component;
- /* CODEC TDM slot masks and params (for fixup) */ - unsigned int tx_mask; - unsigned int rx_mask; - struct list_head list;
/* function mark */ @@ -482,8 +481,7 @@ snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) static inline struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream) { - return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dai->playback_widget : dai->capture_widget; + return dai->stream[stream].widget; }
#define snd_soc_dai_set_widget_playback(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_PLAYBACK, widget) @@ -491,10 +489,7 @@ struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int static inline void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_widget = widget; - else - dai->capture_widget = widget; + dai->stream[stream].widget = widget; }
#define snd_soc_dai_dma_data_get_playback(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_PLAYBACK) @@ -502,8 +497,7 @@ void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_ #define snd_soc_dai_get_dma_data(dai, ss) snd_soc_dai_dma_data_get(dai, ss->stream) static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int stream) { - return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dai->playback_dma_data : dai->capture_dma_data; + return dai->stream[stream].dma_data; }
#define snd_soc_dai_dma_data_set_playback(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_PLAYBACK, data) @@ -511,34 +505,30 @@ static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int #define snd_soc_dai_set_dma_data(dai, ss, data) snd_soc_dai_dma_data_set(dai, ss->stream, data) static inline void snd_soc_dai_dma_data_set(struct snd_soc_dai *dai, int stream, void *data) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = data; - else - dai->capture_dma_data = data; + dai->stream[stream].dma_data = data; }
-static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, - void *playback, void *capture) +static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *playback, void *capture) { - dai->playback_dma_data = playback; - dai->capture_dma_data = capture; + snd_soc_dai_dma_data_set_playback(dai, playback); + snd_soc_dai_dma_data_set_capture(dai, capture); }
static inline unsigned int snd_soc_dai_tdm_mask_get(struct snd_soc_dai *dai, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - return dai->tx_mask; - else - return dai->rx_mask; + return dai->stream[stream].tdm_mask; }
static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream, unsigned int tdm_mask) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->tx_mask = tdm_mask; - else - dai->rx_mask = tdm_mask; + dai->stream[stream].tdm_mask = tdm_mask; +} + +static inline unsigned int snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream) +{ + /* see snd_soc_dai_action() for setup */ + return dai->stream[stream].active; }
static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, @@ -593,10 +583,4 @@ static inline void *snd_soc_dai_get_stream(struct snd_soc_dai *dai, return ERR_PTR(-ENOTSUPP); }
-static inline unsigned int -snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream) -{ - return dai->stream_active[stream]; -} - #endif diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 69f534f0d4bf..0119afbd01fc 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -514,7 +514,7 @@ void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action) { /* see snd_soc_dai_stream_active() */ - dai->stream_active[stream] += action; + dai->stream[stream].active += action;
/* see snd_soc_component_active() */ dai->component->active += action; @@ -527,7 +527,7 @@ int snd_soc_dai_active(struct snd_soc_dai *dai)
active = 0; for_each_pcm_streams(stream) - active += dai->stream_active[stream]; + active += dai->stream[stream].active;
return active; }
On Tue, 24 Jan 2023 02:07:19 +0000, Kuninori Morimoto wrote:
These are v2 patch-set to use helper function on ASoC.
I'm posting about DT schema fixup, but it seems take longer time. This patch-set is 100% independent from it, so I will post it.
struct snd_soc_dai need to have info for playback/capture, but it is using "playback/capture_xxx" or "tx/tx_xxx" or array. This kind of random definition is very difficult to read.
[...]
Applied to
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
Thanks!
[01/25] ASoC: soc-dai.h: add missing snd_soc_dai_set_widget() commit: 37ed123689968fa7f733598d9a6966e8a2db66fb [02/25] ASoC: soc-dai.h: add snd_soc_dai_dma_data_set/get() for low level commit: 60df5935c27e19cc6595bd1740479a60f1880a32 [03/25] ASoC: soc-dai.h: add snd_soc_dai_tdm_mask_set/get() helper commit: d4c0326e6ea038cf99fde6f22296e4e68de4f2b5 [04/25] ASoC: soc-dai.h: add snd_soc_dai_get/set_widget_playback/capture() helper commit: 944d721ee5defb376410c7054a7db0a0ef5ff584 [05/25] ASoC: soc.h: add snd_soc_card_is_instantiated() helper commit: 2e5f86174e31cc1eda92b308f23f3b29684b339e [06/25] ASoC: hdmi-codec: use helper function commit: c59331b39045fcc9ed3c66cb5c80a8ef3ade820f [07/25] ASoC: hda: use helper function commit: 5bf13408fd06068aeef751597748c9d0569c3708 [08/25] ASoC: max: use helper function commit: 812c2852c59f1f532ea9b53d9bbe86231dd83fea [09/25] ASoC: rt: use helper function commit: 8ec352362848d8cc9500ccfb051810597c0abf8a [10/25] ASoC: cirrus: use helper function commit: 0e478b88b257049e1e22077b880419a431595645 [11/25] ASoC: spear: use helper function commit: c4d78c4127f9371dcfc77e71417addddcc5a9533 [12/25] ASoC: sdw-mockup: use helper function commit: e87abb83fc24a481ba490d8596969e1de3a16384 [13/25] ASoC: intel: use helper function commit: ec4b2099c6e01342e5c3289a906615c3abef85b8 [14/25] ASoC: mediatek: use helper function commit: 0df2ec8e4706a6adc36d12cc5cf8e1fa7485fbf5 [15/25] ASoC: meson: use helper function commit: c765ceda78f0bd9df1217f9beaefea58ecf3865c [16/25] ASoC: rockchip: use helper function commit: 67cc242679b106909be52f8476e372dbb03c0a88 [17/25] ASoC: sof: use helper function commit: 323f09a61d43da2aff9b71e8a4764f4c9a88a5e0 [18/25] ASoC: tegra: use helper function commit: 1c1a778401b8ad04ae618eb8baa9a34a8b7e3d50 [19/25] ASoC: ti: use helper function commit: 2abde57fb82b4259b790212135b2194f2fd4dc7c [20/25] ASoC: soc-core.c: use helper function commit: 9024bae4bb229c38522bb8c5dc45b82f9c27e0fc [21/25] ASoC: soc-dapm.c: use helper function commit: 6855ec72102921814aaa43eab305b541af3e9757 [22/25] ASoC: soc-dai.c: use helper function commit: 8ede4b7109f618b3267a9577083b38984857a267 [23/25] ASoC: soc-pcm.c: use helper function commit: e15ff262e212387a05316dc465d754c0a62d9691 [24/25] ASoC: soc-topology.c: use helper function commit: 2b34c1359cca4812d7c8253ff70c394871c22590 [25/25] ASoC: soc-dai.h: cleanup Playback/Capture data for snd_soc_dai commit: 3653480c68120dc16ebfeb80e529200dbbd98f92
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
participants (4)
-
AngeloGioacchino Del Regno -
Jerome Brunet -
Kuninori Morimoto -
Mark Brown