[PATCH 00/25] ASoC: use helper function and cleanup
Hi Mark
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.
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_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 --- 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;
On Fri, Jan 20, 2023 at 07:01:45AM +0000, Kuninori Morimoto wrote:
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
Thanks, Charles
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 --- 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) {
On Fri, Jan 20, 2023 at 07:01:52AM +0000, Kuninori Morimoto wrote:
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
Thanks, Charles
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 --- 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) {
On Fri, Jan 20, 2023 at 07:02:06AM +0000, Kuninori Morimoto wrote:
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
Thanks, Charles
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 --- 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;
On Fri, Jan 20, 2023 at 07:02:13AM +0000, Kuninori Morimoto wrote:
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
Thanks, Charles
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; }
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);
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;
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 --- 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;
/*
On Fri, Jan 20, 2023 at 07:03:44AM +0000, Kuninori Morimoto 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: Charles Keepax ckeepax@opensource.cirrus.com
Thanks, Charles
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-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]) {
On Fri, Jan 20, 2023 at 07:03:51AM +0000, Kuninori Morimoto 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: Charles Keepax ckeepax@opensource.cirrus.com
Thanks, Charles
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-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..e01b87ea04d4 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)
On 1/20/2023 8:03 AM, Kuninori Morimoto 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
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..e01b87ea04d4 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]);
Does this work? There are functions earlier which modify the values of tx_mask/rx_mask and you copy their old values on tdm_mask[] init?
if (dai->driver->ops && dai->driver->ops->set_tdm_slot)
Hi
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
(snip)
Does this work? There are functions earlier which modify the values of tx_mask/rx_mask and you copy their old values on tdm_mask[] init?
Oh, yes, indeed. Thank you for poining it. I will fix it in v2.
Thank you for your help !!
Best regards --- Kuninori Morimoto
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-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);
On Fri, Jan 20, 2023 at 07:04:03AM +0000, Kuninori Morimoto 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: Charles Keepax ckeepax@opensource.cirrus.com
Thanks, Charles
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 --- 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 e01b87ea04d4..906819d441ba 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 Fri, Jan 20, 2023 at 07:04:14AM +0000, Kuninori Morimoto wrote:
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
Thanks, Charles
participants (3)
-
Amadeusz Sławiński -
Charles Keepax -
Kuninori Morimoto