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authorLinus Torvalds <torvalds@linux-foundation.org>2020-04-02 15:50:04 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2020-04-02 15:50:04 -0700
commit848960e576dafc8ed54c691b2f70b92e1fdea9ba (patch)
tree27ea80003da03b81f0b188d3712f0194745126d9 /sound/soc/fsl/fsl-asoc-card.c
parentbc3b3f4bfbded031a11c4284106adddbfacd05bb (diff)
parent5c6cd7021a05a02fcf37f360592d7c18d4d807fb (diff)
Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This became again a busy development cycle. There are few ALSA core updates (merely API cleanups and sparse fixes), with the majority of other changes are found in ASoC scene. Here are some highlights: ALSA core: - More helper macros for sparse warning fixes (e.g. bitwise types) - Slight optimization of PCM OSS locks - Make common handling for PCM / compress buffers (for SOF) ASoC: - Lots of code refactoring and modernization for (still ongoing) componentization works - Conversion of SND_SOC_ALL_CODECS to use imply - Continued refactoring and fixing of the Intel SOF/SST support, including the initial (but still incomplete) SoundWire support - SoundWire and more advanced clocking support for Realtek RT5682 - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140 HD-audio: - Optimizations in HDMI jack handling - A few new quirks and fixups for Realtek codecs USB-audio: - Delayed registration support - New quirks for Motu, Kingston, Presonus" * tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits) ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h" ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - a fake key event is triggered by running shutup ALSA: hda: default enable CA0132 DSP support ASoC: amd: acp3x-pcm-dma: clean up two indentation issues ASoC: tlv320adcx140: Remove undocumented property ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: boards: add sof_sdw machine driver ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms ASoC: rt5682: move DAI clock registry to I2S mode ASoC: pxa: magician: convert to use i2c_new_client_device() ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers ...
Diffstat (limited to 'sound/soc/fsl/fsl-asoc-card.c')
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c10
1 files changed, 5 insertions, 5 deletions
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 9ce55feaac22..bb33601fab84 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -159,7 +159,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
return 0;
/* Specific configurations of DAIs starts from here */
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
cpu_priv->sysclk_freq[tx],
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
@@ -168,7 +168,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
}
if (cpu_priv->slot_width) {
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
cpu_priv->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set TDM slot for cpu dai\n");
@@ -257,7 +257,7 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
@@ -446,14 +446,14 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_pcm_runtime *rtd = list_first_entry(
&card->rtd_list, struct snd_soc_pcm_runtime, list);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = card->dev;
int ret;
if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
/*