Merge tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Many updates in this development cycle are found in ASoC where it got
  a wide range of changes for the continued refactoring.

  Some highlights are below.

  ASoC:

   - Continued refactoring work by Morimoto-san toward the full
     componentization; the changes are seen allover the places

   - Support for force disconnecting muxes in DAPM

   - Continued development of ASoC Intel SOF stuff

   - New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
     CX2072X, Realtek RT1011 and RT1308

  HD-audio:

   - More fixes and adjustments for ASoC SOF HD-audio

   - Fix for resume problem on some Realtek codecs

  USB-audio:

   - A few fixes for the issues reported by syzbot USB fuzzer

   - Fix for UAC2 extension unit parser

   - Quirks for Line6 Helix, Emgaic Unitor 8

  FireWire:

   - Lots of code refactoring and fixes in most of its components"

* tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (626 commits)
  ALSA: firewire-lib: code refactoring for local variables
  ALSA: firewire-lib: code refactoring for post operation to data block counter
  ALSA: firewire-lib: code refactoring for error path of parser for CIP header
  ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet
  ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT
  ALSA: firewire-lib/fireface: fix initial value of data block counter for IR context with CIP_NO_HEADER
  ALSA: firewire-lib: fix invalid length of rx packet payload for tracepoint events
  ALSA: usb-audio: fix Line6 Helix audio format rates
  firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: bebob: fix wrong reference count for stream functionality at error path of rawmidi interface
  ASoC: SOF: Intel: implement runtime idle for CNL/APL
  ASoC: SOF: add runtime idle callback
  ASoC: hdac_hdmi: report codec link up/down status to bus
  ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
  ASoC: sunxi: sun50i-codec-analog: Add earpiece
  ASoC: rt5665: remove redundant assignment to variable idx
  ...

1 files changed, 20 insertions, 12 deletions

diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 613b37172441..33eb72545be6 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -329,41 +329,48 @@ static struct snd_soc_aux_dev cht_max98090_headset_dev = {
 	.codec_name = "i2c-104C227E:00",
 };
 
+SND_SOC_DAILINK_DEF(dummy,
+	DAILINK_COMP_ARRAY(COMP_DUMMY()));
+
+SND_SOC_DAILINK_DEF(media,
+	DAILINK_COMP_ARRAY(COMP_CPU("media-cpu-dai")));
+
+SND_SOC_DAILINK_DEF(deepbuffer,
+	DAILINK_COMP_ARRAY(COMP_CPU("deepbuffer-cpu-dai")));
+
+SND_SOC_DAILINK_DEF(ssp2_port,
+	DAILINK_COMP_ARRAY(COMP_CPU("ssp2-port")));
+SND_SOC_DAILINK_DEF(ssp2_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-193C9890:00", "HiFi")));
+
+SND_SOC_DAILINK_DEF(platform,
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("sst-mfld-platform")));
+
 static struct snd_soc_dai_link cht_dailink[] = {
 	[MERR_DPCM_AUDIO] = {
 		.name = "Audio Port",
 		.stream_name = "Audio",
-		.cpu_dai_name = "media-cpu-dai",
-		.codec_dai_name = "snd-soc-dummy-dai",
-		.codec_name = "snd-soc-dummy",
-		.platform_name = "sst-mfld-platform",
 		.nonatomic = true,
 		.dynamic = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &cht_aif1_ops,
+		SND_SOC_DAILINK_REG(media, dummy, platform),
 	},
 	[MERR_DPCM_DEEP_BUFFER] = {
 		.name = "Deep-Buffer Audio Port",
 		.stream_name = "Deep-Buffer Audio",
-		.cpu_dai_name = "deepbuffer-cpu-dai",
-		.codec_dai_name = "snd-soc-dummy-dai",
-		.codec_name = "snd-soc-dummy",
-		.platform_name = "sst-mfld-platform",
 		.nonatomic = true,
 		.dynamic = 1,
 		.dpcm_playback = 1,
 		.ops = &cht_aif1_ops,
+		SND_SOC_DAILINK_REG(deepbuffer, dummy, platform),
 	},
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
 		.id = 0,
-		.cpu_dai_name = "ssp2-port",
-		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
-		.codec_dai_name = "HiFi",
-		.codec_name = "i2c-193C9890:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 					| SND_SOC_DAIFMT_CBS_CFS,
 		.init = cht_codec_init,
@@ -371,6 +378,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &cht_be_ssp2_ops,
+		SND_SOC_DAILINK_REG(ssp2_port, ssp2_codec, platform),
 	},
 };