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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2019-07-09 19:59:43 +0300 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2019-07-09 19:59:43 +0300 |
| commit | 4cdd5f9186bbe80306e76f11da7ecb0b9720433c (patch) | |
| tree | 23c2f39933cd8253a65385eab00405beaf602f01 /sound/soc/fsl/imx-audmix.c | |
| parent | 2d41ef5432b76ae90dc0db93026f1d981f874ec4 (diff) | |
| parent | 0dcb4efb1095d0a1f5f681c2b94e98b009cc5d77 (diff) | |
| download | linux-4cdd5f9186bbe80306e76f11da7ecb0b9720433c.tar.xz | |
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
...
Diffstat (limited to 'sound/soc/fsl/imx-audmix.c')
| -rw-r--r-- | sound/soc/fsl/imx-audmix.c | 45 |
1 files changed, 35 insertions, 10 deletions
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 9aaf3e5b45b9..9e1cb18859ce 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -205,6 +205,15 @@ static int imx_audmix_probe(struct platform_device *pdev) return -ENOMEM; for (i = 0; i < num_dai; i++) { + struct snd_soc_dai_link_component *dlc; + + /* for CPU/Codec/Platform x 2 */ + dlc = devm_kzalloc(&pdev->dev, 6 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) { + dev_err(&pdev->dev, "failed to allocate dai_link\n"); + return -ENOMEM; + } + ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, &args); if (ret < 0) { @@ -231,13 +240,21 @@ static int imx_audmix_probe(struct platform_device *pdev) dai_name, "CPU-Capture"); } + priv->dai[i].cpus = &dlc[0]; + priv->dai[i].codecs = &dlc[1]; + priv->dai[i].platforms = &dlc[2]; + + priv->dai[i].num_cpus = 1; + priv->dai[i].num_codecs = 1; + priv->dai[i].num_platforms = 1; + priv->dai[i].name = dai_name; priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; - priv->dai[i].codec_dai_name = "snd-soc-dummy-dai"; - priv->dai[i].codec_name = "snd-soc-dummy"; - priv->dai[i].cpu_of_node = args.np; - priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev); - priv->dai[i].platform_of_node = args.np; + priv->dai[i].codecs->dai_name = "snd-soc-dummy-dai"; + priv->dai[i].codecs->name = "snd-soc-dummy"; + priv->dai[i].cpus->of_node = args.np; + priv->dai[i].cpus->dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platforms->of_node = args.np; priv->dai[i].dynamic = 1; priv->dai[i].dpcm_playback = 1; priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); @@ -252,12 +269,20 @@ static int imx_audmix_probe(struct platform_device *pdev) be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, "AUDMIX-Capture-%d", i); + priv->dai[num_dai + i].cpus = &dlc[3]; + priv->dai[num_dai + i].codecs = &dlc[4]; + priv->dai[num_dai + i].platforms = &dlc[5]; + + priv->dai[num_dai + i].num_cpus = 1; + priv->dai[num_dai + i].num_codecs = 1; + priv->dai[num_dai + i].num_platforms = 1; + priv->dai[num_dai + i].name = be_name; - priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai"; - priv->dai[num_dai + i].codec_name = "snd-soc-dummy"; - priv->dai[num_dai + i].cpu_of_node = audmix_np; - priv->dai[num_dai + i].cpu_dai_name = be_name; - priv->dai[num_dai + i].platform_name = "snd-soc-dummy"; + priv->dai[num_dai + i].codecs->dai_name = "snd-soc-dummy-dai"; + priv->dai[num_dai + i].codecs->name = "snd-soc-dummy"; + priv->dai[num_dai + i].cpus->of_node = audmix_np; + priv->dai[num_dai + i].cpus->dai_name = be_name; + priv->dai[num_dai + i].platforms->name = "snd-soc-dummy"; priv->dai[num_dai + i].no_pcm = 1; priv->dai[num_dai + i].dpcm_playback = 1; priv->dai[num_dai + i].dpcm_capture = 1; |
