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authorLinus Torvalds <torvalds@linux-foundation.org>2019-07-09 19:59:43 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2019-07-09 19:59:43 +0300
commit4cdd5f9186bbe80306e76f11da7ecb0b9720433c (patch)
tree23c2f39933cd8253a65385eab00405beaf602f01 /sound/soc/fsl/imx-audmix.c
parent2d41ef5432b76ae90dc0db93026f1d981f874ec4 (diff)
parent0dcb4efb1095d0a1f5f681c2b94e98b009cc5d77 (diff)
downloadlinux-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.c45
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;