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authorMark Brown2012-04-04 23:12:09 +0200
committerMark Brown2012-04-16 20:36:29 +0200
commitc74184ed30ecce2a5e9ae9aa22cb5e3942e0c7c7 (patch)
tree6d0043cf3b0b7734b8b98dd0d632309d94969c47 /sound/soc/soc-core.c
parentASoC: core: Bind DAIs to CODECs at registration time (diff)
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ASoC: core: Support transparent CODEC<->CODEC DAI links
Rather than having the user half start a stream but avoid any DMA to trigger data flow on links which don't pass through the CPU create a DAPM route between the two DAI widgets using a hw_params configuration provided by the machine driver with the new 'params' member of the dai_link struct. If no configuration is provided in the dai_link then use the old style even for CODEC<->CODEC links to avoid breaking systems. This greatly simplifies the userspace usage of such links, making them as simple as analogue connections with the stream configuration being completely transparent to them. This is achieved by defining a new dai_link widget type which is created when CODECs are linked and triggering the configuration of the link via the normal PCM operations from there. It is expected that the bias level callbacks will be used for clock configuration. Currently only the DAI format, rate and channel count can be configured and currently the only DAI operations which can be called are hw_params and digital_mute(). This corresponds well to the majority of CODEC drivers which only use other callbacks for constraint setting but there is obviously much room for extension here. We can't simply call hw_params() on startup as things like the system clocking configuration may change at runtime and in future it will be desirable to offer some configurability of the link parameters. At present we are also restricted to a single DAPM link for the entire DAI. Once we have better support for channel mapping it would also be desirable to extend this feature so that we can propagate per-channel power state over the link. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Diffstat (limited to 'sound/soc/soc-core.c')
-rw-r--r--sound/soc/soc-core.c43
1 files changed, 36 insertions, 7 deletions
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0caa8d9e685f..77f35292f1b0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1189,7 +1189,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dapm_widget *play_w, *capture_w;
int ret;
dev_dbg(card->dev, "probe %s dai link %d late %d\n",
@@ -1270,12 +1272,39 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
if (ret < 0)
pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret);
- /* create the pcm */
- ret = soc_new_pcm(rtd, num);
- if (ret < 0) {
- pr_err("asoc: can't create pcm %s :%d\n",
- dai_link->stream_name, ret);
- return ret;
+ if (!dai_link->params) {
+ /* create the pcm */
+ ret = soc_new_pcm(rtd, num);
+ if (ret < 0) {
+ pr_err("asoc: can't create pcm %s :%d\n",
+ dai_link->stream_name, ret);
+ return ret;
+ }
+ } else {
+ /* link the DAI widgets */
+ play_w = codec_dai->playback_widget;
+ capture_w = cpu_dai->capture_widget;
+ if (play_w && capture_w) {
+ ret = snd_soc_dapm_new_pcm(card, dai_link->params,
+ capture_w, play_w);
+ if (ret != 0) {
+ dev_err(card->dev, "Can't link %s to %s: %d\n",
+ play_w->name, capture_w->name, ret);
+ return ret;
+ }
+ }
+
+ play_w = cpu_dai->playback_widget;
+ capture_w = codec_dai->capture_widget;
+ if (play_w && capture_w) {
+ ret = snd_soc_dapm_new_pcm(card, dai_link->params,
+ capture_w, play_w);
+ if (ret != 0) {
+ dev_err(card->dev, "Can't link %s to %s: %d\n",
+ play_w->name, capture_w->name, ret);
+ return ret;
+ }
+ }
}
/* add platform data for AC97 devices */