Revert "audio: use same card and device for PCM amd compress offload path"
This reverts commit 476a97240a37b5ad79f2db295a7f1eb58959982a.
diff --git a/audio/hal/audio_hw.c b/audio/hal/audio_hw.c
index 4273dc1..2557619 100644
--- a/audio/hal/audio_hw.c
+++ b/audio/hal/audio_hw.c
@@ -50,7 +50,7 @@
#define MIXER_CTL_COMPRESS_PLAYBACK_VOLUME "Compress Playback Volume"
/* TODO: the following PCM device profiles could be read from a config file */
-struct pcm_device_profile pcm_device_playback = {
+struct pcm_device_profile pcm_device_playback_hs = {
.config = {
.channels = PLAYBACK_DEFAULT_CHANNEL_COUNT,
.rate = PLAYBACK_DEFAULT_SAMPLING_RATE,
@@ -63,10 +63,9 @@
.avail_min = PLAYBACK_AVAILABLE_MIN,
},
.card = SOUND_CARD,
- .id = 9,
+ .id = 0,
.type = PCM_PLAYBACK,
- .devices = AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|
- AUDIO_DEVICE_OUT_SPEAKER,
+ .devices = AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE,
};
struct pcm_device_profile pcm_device_capture = {
@@ -105,6 +104,24 @@
.devices = SND_DEVICE_IN_LOOPBACK_AEC,
};
+struct pcm_device_profile pcm_device_playback_spk = {
+ .config = {
+ .channels = PLAYBACK_DEFAULT_CHANNEL_COUNT,
+ .rate = PLAYBACK_DEFAULT_SAMPLING_RATE,
+ .period_size = PLAYBACK_PERIOD_SIZE,
+ .period_count = PLAYBACK_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = PLAYBACK_START_THRESHOLD,
+ .stop_threshold = PLAYBACK_STOP_THRESHOLD,
+ .silence_threshold = 0,
+ .avail_min = PLAYBACK_AVAILABLE_MIN,
+ },
+ .card = SOUND_CARD,
+ .id = 1,
+ .type = PCM_PLAYBACK,
+ .devices = AUDIO_DEVICE_OUT_SPEAKER,
+};
+
struct pcm_device_profile pcm_device_playback_sco = {
.config = {
.channels = SCO_DEFAULT_CHANNEL_COUNT,
@@ -162,8 +179,9 @@
};
struct pcm_device_profile *pcm_devices[] = {
- &pcm_device_playback,
+ &pcm_device_playback_hs,
&pcm_device_capture,
+ &pcm_device_playback_spk,
&pcm_device_playback_sco,
&pcm_device_capture_sco,
&pcm_device_capture_loopback_aec,
@@ -4314,7 +4332,13 @@
struct pcm_device_profile *profile = NULL;
memset(&config, 0, sizeof(struct pcm_config));
- profile = &pcm_device_playback;
+ if (adev->dummybuf_thread_devices == AUDIO_DEVICE_OUT_WIRED_HEADPHONE) {
+ profile = &pcm_device_playback_hs;
+ mixer = mixer_open(profile->card);
+ }
+ else
+ profile = &pcm_device_playback_spk;
+
memcpy(&config, &profile->config, sizeof(struct pcm_config));
/* Use large value for stop_threshold so that automatic
trigger for stop is avoided, when this thread fails to write data */
@@ -4330,9 +4354,6 @@
ALOGD("pcm_open: card=%d, id=%d", profile->card, profile->id);
}
- if (adev->dummybuf_thread_devices == AUDIO_DEVICE_OUT_WIRED_HEADPHONE)
- mixer = mixer_open(profile->card);
-
if (mixer) {
ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
if (ctl != NULL)
diff --git a/audio/hal/audio_hw.h b/audio/hal/audio_hw.h
index 9923d77..15ed4bd 100644
--- a/audio/hal/audio_hw.h
+++ b/audio/hal/audio_hw.h
@@ -167,8 +167,8 @@
#define CAPTURE_DEFAULT_SAMPLING_RATE 48000
#define CAPTURE_START_THRESHOLD 1
-#define COMPRESS_CARD 0
-#define COMPRESS_DEVICE 5
+#define COMPRESS_CARD 2
+#define COMPRESS_DEVICE 0
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/* ToDo: Check and update a proper value in msec */
diff --git a/mixer_paths_0.xml b/mixer_paths_0.xml
index 168cbff..f7b15df 100644
--- a/mixer_paths_0.xml
+++ b/mixer_paths_0.xml
@@ -11,17 +11,13 @@
<!-- TODO other defaults -->
<path name="speaker">
- <ctl name="SPK VMixer SPK Switch" value="1"/>
</path>
<path name="headphones">
<ctl name="Headphone Jack Switch" value="1"/>
- <ctl name="Codec VMixer Codec Switch" value="1"/>
</path>
<path name="speaker-and-headphones">
- <ctl name="SPK VMixer SPK Switch" value="1"/>
- <ctl name="Codec VMixer Codec Switch" value="1"/>
<ctl name="Headphone Jack Switch" value="1"/>
</path>
