Merge "camera: Fix setParameters for Preview FPS single/range values" into lmp-dev
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index a706987..2c48306 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1898,7 +1898,8 @@
me, buffer->raw, buffer->size, me->mCallbackCookie,
CB_EVENT_FILL_BUFFER);
- if (actualSize == 0 && buffer->size > 0 && me->mNextOutput == NULL) {
+ if ((me->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0 &&
+ actualSize == 0 && buffer->size > 0 && me->mNextOutput == NULL) {
// We've reached EOS but the audio track is not stopped yet,
// keep playing silence.
diff --git a/media/libstagefright/data/media_codecs_google_audio.xml b/media/libstagefright/data/media_codecs_google_audio.xml
index f6db0cc..f599004 100644
--- a/media/libstagefright/data/media_codecs_google_audio.xml
+++ b/media/libstagefright/data/media_codecs_google_audio.xml
@@ -16,21 +16,76 @@
<Included>
<Decoders>
- <MediaCodec name="OMX.google.mp3.decoder" type="audio/mpeg" />
- <MediaCodec name="OMX.google.amrnb.decoder" type="audio/3gpp" />
- <MediaCodec name="OMX.google.amrwb.decoder" type="audio/amr-wb" />
- <MediaCodec name="OMX.google.aac.decoder" type="audio/mp4a-latm" />
- <MediaCodec name="OMX.google.g711.alaw.decoder" type="audio/g711-alaw" />
- <MediaCodec name="OMX.google.g711.mlaw.decoder" type="audio/g711-mlaw" />
- <MediaCodec name="OMX.google.vorbis.decoder" type="audio/vorbis" />
- <MediaCodec name="OMX.google.opus.decoder" type="audio/opus" />
- <MediaCodec name="OMX.google.raw.decoder" type="audio/raw" />
+ <MediaCodec name="OMX.google.mp3.decoder" type="audio/mpeg">
+ <Limit name="channel-count" max="2" />
+ <Limit name="sample-rate" ranges="8000,11025,12000,16000,22050,24000,32000,44100,48000" />
+ <Limit name="bitrate" range="8000-320000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.amrnb.decoder" type="audio/3gpp">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="8000" />
+ <Limit name="bitrate" range="4750-12200" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.amrwb.decoder" type="audio/amr-wb">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="16000" />
+ <Limit name="bitrate" range="6600-23850" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.aac.decoder" type="audio/mp4a-latm">
+ <Limit name="channel-count" max="8" />
+ <Limit name="sample-rate" ranges="7350,8000,11025,12000,16000,22050,24000,32000,44100,48000" />
+ <Limit name="bitrate" range="8000-960000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.g711.alaw.decoder" type="audio/g711-alaw">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="8000" />
+ <Limit name="bitrate" range="64000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.g711.mlaw.decoder" type="audio/g711-mlaw">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="8000" />
+ <Limit name="bitrate" range="64000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.vorbis.decoder" type="audio/vorbis">
+ <Limit name="channel-count" max="8" />
+ <Limit name="sample-rate" ranges="8000,11025,12000,16000,22050,24000,32000,44100,48000,96000" />
+ <Limit name="bitrate" range="32000-500000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.opus.decoder" type="audio/opus">
+ <Limit name="channel-count" max="8" />
+ <Limit name="sample-rate" ranges="48000" />
+ <Limit name="bitrate" range="6000-510000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.raw.decoder" type="audio/raw">
+ <Limit name="channel-count" max="8" />
+ <Limit name="sample-rate" ranges="8000-96000" />
+ <Limit name="bitrate" range="1-10000000" />
+ </MediaCodec>
</Decoders>
-
<Encoders>
- <MediaCodec name="OMX.google.aac.encoder" type="audio/mp4a-latm" />
- <MediaCodec name="OMX.google.amrnb.encoder" type="audio/3gpp" />
- <MediaCodec name="OMX.google.amrwb.encoder" type="audio/amr-wb" />
- <MediaCodec name="OMX.google.flac.encoder" type="audio/flac" />
+ <MediaCodec name="OMX.google.aac.encoder" type="audio/mp4a-latm">
+ <Limit name="channel-count" max="6" />
+ <Limit name="sample-rate" ranges="11025,12000,16000,22050,24000,32000,44100,48000" />
+ <Limit name="bitrate" range="8000-960000" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.amrnb.encoder" type="audio/3gpp">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="8000" />
+ <Limit name="bitrate" range="4750-12200" />
+ <Feature name="bitrate-modes" value="CBR" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.amrwb.encoder" type="audio/amr-wb">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="16000" />
+ <Limit name="bitrate" range="6600-23850" />
+ <Feature name="bitrate-modes" value="CBR" />
+ </MediaCodec>
+ <MediaCodec name="OMX.google.flac.encoder" type="audio/flac">
+ <Limit name="channel-count" max="2" />
+ <Limit name="sample-rate" ranges="1-655350" />
+ <Limit name="bitrate" range="1-21000000" />
+ <Limit name="complexity" range="0-8" default="5" />
+ <Feature name="bitrate-modes" value="CQ" />
+ </MediaCodec>
</Encoders>
</Included>
diff --git a/media/libstagefright/data/media_codecs_google_telephony.xml b/media/libstagefright/data/media_codecs_google_telephony.xml
index 28f5ffc..5ad90d9 100644
--- a/media/libstagefright/data/media_codecs_google_telephony.xml
+++ b/media/libstagefright/data/media_codecs_google_telephony.xml
@@ -16,6 +16,10 @@
<Included>
<Decoders>
- <MediaCodec name="OMX.google.gsm.decoder" type="audio/gsm" />
+ <MediaCodec name="OMX.google.gsm.decoder" type="audio/gsm">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="8000" />
+ <Limit name="bitrate" range="13000" />
+ </MediaCodec>
</Decoders>
</Included>
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 193f8e4..f721d5c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5017,9 +5017,12 @@
// activeTracks accumulates a copy of a subset of mActiveTracks
Vector< sp<RecordTrack> > activeTracks;
- // reference to the (first and only) fast track
+ // reference to the (first and only) active fast track
sp<RecordTrack> fastTrack;
+ // reference to a fast track which is about to be removed
+ sp<RecordTrack> fastTrackToRemove;
+
{ // scope for mLock
Mutex::Autolock _l(mLock);
@@ -5058,6 +5061,10 @@
activeTrack = mActiveTracks[i];
if (activeTrack->isTerminated()) {
+ if (activeTrack->isFastTrack()) {
+ ALOG_ASSERT(fastTrackToRemove == 0);
+ fastTrackToRemove = activeTrack;
+ }
removeTrack_l(activeTrack);
mActiveTracks.remove(activeTrack);
mActiveTracksGen++;
@@ -5130,10 +5137,12 @@
effectChains[i]->process_l();
}
- // Start the fast capture if it's not already running
+ // Push a new fast capture state if fast capture is not already running, or cblk change
if (mFastCapture != 0) {
FastCaptureStateQueue *sq = mFastCapture->sq();
FastCaptureState *state = sq->begin();
+ bool didModify = false;
+ FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
(kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
if (state->mCommand == FastCaptureState::COLD_IDLE) {
@@ -5147,19 +5156,32 @@
mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
#endif
- state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL;
- sq->end();
- sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+ didModify = true;
+ }
+ audio_track_cblk_t *cblkOld = state->mCblk;
+ audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
+ if (cblkNew != cblkOld) {
+ state->mCblk = cblkNew;
+ // block until acked if removing a fast track
+ if (cblkOld != NULL) {
+ block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
+ }
+ didModify = true;
+ }
+ sq->end(didModify);
+ if (didModify) {
+ sq->push(block);
#if 0
if (kUseFastCapture == FastCapture_Dynamic) {
mNormalSource = mPipeSource;
}
#endif
- } else {
- sq->end(false /*didModify*/);
}
}
+ // now run the fast track destructor with thread mutex unlocked
+ fastTrackToRemove.clear();
+
// Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
// Only the client(s) that are too slow will overrun. But if even the fastest client is too
// slow, then this RecordThread will overrun by not calling HAL read often enough.