am fff97293: am cb66d22f: Merge "Camera2: Use FORMAT_ZSL until transition to new gralloc scheme." into jb-mr1-dev
* commit 'fff97293d56ca9869004f89ce46c2b337d7f57e7':
Camera2: Use FORMAT_ZSL until transition to new gralloc scheme.
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index dc1e351..c09f86f 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -513,8 +513,6 @@
}
}
} else if (what == Renderer::kWhatFlushComplete) {
- CHECK_EQ(what, (int32_t)Renderer::kWhatFlushComplete);
-
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index ab75dd0..1f13625 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -108,6 +108,8 @@
LocalClock lc;
+ pthread_once(&sOnceControl, &sInitRoutine);
+
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
@@ -121,8 +123,6 @@
// and mTrackNames is initially 0. However, leave it here until that's verified.
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
- // FIXME redundant per track
- t->localTimeFreq = lc.getLocalFreq();
t->resampler = NULL;
t->downmixerBufferProvider = NULL;
t++;
@@ -192,7 +192,6 @@
t->sessionId = sessionId;
// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
t->bufferProvider = NULL;
- t->downmixerBufferProvider = NULL;
t->buffer.raw = NULL;
// no initialization needed
// t->buffer.frameCount
@@ -203,7 +202,7 @@
// setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
t->mainBuffer = NULL;
t->auxBuffer = NULL;
- // see t->localTimeFreq in constructor above
+ t->downmixerBufferProvider = NULL;
status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
if (status == OK) {
@@ -544,7 +543,7 @@
// the resampler sees the number of channels after the downmixer, if any
downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
devSampleRate);
- resampler->setLocalTimeFreq(localTimeFreq);
+ resampler->setLocalTimeFreq(sLocalTimeFreq);
}
return true;
}
@@ -1411,7 +1410,16 @@
if (AudioBufferProvider::kInvalidPTS == basePTS)
return AudioBufferProvider::kInvalidPTS;
- return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
+ return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
+}
+
+/*static*/ uint64_t AudioMixer::sLocalTimeFreq;
+/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
+
+/*static*/ void AudioMixer::sInitRoutine()
+{
+ LocalClock lc;
+ sLocalTimeFreq = lc.getLocalFreq();
}
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 6333357..dc468ff 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -188,12 +188,12 @@
// 16-byte boundary
- uint64_t localTimeFreq;
-
DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
int32_t sessionId;
+ int32_t padding[2];
+
// 16-byte boundary
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
@@ -274,6 +274,10 @@
static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex);
+
+ static uint64_t sLocalTimeFreq;
+ static pthread_once_t sOnceControl;
+ static void sInitRoutine();
};
// ----------------------------------------------------------------------------