Snap for 8986256 from 7c84f62f223de56ed03b32a435db206694e99bc3 to tm-platform-release
Change-Id: Ic1adabd0add6ad5bfcd983efec0965582a6c4934
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index cb050f9..45fac76 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1946,12 +1946,27 @@
int32_t numChannels;
CHECK(format->findInt32("channel-count", &numChannels));
- int32_t rawChannelMask;
- audio_channel_mask_t channelMask =
- format->findInt32("channel-mask", &rawChannelMask) ?
- static_cast<audio_channel_mask_t>(rawChannelMask)
- // signal to the AudioSink to derive the mask from count.
- : CHANNEL_MASK_USE_CHANNEL_ORDER;
+ // channel mask info as read from the audio format
+ int32_t channelMaskFromFormat;
+ // channel mask to use for native playback
+ audio_channel_mask_t channelMask;
+ if (format->findInt32("channel-mask", &channelMaskFromFormat)) {
+ // KEY_CHANNEL_MASK follows the android.media.AudioFormat java mask
+ // which is left-bitshifted by 2 relative to the native mask
+ if ((channelMaskFromFormat & 0b11) != 0) {
+ // received an unexpected mask (supposed to follow AudioFormat constants
+ // for output masks with the 2 least-significant bits at 0), but
+ // it may come from an extractor that uses native masks: keeping
+ // the mask as given is ok as it contains at least mono or stereo
+ // and potentially the haptic channels
+ channelMask = static_cast<audio_channel_mask_t>(channelMaskFromFormat);
+ } else {
+ channelMask = static_cast<audio_channel_mask_t>(channelMaskFromFormat >> 2);
+ }
+ } else {
+ // no mask found: the mask will be derived from the channel count
+ channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
+ }
int32_t sampleRate;
CHECK(format->findInt32("sample-rate", &sampleRate));
diff --git a/media/libstagefright/rtsp/AAVCAssembler.cpp b/media/libstagefright/rtsp/AAVCAssembler.cpp
index 2f516d5..ddf797c 100644
--- a/media/libstagefright/rtsp/AAVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AAVCAssembler.cpp
@@ -618,13 +618,14 @@
int32_t AAVCAssembler::pickStartSeq(const Queue *queue,
uint32_t first, int64_t play, int64_t jit) {
+ CHECK(!queue->empty());
// pick the first sequence number has the start bit.
sp<ABuffer> buffer = *(queue->begin());
int32_t firstSeqNo = buffer->int32Data();
// This only works for FU-A type & non-start sequence
- unsigned nalType = buffer->data()[0] & 0x1f;
- if (nalType != 28 || buffer->data()[1] & 0x80) {
+ int32_t nalType = buffer->size() >= 1 ? buffer->data()[0] & 0x1f : -1;
+ if (nalType != 28 || (buffer->size() >= 2 && buffer->data()[1] & 0x80)) {
return firstSeqNo;
}
@@ -634,7 +635,7 @@
if (rtpTime + jit >= play) {
break;
}
- if ((data[1] & 0x80)) {
+ if (it->size() >= 2 && (data[1] & 0x80)) {
const int32_t seqNo = it->int32Data();
ALOGE("finding [HEAD] pkt. \t Seq# (%d ~ )[%d", firstSeqNo, seqNo);
firstSeqNo = seqNo;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 56ebb6e..07e82a8 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -9560,6 +9560,12 @@
if (isOutput()) {
ret = AudioSystem::startOutput(portId);
} else {
+ {
+ // Add the track record before starting input so that the silent status for the
+ // client can be cached.
+ Mutex::Autolock _l(mLock);
+ setClientSilencedState_l(portId, false /*silenced*/);
+ }
ret = AudioSystem::startInput(portId);
}
@@ -9578,6 +9584,7 @@
} else {
mHalStream->stop();
}
+ eraseClientSilencedState_l(portId);
return PERMISSION_DENIED;
}
@@ -9586,6 +9593,9 @@
mChannelMask, mSessionId, isOutput(),
client.attributionSource,
IPCThreadState::self()->getCallingPid(), portId);
+ if (!isOutput()) {
+ track->setSilenced_l(isClientSilenced_l(portId));
+ }
if (isOutput()) {
// force volume update when a new track is added
@@ -9643,6 +9653,7 @@
}
mActiveTracks.remove(track);
+ eraseClientSilencedState_l(track->portId());
mLock.unlock();
if (isOutput()) {
@@ -10433,6 +10444,7 @@
broadcast_l();
}
}
+ setClientSilencedIfExists_l(portId, silenced);
}
void AudioFlinger::MmapCaptureThread::toAudioPortConfig(struct audio_port_config *config)
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index b2962ed8..074ae8f 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -2057,6 +2057,26 @@
virtual bool isStreamInitialized() { return false; }
+ void setClientSilencedState_l(audio_port_handle_t portId, bool silenced) {
+ mClientSilencedStates[portId] = silenced;
+ }
+
+ size_t eraseClientSilencedState_l(audio_port_handle_t portId) {
+ return mClientSilencedStates.erase(portId);
+ }
+
+ bool isClientSilenced_l(audio_port_handle_t portId) const {
+ const auto it = mClientSilencedStates.find(portId);
+ return it != mClientSilencedStates.end() ? it->second : false;
+ }
+
+ void setClientSilencedIfExists_l(audio_port_handle_t portId, bool silenced) {
+ const auto it = mClientSilencedStates.find(portId);
+ if (it != mClientSilencedStates.end()) {
+ it->second = silenced;
+ }
+ }
+
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
@@ -2076,6 +2096,7 @@
AudioHwDevice* const mAudioHwDev;
ActiveTracks<MmapTrack> mActiveTracks;
float mHalVolFloat;
+ std::map<audio_port_handle_t, bool> mClientSilencedStates;
int32_t mNoCallbackWarningCount;
static constexpr int32_t kMaxNoCallbackWarnings = 5;
