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-rw-r--r--include/media/stagefright/ACodec.h2
-rw-r--r--media/libmedia/AudioRecord.cpp2
-rw-r--r--media/libmedia/AudioTrack.cpp2
-rw-r--r--media/libstagefright/ACodec.cpp28
-rw-r--r--media/libstagefright/wifi-display/source/TSPacketizer.cpp2
-rw-r--r--services/audioflinger/AudioFlinger.cpp4
-rw-r--r--services/audioflinger/Threads.cpp8
-rw-r--r--services/audioflinger/Tracks.cpp2
8 files changed, 42 insertions, 8 deletions
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index e796ab3..7395055 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -116,6 +116,7 @@ private:
kWhatStart = 'star',
kWhatRequestIDRFrame = 'ridr',
kWhatSetParameters = 'setP',
+ kWhatSubmitOutputMetaDataBufferIfEOS = 'subm',
};
enum {
@@ -212,6 +213,7 @@ private:
OMX_U32 *nMinUndequeuedBuffers);
status_t allocateOutputMetaDataBuffers();
status_t submitOutputMetaDataBuffer();
+ void signalSubmitOutputMetaDataBufferIfEOS_workaround();
status_t allocateOutputBuffersFromNativeWindow();
status_t cancelBufferToNativeWindow(BufferInfo *info);
status_t freeOutputBuffersNotOwnedByComponent();
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 666fafa..ccbc5a3 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -545,13 +545,13 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index a9d6993..772612a 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1113,13 +1113,13 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 76a3358..3ee32ea 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -452,6 +452,18 @@ void ACodec::signalRequestIDRFrame() {
(new AMessage(kWhatRequestIDRFrame, id()))->post();
}
+// *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+// Some codecs may return input buffers before having them processed.
+// This causes a halt if we already signaled an EOS on the input
+// port. For now keep submitting an output buffer if there was an
+// EOS on the input port, but not yet on the output port.
+void ACodec::signalSubmitOutputMetaDataBufferIfEOS_workaround() {
+ if (mPortEOS[kPortIndexInput] && !mPortEOS[kPortIndexOutput] &&
+ mMetaDataBuffersToSubmit > 0) {
+ (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, id()))->post();
+ }
+}
+
status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) {
CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
@@ -4036,6 +4048,9 @@ void ACodec::ExecutingState::submitOutputMetaBuffers() {
break;
}
}
+
+ // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+ mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
}
void ACodec::ExecutingState::submitRegularOutputBuffers() {
@@ -4184,6 +4199,19 @@ bool ACodec::ExecutingState::onMessageReceived(const sp<AMessage> &msg) {
break;
}
+ // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+ case kWhatSubmitOutputMetaDataBufferIfEOS:
+ {
+ if (mCodec->mPortEOS[kPortIndexInput] &&
+ !mCodec->mPortEOS[kPortIndexOutput]) {
+ status_t err = mCodec->submitOutputMetaDataBuffer();
+ if (err == OK) {
+ mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
+ }
+ }
+ return true;
+ }
+
default:
handled = BaseState::onMessageReceived(msg);
break;
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index edcc087..50d317a 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@ sp<ABuffer> TSPacketizer::Track::prependADTSHeader(
uint8_t *ptr = dup->data();
*ptr++ = 0xff;
- *ptr++ = 0xf1; // b11110001, ID=0, layer=0, protection_absent=1
+ *ptr++ = 0xf9; // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
*ptr++ =
profile << 6
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index e9c38e3..26dac95 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -513,6 +513,8 @@ sp<IAudioTrack> AudioFlinger::createTrack(
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
+ LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
+ // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -1291,7 +1293,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
frameCount, lSessionId,
IPCThreadState::self()->getCallingUid(),
flags, tid, &lStatus);
- LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
+ LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
}
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the RecordTrack so that the
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 498ddb6..cac785a 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1324,8 +1324,10 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
track = TimedTrack::create(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId, uid);
}
+
if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
+ // track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
@@ -1915,7 +1917,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
- size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t);
+ size_t offset = (mCurrentWriteLength - mBytesRemaining);
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
@@ -1926,7 +1928,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->stream->write(mOutput->stream,
- mMixBuffer + offset, mBytesRemaining);
+ (char *)mMixBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
@@ -4741,7 +4743,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR
if (track->getCblk() == 0) {
ALOGE("createRecordTrack_l() no control block");
lStatus = NO_MEMORY;
- track.clear();
+ // track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index fccc7b8..813ec8a 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -551,7 +551,7 @@ size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
// Don't call for fast tracks; the framesReady() could result in priority inversion
bool AudioFlinger::PlaybackThread::Track::isReady() const {
- if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+ if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing() || isStopping()) {
return true;
}
generated by cgit v1.1 at 2024-05-09 02:38:39 +0000