Merge "camera: Fix deadlock when sending callbacks during disconnect" into klp-dev
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
new file mode 100644
index 0000000..c29c7e5
--- /dev/null
+++ b/include/media/AudioTimestamp.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_TIMESTAMP_H
+#define ANDROID_AUDIO_TIMESTAMP_H
+
+#include <time.h>
+
+class AudioTimestamp {
+public:
+ AudioTimestamp() : mPosition(0) {
+ mTime.tv_sec = 0;
+ mTime.tv_nsec = 0;
+ }
+ // FIXME change type to match android.media.AudioTrack
+ uint32_t mPosition; // a frame position in AudioTrack::getPosition() units
+ struct timespec mTime; // corresponding CLOCK_MONOTONIC when frame is expected to present
+};
+
+#endif // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index ae92cdd..453c106 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -19,6 +19,7 @@
#include <cutils/sched_policy.h>
#include <media/AudioSystem.h>
+#include <media/AudioTimestamp.h>
#include <media/IAudioTrack.h>
#include <utils/threads.h>
@@ -62,6 +63,9 @@
// voluntary invalidation by mediaserver, or mediaserver crash.
EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played
// back (after stop is called)
+ EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change
+ // in the mapping from frame position to presentation time.
+ // See AudioTimestamp for the information included with event.
};
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -107,6 +111,8 @@
* - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
* - EVENT_BUFFER_END: unused.
* - EVENT_NEW_IAUDIOTRACK: unused.
+ * - EVENT_STREAM_END: unused.
+ * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
*/
typedef void (*callback_t)(int event, void* user, void *info);
@@ -221,6 +227,7 @@
* - INVALID_OPERATION: AudioTrack is already initialized
* - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
+ * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack.
* If sharedBuffer is non-0, the frameCount parameter is ignored and
* replaced by the shared buffer's total allocated size in frame units.
*
@@ -243,7 +250,7 @@
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL);
- /* Result of constructing the AudioTrack. This must be checked
+ /* Result of constructing the AudioTrack. This must be checked for successful initialization
* before using any AudioTrack API (except for set()), because using
* an uninitialized AudioTrack produces undefined results.
* See set() method above for possible return codes.
@@ -564,6 +571,16 @@
/* Get parameters */
String8 getParameters(const String8& keys);
+ /* Poll for a timestamp on demand.
+ * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs,
+ * or if you need to get the most recent timestamp outside of the event callback handler.
+ * Caution: calling this method too often may be inefficient;
+ * if you need a high resolution mapping between frame position and presentation time,
+ * consider implementing that at application level, based on the low resolution timestamps.
+ * Returns NO_ERROR if timestamp is valid.
+ */
+ status_t getTimestamp(AudioTimestamp& timestamp);
+
protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
@@ -630,7 +647,7 @@
bool isOffloaded() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
- // may be changed if IAudioTrack is re-created
+ // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 1014403..afac4ae 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -26,6 +26,7 @@
#include <binder/IMemory.h>
#include <utils/LinearTransform.h>
#include <utils/String8.h>
+#include <media/AudioTimestamp.h>
namespace android {
@@ -86,6 +87,9 @@
/* Send parameters to the audio hardware */
virtual status_t setParameters(const String8& keyValuePairs) = 0;
+
+ /* Return NO_ERROR if timestamp is valid */
+ virtual status_t getTimestamp(AudioTimestamp& timestamp) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 1379379..ad7409d 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -422,6 +422,9 @@
// Return the total number of frames which AudioFlinger desired but were unavailable,
// and thus which resulted in an underrun.
virtual uint32_t getUnderrunFrames() const { return mCblk->u.mStreaming.mUnderrunFrames; }
+
+ // Return the total number of frames that AudioFlinger has obtained and released
+ virtual size_t framesReleased() const { return mCblk->mServer; }
};
class StaticAudioTrackServerProxy : public AudioTrackServerProxy {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7aa9e35..176197c 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -148,10 +148,8 @@
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
- if (mAudioTrack != 0) {
- mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
- mAudioTrack.clear();
- }
+ mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
+ mAudioTrack.clear();
IPCThreadState::self()->flushCommands();
AudioSystem::releaseAudioSessionId(mSessionId);
}
@@ -222,6 +220,7 @@
AutoMutex lock(mLock);
+ // invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
@@ -968,6 +967,7 @@
ALOGE("Could not get control block");
return NO_INIT;
}
+ // invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
@@ -1708,11 +1708,13 @@
status_t AudioTrack::setParameters(const String8& keyValuePairs)
{
AutoMutex lock(mLock);
- if (mAudioTrack != 0) {
- return mAudioTrack->setParameters(keyValuePairs);
- } else {
- return NO_INIT;
- }
+ return mAudioTrack->setParameters(keyValuePairs);
+}
+
+status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
+{
+ AutoMutex lock(mLock);
+ return mAudioTrack->getTimestamp(timestamp);
}
String8 AudioTrack::getParameters(const String8& keys)
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index a2b49a3..f0d75ba 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -39,7 +39,8 @@
ALLOCATE_TIMED_BUFFER,
QUEUE_TIMED_BUFFER,
SET_MEDIA_TIME_TRANSFORM,
- SET_PARAMETERS
+ SET_PARAMETERS,
+ GET_TIMESTAMP,
};
class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -166,6 +167,21 @@
}
return status;
}
+
+ virtual status_t getTimestamp(AudioTimestamp& timestamp) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_TIMESTAMP, data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ if (status == NO_ERROR) {
+ timestamp.mPosition = reply.readInt32();
+ timestamp.mTime.tv_sec = reply.readInt32();
+ timestamp.mTime.tv_nsec = reply.readInt32();
+ }
+ }
+ return status;
+ }
};
IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -241,6 +257,18 @@
reply->writeInt32(setParameters(keyValuePairs));
return NO_ERROR;
} break;
+ case GET_TIMESTAMP: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ AudioTimestamp timestamp;
+ status_t status = getTimestamp(timestamp);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeInt32(timestamp.mPosition);
+ reply->writeInt32(timestamp.mTime.tv_sec);
+ reply->writeInt32(timestamp.mTime.tv_nsec);
+ }
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 2418aab..e38e261 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -680,6 +680,14 @@
int64_t AudioPlayer::getRealTimeUs() {
Mutex::Autolock autoLock(mLock);
+ if (useOffload()) {
+ if (mSeeking) {
+ return mSeekTimeUs;
+ }
+ mPositionTimeRealUs = getOutputPlayPositionUs_l();
+ return mPositionTimeRealUs;
+ }
+
return getRealTimeUsLocked();
}
@@ -741,11 +749,6 @@
return 0;
}
- if (useOffload()) {
- mPositionTimeRealUs = getOutputPlayPositionUs_l();
- return mPositionTimeRealUs;
- }
-
int64_t realTimeOffset = getRealTimeUsLocked() - mPositionTimeRealUs;
if (realTimeOffset < 0) {
realTimeOffset = 0;
@@ -758,8 +761,14 @@
int64_t *realtime_us, int64_t *mediatime_us) {
Mutex::Autolock autoLock(mLock);
- *realtime_us = mPositionTimeRealUs;
- *mediatime_us = mPositionTimeMediaUs;
+ if (useOffload()) {
+ mPositionTimeRealUs = getOutputPlayPositionUs_l();
+ *realtime_us = mPositionTimeRealUs;
+ *mediatime_us = mPositionTimeRealUs;
+ } else {
+ *realtime_us = mPositionTimeRealUs;
+ *mediatime_us = mPositionTimeMediaUs;
+ }
return mPositionTimeRealUs != -1 && mPositionTimeMediaUs != -1;
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 1b20cbb..c9b5d26 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -58,6 +58,8 @@
mIsADTS(false),
mInputBufferCount(0),
mSignalledError(false),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mAnchorTimeUs(0),
mNumSamplesOutput(0),
mOutputPortSettingsChange(NONE) {
@@ -350,115 +352,83 @@
return;
}
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ if (inHeader) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- if (mDecoderHasData) {
- // flush out the decoder's delayed data by calling DecodeFrame
- // one more time, with the AACDEC_FLUSH flag set
- INT_PCM *outBuffer =
- reinterpret_cast<INT_PCM *>(
- outHeader->pBuffer + outHeader->nOffset);
+ if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumSamplesOutput = 0;
+ }
- AAC_DECODER_ERROR decoderErr =
- aacDecoder_DecodeFrame(mAACDecoder,
- outBuffer,
- outHeader->nAllocLen,
- AACDEC_FLUSH);
- mDecoderHasData = false;
+ if (mIsADTS) {
+ size_t adtsHeaderSize = 0;
+ // skip 30 bits, aac_frame_length follows.
+ // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
- if (decoderErr != AAC_DEC_OK) {
+ const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
+
+ bool signalError = false;
+ if (inHeader->nFilledLen < 7) {
+ ALOGE("Audio data too short to contain even the ADTS header. "
+ "Got %ld bytes.", inHeader->nFilledLen);
+ hexdump(adtsHeader, inHeader->nFilledLen);
+ signalError = true;
+ } else {
+ bool protectionAbsent = (adtsHeader[1] & 1);
+
+ unsigned aac_frame_length =
+ ((adtsHeader[3] & 3) << 11)
+ | (adtsHeader[4] << 3)
+ | (adtsHeader[5] >> 5);
+
+ if (inHeader->nFilledLen < aac_frame_length) {
+ ALOGE("Not enough audio data for the complete frame. "
+ "Got %ld bytes, frame size according to the ADTS "
+ "header is %u bytes.",
+ inHeader->nFilledLen, aac_frame_length);
+ hexdump(adtsHeader, inHeader->nFilledLen);
+ signalError = true;
+ } else {
+ adtsHeaderSize = (protectionAbsent ? 7 : 9);
+
+ inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
+ inBufferLength[0] = aac_frame_length - adtsHeaderSize;
+
+ inHeader->nOffset += adtsHeaderSize;
+ inHeader->nFilledLen -= adtsHeaderSize;
+ }
+ }
+
+ if (signalError) {
mSignalledError = true;
- notify(OMX_EventError, OMX_ErrorUndefined, decoderErr,
+ notify(OMX_EventError,
+ OMX_ErrorStreamCorrupt,
+ ERROR_MALFORMED,
NULL);
return;
}
-
- outHeader->nFilledLen =
- mStreamInfo->frameSize
- * sizeof(int16_t)
- * mStreamInfo->numChannels;
} else {
- // we never submitted any data to the decoder, so there's nothing to flush out
- outHeader->nFilledLen = 0;
- }
-
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
- }
-
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumSamplesOutput = 0;
- }
-
- size_t adtsHeaderSize = 0;
- if (mIsADTS) {
- // skip 30 bits, aac_frame_length follows.
- // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
-
- const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
-
- bool signalError = false;
- if (inHeader->nFilledLen < 7) {
- ALOGE("Audio data too short to contain even the ADTS header. "
- "Got %ld bytes.", inHeader->nFilledLen);
- hexdump(adtsHeader, inHeader->nFilledLen);
- signalError = true;
- } else {
- bool protectionAbsent = (adtsHeader[1] & 1);
-
- unsigned aac_frame_length =
- ((adtsHeader[3] & 3) << 11)
- | (adtsHeader[4] << 3)
- | (adtsHeader[5] >> 5);
-
- if (inHeader->nFilledLen < aac_frame_length) {
- ALOGE("Not enough audio data for the complete frame. "
- "Got %ld bytes, frame size according to the ADTS "
- "header is %u bytes.",
- inHeader->nFilledLen, aac_frame_length);
- hexdump(adtsHeader, inHeader->nFilledLen);
- signalError = true;
- } else {
- adtsHeaderSize = (protectionAbsent ? 7 : 9);
-
- inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
- inBufferLength[0] = aac_frame_length - adtsHeaderSize;
-
- inHeader->nOffset += adtsHeaderSize;
- inHeader->nFilledLen -= adtsHeaderSize;
- }
- }
-
- if (signalError) {
- mSignalledError = true;
-
- notify(OMX_EventError,
- OMX_ErrorStreamCorrupt,
- ERROR_MALFORMED,
- NULL);
-
- return;
+ inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+ inBufferLength[0] = inHeader->nFilledLen;
}
} else {
- inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
- inBufferLength[0] = inHeader->nFilledLen;
+ inBufferLength[0] = 0;
}
// Fill and decode
@@ -471,50 +441,66 @@
int prevNumChannels = mStreamInfo->numChannels;
AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;
- while (bytesValid[0] > 0 && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+ while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+ mDecoderHasData |= (bytesValid[0] > 0);
aacDecoder_Fill(mAACDecoder,
inBuffer,
inBufferLength,
bytesValid);
- mDecoderHasData = true;
decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
outBuffer,
outHeader->nAllocLen,
0 /* flags */);
-
if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
- ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+ if (mSawInputEos && bytesValid[0] <= 0) {
+ if (mDecoderHasData) {
+ // flush out the decoder's delayed data by calling DecodeFrame
+ // one more time, with the AACDEC_FLUSH flag set
+ decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
+ outBuffer,
+ outHeader->nAllocLen,
+ AACDEC_FLUSH);
+ mDecoderHasData = false;
+ }
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ break;
+ } else {
+ ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+ }
}
}
size_t numOutBytes =
mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
- if (decoderErr == AAC_DEC_OK) {
- UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
- inHeader->nFilledLen -= inBufferUsedLength;
- inHeader->nOffset += inBufferUsedLength;
- } else {
- ALOGW("AAC decoder returned error %d, substituting silence",
- decoderErr);
+ if (inHeader) {
+ if (decoderErr == AAC_DEC_OK) {
+ UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
+ inHeader->nFilledLen -= inBufferUsedLength;
+ inHeader->nOffset += inBufferUsedLength;
+ } else {
+ ALOGW("AAC decoder returned error %d, substituting silence",
+ decoderErr);
- memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
+ memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
- // Discard input buffer.
- inHeader->nFilledLen = 0;
+ // Discard input buffer.
+ inHeader->nFilledLen = 0;
- aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
+ aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
- // fall through
- }
+ // fall through
+ }
- if (inHeader->nFilledLen == 0) {
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
+ if (inHeader->nFilledLen == 0) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
}
/*
@@ -555,7 +541,6 @@
// we've previously decoded valid data, in the latter case
// (decode failed) we'll output a silent frame.
outHeader->nFilledLen = numOutBytes;
- outHeader->nFlags = 0;
outHeader->nTimeStamp =
mAnchorTimeUs
@@ -606,6 +591,8 @@
mStreamInfo->sampleRate = 0;
mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index 2d960ab..a7ea1e2 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -55,6 +55,8 @@
bool mDecoderHasData;
size_t mInputBufferCount;
bool mSignalledError;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
int64_t mAnchorTimeUs;
int64_t mNumSamplesOutput;
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 7c382fb..877e3cb 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -49,6 +49,8 @@
mNumChannels(2),
mSamplingRate(44100),
mSignalledError(false),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mOutputPortSettingsChange(NONE) {
initPorts();
initDecoder();
@@ -194,48 +196,36 @@
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
-
- if (!mIsFirst) {
- // pad the end of the stream with 529 samples, since that many samples
- // were trimmed off the beginning when decoding started
- outHeader->nFilledLen =
- kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
-
- memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
- } else {
- // Since we never discarded frames from the start, we won't have
- // to add any padding at the end either.
- outHeader->nFilledLen = 0;
+ if (inHeader) {
+ if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumFramesOutput = 0;
}
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ mConfig->pInputBuffer =
+ inHeader->pBuffer + inHeader->nOffset;
+
+ mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
+ } else {
+ mConfig->pInputBuffer = NULL;
+ mConfig->inputBufferCurrentLength = 0;
}
-
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumFramesOutput = 0;
- }
-
- mConfig->pInputBuffer =
- inHeader->pBuffer + inHeader->nOffset;
-
- mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
mConfig->inputBufferMaxLength = 0;
mConfig->inputBufferUsedLength = 0;
@@ -262,13 +252,28 @@
mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t);
}
- // This is recoverable, just ignore the current frame and
- // play silence instead.
- memset(outHeader->pBuffer,
- 0,
- mConfig->outputFrameSize * sizeof(int16_t));
+ if (decoderErr == NO_ENOUGH_MAIN_DATA_ERROR && mSawInputEos) {
+ if (!mIsFirst) {
+ // pad the end of the stream with 529 samples, since that many samples
+ // were trimmed off the beginning when decoding started
+ outHeader->nOffset = 0;
+ outHeader->nFilledLen = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
- mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+ memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
+ }
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ } else {
+ // This is recoverable, just ignore the current frame and
+ // play silence instead.
+ memset(outHeader->pBuffer,
+ 0,
+ mConfig->outputFrameSize * sizeof(int16_t));
+
+ if (inHeader) {
+ mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+ }
+ }
} else if (mConfig->samplingRate != mSamplingRate
|| mConfig->num_channels != mNumChannels) {
mSamplingRate = mConfig->samplingRate;
@@ -289,7 +294,7 @@
outHeader->nFilledLen =
mConfig->outputFrameSize * sizeof(int16_t) - outHeader->nOffset;
- } else {
+ } else if (!mSignalledOutputEos) {
outHeader->nOffset = 0;
outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t);
}
@@ -298,23 +303,24 @@
mAnchorTimeUs
+ (mNumFramesOutput * 1000000ll) / mConfig->samplingRate;
- outHeader->nFlags = 0;
+ if (inHeader) {
+ CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
- CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
+ inHeader->nOffset += mConfig->inputBufferUsedLength;
+ inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
- inHeader->nOffset += mConfig->inputBufferUsedLength;
- inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
+
+ if (inHeader->nFilledLen == 0) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
+ }
mNumFramesOutput += mConfig->outputFrameSize / mNumChannels;
- if (inHeader->nFilledLen == 0) {
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
- }
-
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
outInfo = NULL;
@@ -362,6 +368,8 @@
pvmp3_InitDecoder(mConfig, mDecoderBuf);
mIsFirst = true;
mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.h b/media/libstagefright/codecs/mp3dec/SoftMP3.h
index 4af91ea..f9e7b53 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.h
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.h
@@ -61,6 +61,8 @@
bool mIsFirst;
bool mSignalledError;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
enum {
NONE,
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 51bb958..a377b23 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -54,6 +54,8 @@
mAnchorTimeUs(0),
mNumFramesOutput(0),
mNumFramesLeftOnPage(-1),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mOutputPortSettingsChange(NONE) {
initPorts();
CHECK_EQ(initDecoder(), (status_t)OK);
@@ -290,48 +292,47 @@
return;
}
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ int32_t numPageSamples = 0;
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ if (inHeader) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ if (inHeader->nFilledLen || !mSawInputEos) {
+ CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
+ memcpy(&numPageSamples,
+ inHeader->pBuffer
+ + inHeader->nOffset + inHeader->nFilledLen - 4,
+ sizeof(numPageSamples));
+
+ if (inHeader->nOffset == 0) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumFramesOutput = 0;
+ }
+
+ inHeader->nFilledLen -= sizeof(numPageSamples);;
+ }
}
- int32_t numPageSamples;
- CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
- memcpy(&numPageSamples,
- inHeader->pBuffer
- + inHeader->nOffset + inHeader->nFilledLen - 4,
- sizeof(numPageSamples));
-
if (numPageSamples >= 0) {
mNumFramesLeftOnPage = numPageSamples;
}
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumFramesOutput = 0;
- }
-
- inHeader->nFilledLen -= sizeof(numPageSamples);;
-
ogg_buffer buf;
- buf.data = inHeader->pBuffer + inHeader->nOffset;
- buf.size = inHeader->nFilledLen;
+ buf.data = inHeader ? inHeader->pBuffer + inHeader->nOffset : NULL;
+ buf.size = inHeader ? inHeader->nFilledLen : 0;
buf.refcount = 1;
buf.ptr.owner = NULL;
@@ -384,11 +385,13 @@
mNumFramesOutput += numFrames;
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
+ if (inHeader) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
@@ -425,6 +428,8 @@
mVi = NULL;
}
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
index cb628a0..1d00816 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
@@ -59,6 +59,8 @@
int64_t mAnchorTimeUs;
int64_t mNumFramesOutput;
int32_t mNumFramesLeftOnPage;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
enum {
NONE,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index e5e4113..5df04f4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -411,6 +411,7 @@
virtual status_t setMediaTimeTransform(const LinearTransform& xform,
int target);
virtual status_t setParameters(const String8& keyValuePairs);
+ virtual status_t getTimestamp(AudioTimestamp& timestamp);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 5600411c..d34833f 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -59,6 +59,7 @@
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
int16_t *mainBuffer() const { return mMainBuffer; }
int auxEffectId() const { return mAuxEffectId; }
+ virtual status_t getTimestamp(AudioTimestamp& timestamp);
// implement FastMixerState::VolumeProvider interface
virtual uint32_t getVolumeLR();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2c2931f..bc01ede 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -943,7 +943,9 @@
mDraining(false),
mScreenState(AudioFlinger::mScreenState),
// index 0 is reserved for normal mixer's submix
- mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
+ mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1),
+ // mLatchD, mLatchQ,
+ mLatchDValid(false), mLatchQValid(false)
{
snprintf(mName, kNameLength, "AudioOut_%X", id);
mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
@@ -1818,6 +1820,14 @@
} else {
bytesWritten = framesWritten;
}
+ status_t status = INVALID_OPERATION; // mLatchD.mTimestamp is invalid
+ if (status == NO_ERROR) {
+ size_t totalFramesWritten = mNormalSink->framesWritten();
+ if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) {
+ mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition;
+ mLatchDValid = true;
+ }
+ }
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
@@ -2096,6 +2106,12 @@
logString = NULL;
}
+ if (mLatchDValid) {
+ mLatchQ = mLatchD;
+ mLatchDValid = false;
+ mLatchQValid = true;
+ }
+
if (checkForNewParameters_l()) {
cacheParameters_l();
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 31d5323..1333de2 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -606,6 +606,17 @@
// accessed by both binder threads and within threadLoop(), lock on mutex needed
unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
virtual void flushOutput_l();
+
+private:
+ // timestamp latch:
+ // D input is written by threadLoop_write while mutex is unlocked, and read while locked
+ // Q output is written while locked, and read while locked
+ struct {
+ AudioTimestamp mTimestamp;
+ uint32_t mUnpresentedFrames;
+ } mLatchD, mLatchQ;
+ bool mLatchDValid; // true means mLatchD is valid, and clock it into latch at next opportunity
+ bool mLatchQValid; // true means mLatchQ is valid
};
class MixerThread : public PlaybackThread {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e676365..9622709 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -283,6 +283,11 @@
return mTrack->setParameters(keyValuePairs);
}
+status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
+{
+ return mTrack->getTimestamp(timestamp);
+}
+
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -711,6 +716,29 @@
}
}
+status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
+{
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread == 0) {
+ return false;
+ }
+ Mutex::Autolock _l(thread->mLock);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ if (!playbackThread->mLatchQValid) {
+ return INVALID_OPERATION;
+ }
+ uint32_t unpresentedFrames =
+ ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
+ playbackThread->mSampleRate;
+ uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
+ if (framesWritten < unpresentedFrames) {
+ return INVALID_OPERATION;
+ }
+ timestamp.mPosition = framesWritten - unpresentedFrames;
+ timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
+ return NO_ERROR;
+}
+
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
status_t status = DEAD_OBJECT;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 7f2ec7a..47321e0 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -1424,6 +1424,8 @@
Mutex::Autolock l(mRequestLock);
mRequestQueue.push_back(request);
+ unpauseForNewRequests();
+
return OK;
}
@@ -1489,6 +1491,9 @@
mRepeatingRequests.clear();
mRepeatingRequests.insert(mRepeatingRequests.begin(),
requests.begin(), requests.end());
+
+ unpauseForNewRequests();
+
return OK;
}
@@ -1791,7 +1796,9 @@
mRequestQueue.erase(firstRequest);
}
- // Not paused
+ // In case we've been unpaused by setPaused clearing mDoPause, need to
+ // update internal pause state (capture/setRepeatingRequest unpause
+ // directly).
Mutex::Autolock pl(mPauseLock);
mPaused = false;
@@ -1824,6 +1831,16 @@
return false;
}
+void Camera3Device::RequestThread::unpauseForNewRequests() {
+ // With work to do, mark thread as unpaused.
+ // If paused by request (setPaused), don't resume, to avoid
+ // extra signaling/waiting overhead to waitUntilPaused
+ Mutex::Autolock p(mPauseLock);
+ if (!mDoPause) {
+ mPaused = false;
+ }
+}
+
void Camera3Device::RequestThread::setErrorState(const char *fmt, ...) {
sp<Camera3Device> parent = mParent.promote();
if (parent != NULL) {
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 99e1cc8..6565048 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -314,6 +314,7 @@
// Pause handling
bool waitIfPaused();
+ void unpauseForNewRequests();
// Relay error to parent device object setErrorState
void setErrorState(const char *fmt, ...);