Merge "Audio policy manager changes for audio effects" into gingerbread
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 9af5871..9a09586 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -86,6 +86,10 @@
bool mStarted;
+ bool mIsFirstBuffer;
+ status_t mFirstBufferResult;
+ MediaBuffer *mFirstBuffer;
+
sp<MediaPlayerBase::AudioSink> mAudioSink;
static void AudioCallback(int event, void *user, void *info);
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index b79ba13..b7bde6b 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -41,6 +41,9 @@
mReachedEOS(false),
mFinalStatus(OK),
mStarted(false),
+ mIsFirstBuffer(false),
+ mFirstBufferResult(OK),
+ mFirstBuffer(NULL),
mAudioSink(audioSink) {
}
@@ -68,6 +71,24 @@
}
}
+ // We allow an optional INFO_FORMAT_CHANGED at the very beginning
+ // of playback, if there is one, getFormat below will retrieve the
+ // updated format, if there isn't, we'll stash away the valid buffer
+ // of data to be used on the first audio callback.
+
+ CHECK(mFirstBuffer == NULL);
+
+ mFirstBufferResult = mSource->read(&mFirstBuffer);
+ if (mFirstBufferResult == INFO_FORMAT_CHANGED) {
+ LOGV("INFO_FORMAT_CHANGED!!!");
+
+ CHECK(mFirstBuffer == NULL);
+ mFirstBufferResult = OK;
+ mIsFirstBuffer = false;
+ } else {
+ mIsFirstBuffer = true;
+ }
+
sp<MetaData> format = mSource->getFormat();
const char *mime;
bool success = format->findCString(kKeyMIMEType, &mime);
@@ -87,6 +108,11 @@
DEFAULT_AUDIOSINK_BUFFERCOUNT,
&AudioPlayer::AudioSinkCallback, this);
if (err != OK) {
+ if (mFirstBuffer != NULL) {
+ mFirstBuffer->release();
+ mFirstBuffer = NULL;
+ }
+
if (!sourceAlreadyStarted) {
mSource->stop();
}
@@ -110,6 +136,11 @@
delete mAudioTrack;
mAudioTrack = NULL;
+ if (mFirstBuffer != NULL) {
+ mFirstBuffer->release();
+ mFirstBuffer = NULL;
+ }
+
if (!sourceAlreadyStarted) {
mSource->stop();
}
@@ -163,6 +194,12 @@
// Make sure to release any buffer we hold onto so that the
// source is able to stop().
+
+ if (mFirstBuffer != NULL) {
+ mFirstBuffer->release();
+ mFirstBuffer = NULL;
+ }
+
if (mInputBuffer != NULL) {
LOGV("AudioPlayer releasing input buffer.");
@@ -247,6 +284,14 @@
Mutex::Autolock autoLock(mLock);
if (mSeeking) {
+ if (mIsFirstBuffer) {
+ if (mFirstBuffer != NULL) {
+ mFirstBuffer->release();
+ mFirstBuffer = NULL;
+ }
+ mIsFirstBuffer = false;
+ }
+
options.setSeekTo(mSeekTimeUs);
if (mInputBuffer != NULL) {
@@ -259,7 +304,17 @@
}
if (mInputBuffer == NULL) {
- status_t err = mSource->read(&mInputBuffer, &options);
+ status_t err;
+
+ if (mIsFirstBuffer) {
+ mInputBuffer = mFirstBuffer;
+ mFirstBuffer = NULL;
+ err = mFirstBufferResult;
+
+ mIsFirstBuffer = false;
+ } else {
+ err = mSource->read(&mInputBuffer, &options);
+ }
CHECK((err == OK && mInputBuffer != NULL)
|| (err != OK && mInputBuffer == NULL));
diff --git a/media/libstagefright/codecs/aacdec/AACDecoder.cpp b/media/libstagefright/codecs/aacdec/AACDecoder.cpp
index f3b281f..8ae1135 100644
--- a/media/libstagefright/codecs/aacdec/AACDecoder.cpp
+++ b/media/libstagefright/codecs/aacdec/AACDecoder.cpp
@@ -99,15 +99,6 @@
!= MP4AUDEC_SUCCESS) {
return ERROR_UNSUPPORTED;
}
-
- // Check on the sampling rate to see whether it is changed.
- int32_t sampleRate;
- CHECK(mMeta->findInt32(kKeySampleRate, &sampleRate));
- if (mConfig->samplingRate != sampleRate) {
- mMeta->setInt32(kKeySampleRate, mConfig->samplingRate);
- LOGW("Sample rate was %d, but now is %d",
- sampleRate, mConfig->samplingRate);
- }
}
return OK;
}
@@ -215,6 +206,19 @@
Int decoderErr = PVMP4AudioDecodeFrame(mConfig, mDecoderBuf);
+ // Check on the sampling rate to see whether it is changed.
+ int32_t sampleRate;
+ CHECK(mMeta->findInt32(kKeySampleRate, &sampleRate));
+ if (mConfig->samplingRate != sampleRate) {
+ mMeta->setInt32(kKeySampleRate, mConfig->samplingRate);
+ LOGW("Sample rate was %d, but now is %d",
+ sampleRate, mConfig->samplingRate);
+ buffer->release();
+ mInputBuffer->release();
+ mInputBuffer = NULL;
+ return INFO_FORMAT_CHANGED;
+ }
+
size_t numOutBytes =
mConfig->frameLength * sizeof(int16_t) * mConfig->desiredChannels;
if (mConfig->aacPlusUpsamplingFactor == 2) {