Merge "set AAC MPEG version to MPEG-2"
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 9c9721a..56896b9 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -52,7 +52,10 @@
// the combinations that are actually needed within AudioFlinger. If the list of combinations grows
// too large, then this decision should be re-visited.
// Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
-typedef unsigned NBAIO_Format;
+struct NBAIO_Format {
+//private:
+ unsigned mPacked;
+};
extern const NBAIO_Format Format_Invalid;
@@ -60,9 +63,11 @@
size_t Format_frameSize(const NBAIO_Format& format);
// Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(const NBAIO_Format& format);
+// or -1 if frame size is not a power of 2
+int Format_frameBitShift(const NBAIO_Format& format);
// Convert a sample rate in Hz and channel count to an NBAIO_Format
+// FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
// Return the sample rate in Hz of an NBAIO_Format
@@ -76,6 +81,12 @@
typedef ssize_t (*readVia_t)(void *user, const void *buffer,
size_t count, int64_t readPTS);
+// Check whether an NBAIO_Format is valid
+bool Format_isValid(const NBAIO_Format& format);
+
+// Compare two NBAIO_Format values
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2);
+
// Abstract class (interface) representing a data port.
class NBAIO_Port : public RefBase {
diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp
index e952a00..4a69104 100644
--- a/media/libnbaio/AudioBufferProviderSource.cpp
+++ b/media/libnbaio/AudioBufferProviderSource.cpp
@@ -28,7 +28,7 @@
NBAIO_Source(format), mProvider(provider), mConsumed(0)
{
ALOG_ASSERT(provider != NULL);
- ALOG_ASSERT(format != Format_Invalid);
+ ALOG_ASSERT(Format_isValid(format));
}
AudioBufferProviderSource::~AudioBufferProviderSource()
diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp
index 05273f6..ae8fac8 100644
--- a/media/libnbaio/AudioStreamInSource.cpp
+++ b/media/libnbaio/AudioStreamInSource.cpp
@@ -40,7 +40,7 @@
ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOffers,
NBAIO_Format counterOffers[], size_t& numCounterOffers)
{
- if (mFormat == Format_Invalid) {
+ if (!Format_isValid(mFormat)) {
mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -67,7 +67,7 @@
ssize_t AudioStreamInSource::read(void *buffer, size_t count)
{
- if (CC_UNLIKELY(mFormat == Format_Invalid)) {
+ if (CC_UNLIKELY(!Format_isValid(mFormat))) {
return NEGOTIATE;
}
ssize_t bytesRead = mStream->read(mStream, buffer, count << mBitShift);
diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp
index e4341d7..aa9810e 100644
--- a/media/libnbaio/AudioStreamOutSink.cpp
+++ b/media/libnbaio/AudioStreamOutSink.cpp
@@ -37,7 +37,7 @@
ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOffers,
NBAIO_Format counterOffers[], size_t& numCounterOffers)
{
- if (mFormat == Format_Invalid) {
+ if (!Format_isValid(mFormat)) {
mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -56,7 +56,7 @@
if (!mNegotiated) {
return NEGOTIATE;
}
- ALOG_ASSERT(mFormat != Format_Invalid);
+ ALOG_ASSERT(Format_isValid(mFormat));
ssize_t ret = mStream->write(mStream, buffer, count << mBitShift);
if (ret > 0) {
ret >>= mBitShift;
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index 8a8155d..51514de 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -24,13 +24,16 @@
size_t Format_frameSize(const NBAIO_Format& format)
{
+ // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
return Format_channelCount(format) * sizeof(short);
}
-size_t Format_frameBitShift(const NBAIO_Format& format)
+int Format_frameBitShift(const NBAIO_Format& format)
{
+ // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
// sizeof(short) == 2, so frame size == 1 << channels
return Format_channelCount(format);
+ // FIXME must return -1 for non-power of 2
}
const NBAIO_Format Format_Invalid = { 0 };
@@ -55,10 +58,10 @@
unsigned Format_sampleRate(const NBAIO_Format& format)
{
- if (format == Format_Invalid) {
+ if (!Format_isValid(format)) {
return 0;
}
- switch (format & Format_SR_Mask) {
+ switch (format.mPacked & Format_SR_Mask) {
case Format_SR_8000:
return 8000;
case Format_SR_11025:
@@ -82,10 +85,10 @@
unsigned Format_channelCount(const NBAIO_Format& format)
{
- if (format == Format_Invalid) {
+ if (!Format_isValid(format)) {
return 0;
}
- switch (format & Format_C_Mask) {
+ switch (format.mPacked & Format_C_Mask) {
case Format_C_1:
return 1;
case Format_C_2:
@@ -97,7 +100,7 @@
NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
{
- NBAIO_Format format;
+ unsigned format;
switch (sampleRate) {
case 8000:
format = Format_SR_8000;
@@ -136,7 +139,9 @@
default:
return Format_Invalid;
}
- return format;
+ NBAIO_Format ret;
+ ret.mPacked = format;
+ return ret;
}
// This is a default implementation; it is expected that subclasses will optimize this.
@@ -218,9 +223,9 @@
{
ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
offers, numOffers, counterOffers, numCounterOffers);
- if (mFormat != Format_Invalid) {
+ if (Format_isValid(mFormat)) {
for (size_t i = 0; i < numOffers; ++i) {
- if (offers[i] == mFormat) {
+ if (Format_isEqual(offers[i], mFormat)) {
mNegotiated = true;
return i;
}
@@ -235,4 +240,14 @@
return (ssize_t) NEGOTIATE;
}
+bool Format_isValid(const NBAIO_Format& format)
+{
+ return format.mPacked != Format_Invalid.mPacked;
+}
+
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2)
+{
+ return format1.mPacked == format2.mPacked;
+}
+
} // namespace android
diff --git a/media/libstagefright/http/Android.mk b/media/libstagefright/http/Android.mk
index baef9ab..a6b481f 100644
--- a/media/libstagefright/http/Android.mk
+++ b/media/libstagefright/http/Android.mk
@@ -1,5 +1,7 @@
LOCAL_PATH:= $(call my-dir)
+ifneq ($(TARGET_BUILD_PDK), true)
+
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
@@ -20,3 +22,5 @@
LOCAL_CFLAGS += -Wno-multichar
include $(BUILD_SHARED_LIBRARY)
+
+endif
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 7c73b19..81f046f 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1783,7 +1783,7 @@
popcount(inStream->common.get_channels(&inStream->common)));
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
- } else if (format == Format_Invalid) {
+ } else if (!Format_isValid(format)) {
kind = TEE_SINK_NO;
} else if (mRecordTeeSink == 0) {
kind = TEE_SINK_NEW;
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 7126e92..28b9d77 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -239,7 +239,7 @@
dumpState->mSampleRate = sampleRate;
}
- if ((format != previousFormat) || (frameCount != previous->mFrameCount)) {
+ if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
// FIXME to avoid priority inversion, don't delete here
delete mixer;
mixer = NULL;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 4b6c74d..6600161 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -94,6 +94,8 @@
bool isReady() const;
void setPaused() { mState = PAUSED; }
void reset();
+ bool isFlushPending() const { return mFlushHwPending; }
+ void flushAck();
bool isOutputTrack() const {
return (mStreamType == AUDIO_STREAM_CNT);
@@ -155,6 +157,7 @@
bool mIsInvalid; // non-resettable latch, set by invalidate()
AudioTrackServerProxy* mAudioTrackServerProxy;
bool mResumeToStopping; // track was paused in stopping state.
+ bool mFlushHwPending; // track requests for thread flush
}; // end of Track
class TimedTrack : public Track {
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 24df473..7de688e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1490,9 +1490,7 @@
status = NO_ERROR;
}
- ALOGV("signal playback thread");
- broadcast_l();
-
+ onAddNewTrack_l();
return status;
}
@@ -2727,12 +2725,6 @@
PlaybackThread::threadLoop_standby();
}
-// Empty implementation for standard mixer
-// Overridden for offloaded playback
-void AudioFlinger::PlaybackThread::flushOutput_l()
-{
-}
-
bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
{
return false;
@@ -2764,6 +2756,12 @@
}
}
+void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+{
+ ALOGV("signal playback thread");
+ broadcast_l();
+}
+
void AudioFlinger::MixerThread::threadLoop_mix()
{
// obtain the presentation timestamp of the next output buffer
@@ -4003,6 +4001,17 @@
sp<Track> l = mLatestActiveTrack.promote();
bool last = l.get() == track;
+ if (track->isInvalid()) {
+ ALOGW("An invalidated track shouldn't be in active list");
+ tracksToRemove->add(track);
+ continue;
+ }
+
+ if (track->mState == TrackBase::IDLE) {
+ ALOGW("An idle track shouldn't be in active list");
+ continue;
+ }
+
if (track->isPausing()) {
track->setPaused();
if (last) {
@@ -4021,6 +4030,11 @@
mBytesRemaining = 0; // stop writing
}
tracksToRemove->add(track);
+ } else if (track->isFlushPending()) {
+ track->flushAck();
+ if (last) {
+ mFlushPending = true;
+ }
} else if (track->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -4070,7 +4084,6 @@
// seek when resuming.
if (previousTrack->sessionId() != track->sessionId()) {
previousTrack->invalidate();
- mFlushPending = true;
}
}
}
@@ -4146,9 +4159,6 @@
// if resume is received before pause is executed.
if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
mOutput->stream->pause(mOutput->stream);
- if (!doHwPause) {
- doHwResume = true;
- }
}
if (mFlushPending) {
flushHw_l();
@@ -4164,11 +4174,6 @@
return mixerStatus;
}
-void AudioFlinger::OffloadThread::flushOutput_l()
-{
- mFlushPending = true;
-}
-
// must be called with thread mutex locked
bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
{
@@ -4209,6 +4214,8 @@
mBytesRemaining = 0;
mPausedWriteLength = 0;
mPausedBytesRemaining = 0;
+ mHwPaused = false;
+
if (mUseAsyncWrite) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -4219,6 +4226,18 @@
}
}
+void AudioFlinger::OffloadThread::onAddNewTrack_l()
+{
+ sp<Track> previousTrack = mPreviousTrack.promote();
+ sp<Track> latestTrack = mLatestActiveTrack.promote();
+
+ if (previousTrack != 0 && latestTrack != 0 &&
+ (previousTrack->sessionId() != latestTrack->sessionId())) {
+ mFlushPending = true;
+ }
+ PlaybackThread::onAddNewTrack_l();
+}
+
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 1745ea1..9bf59d8 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -65,7 +65,7 @@
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(int event, int param) :
- ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
+ ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {}
virtual ~IoConfigEvent() {}
int event() const { return mEvent; }
@@ -394,7 +394,7 @@
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
-
+ virtual void onAddNewTrack_l();
// ThreadBase virtuals
virtual void preExit();
@@ -631,7 +631,6 @@
protected:
// 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:
@@ -750,11 +749,11 @@
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_exit();
- virtual void flushOutput_l();
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
+ virtual void onAddNewTrack_l();
private:
void flushHw_l();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ecedca9..8004b99 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -146,7 +146,7 @@
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
- if (pipeFormat != Format_Invalid) {
+ if (Format_isValid(pipeFormat)) {
Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {pipeFormat};
@@ -347,7 +347,8 @@
mCachedVolume(1.0),
mIsInvalid(false),
mAudioTrackServerProxy(NULL),
- mResumeToStopping(false)
+ mResumeToStopping(false),
+ mFlushHwPending(false)
{
if (mCblk != NULL) {
if (sharedBuffer == 0) {
@@ -731,6 +732,7 @@
mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
}
+ mFlushHwPending = true;
mResumeToStopping = false;
} else {
if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
@@ -751,11 +753,19 @@
// Prevent flush being lost if the track is flushed and then resumed
// before mixer thread can run. This is important when offloading
// because the hardware buffer could hold a large amount of audio
- playbackThread->flushOutput_l();
playbackThread->broadcast_l();
}
}
+// must be called with thread lock held
+void AudioFlinger::PlaybackThread::Track::flushAck()
+{
+ if (!isOffloaded())
+ return;
+
+ mFlushHwPending = false;
+}
+
void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before