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-rw-r--r--media/libmedia/AudioTrack.cpp171
1 files changed, 113 insertions, 58 deletions
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index ff5fe1d..ae016ef 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -30,6 +30,8 @@
#include <media/IAudioFlinger.h>
#include <media/AudioPolicyHelper.h>
#include <media/AudioResamplerPublic.h>
+#include "media/AVMediaExtensions.h"
+#include <cutils/properties.h>
#define WAIT_PERIOD_MS 10
#define WAIT_STREAM_END_TIMEOUT_SEC 120
@@ -163,11 +165,13 @@ status_t AudioTrack::getMinFrameCount(
AudioTrack::AudioTrack()
: mStatus(NO_INIT),
+ mState(STATE_STOPPED),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
- mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mPlaybackRateSet(false)
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
@@ -193,11 +197,13 @@ AudioTrack::AudioTrack(
const audio_attributes_t* pAttributes,
bool doNotReconnect)
: mStatus(NO_INIT),
+ mState(STATE_STOPPED),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
- mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mPlaybackRateSet(false)
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
@@ -223,11 +229,13 @@ AudioTrack::AudioTrack(
const audio_attributes_t* pAttributes,
bool doNotReconnect)
: mStatus(NO_INIT),
+ mState(STATE_STOPPED),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
- mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
+ mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+ mPlaybackRateSet(false)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
@@ -473,7 +481,6 @@ status_t AudioTrack::set(
}
mStatus = NO_ERROR;
- mState = STATE_STOPPED;
mUserData = user;
mLoopCount = 0;
mLoopStart = 0;
@@ -541,6 +548,12 @@ status_t AudioTrack::start()
// force refresh of remaining frames by processAudioBuffer() as last
// write before stop could be partial.
mRefreshRemaining = true;
+
+ // for static track, clear the old flags when start from stopped state
+ if (mSharedBuffer != 0)
+ android_atomic_and(
+ ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
+ &mCblk->mFlags);
}
mNewPosition = mPosition + mUpdatePeriod;
int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
@@ -706,7 +719,7 @@ status_t AudioTrack::setVolume(float left, float right)
mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
- if (isOffloaded_l()) {
+ if (isOffloaded_l() && mAudioTrack != NULL) {
mAudioTrack->signal();
}
return NO_ERROR;
@@ -831,13 +844,13 @@ status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
}
// Check resampler ratios are within bounds
- if (effectiveRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
+ if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
}
- if (effectiveRate * AUDIO_RESAMPLER_UP_RATIO_MAX < mSampleRate) {
+ if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
@@ -846,6 +859,14 @@ status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
//set effective rates
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
+
+ // fallback out of Direct PCM if setPlaybackRate is called on a track offloaded
+ // session. Do this by setting mPlaybackRateSet to true
+ if (mTrackOffloaded) {
+ mPlaybackRateSet = true;
+ android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
+ }
+
return NO_ERROR;
}
@@ -1001,10 +1022,18 @@ status_t AudioTrack::getPosition(uint32_t *position)
return NO_ERROR;
}
+ if (AVMediaUtils::get()->AudioTrackIsPcmOffloaded(mFormat) &&
+ AVMediaUtils::get()->AudioTrackGetPosition(this, position) == NO_ERROR) {
+ return NO_ERROR;
+ }
+
if (mOutput != AUDIO_IO_HANDLE_NONE) {
uint32_t halFrames; // actually unused
- (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
- // FIXME: on getRenderPosition() error, we return OK with frame position 0.
+ status_t status = AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
+ if (status != NO_ERROR) {
+ ALOGW("failed to getRenderPosition for offload session status %d", status);
+ return INVALID_OPERATION;
+ }
}
// FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
// due to hardware latency. We leave this behavior for now.
@@ -1129,11 +1158,16 @@ status_t AudioTrack::createTrack_l()
audio_stream_type_t streamType = mStreamType;
audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;
- status_t status;
- status = AudioSystem::getOutputForAttr(attr, &output,
+ audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
+ if (mPlaybackRateSet == true && mOffloadInfo == NULL && mFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ mOffloadInfo = &tOffloadInfo;
+ }
+ status_t status = AudioSystem::getOutputForAttr(attr, &output,
(audio_session_t)mSessionId, &streamType, mClientUid,
mSampleRate, mFormat, mChannelMask,
mFlags, mSelectedDeviceId, mOffloadInfo);
+ //reset offload info if forced
+ mOffloadInfo = (mOffloadInfo == &tOffloadInfo) ? NULL : mOffloadInfo;
if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x,"
@@ -1141,6 +1175,7 @@ status_t AudioTrack::createTrack_l()
mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
+ mTrackOffloaded = AVMediaUtils::get()->AudioTrackIsTrackOffloaded(output);
{
// Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
// we must release it ourselves if anything goes wrong.
@@ -1203,6 +1238,7 @@ status_t AudioTrack::createTrack_l()
frameCount = mSharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mAfFrameCount;
+ frameCount = AVMediaUtils::get()->AudioTrackGetOffloadFrameCount(frameCount);
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
@@ -1263,7 +1299,7 @@ status_t AudioTrack::createTrack_l()
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
- if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if ((mFlags & AUDIO_OUTPUT_FLAG_DIRECT) || mTrackOffloaded) {
trackFlags |= IAudioFlinger::TRACK_DIRECT;
}
@@ -1838,6 +1874,36 @@ nsecs_t AudioTrack::processAudioBuffer()
// get anchor time to account for callbacks.
const nsecs_t timeBeforeCallbacks = systemTime();
+ // perform callbacks while unlocked
+ if (newUnderrun) {
+ mCbf(EVENT_UNDERRUN, mUserData, NULL);
+ }
+ while (loopCountNotifications > 0) {
+ mCbf(EVENT_LOOP_END, mUserData, NULL);
+ --loopCountNotifications;
+ }
+ if (flags & CBLK_BUFFER_END) {
+ mCbf(EVENT_BUFFER_END, mUserData, NULL);
+ }
+ if (markerReached) {
+ mCbf(EVENT_MARKER, mUserData, &markerPosition);
+ }
+ while (newPosCount > 0) {
+ size_t temp = newPosition;
+ mCbf(EVENT_NEW_POS, mUserData, &temp);
+ newPosition += updatePeriod;
+ newPosCount--;
+ }
+
+ if (mObservedSequence != sequence) {
+ mObservedSequence = sequence;
+ mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
+ // for offloaded tracks, just wait for the upper layers to recreate the track
+ if (isOffloadedOrDirect()) {
+ return NS_INACTIVE;
+ }
+ }
+
if (waitStreamEnd) {
// FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
// should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
@@ -1852,6 +1918,12 @@ nsecs_t AudioTrack::processAudioBuffer()
case NO_ERROR:
case DEAD_OBJECT:
case TIMED_OUT:
+ if (isOffloaded_l()) {
+ if (mCblk->mFlags & (CBLK_INVALID)){
+ // will trigger EVENT_STREAM_END in next iteration
+ return 0;
+ }
+ }
if (status != DEAD_OBJECT) {
// for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
// instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
@@ -1876,36 +1948,6 @@ nsecs_t AudioTrack::processAudioBuffer()
return 0;
}
- // perform callbacks while unlocked
- if (newUnderrun) {
- mCbf(EVENT_UNDERRUN, mUserData, NULL);
- }
- while (loopCountNotifications > 0) {
- mCbf(EVENT_LOOP_END, mUserData, NULL);
- --loopCountNotifications;
- }
- if (flags & CBLK_BUFFER_END) {
- mCbf(EVENT_BUFFER_END, mUserData, NULL);
- }
- if (markerReached) {
- mCbf(EVENT_MARKER, mUserData, &markerPosition);
- }
- while (newPosCount > 0) {
- size_t temp = newPosition;
- mCbf(EVENT_NEW_POS, mUserData, &temp);
- newPosition += updatePeriod;
- newPosCount--;
- }
-
- if (mObservedSequence != sequence) {
- mObservedSequence = sequence;
- mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
- // for offloaded tracks, just wait for the upper layers to recreate the track
- if (isOffloadedOrDirect()) {
- return NS_INACTIVE;
- }
- }
-
// if inactive, then don't run me again until re-started
if (!active) {
return NS_INACTIVE;
@@ -2161,8 +2203,7 @@ uint32_t AudioTrack::updateAndGetPosition_l()
{
// This is the sole place to read server consumed frames
uint32_t newServer = mProxy->getPosition();
- int32_t delta = newServer - mServer;
- mServer = newServer;
+ uint32_t delta = newServer > mServer ? newServer - mServer : 0;
// TODO There is controversy about whether there can be "negative jitter" in server position.
// This should be investigated further, and if possible, it should be addressed.
// A more definite failure mode is infrequent polling by client.
@@ -2171,11 +2212,12 @@ uint32_t AudioTrack::updateAndGetPosition_l()
// That should ensure delta never goes negative for infrequent polling
// unless the server has more than 2^31 frames in its buffer,
// in which case the use of uint32_t for these counters has bigger issues.
- if (delta < 0) {
- ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta);
- delta = 0;
+ if (newServer < mServer) {
+ ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d",
+ (int32_t) newServer - mServer);
}
- return mPosition += (uint32_t) delta;
+ mServer = newServer;
+ return mPosition += delta;
}
bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const
@@ -2237,14 +2279,22 @@ status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
}
}
- // The presented frame count must always lag behind the consumed frame count.
- // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
- status_t status = mAudioTrack->getTimestamp(timestamp);
- if (status != NO_ERROR) {
- ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status);
- return status;
+ status_t status = UNKNOWN_ERROR;
+ //call Timestamp only if its NOT PCM offloaded and NOT Track Offloaded
+ if (!AVMediaUtils::get()->AudioTrackIsPcmOffloaded(mFormat) && !mTrackOffloaded) {
+ // The presented frame count must always lag behind the consumed frame count.
+ // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
+
+ status = mAudioTrack->getTimestamp(timestamp);
+ if (status != NO_ERROR) {
+ ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status);
+ return status;
+ }
+
}
- if (isOffloadedOrDirect_l()) {
+
+ if (isOffloadedOrDirect_l() && !AVMediaUtils::get()->AudioTrackIsPcmOffloaded(mFormat)
+ && !mTrackOffloaded) {
if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
// use cached paused position in case another offloaded track is running.
timestamp.mPosition = mPausedPosition;
@@ -2302,6 +2352,11 @@ status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
}
} else {
// Update the mapping between local consumed (mPosition) and server consumed (mServer)
+
+ if (AVMediaUtils::get()->AudioTrackGetTimestamp(this, &timestamp) == NO_ERROR) {
+ return NO_ERROR;
+ }
+
(void) updateAndGetPosition_l();
// Server consumed (mServer) and presented both use the same server time base,
// and server consumed is always >= presented.
@@ -2315,9 +2370,9 @@ status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
// Convert timestamp position from server time base to client time base.
// TODO The following code should work OK now because timestamp.mPosition is 32-bit.
// But if we change it to 64-bit then this could fail.
- // If (mPosition - mServer) can be negative then should use:
- // (int32_t)(mPosition - mServer)
- timestamp.mPosition += mPosition - mServer;
+ // Split this out instead of using += to prevent unsigned overflow
+ // checks in the outer sum.
+ timestamp.mPosition = timestamp.mPosition + static_cast<int32_t>(mPosition) - mServer;
// Immediately after a call to getPosition_l(), mPosition and
// mServer both represent the same frame position. mPosition is
// in client's point of view, and mServer is in server's point of
generated by cgit v1.1 at 2024-05-07 05:26:55 +0000