AudioTrack的write方法有多个重载方法：
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes) {
return write(audioData, offsetInBytes, sizeInBytes, WRITE_BLOCKING);
}
}
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes,
@WriteMode int writeMode) {
// Note: we allow writes of extended integers and compressed formats from a byte array.
if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) {
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes < 0) // detect integer overflow
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
final int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat,
writeMode == WRITE_BLOCKING);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
}
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts) {
return write(audioData, offsetInShorts, sizeInShorts, WRITE_BLOCKING);
}
}
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts,
@WriteMode int writeMode) {
if (mState == STATE_UNINITIALIZED
|| mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT
// use ByteBuffer or byte[] instead for later encodings
|| mAudioFormat > AudioFormat.ENCODING_LEGACY_SHORT_ARRAY_THRESHOLD) {
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0)
|| (offsetInShorts + sizeInShorts < 0) // detect integer overflow
|| (offsetInShorts + sizeInShorts > audioData.length)) {
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
final int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat,
writeMode == WRITE_BLOCKING);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
}
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull float[] audioData, int offsetInFloats, int sizeInFloats,
@WriteMode int writeMode) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) {
Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT");
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0)
|| (offsetInFloats + sizeInFloats < 0) // detect integer overflow
|| (offsetInFloats + sizeInFloats > audioData.length)) {
Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size");
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
final int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat,
writeMode == WRITE_BLOCKING);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
}
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull ByteBuffer audioData, int sizeInBytes,
@WriteMode int writeMode) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) {
Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value");
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
int ret = 0;
if (audioData.isDirect()) {
ret = native_write_native_bytes(audioData,
audioData.position(), sizeInBytes, mAudioFormat,
writeMode == WRITE_BLOCKING);
} else {
ret = native_write_byte(NioUtils.unsafeArray(audioData),
NioUtils.unsafeArrayOffset(audioData) + audioData.position(),
sizeInBytes, mAudioFormat,
writeMode == WRITE_BLOCKING);
}
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
if (ret > 0) {
audioData.position(audioData.position() + ret);
}
return ret;
}
}
//frameworks/base/media/java/android/media/AudioTrack.java
public class AudioTrack extends PlayerBase implements AudioRouting, VolumeAutomation
{
public int write(@NonNull ByteBuffer audioData, int sizeInBytes,
@WriteMode int writeMode, long timestamp) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if (mDataLoadMode != MODE_STREAM) {
Log.e(TAG, "AudioTrack.write() with timestamp called for non-streaming mode track");
return ERROR_INVALID_OPERATION;
}
if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) == 0) {
Log.d(TAG, "AudioTrack.write() called on a regular AudioTrack. Ignoring pts...");
return write(audioData, sizeInBytes, writeMode);
}
if ((audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) {
Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value");
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
// create timestamp header if none exists
if (mAvSyncHeader == null) {
mAvSyncHeader = ByteBuffer.allocate(mOffset);
mAvSyncHeader.order(ByteOrder.BIG_ENDIAN);
mAvSyncHeader.putInt(0x55550002);
}
if (mAvSyncBytesRemaining == 0) {
mAvSyncHeader.putInt(4, sizeInBytes);
mAvSyncHeader.putLong(8, timestamp);
mAvSyncHeader.putInt(16, mOffset);
mAvSyncHeader.position(0);
mAvSyncBytesRemaining = sizeInBytes;
}
// write timestamp header if not completely written already
int ret = 0;
if (mAvSyncHeader.remaining() != 0) {
ret = write(mAvSyncHeader, mAvSyncHeader.remaining(), writeMode);
if (ret < 0) {
Log.e(TAG, "AudioTrack.write() could not write timestamp header!");
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
return ret;
}
if (mAvSyncHeader.remaining() > 0) {
Log.v(TAG, "AudioTrack.write() partial timestamp header written.");
return 0;
}
}
// write audio data
int sizeToWrite = Math.min(mAvSyncBytesRemaining, sizeInBytes);
ret = write(audioData, sizeToWrite, writeMode);
if (ret < 0) {
Log.e(TAG, "AudioTrack.write() could not write audio data!");
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
return ret;
}
mAvSyncBytesRemaining -= ret;
return ret;
}
}
上述方法主要调用如下方法：
native_write_byte
native_write_short
native_write_float
下面分别进行分析：
native_write_byte
native_write_byte为native方法：
private native final int native_write_byte(byte[] audioData,
int offsetInBytes, int sizeInBytes, int format,
boolean isBlocking);
通过查询android_media_AudioTrack.cpp得到：
{"native_write_byte", "([BIIIZ)I", (void *)android_media_AudioTrack_writeArray<jbyteArray>},因此会调用android_media_AudioTrack_writeArray方法：
//frameworks/base/core/jni/android_media_AudioTrack.cpp
static jint android_media_AudioTrack_writeArray(JNIEnv *env, jobject thiz,
T javaAudioData,
jint offsetInSamples, jint sizeInSamples,
jint javaAudioFormat,
jboolean isWriteBlocking) {
//ALOGV("android_media_AudioTrack_writeArray(offset=%d, sizeInSamples=%d) called",
// offsetInSamples, sizeInSamples);
sp<AudioTrack> lpTrack = getAudioTrack(env, thiz);
if (lpTrack == NULL) {
jniThrowException(env, "java/lang/IllegalStateException",
"Unable to retrieve AudioTrack pointer for write()");
return (jint)AUDIO_JAVA_INVALID_OPERATION;
}
if (javaAudioData == NULL) {
ALOGE("NULL java array of audio data to play");
return (jint)AUDIO_JAVA_BAD_VALUE;
}
// NOTE: We may use GetPrimitiveArrayCritical() when the JNI implementation changes in such
// a way that it becomes much more efficient. When doing so, we will have to prevent the
// AudioSystem callback to be called while in critical section (in case of media server
// process crash for instance)
// get the pointer for the audio data from the java array
auto cAudioData = envGetArrayElements(env, javaAudioData, NULL);
if (cAudioData == NULL) {
ALOGE("Error retrieving source of audio data to play");
return (jint)AUDIO_JAVA_BAD_VALUE; // out of memory or no data to load
}
jint samplesWritten = writeToTrack(lpTrack, javaAudioFormat, cAudioData,
offsetInSamples, sizeInSamples, isWriteBlocking == JNI_TRUE /* blocking */); //调用writeToTrack方法
envReleaseArrayElements(env, javaAudioData, cAudioData, 0);
//ALOGV("write wrote %d (tried %d) samples in the native AudioTrack with offset %d",
// (int)samplesWritten, (int)(sizeInSamples), (int)offsetInSamples);
return samplesWritten;
}
native_write_short
native_write_short为native方法：
private native final int native_write_short(short[] audioData,
int offsetInShorts, int sizeInShorts, int format,
boolean isBlocking);
通过查询android_media_AudioTrack.cpp得到：
{"native_write_short", "([SIIIZ)I",(void *)android_media_AudioTrack_writeArray<jshortArray>},因此同样调用android_media_AudioTrack_writeArray方法。
native_write_float
native_write_float为native方法：
{"native_write_float", "([FIIIZ)I",(void *)android_media_AudioTrack_writeArray<jfloatArray>},因此同样调用android_media_AudioTrack_writeArray方法。
writeToTrack
在android_media_AudioTrack_writeArray方法中调用了writeToTrack方法:
//frameworks/base/core/jni/android_media_AudioTrack.cpp
static jint writeToTrack(const sp<AudioTrack>& track, jint audioFormat, const T *data,
jint offsetInSamples, jint sizeInSamples, bool blocking) {
// give the data to the native AudioTrack object (the data starts at the offset)
ssize_t written = 0;
// regular write() or copy the data to the AudioTrack's shared memory?
size_t sizeInBytes = sizeInSamples * sizeof(T);
if (track->sharedBuffer() == 0) {
written = track->write(data + offsetInSamples, sizeInBytes, blocking); //调用AudioTrack的write方法
// for compatibility with earlier behavior of write(), return 0 in this case
if (written == (ssize_t) WOULD_BLOCK) {
written = 0;
}
} else {
// writing to shared memory, check for capacity
if ((size_t)sizeInBytes > track->sharedBuffer()->size()) {
sizeInBytes = track->sharedBuffer()->size();
}
memcpy(track->sharedBuffer()->unsecurePointer(), data + offsetInSamples, sizeInBytes);
written = sizeInBytes;
}
if (written >= 0) {
return written / sizeof(T);
}
return interpretWriteSizeError(written);
}
write
调用AudioTrack的write方法：
//frameworks/av/media/libaudioclient/AudioTrack.cpp
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
return INVALID_OPERATION;
}
if (isDirect()) {
AutoMutex lock(mLock);
int32_t flags = android_atomic_and(
~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
&mCblk->mFlags);
if (flags & CBLK_INVALID) {
return DEAD_OBJECT;
}
}
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Validation: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
__func__, mPortId, buffer, userSize, userSize);
return BAD_VALUE;
}
size_t written = 0;
Buffer audioBuffer;
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
status_t err = obtainBuffer(&audioBuffer,
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); //调用obtainBuffer获取缓冲区
if (err < 0) {
if (written > 0) {
break;
}
if (err == TIMED_OUT || err == -EINTR) {
err = WOULD_BLOCK;
}
return ssize_t(err);
}
size_t toWrite = audioBuffer.size();
memcpy(audioBuffer.raw, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
}
if (written > 0) {
mFramesWritten += written / mFrameSize;
if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
const sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
// causes wake up of the playback thread, that will callback the client for
// more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
t->wake(); //唤醒playback线程
}
}
}
return written;
}
调用obtainBuffer方法：
//frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
{
if (audioBuffer == NULL) {
if (nonContig != NULL) {
*nonContig = 0;
}
return BAD_VALUE;
}
if (mTransfer != TRANSFER_OBTAIN) {
audioBuffer->frameCount = 0;
audioBuffer->mSize = 0;
audioBuffer->raw = NULL;
if (nonContig != NULL) {
*nonContig = 0;
}
return INVALID_OPERATION;
}
const struct timespec *requested;
struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (ms % 1000) * 1000000;
requested = &timeout;
} else {
ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
requested = NULL;
}
return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
}
调用obtainBuffer方法：
sp<AudioTrackClientProxy> mProxy; // primary owner of the memory
//frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed, size_t *nonContig)
{
// previous and new IAudioTrack sequence numbers are used to detect track re-creation
uint32_t oldSequence = 0;
Proxy::Buffer buffer;
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
do {
// obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
// keep them from going away if another thread re-creates the track during obtainBuffer()
sp<AudioTrackClientProxy> proxy;
sp<IMemory> iMem;
{ // start of lock scope
AutoMutex lock(mLock);
uint32_t newSequence = mSequence;
// did previous obtainBuffer() fail due to media server death or voluntary invalidation?
if (status == DEAD_OBJECT) {
// re-create track, unless someone else has already done so
if (newSequence == oldSequence) {
status = restoreTrack_l("obtainBuffer");
if (status != NO_ERROR) {
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
}
}
oldSequence = newSequence;
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
}
// Keep the extra references
proxy = mProxy;
iMem = mCblkMemory;
if (mState == STATE_STOPPING) {
status = -EINTR;
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
// Non-blocking if track is stopped or paused
if (mState != STATE_ACTIVE) {
requested = &ClientProxy::kNonBlocking;
}
} // end of lock scope
buffer.mFrameCount = audioBuffer->frameCount;
// FIXME starts the requested timeout and elapsed over from scratch
status = proxy->obtainBuffer(&buffer, requested, elapsed);
} while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
audioBuffer->sequence = oldSequence;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
}
return status;
}
调用AudioTrack的restoreTrack_l方法：
sp<media::IAudioTrack> mAudioTrack;
//frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::restoreTrack_l(const char *from)
{
status_t result = NO_ERROR; // logged: make sure to set this before returning.
const int64_t beginNs = systemTime();
mediametrics::Defer defer([&] {
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
.set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
.set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
.set(AMEDIAMETRICS_PROP_WHERE, from)
.record(); });
ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
__func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
++mSequence;
// refresh the audio configuration cache in this process to make sure we get new
// output parameters and new IAudioFlinger in createTrack_l()
AudioSystem::clearAudioConfigCache();
if (isOffloadedOrDirect_l() || mDoNotReconnect) {
// FIXME re-creation of offloaded and direct tracks is not yet implemented;
// reconsider enabling for linear PCM encodings when position can be preserved.
result = DEAD_OBJECT;
return result;
}
// Save so we can return count since creation.
mUnderrunCountOffset = getUnderrunCount_l();
// save the old static buffer position
uint32_t staticPosition = 0;
size_t bufferPosition = 0;
int loopCount = 0;
if (mStaticProxy != 0) {
mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
staticPosition = mStaticProxy->getPosition().unsignedValue();
}
// save the old startThreshold and framecount
const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
const uint32_t originalFrameCount = mProxy->frameCount();
// See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
// causes a lot of churn on the service side, and it can reject starting
// playback of a previously created track. May also apply to other cases.
const int INITIAL_RETRIES = 3;
int retries = INITIAL_RETRIES;
retry:
if (retries < INITIAL_RETRIES) {
// See the comment for clearAudioConfigCache at the start of the function.
AudioSystem::clearAudioConfigCache();
}
mFlags = mOrigFlags;
// If a new IAudioTrack is successfully created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory.
// If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
result = createTrack_l();
if (result == NO_ERROR) {
// take the frames that will be lost by track recreation into account in saved position
// For streaming tracks, this is the amount we obtained from the user/client
// (not the number actually consumed at the server - those are already lost).
if (mStaticProxy == 0) {
mPosition = mReleased;
}
// Continue playback from last known position and restore loop.
if (mStaticProxy != 0) {
if (loopCount != 0) {
mStaticProxy->setBufferPositionAndLoop(bufferPosition,
mLoopStart, mLoopEnd, loopCount);
} else {
mStaticProxy->setBufferPosition(bufferPosition);
if (bufferPosition == mFrameCount) {
ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
}
}
}
// restore volume handler
mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
sp<VolumeShaper::Operation> operationToEnd =
new VolumeShaper::Operation(shaper.mOperation);
// TODO: Ideally we would restore to the exact xOffset position
// as returned by getVolumeShaperState(), but we don't have that
// information when restoring at the client unless we periodically poll
// the server or create shared memory state.
//
// For now, we simply advance to the end of the VolumeShaper effect
// if it has been started.
if (shaper.isStarted()) {
operationToEnd->setNormalizedTime(1.f);
}
media::VolumeShaperConfiguration config;
shaper.mConfiguration->writeToParcelable(&config);
media::VolumeShaperOperation operation;
operationToEnd->writeToParcelable(&operation);
status_t status;
mAudioTrack->applyVolumeShaper(config, operation, &status);
return status;
});
// restore the original start threshold if different than frameCount.
if (originalStartThresholdInFrames != originalFrameCount) {
// Note: mProxy->setStartThresholdInFrames() call is in the Proxy
// and does not trigger a restart.
// (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
// Any start would be triggered on the mState == ACTIVE check below.
const uint32_t currentThreshold =
mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
"%s(%d) startThresholdInFrames changing from %u to %u",
__func__, mPortId, originalStartThresholdInFrames, currentThreshold);
}
if (mState == STATE_ACTIVE) {
mAudioTrack->start(&result);
}
// server resets to zero so we offset
mFramesWrittenServerOffset =
mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
mFramesWrittenAtRestore = mFramesWrittenServerOffset;
}
if (result != NO_ERROR) {
ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
if (--retries > 0) {
// leave time for an eventual race condition to clear before retrying
usleep(500000);
goto retry;
}
// if no retries left, set invalid bit to force restoring at next occasion
// and avoid inconsistent active state on client and server sides
if (mCblk != nullptr) {
android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
}
}
return result;
}
这里最重要的代码就是status = mAudioTrack->start(); ，其中mAudioTrack的定义为sp<IAudioTrack> mAudioTrack;，IAudioTrack接口由AudioFlinger的BnAudioTrack实现， mAudioTrack->start()会调用BnAudioTrack的start函数。
/frameworks/av/media/libaudioclient/IAudioTrack.aidl
interface IAudioTrack {
......
int start();
}
AudioFlinger TrackHandle start
之后就是AudioFlinger的处理了：
9799
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
