进行AudioTrack相关操作前需要先创建一个Track，如下为CreateTrack的流程：
首先在AudioTrack的构造函数中会调用native_setup方法：
//frameworks/base/media/java/android/media/AudioTrack.java
public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId)
throws IllegalArgumentException {
this(attributes, format, bufferSizeInBytes, mode, sessionId, false /*offload*/,
ENCAPSULATION_MODE_NONE, null /* tunerConfiguration */);
}
private AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId, boolean offload, int encapsulationMode,
@Nullable TunerConfiguration tunerConfiguration)
throws IllegalArgumentException {
super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
// mState already == STATE_UNINITIALIZED
mConfiguredAudioAttributes = attributes; // object copy not needed, immutable.
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat");
}
// Check if we should enable deep buffer mode
if (shouldEnablePowerSaving(mAttributes, format, bufferSizeInBytes, mode)) {
mAttributes = new AudioAttributes.Builder(mAttributes)
.replaceFlags((mAttributes.getAllFlags()
| AudioAttributes.FLAG_DEEP_BUFFER)
& ~AudioAttributes.FLAG_LOW_LATENCY)
.build();
}
// remember which looper is associated with the AudioTrack instantiation
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
int rate = format.getSampleRate();
if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
rate = 0;
}
int channelIndexMask = 0;
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
channelIndexMask = format.getChannelIndexMask();
}
int channelMask = 0;
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
channelMask = format.getChannelMask();
} else if (channelIndexMask == 0) { // if no masks at all, use stereo
channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT
| AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
}
int encoding = AudioFormat.ENCODING_DEFAULT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
encoding = format.getEncoding();
}
audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);
mOffloaded = offload;
mStreamType = AudioSystem.STREAM_DEFAULT;
audioBuffSizeCheck(bufferSizeInBytes);
mInitializationLooper = looper;
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
}
int[] sampleRate = new int[] {mSampleRate};
int[] session = new int[1];
session[0] = sessionId;
// native initialization
int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,
sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/,
offload, encapsulationMode, tunerConfiguration,
getCurrentOpPackageName()); //调用native_setup方法
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing AudioTrack.");
return; // with mState == STATE_UNINITIALIZED
}
mSampleRate = sampleRate[0];
mSessionId = session[0];
// TODO: consider caching encapsulationMode and tunerConfiguration in the Java object.
if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) != 0) {
int frameSizeInBytes;
if (AudioFormat.isEncodingLinearFrames(mAudioFormat)) {
frameSizeInBytes = mChannelCount * AudioFormat.getBytesPerSample(mAudioFormat);
} else {
frameSizeInBytes = 1;
}
mOffset = ((int) Math.ceil(HEADER_V2_SIZE_BYTES / frameSizeInBytes)) * frameSizeInBytes;
}
if (mDataLoadMode == MODE_STATIC) {
mState = STATE_NO_STATIC_DATA;
} else {
mState = STATE_INITIALIZED;
}
baseRegisterPlayer(mSessionId);
native_setPlayerIId(mPlayerIId); // mPlayerIId now ready to send to native AudioTrack.
}
通过查询JNI代码android_media_AudioTrack.cpp得出调用的函数是android_media_AudioTrack_setup：
//frameworks/base/core/jni/android_media_AudioTrack.cpp
static jint android_media_AudioTrack_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jobject jaa, jintArray jSampleRate,
jint channelPositionMask, jint channelIndexMask,
jint audioFormat, jint buffSizeInBytes, jint memoryMode,
jintArray jSession, jlong nativeAudioTrack,
jboolean offload, jint encapsulationMode,
jobject tunerConfiguration, jstring opPackageName) {
ALOGV("sampleRates=%p, channel mask=%x, index mask=%x, audioFormat(Java)=%d, buffSize=%d,"
" nativeAudioTrack=0x%" PRIX64 ", offload=%d encapsulationMode=%d tuner=%p",
jSampleRate, channelPositionMask, channelIndexMask, audioFormat, buffSizeInBytes,
nativeAudioTrack, offload, encapsulationMode, tunerConfiguration);
if (jSession == NULL) {
ALOGE("Error creating AudioTrack: invalid session ID pointer");
return (jint) AUDIO_JAVA_ERROR;
}
const TunerConfigurationHelper tunerHelper(env, tunerConfiguration);
jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
if (nSession == NULL) {
ALOGE("Error creating AudioTrack: Error retrieving session id pointer");
return (jint) AUDIO_JAVA_ERROR;
}
audio_session_t sessionId = (audio_session_t) nSession[0];
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
nSession = NULL;
jclass clazz = env->GetObjectClass(thiz);
if (clazz == NULL) {
ALOGE("Can't find %s when setting up callback.", kClassPathName);
return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
}
// if we pass in an existing *Native* AudioTrack, we don't need to create/initialize one.
sp<AudioTrack> lpTrack;
const auto lpJniStorage = sp<AudioTrackJniStorage>::make(clazz, weak_this, offload);
if (nativeAudioTrack == 0) {
if (jaa == 0) {
ALOGE("Error creating AudioTrack: invalid audio attributes");
return (jint) AUDIO_JAVA_ERROR;
}
if (jSampleRate == 0) {
ALOGE("Error creating AudioTrack: invalid sample rates");
return (jint) AUDIO_JAVA_ERROR;
}
int* sampleRates = env->GetIntArrayElements(jSampleRate, NULL);
int sampleRateInHertz = sampleRates[0];
env->ReleaseIntArrayElements(jSampleRate, sampleRates, JNI_ABORT);
// Invalid channel representations are caught by !audio_is_output_channel() below.
audio_channel_mask_t nativeChannelMask = nativeChannelMaskFromJavaChannelMasks(
channelPositionMask, channelIndexMask);
if (!audio_is_output_channel(nativeChannelMask)) {
ALOGE("Error creating AudioTrack: invalid native channel mask %#x.", nativeChannelMask);
return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK;
}
uint32_t channelCount = audio_channel_count_from_out_mask(nativeChannelMask);
// check the format.
// This function was called from Java, so we compare the format against the Java constants
audio_format_t format = audioFormatToNative(audioFormat);
if (format == AUDIO_FORMAT_INVALID) {
ALOGE("Error creating AudioTrack: unsupported audio format %d.", audioFormat);
return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT;
}
// compute the frame count
size_t frameCount;
if (audio_has_proportional_frames(format)) {
const size_t bytesPerSample = audio_bytes_per_sample(format);
frameCount = buffSizeInBytes / (channelCount * bytesPerSample);
} else {
frameCount = buffSizeInBytes;
}
// create the native AudioTrack object
ScopedUtfChars opPackageNameStr(env, opPackageName);
// TODO b/182469354: make consistent with AudioRecord
AttributionSourceState attributionSource;
attributionSource.packageName = std::string(opPackageNameStr.c_str());
attributionSource.token = sp<BBinder>::make();
lpTrack = sp<AudioTrack>::make(attributionSource); //创建AudioTrack
// read the AudioAttributes values
auto paa = JNIAudioAttributeHelper::makeUnique();
jint jStatus = JNIAudioAttributeHelper::nativeFromJava(env, jaa, paa.get());
if (jStatus != (jint)AUDIO_JAVA_SUCCESS) {
return jStatus;
}
ALOGV("AudioTrack_setup for usage=%d content=%d flags=0x%#x tags=%s",
paa->usage, paa->content_type, paa->flags, paa->tags);
// initialize the callback information:
// this data will be passed with every AudioTrack callback
audio_offload_info_t offloadInfo;
if (offload == JNI_TRUE) {
offloadInfo = AUDIO_INFO_INITIALIZER;
offloadInfo.format = format;
offloadInfo.sample_rate = sampleRateInHertz;
offloadInfo.channel_mask = nativeChannelMask;
offloadInfo.has_video = false;
offloadInfo.stream_type = AUDIO_STREAM_MUSIC; //required for offload
}
if (encapsulationMode != 0) {
offloadInfo = AUDIO_INFO_INITIALIZER;
offloadInfo.format = format;
offloadInfo.sample_rate = sampleRateInHertz;
offloadInfo.channel_mask = nativeChannelMask;
offloadInfo.stream_type = AUDIO_STREAM_MUSIC;
offloadInfo.encapsulation_mode =
static_cast<audio_encapsulation_mode_t>(encapsulationMode);
offloadInfo.content_id = tunerHelper.getContentId();
offloadInfo.sync_id = tunerHelper.getSyncId();
}
// initialize the native AudioTrack object
status_t status = NO_ERROR;
switch (memoryMode) {
case MODE_STREAM:
//调用AudioTrack的set函数
status = lpTrack->set(AUDIO_STREAM_DEFAULT, // stream type, but more info conveyed
// in paa (last argument)
sampleRateInHertz,
format, // word length, PCM
nativeChannelMask, offload ? 0 : frameCount,
offload ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
: AUDIO_OUTPUT_FLAG_NONE,
lpJniStorage,
0, // notificationFrames == 0 since not using EVENT_MORE_DATA
// to feed the AudioTrack
0, // shared mem
true, // thread can call Java
sessionId, // audio session ID
offload ? AudioTrack::TRANSFER_SYNC_NOTIF_CALLBACK
: AudioTrack::TRANSFER_SYNC,
(offload || encapsulationMode) ? &offloadInfo : NULL,
AttributionSourceState(), // default uid, pid values
paa.get());
break;
case MODE_STATIC:
{
// AudioTrack is using shared memory
const auto iMem = allocSharedMem(buffSizeInBytes);
if (iMem == nullptr) {
ALOGE("Error creating AudioTrack in static mode: error creating mem heap base");
goto native_init_failure;
}
status = lpTrack->set(AUDIO_STREAM_DEFAULT, // stream type, but more info conveyed
// in paa (last argument)
sampleRateInHertz,
format, // word length, PCM
nativeChannelMask, frameCount, AUDIO_OUTPUT_FLAG_NONE,
lpJniStorage,
0, // notificationFrames == 0 since not using EVENT_MORE_DATA
// to feed the AudioTrack
iMem, // shared mem
true, // thread can call Java
sessionId, // audio session ID
AudioTrack::TRANSFER_SHARED,
nullptr , // default offloadInfo
AttributionSourceState(), // default uid, pid values
paa.get());
break;
}
default:
ALOGE("Unknown mode %d", memoryMode);
goto native_init_failure;
}
if (status != NO_ERROR) {
ALOGE("Error %d initializing AudioTrack", status);
goto native_init_failure;
}
// Set caller name so it can be logged in destructor.
// MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_JAVA
lpTrack->setCallerName("java");
} else { // end if (nativeAudioTrack == 0)
lpTrack = sp<AudioTrack>::fromExisting(reinterpret_cast<AudioTrack*>(nativeAudioTrack));
// TODO: We need to find out which members of the Java AudioTrack might
// need to be initialized from the Native AudioTrack
// these are directly returned from getters:
// mSampleRate
// mAudioFormat
// mStreamType
// mChannelConfiguration
// mChannelCount
// mState (?)
// mPlayState (?)
// these may be used internally (Java AudioTrack.audioParamCheck():
// mChannelMask
// mChannelIndexMask
// mDataLoadMode
// initialize the callback information:
// this data will be passed with every AudioTrack callback
// TODO this callback information is useless, it isn't passed to the
// native AudioTrack object
/*
lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
// we use a weak reference so the AudioTrack object can be garbage collected.
lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
lpJniStorage->mCallbackData.busy = false;
*/
}
lpJniStorage->mAudioTrackCallback =
sp<JNIAudioTrackCallback>::make(env, thiz, lpJniStorage->getAudioTrackWeakRef(),
javaAudioTrackFields.postNativeEventInJava);
lpTrack->setAudioTrackCallback(lpJniStorage->mAudioTrackCallback); //设置回调
nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
if (nSession == NULL) {
ALOGE("Error creating AudioTrack: Error retrieving session id pointer");
goto native_init_failure;
}
// read the audio session ID back from AudioTrack in case we create a new session
nSession[0] = lpTrack->getSessionId();
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
nSession = NULL;
{
const jint elements[1] = { (jint) lpTrack->getSampleRate() };
env->SetIntArrayRegion(jSampleRate, 0, 1, elements);
}
// save our newly created C++ AudioTrack in the "nativeTrackInJavaObj" field
// of the Java object (in mNativeTrackInJavaObj)
setFieldSp(env, thiz, lpTrack, javaAudioTrackFields.nativeTrackInJavaObj);
// save the JNI resources so we can free them later
//ALOGV("storing lpJniStorage: %x\n", (long)lpJniStorage);
setFieldSp(env, thiz, lpJniStorage, javaAudioTrackFields.jniData);
// since we had audio attributes, the stream type was derived from them during the
// creation of the native AudioTrack: push the same value to the Java object
env->SetIntField(thiz, javaAudioTrackFields.fieldStreamType, (jint) lpTrack->streamType());
return (jint) AUDIO_JAVA_SUCCESS;
// failures:
native_init_failure:
if (nSession != NULL) {
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
}
setFieldSp(env, thiz, sp<AudioTrack>{}, javaAudioTrackFields.nativeTrackInJavaObj);
setFieldSp(env, thiz, sp<AudioTrackJniStorage>{}, javaAudioTrackFields.jniData);
// lpTrack goes out of scope, so reference count drops to zero
return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
}
在android_media_AudioTrack_setup函数中会创建AudioTrack并调用AudioTrack的set函数：
//frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
const wp<IAudioTrackCallback>& callback,
int32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
const AttributionSourceState& attributionSource,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
{
LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__);
mInitialized = true;
status_t status;
uint32_t channelCount;
pid_t callingPid;
pid_t myPid;
uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
std::string errorMessage;
// Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
__func__,
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
sessionId, transferType, attributionSource.uid, attributionSource.pid);
mThreadCanCallJava = threadCanCallJava;
// These variables are pulled in an error report, so we initialize them early.
mSelectedDeviceId = selectedDeviceId;
mSessionId = sessionId;
mChannelMask = channelMask;
mReqFrameCount = mFrameCount = frameCount;
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// update format and flags before storing them in mFormat, mOrigFlags and mFlags
if (pAttributes != NULL) {
// stream type shouldn't be looked at, this track has audio attributes
ALOGV("%s(): Building AudioTrack with attributes:"
" usage=%d content=%d flags=0x%x tags=[%s]",
__func__,
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
}
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
} else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
}
// force direct flag if format is not linear PCM
// or offload was requested
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
|| !audio_is_linear_pcm(format)) {
ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
? "%s(): Offload request, forcing to Direct Output"
: "%s(): Not linear PCM, forcing to Direct Output",
__func__);
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// force direct flag if HW A/V sync requested
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
mFormat = format;
mOrigFlags = mFlags = flags;
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
transferType = TRANSFER_SHARED;
} else if (callback == nullptr|| threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
case TRANSFER_SYNC_NOTIF_CALLBACK:
if (callback == nullptr || sharedBuffer != 0) {
errorMessage = StringPrintf(
"%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
convertTransferToText(transferType), __func__);
status = BAD_VALUE;
goto error;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
errorMessage = StringPrintf(
"%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
status = BAD_VALUE;
goto error;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
errorMessage = StringPrintf(
"%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
status = BAD_VALUE;
goto error;
}
break;
default:
errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
status = BAD_VALUE;
goto error;
}
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
__func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
errorMessage = StringPrintf("%s: Track already in use", __func__);
status = INVALID_OPERATION;
goto error;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
status = BAD_VALUE;
goto error;
}
mOriginalStreamType = streamType;
} else {
mOriginalStreamType = AUDIO_STREAM_DEFAULT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
status = BAD_VALUE;
goto error;
}
if (!audio_is_output_channel(channelMask)) {
errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
status = BAD_VALUE;
goto error;
}
channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (audio_has_proportional_frames(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
} else {
ALOG_ASSERT(audio_has_proportional_frames(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
// sampling rate must be specified for direct outputs
if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
errorMessage = StringPrintf(
"%s: sample rate must be specified for direct outputs", __func__);
status = BAD_VALUE;
goto error;
}
// 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
// (b) we can support re-creation of offloaded tracks
if (offloadInfo != NULL) {
mOffloadInfoCopy = *offloadInfo;
} else {
memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
mOffloadInfoCopy.format = format;
mOffloadInfoCopy.sample_rate = sampleRate;
mOffloadInfoCopy.channel_mask = channelMask;
mOffloadInfoCopy.stream_type = streamType;
}
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
if (notificationFrames >= 0) {
mNotificationFramesReq = notificationFrames;
mNotificationsPerBufferReq = 0;
} else {
if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
errorMessage = StringPrintf(
"%s: notificationFrames=%d not permitted for non-fast track",
__func__, notificationFrames);
status = BAD_VALUE;
goto error;
}
if (frameCount > 0) {
ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
__func__, notificationFrames, frameCount);
status = BAD_VALUE;
goto error;
}
mNotificationFramesReq = 0;
const uint32_t minNotificationsPerBuffer = 1;
const uint32_t maxNotificationsPerBuffer = 8;
mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
"%s(): notificationFrames=%d clamped to the range -%u to -%u",
__func__,
notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
}
mNotificationFramesAct = 0;
// TODO b/182392553: refactor or remove
mClientAttributionSource = AttributionSourceState(attributionSource);
callingPid = IPCThreadState::self()->getCallingPid();
myPid = getpid();
if (uid == -1 || (callingPid != myPid)) {
mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
IPCThreadState::self()->getCallingUid()));
}
if (pid == (pid_t)-1 || (callingPid != myPid)) {
mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
}
mAuxEffectId = 0;
mCallback = callback;
if (callback != nullptr) {
mAudioTrackThread = sp<AudioTrackThread>::make(*this); //创建AudioTrack线程
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); //运行AudioTrack线程
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
{
AutoMutex lock(mLock);
status = createTrack_l(); 创建IAudioTrack
}
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
// We do not goto error to prevent double-logging errors.
goto exit;
}
mLoopCount = 0;
mLoopStart = 0;
mLoopEnd = 0;
mLoopCountNotified = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mPosition = 0;
mReleased = 0;
mStartNs = 0;
mStartFromZeroUs = 0;
AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
mPreviousTimestampValid = false;
mTimestampStartupGlitchReported = false;
mTimestampRetrogradePositionReported = false;
mTimestampRetrogradeTimeReported = false;
mTimestampStallReported = false;
mTimestampStaleTimeReported = false;
mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
mStartTs.mPosition = 0;
mUnderrunCountOffset = 0;
mFramesWritten = 0;
mFramesWrittenServerOffset = 0;
mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
mVolumeHandler = new media::VolumeHandler();
error:
if (status != NO_ERROR) {
ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
}
// fall through
exit:
mStatus = status;
return status;
}
这个函数会创建并运行AudioTrack线程，然后调用createTrack_l创建IAudioTrack：
//frameworks/av/media/libaudioclient/AudioTrack.cpp
status_t AudioTrack::createTrack_l()
{
status_t status;
bool callbackAdded = false;
std::string errorMessage;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); IAudioFlinger接口的服务端为audioFlinger ，通过AudioSystem取得
if (audioFlinger == 0) {
errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
__func__, mPortId);
status = DEAD_OBJECT;
goto exit;
}
{
// mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
// After fast request is denied, we will request again if IAudioTrack is re-created.
// Client can only express a preference for FAST. Server will perform additional tests.
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
// either of these use cases:
// use case 1: shared buffer
bool sharedBuffer = mSharedBuffer != 0;
bool transferAllowed =
// use case 2: callback transfer mode
(mTransfer == TRANSFER_CALLBACK) ||
// use case 3: obtain/release mode
(mTransfer == TRANSFER_OBTAIN) ||
// use case 4: synchronous write
((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
&& mThreadCanCallJava);
bool fastAllowed = sharedBuffer || transferAllowed;
if (!fastAllowed) {
ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
" not shared buffer and transfer = %s",
__func__, mPortId,
convertTransferToText(mTransfer));
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
IAudioFlinger::CreateTrackInput input; //配置Track输入
if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
// Legacy: This is based on original parameters even if the track is recreated.
input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
} else {
input.attr = mAttributes;
}
input.config = AUDIO_CONFIG_INITIALIZER;
input.config.sample_rate = mSampleRate;
input.config.channel_mask = mChannelMask;
input.config.format = mFormat;
input.config.offload_info = mOffloadInfoCopy;
input.clientInfo.attributionSource = mClientAttributionSource;
input.clientInfo.clientTid = -1;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
// It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
// application-level code follows all non-blocking design rules, the language runtime
// doesn't also follow those rules, so the thread will not benefit overall.
if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
input.clientInfo.clientTid = mAudioTrackThread->getTid();
}
}
input.sharedBuffer = mSharedBuffer;
input.notificationsPerBuffer = mNotificationsPerBufferReq;
input.speed = 1.0;
if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
(mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
}
input.flags = mFlags;
input.frameCount = mReqFrameCount;
input.notificationFrameCount = mNotificationFramesReq;
input.selectedDeviceId = mSelectedDeviceId;
input.sessionId = mSessionId;
input.audioTrackCallback = mAudioTrackCallback;
media::CreateTrackResponse response;
status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response); 调用audioFlinger的createTrack函数创建PlaybackTrack
IAudioFlinger::CreateTrackOutput output{}; //创建Track输出
if (status == NO_ERROR) {
output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
}
if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
errorMessage = StringPrintf(
"%s(%d): AudioFlinger could not create track, status: %d output %d",
__func__, mPortId, status, output.outputId);
if (status == NO_ERROR) {
status = INVALID_OPERATION; // device not ready
}
goto exit;
}
ALOG_ASSERT(output.audioTrack != 0);
mFrameCount = output.frameCount;
mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
mRoutedDeviceId = output.selectedDeviceId;
mSessionId = output.sessionId;
mStreamType = output.streamType;
mSampleRate = output.sampleRate;
if (mOriginalSampleRate == 0) {
mOriginalSampleRate = mSampleRate;
}
mAfFrameCount = output.afFrameCount;
mAfSampleRate = output.afSampleRate;
mAfLatency = output.afLatencyMs;
mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
// FIXME compare to AudioRecord
std::optional<media::SharedFileRegion> sfr;
output.audioTrack->getCblk(&sfr);
sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
if (iMem == 0) {
errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
status = FAILED_TRANSACTION;
goto exit;
}
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
void *iMemPointer = iMem->unsecurePointer();
if (iMemPointer == NULL) {
errorMessage = StringPrintf(
"%s(%d): Could not get control block pointer", __func__, mPortId);
status = FAILED_TRANSACTION;
goto exit;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
mAudioTrack = output.audioTrack;
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
__func__, mPortId, mReqFrameCount, mFrameCount);
if (!mThreadCanCallJava) {
mAwaitBoost = true;
}
} else {
ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
__func__, mPortId, mReqFrameCount, mFrameCount);
}
}
mFlags = output.flags;
//mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
if (mDeviceCallback != 0) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
}
AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
callbackAdded = true;
}
mPortId = output.portId;
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output.outputId;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
// is the value of pointer() for the shared buffer, otherwise buffers points
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
if (mSharedBuffer == 0) {
buffers = cblk + 1;
} else {
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
// issue (e.g. by copying).
buffers = mSharedBuffer->unsecurePointer();
if (buffers == NULL) {
errorMessage = StringPrintf(
"%s(%d): Could not get buffer pointer", __func__, mPortId);
ALOGE("%s", errorMessage.c_str());
status = FAILED_TRANSACTION;
goto exit;
}
}
mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (mFrameCount > mReqFrameCount) {
mReqFrameCount = mFrameCount;
}
// reset server position to 0 as we have new cblk.
mServer = 0;
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy = mStaticProxy;
}
mProxy->setVolumeLR(gain_minifloat_pack(
gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
mProxy->setSendLevel(mSendLevel);
const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
mProxy->setSampleRate(effectiveSampleRate);
AudioPlaybackRate playbackRateTemp = mPlaybackRate;
playbackRateTemp.mSpeed = effectiveSpeed;
playbackRateTemp.mPitch = effectivePitch;
mProxy->setPlaybackRate(playbackRateTemp);
mProxy->setMinimum(mNotificationFramesAct);
if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
setDualMonoMode_l(mDualMonoMode);
}
if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
}
mDeathNotifier = new DeathNotifier(this);
IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
// This is the first log sent from the AudioTrack client.
// The creation of the audio track by AudioFlinger (in the code above)
// is the first log of the AudioTrack and must be present before
// any AudioTrack client logs will be accepted.
mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
// the following are immutable
.set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
.set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
.set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
.set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
.set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
.set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
.set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
.set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
.set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
.set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
.set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
.set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
.set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
.set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
// the following are NOT immutable
.set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
.set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
.set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
.set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
.set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
.set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
.set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
.set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
.set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
.set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
.set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
.record();
// mSendLevel
// mReqFrameCount?
// mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
// mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
}
exit:
if (status != NO_ERROR) {
if (callbackAdded) {
// note: mOutput is always valid is callbackAdded is true
AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
}
ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
}
mStatus = status;
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;
}
调用audioFlinger的createTrack函数创建PlaybackTrack，是通过IAudioFlinger接口：
//frameworks/av/media/libaudioclient/include/media/IAudioFlinger.h
class IAudioFlinger : public virtual RefBase {
......
virtual status_t createTrack(const media::CreateTrackRequest& input,
media::CreateTrackResponse& output) = 0;
......
}
AudioFlinger createTrack
调用audioFlinger的createTrack方法：
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