/*cts/tests/tests/media/src/android/media/cts*/
public voidtestSetStereoVolumeMax() throwsException {
final String TEST_NAME= "testSetStereoVolumeMax";
final int TEST_SR =22050;
final int TEST_CONF =AudioFormat.CHANNEL_CONFIGURATION_STEREO;
final int TEST_FORMAT= AudioFormat.ENCODING_PCM_16BIT;
final int TEST_MODE =AudioTrack.MODE_STREAM;
final intTEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
// --------initialization --------------
/*Step1.*/
//根據Hal buff size計算需要爲AudioTrack分配的buff size,
//後續會將這些buff分配給track的mblck，用來進行數據傳輸
int minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT);
/*Step 2.*/
AudioTrack track = newAudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF,
TEST_FORMAT, 2 * minBuffSize,TEST_MODE);
byte data[] = newbyte[minBuffSize];
// -------- test--------------
track.write(data, OFFSET_DEFAULT, data.length);
track.write(data, OFFSET_DEFAULT, data.length);
track.play();
float maxVol =AudioTrack.getMaxVolume();
assertTrue(TEST_NAME, track.setStereoVolume(maxVol, maxVol) == AudioTrack.SUCCESS);
// -------- tear down--------------
track.release();
}
//默認sessionId設置爲0
public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode)
throws IllegalArgumentException {
this(streamType, sampleRateInHz, channelConfig, audioFormat,
bufferSizeInBytes, mode, 0 /*session*/);
}
public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode, int sessionId)
throws IllegalArgumentException {
// mState already == STATE_UNINITIALIZED
// remember which looper is associated with the AudioTrack instantiation
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
//保留創建該AudioTrack的loop，後續可以進行message傳遞
mInitializationLooper = looper;
/**
* 參數檢查
* 1.檢查streamType是合法，並賦值給mStreamType
* 2.檢查sampleRateInHz是否在4000到48000之間，並賦值給mSampleRate
* 3.設置mChannels，將聲道轉換爲 CHANNEL_OUT_MONO(單聲道)
或CHANNEL_OUT_STEREO(雙聲道)
* 4.設置mAudioFormat：
* ENCODING_PCM_16BIT、ENCODING_DEFAULT ---> ENCODING_PCM_16BIT
* ENCODING_PCM_8BIT ---> ENCODING_PCM_8BIT
* 5.設置mDataLoadMode： MODE_STREAM 或MODE_STATIC
*/
audioParamCheck(streamType, sampleRateInHz, channelConfig, audioFormat, mode);
/ *根據採樣精度計算每幀字節大小和緩衝區幀數=緩衝區/每幀大小 ，
*且緩衝區幀數必須是每幀大小的整數倍
*/
audioBuffSizeCheck(bufferSizeInBytes);
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
}
int[] session = new int[1];
session[0] = sessionId;
// native initialization
int initResult = native_setup(new WeakReference<AudioTrack>(this),
mStreamType, mSampleRate, mChannels, mAudioFormat,
mNativeBufferSizeInBytes, mDataLoadMode, session);
if (initResult != SUCCESS) {
loge("Error code "+initResult+" when initializing AudioTrack.");
return; // with mState == STATE_UNINITIALIZED
}
//更新爲native層返回的sessionId
mSessionId = session[0];
if (mDataLoadMode == MODE_STATIC) {
mState = STATE_NO_STATIC_DATA;
} else {
mState = STATE_INITIALIZED;
}
}
static jint
android_media_AudioTrack_native_setup(JNIEnv *env, jobject thiz, jobject weak_this,
jint streamType, jint sampleRateInHertz, jint javaChannelMask,
jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession)
{
ALOGV("sampleRate=%d, audioFormat(from Java)=%d, channel mask=%x, buffSize=%d",
sampleRateInHertz, audioFormat, javaChannelMask, buffSizeInBytes);
uint32_t afSampleRate;
size_t afFrameCount;
//通過AudioSystem從AudioPolicyService中讀取對應音頻流類型的幀數
//需要強調的是，getOutputxxxx函數其實很複雜，以getOutputFrameCount爲例
//具體過程如下：
//1、通過streamType找到strategy，
//2、然後通過strategy找到device，
//3、最後通過device找到output
//4、返回output的FrameCount(硬件緩衝區包含的幀數)
if (AudioSystem::getOutputFrameCount(&afFrameCount, (audio_stream_type_t) streamType) != NO_ERROR) {
ALOGE("Error creating AudioTrack: Could not get AudioSystem frame count.");
return (jint) AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
}
if (AudioSystem::getOutputSamplingRate(&afSampleRate, (audio_stream_type_t) streamType) != NO_ERROR) {
ALOGE("Error creating AudioTrack: Could not get AudioSystem sampling rate.");
return (jint) AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
}
// Java channel masks don't map directly to the native definition, but it's a simple shift
// to skip the two deprecated channel configurations "default" and "mono".
// native層沒有default和mono類型的channel定義，需要轉化
uint32_t nativeChannelMask = ((uint32_t)javaChannelMask) >> 2;
if (!audio_is_output_channel(nativeChannelMask)) {
ALOGE("Error creating AudioTrack: invalid channel mask %#x.", javaChannelMask);
return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK;
}
//計算bit 1的數目，每個bit代表一個聲道
int nbChannels = popcount(nativeChannelMask);
// check the stream type
audio_stream_type_t atStreamType;
switch (streamType) {
case AUDIO_STREAM_VOICE_CALL:
case AUDIO_STREAM_SYSTEM:
case AUDIO_STREAM_RING:
case AUDIO_STREAM_MUSIC:
case AUDIO_STREAM_ALARM:
case AUDIO_STREAM_NOTIFICATION:
case AUDIO_STREAM_BLUETOOTH_SCO:
case AUDIO_STREAM_DTMF:
atStreamType = (audio_stream_type_t) streamType;
break;
default:
ALOGE("Error creating AudioTrack: unknown stream type.");
return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDSTREAMTYPE;
}
// check the format.
// This function was called from Java, so we compare the format against the Java constants
if ((audioFormat != ENCODING_PCM_16BIT) && (audioFormat != ENCODING_PCM_8BIT)) {
ALOGE("Error creating AudioTrack: unsupported audio format.");
return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT;
}
// for the moment 8bitPCM in MODE_STATIC is not supported natively in the AudioTrack C++ class
// so we declare everything as 16bitPCM, the 8->16bit conversion for MODE_STATIC will be handled
// in android_media_AudioTrack_native_write_byte()
// 擴展精度8BIT->16BIT和緩衝區*2，通過將精度設置爲16BIT
//因爲在static模式下，底層只支持16BIT，
//後續寫數據的過程中也要做對應的處理
if ((audioFormat == ENCODING_PCM_8BIT)
&& (memoryMode == MODE_STATIC)) {
ALOGV("android_media_AudioTrack_native_setup(): requesting MODE_STATIC for 8bit \
buff size of %dbytes, switching to 16bit, buff size of %dbytes",
buffSizeInBytes, 2*buffSizeInBytes);
audioFormat = ENCODING_PCM_16BIT;
// we will need twice the memory to store the data
buffSizeInBytes *= 2;
}
// /根據不同的採樣方式得到一個採樣點的字節數,然後計算緩衝區幀數
int bytesPerSample = audioFormat == ENCODING_PCM_16BIT ? 2 : 1;
audio_format_t format = audioFormat == ENCODING_PCM_16BIT ?
AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_8_BIT;
int frameCount = buffSizeInBytes / (nbChannels * bytesPerSample);
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 (jSession == NULL) {
ALOGE("Error creating AudioTrack: invalid session ID pointer");
return (jint) AUDIOTRACK_ERROR;
}
jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
if (nSession == NULL) {
ALOGE("Error creating AudioTrack: Error retrieving session id pointer");
return (jint) AUDIOTRACK_ERROR;
}
int sessionId = nSession[0];
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
nSession = NULL;
// create the native AudioTrack object
sp<AudioTrack> lpTrack = new AudioTrack();
// initialize the callback information:
// this data will be passed with every AudioTrack callback
AudioTrackJniStorage* lpJniStorage = new AudioTrackJniStorage();
lpJniStorage->mStreamType = atStreamType;
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;
// initialize the native AudioTrack object
switch (memoryMode) {
case MODE_STREAM:
//STREAM模式，現在沒有申請共享內存，後續Track對象通過
//AudioFlinger中的Client對象申請heap空間給mCblk進行數據傳輸
lpTrack->set(
atStreamType,// stream type
sampleRateInHertz,
format,// word length, PCM
nativeChannelMask,
frameCount,
AUDIO_OUTPUT_FLAG_NONE,
audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user)
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
break;
case MODE_STATIC:
// STATIC模式，爲AudioTrack分配共享內存區域，大小爲buffSizeInBytes
//後續會將這些buff分配給track的mCblk進行數據傳輸
if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {
ALOGE("Error creating AudioTrack in static mode: error creating mem heap base");
goto native_init_failure;
}
lpTrack->set(
atStreamType,// stream type
sampleRateInHertz,
format,// word length, PCM
nativeChannelMask,
frameCount,
AUDIO_OUTPUT_FLAG_NONE,
audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user));
0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
lpJniStorage->mMemBase,// shared mem
true,// thread can call Java
sessionId);// audio session ID
break;
default:
ALOGE("Unknown mode %d", memoryMode);
goto native_init_failure;
}
if (lpTrack->initCheck() != NO_ERROR) {
ALOGE("Error initializing AudioTrack");
goto native_init_failure;
}
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
//從C層的AudioTrack中更新sessionId
nSession[0] = lpTrack->getSessionId();
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
nSession = NULL;
{ // scope for the lock
Mutex::Autolock l(sLock);
sAudioTrackCallBackCookies.add(&lpJniStorage->mCallbackData);
}
// save our newly created C++ AudioTrack in the "nativeTrackInJavaObj" field
// of the Java object (in mNativeTrackInJavaObj)
setAudioTrack(env, thiz, lpTrack);
// save the JNI resources so we can free them later
//ALOGV("storing lpJniStorage: %x\n", (long)lpJniStorage);
env->SetLongField(thiz, javaAudioTrackFields.jniData, (jlong)lpJniStorage);
return (jint) AUDIOTRACK_SUCCESS;
// failures:
native_init_failure:
if (nSession != NULL) {
env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
}
env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_class);
env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_ref);
delete lpJniStorage;
env->SetLongField(thiz, javaAudioTrackFields.jniData, 0);
return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
}
static void audioCallback(int event, void* user, void *info) {
audiotrack_callback_cookie *callbackInfo = (audiotrack_callback_cookie *)user;
{
Mutex::Autolock l(sLock);
if (sAudioTrackCallBackCookies.indexOf(callbackInfo) < 0) {
return;
}
callbackInfo->busy = true;
}
switch (event) {
case AudioTrack::EVENT_MARKER: {
JNIEnv *env = AndroidRuntime::getJNIEnv();
if (user != NULL && env != NULL) {
env->CallStaticVoidMethod(
callbackInfo->audioTrack_class,
javaAudioTrackFields.postNativeEventInJava,
callbackInfo->audioTrack_ref, event, 0,0, NULL);
if (env->ExceptionCheck()) {
env->ExceptionDescribe();
env->ExceptionClear();
}
}
} break;
case AudioTrack::EVENT_NEW_POS: {
JNIEnv *env = AndroidRuntime::getJNIEnv();
if (user != NULL && env != NULL) {
env->CallStaticVoidMethod(
callbackInfo->audioTrack_class,
javaAudioTrackFields.postNativeEventInJava,
callbackInfo->audioTrack_ref, event, 0,0, NULL);
if (env->ExceptionCheck()) {
env->ExceptionDescribe();
env->ExceptionClear();
}
}
} break;
}
{
Mutex::Autolock l(sLock);
callbackInfo->busy = false;
callbackInfo->cond.broadcast();
}
}
status_t set(audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCount = 0,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
int sessionId = 0,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCountInt,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid)
{
//根據transferType、共享內存、回調函數、threadCanCallJava參數
//設置音頻數據傳輸類型 ，一共有4中傳輸類型，不過常用就只有
// CALLBACK和SHARED
switch (transferType) {
case TRANSFER_DEFAULT:
if (sharedBuffer != 0) {
transferType = TRANSFER_SHARED;
} else if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
return BAD_VALUE;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
return BAD_VALUE;
}
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
// FIXME "int" here is legacy and will be replaced by size_t later
if (frameCountInt < 0) {
ALOGE("Invalid frame count %d", frameCountInt);
return BAD_VALUE;
}
size_t frameCount = frameCountInt;
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
sharedBuffer->size());
ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
mOutput = 0;
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (sampleRate == 0) {
uint32_t afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
sampleRate = afSampleRate;
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
channelMask = AUDIO_CHANNEL_OUT_STEREO;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %d", format);
return BAD_VALUE;
}
// AudioFlinger does not currently support 8-bit data in shared memory
//STATCI模式時，不支持8BIT精度，JNI層已轉換到16BIT
if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
ALOGE("8-bit data in shared memory is not supported");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
// or offload was requested
//如果是offload類型或者非PCM類型的數據，強制設爲direct類型
//即如果不是PCM類型，則直接輸出，由硬件進行解析
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
|| !audio_is_linear_pcm(format)) {
ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
? "Offload request, forcing to Direct Output"
: "Not linear PCM, forcing to Direct Output");
flags = (audio_output_flags_t)
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// only allow deep buffering for music stream type
if (streamType != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
if (!audio_is_output_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
mFrameSizeAF = channelCount * sizeof(int16_t);
} else {
mFrameSize = sizeof(uint8_t);
mFrameSizeAF = sizeof(uint8_t);
}
audio_io_handle_t output = AudioSystem::getOutput(
streamType,
sampleRate, format, channelMask,
flags,
offloadInfo);
if (output == 0) {
ALOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0.0f;
mFrameCount = frameCount;
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
mSessionId = sessionId;
if (uid == -1 || (IPCThreadState::self()->getCallingPid() != getpid())) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
mAuxEffectId = 0;
mFlags = flags;
mCbf = cbf;
/*cbf爲JNI層設置的回調函數
因爲cbf是audioCallback不爲空，所以這裏會啓動一個AudioTrackThread線程。
這個線程是用於AudioTrack(native)與AudioTrack(java)間的數據事件通知的，
這就爲上層應用處理事件提供了一個入口，包括：
EVENT_MORE_DATA = 0, /*請求寫入更多數據*/
EVENT_UNDERRUN = 1, /*PCM 緩衝發生了underrun*/
EVENT_LOOP_END = 2, /*到達loop end，loop count!=null從loop start重新開始回放*/
EVENT_MARKER = 3, /*Playback head在指定的位置，參考setMarkerPosition*/
EVENT_NEW_POS = 4, /*Playback head在一個新的位置，參考setPositionUpdatePeriod */
EVENT_BUFFER_END = 5 /*Playback head在buffer末尾*/
*/
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
}
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
format,
frameCount,
flags,
sharedBuffer,
output,
0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
//Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
AudioSystem::releaseOutput(output);
return status;
}
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
mSharedBuffer = sharedBuffer;
mState = STATE_STOPPED;
mUserData = user;
mLoopPeriod = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId);
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
mOutput = output;
return NO_ERROR;
}
// must be called with mLock held
status_t AudioTrack::createTrack_l(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
size_t frameCount,
audio_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
size_t epoch)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
return NO_INIT;
}
// Not all of these values are needed under all conditions, but it is easier to get them all
// 讀取Hal層的各類參數
uint32_t afLatency;
status = AudioSystem::getLatency(output, streamType, &afLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
return NO_INIT;
}
size_t afFrameCount;
status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
if (status != NO_ERROR) {
ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
return NO_INIT;
}
uint32_t afSampleRate;
status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
if (status != NO_ERROR) {
ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
return NO_INIT;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
// either of these use cases:
// use case 1: shared buffer
(sharedBuffer != 0) ||
// use case 2: callback handler
(mCbf != NULL))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
if ((flags & AUDIO_OUTPUT_FLAG_FAST) && sampleRate != afSampleRate) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client due to mismatching sample rate (%d vs %d)",
sampleRate, afSampleRate);
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
}
// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
// n = 1 fast track with single buffering; nBuffering is ignored
// n = 2 fast track with double buffering
// n = 2 normal track, no sample rate conversion
// n = 3 normal track, with sample rate conversion
// (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
// n > 3 very high latency or very small notification interval; nBuffering is ignored
// 根據硬件採樣率來設置緩衝區，最少爲硬件緩衝的兩倍
const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
mNotificationFramesAct = mNotificationFramesReq;
//以下三種情況需要更新frameCount
//1、非PCM類型數據，如果是STATIC類型，則通過緩衝區來
// 獲得緩衝區幀數，否則使用Hal層參數更新
//2、PCM類型數據，如果是STATIC類型，對齊緩衝區buffer，
// 則通過緩衝區來獲得緩衝區幀數
//3、PCM類型數據，如果是STREAM類型，且非FAST TRACK時，
// 通過硬件延時來計算最小緩衝區，比較當前緩衝區和最小緩衝區
// 如果最小緩衝區大於當前緩衝區，則更新緩衝區幀數
//4、如果是FAST TRACK類型，由服務端進行緩衝區幀數的計算和校驗
if (!audio_is_linear_pcm(format)) {
if (sharedBuffer != 0) { //static模式
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
frameCount = afFrameCount;
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
}
} else if (sharedBuffer != 0) { // static模式
// Ensure that buffer alignment matches channel count
// 8-bit data in shared memory is not currently supported by AudioFlinger
// buffer對齊，原理未知
size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
if (mChannelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
sharedBuffer->pointer(), mChannelCount);
return BAD_VALUE;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
//因爲AudioTrack的構造函數沒有傳入frameCount，
frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
// Ensure that buffer depth covers at least audio hardware latency
//根據硬件延遲計算最少需要幾個硬件buffer塊大小的緩衝區
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
afFrameCount, minBufCount, afSampleRate, afLatency);
if (minBufCount <= nBuffering) {
minBufCount = nBuffering;
}
//相當於minFrameCount/ sampleRate = (afFrameCount/ afSampleRate)* minBufCount
//將minBufCount代入，即minFrameCount = (afLatency* sampleRate)/1000
//當硬件有延遲時，需要一定的內存緩衝區來進行緩衝，避免數據丟失，
//值得說明的是理想情況下，SIZE(內存緩衝區)==SIZE(硬件緩衝區)
size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
ALOGV("Minimum buffer size corrected from %d to %d",
frameCount, minFrameCount);
frameCount = minFrameCount;
}
// Make sure that application is notified with sufficient margin before underrun
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
mNotificationFramesAct = frameCount/nBuffering;
}
} else {
// For fast tracks, the frame count calculations and checks are done by server
}
IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
if (mIsTimed) {
trackFlags |= IAudioFlinger::TRACK_TIMED;
}
pid_t tid = -1;
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
//調用Flinger構造Track對象，並返回TrackHand的句柄，需要繼續分析
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
sampleRate,
// AudioFlinger only sees 16-bit PCM
format == AUDIO_FORMAT_PCM_8_BIT ?
AUDIO_FORMAT_PCM_16_BIT : format,
mChannelMask,
frameCount,
&trackFlags,
sharedBuffer,
output,
tid,
&mSessionId,
mName,
mClientUid,
&status);
if (track == 0) {
ALOGE("AudioFlinger could not create track, status: %d", status);
return status;
}
//保存AudioFlinge端申請的track對象(其實是TrackHandl)對象，及Cblk管理的內存
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
return NO_INIT;
}
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
mAudioTrack = track;
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
mCblk = cblk;
size_t temp = cblk->frameCount_;
//比較請求的緩衝區幀數frameCount和實際分配的緩衝區幀數temp，並將請求的幀數更新爲實際幀數
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
// for fast track as it uses a special method of assigning frame count.
ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
//flags中包含創建fast track的標誌位
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
//audioflinge創建完fast track之後返回的標誌位，如果該bit爲1，表示創建成功，否則失敗
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
mAwaitBoost = true;
if (sharedBuffer == 0) {
// Theoretically double-buffering is not required for fast tracks,
// due to tighter scheduling. But in practice, to accommodate kernels with
// scheduling jitter, and apps with computation jitter, we use double-buffering.
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
mNotificationFramesAct = frameCount/nBuffering;
}
}
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
// once denied, do not request again if IAudioTrack is re-created
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
if (sharedBuffer == 0) {
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
mNotificationFramesAct = frameCount/nBuffering;
}
}
}
}
//flags中包含創建offload的標誌位
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
mFlags = flags;
return NO_INIT;
}
}
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.
// 如果是SREAM模式，則需要跳過cblk的地址，如果是STATIC模式，則直接使用shareBuffer的首地址
void* buffers;
if (sharedBuffer == 0) {
buffers = (char*)cblk + sizeof(audio_track_cblk_t);
} else {
buffers = sharedBuffer->pointer();
}
//操作audioflinge端與該effect關聯的track，由於effectId==0，實際上是將該
//track對應的mAuxBuffer置爲unll，相當於初始化該track，具體過程後續分析
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
mLatency = afLatency + (1000*frameCount) / sampleRate;
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
if (frameCount > mReqFrameCount) {
mReqFrameCount = frameCount;
}
// update proxy
// 根據STREAM和STATIC類型構造AudioTrack的代理對象，供客戶端調用
// 保存緩衝區控制塊、緩衝區數據塊首地址、已申請緩衝區幀數、AudioFlinge端每幀大小
if (sharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
} else {
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
mProxy = mStaticProxy;
}
mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
uint16_t(mVolume[LEFT] * 0x1000));
mProxy->setSendLevel(mSendLevel);
mProxy->setSampleRate(mSampleRate);
mProxy->setEpoch(epoch);
mProxy->setMinimum(mNotificationFramesAct);
mDeathNotifier = new DeathNotifier(this);
mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
}
Audio筆記之AudioTrack
播放聲音可以用MediaPlayer和AudioTrack，兩者都提供了java API供應用開發者使用。雖然都可以播放聲音，但兩者還是有很大的區別的。其中最大的區別是MediaPlayer可以播放多種格式的聲音文件，例如MP3，AAC，WAV，OGG，MIDI等。MediaPlayer會在framework層創建對應的音頻解碼器。而AudioTrack只能播放已經解碼的PCM流，如果是文件的話只支持wav格式的音頻文件，因爲wav格式的音頻文件大部分都是PCM流。AudioTrack不創建解碼器，所以只能播放不需要解碼的wav文件。當然兩者之間還是有緊密的聯繫，MediaPlayer在framework層還是會創建AudioTrack，把解碼後的PCM數流傳遞給AudioTrack，AudioTrack再傳遞給AudioFlinger進行混音，然後才傳遞給硬件播放,所以是MediaPlayer包含了AudioTrack。使用AudioTrack播放音樂示例：
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