播放声音可以用MediaPlayer和AudioTrack，两者都提供了java API供应用开发者使用。虽然都可以播放声音，但两者还是有很大的区别的。其中最大的区别是MediaPlayer可以播放多种格式的声音文件，例如MP3，AAC，WAV，OGG，MIDI等。MediaPlayer会在framework层创建对应的音频解码器。而AudioTrack只能播放已经解码的PCM流，如果是文件的话只支持wav格式的音频文件，因为wav格式的音频文件大部分都是PCM流。AudioTrack不创建解码器，所以只能播放不需要解码的wav文件。当然两者之间还是有紧密的联系，MediaPlayer在framework层还是会创建AudioTrack，把解码后的PCM数流传递给AudioTrack，AudioTrack再传递给AudioFlinger进行混音，然后才传递给硬件播放,所以是MediaPlayer包含了AudioTrack。使用AudioTrack播放音乐示例：
/*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.*/
//根据Hal buff size计算需要为AudioTrack分配的buff size,
//后续会将这些buff分配给track的mblck，用来进行数据传输
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();
}
//默认sessionId设置为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();
}
//保留创建该AudioTrack的loop，后续可以进行message传递
mInitializationLooper = looper;
/**
* 参数检查
* 1.检查streamType是合法，并赋值给mStreamType
* 2.检查sampleRateInHz是否在4000到48000之间，并赋值给mSampleRate
* 3.设置mChannels，将声道转换为 CHANNEL_OUT_MONO(单声道)
或CHANNEL_OUT_STEREO(双声道)
* 4.设置mAudioFormat：
* ENCODING_PCM_16BIT、ENCODING_DEFAULT ---> ENCODING_PCM_16BIT
* ENCODING_PCM_8BIT ---> ENCODING_PCM_8BIT
* 5.设置mDataLoadMode： MODE_STREAM 或MODE_STATIC
*/
audioParamCheck(streamType, sampleRateInHz, channelConfig, audioFormat, mode);
/ *根据采样精度计算每帧字节大小和缓冲区帧数=缓冲区/每帧大小 ，
*且缓冲区帧数必须是每帧大小的整数倍
*/
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
}
//更新为native层返回的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;
//通过AudioSystem从AudioPolicyService中读取对应音频流类型的帧数
//需要强调的是，getOutputxxxx函数其实很复杂，以getOutputFrameCount为例
//具体过程如下：
//1、通过streamType找到strategy，
//2、然后通过strategy找到device，
//3、最后通过device找到output
//4、返回output的FrameCount(硬件缓冲区包含的帧数)
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层没有default和mono类型的channel定义，需要转化
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;
}
//计算bit 1的数目，每个bit代表一个声道
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()
// 扩展精度8BIT->16BIT和缓冲区*2，通过将精度设置为16BIT
//因为在static模式下，底层只支持16BIT，
//后续写数据的过程中也要做对应的处理
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;
}
// /根据不同的采样方式得到一个采样点的字节数,然后计算缓冲区帧数
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，现在没有申请共享内存，后续Track对象通过
//AudioFlinger中的Client对象申请heap空间给mCblk进行数据传输
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，为AudioTrack分配共享内存区域，大小为buffSizeInBytes
//后续会将这些buff分配给track的mCblk进行数据传输
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
//从C层的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)
{
//根据transferType、共享内存、回调函数、threadCanCallJava参数
//设置音频数据传输类型 ，一共有4中传输类型，不过常用就只有
// 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模式时，不支持8BIT精度，JNI层已转换到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类型或者非PCM类型的数据，强制设为direct类型
//即如果不是PCM类型，则直接输出，由硬件进行解析
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为JNI层设置的回调函数
因为cbf是audioCallback不为空，所以这里会启动一个AudioTrackThread线程。
这个线程是用于AudioTrack(native)与AudioTrack(java)间的数据事件通知的，
这就为上层应用处理事件提供了一个入口，包括：
EVENT_MORE_DATA = 0, /*请求写入更多数据*/
EVENT_UNDERRUN = 1, /*PCM 缓冲发生了underrun*/
EVENT_LOOP_END = 2, /*到达loop end，loop count!=null从loop start重新开始回放*/
EVENT_MARKER = 3, /*Playback head在指定的位置，参考setMarkerPosition*/
EVENT_NEW_POS = 4, /*Playback head在一个新的位置，参考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
// 读取Hal层的各类参数
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
// 根据硬件采样率来设置缓冲区，最少为硬件缓冲的两倍
const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
mNotificationFramesAct = mNotificationFramesReq;
//以下三种情况需要更新frameCount
//1、非PCM类型数据，如果是STATIC类型，则通过缓冲区来
// 获得缓冲区帧数，否则使用Hal层参数更新
//2、PCM类型数据，如果是STATIC类型，对齐缓冲区buffer，
// 则通过缓冲区来获得缓冲区帧数
//3、PCM类型数据，如果是STREAM类型，且非FAST TRACK时，
// 通过硬件延时来计算最小缓冲区，比较当前缓冲区和最小缓冲区
// 如果最小缓冲区大于当前缓冲区，则更新缓冲区帧数
//4、如果是FAST TRACK类型，由服务端进行缓冲区帧数的计算和校验
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对齐，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.
//因为AudioTrack的构造函数没有传入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
//根据硬件延迟计算最少需要几个硬件buffer块大小的缓冲区
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;
}
//相当于minFrameCount/ sampleRate = (afFrameCount/ afSampleRate)* minBufCount
//将minBufCount代入，即minFrameCount = (afLatency* sampleRate)/1000
//当硬件有延迟时，需要一定的内存缓冲区来进行缓冲，避免数据丢失，
//值得说明的是理想情况下，SIZE(内存缓冲区)==SIZE(硬件缓冲区)
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;
}
//调用Flinger构造Track对象，并返回TrackHand的句柄，需要继续分析
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端申请的track对象(其实是TrackHandl)对象，及Cblk管理的内存
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_;
//比较请求的缓冲区帧数frameCount和实际分配的缓冲区帧数temp，并将请求的帧数更新为实际帧数
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中包含创建fast track的标志位
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
//audioflinge创建完fast track之后返回的标志位，如果该bit为1，表示创建成功，否则失败
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中包含创建offload的标志位
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，则需要跳过cblk的地址，如果是STATIC模式，则直接使用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端与该effect关联的track，由于effectId==0，实际上是将该
//track对应的mAuxBuffer置为unll，相当于初始化该track，具体过程后续分析
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
// 根据STREAM和STATIC类型构造AudioTrack的代理对象，供客户端调用
// 保存缓冲区控制块、缓冲区数据块首地址、已申请缓冲区帧数、AudioFlinge端每帧大小
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;
}