【很多同学读 Android 系统的源码时感觉比较费力，一定会觉得是自己水平不够见识有限认知水平不足，觉得自己需要多学习多努力多下功夫，但 Android 系统源码质量之烂简直超乎想象。尽管 Android 系统确实实现了很多功能、特性，提供了很多亮点，但不可否认，把 Android 系统的代码称为屎山，可能都有点侮辱屎山了。】
关于 Android 的音频数据处理，音频播放和采集是两个紧密相关的过程，但这两个过程又有着巨大的差别。播放过程的特别之处在于，Android 音频子系统支持多个应用程序共享音频输出设备，多个应用程序的音频数据可以被混音之后再播放出来，Android 音频子系统支持应用程序独占并直接输出音频数据到音频输出设备，此外，Android 音频子系统支持应用程序传入一个编码的音频流，由 Android 音频子系统通过硬件的能力解码并播放出来。音频采集设备则总是应用程序独占的，应用程序要么抢到设备，可以采集音频数据，要么没抢到，什么都做不了。Android 音频子系统音频数据播放部分的数据处理管线形态要更加复杂一点。
在 Android Java 应用中，一般用 AudioTrack 管理并播放单个音频资源，用 AudioRecord 管理从平台的音频输入设备录制音频所需的音频资源。AudioTrack 和 AudioRecord 的 Java 代码位于 frameworks/base/media/java/android/media，它们的 JNI 代码位于 frameworks/base/core/jni，而不是 frameworks/base/media/jni。这样的代码目录组织显得颇为奇怪，令人困扰。
AudioTrack 和 AudioRecord 的实现基于 libaudioclient 库完成，libaudioclient 库的代码位于 frameworks/av/media/libaudioclient。
Java AudioTrack 的基本使用过程是这样的：
- 创建
AudioTrack对象实例
@TargetApi(21)
private static AudioTrack createAudioTrackOnLollipopOrHigher(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes, int streamType) {
Logging.d(TAG, "createAudioTrackOnLollipopOrHigher");
// TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
// performance when Android O is supported. Add some logging in the mean time.
final int nativeOutputSampleRate = AudioTrack.getNativeOutputSampleRate(streamType);
Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
if (sampleRateInHz != nativeOutputSampleRate) {
Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
}
if (usageAttribute != DEFAULT_USAGE) {
Logging.w(TAG, "A non default usage attribute is used: " + usageAttribute);
}
int contentType = AudioAttributes.CONTENT_TYPE_SPEECH;
if (streamType == AudioManager.STREAM_MUSIC) {
contentType = AudioAttributes.CONTENT_TYPE_MUSIC;
}
// Create an audio track where the audio usage is for VoIP and the content type is speech.
return new AudioTrack(
new AudioAttributes.Builder()
.setUsage(usageAttribute)
.setContentType(contentType)
.build(),
new AudioFormat.Builder()
.setEncoding(AudioFormat.ENCODING_PCM_16BIT)
.setSampleRate(sampleRateInHz)
.setChannelMask(channelConfig)
.build(),
bufferSizeInBytes,
AudioTrack.MODE_STREAM,
AudioManager.AUDIO_SESSION_ID_GENERATE);
}
@SuppressWarnings("deprecation") // Deprecated in API level 25.
private static AudioTrack createAudioTrackOnLowerThanLollipop(
int sampleRateInHz, int channelConfig, int bufferSizeInBytes, int streamType) {
return new AudioTrack(streamType, sampleRateInHz, channelConfig,
AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes, AudioTrack.MODE_STREAM);
}
- 启动播放
try {
audioTrack.play();
} catch (IllegalStateException e) {
reportAwAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_EXCEPTION,
"AudioTrack.play failed: " + e.getMessage());
releaseAudioResources();
return false;
}
- 循环向
AudioTrack写入要播放的音频 PCM 数据
private int writeBytes(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
if (Build.VERSION.SDK_INT >= 21) {
return audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
} else {
return audioTrack.write(byteBuffer.array(), byteBuffer.arrayOffset(), sizeInBytes);
}
}
- 音频数据播放结束之后，停止播放
if (audioTrack != null) {
Logging.d(TAG, "Calling AudioTrack.stop...");
try {
audioTrack.stop();
Logging.d(TAG, "AudioTrack.stop is done.");
} catch (IllegalStateException e) {
Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
}
}
- 释放音频资源
private void releaseAudioResources() {
Logging.d(TAG, "releaseAudioResources");
if (audioTrack != null) {
audioTrack.release();
audioTrack = null;
}
}
AudioTrack 对象用来在应用程序中表示一条音频输出流。AudioTrack 的实现中，会与 AudioFlinger 建立一条跨进程的数据通路，应用程序向 AudioTrack 写入的数据会先给到 AudioFlinger，AudioFlinger 对这些数据做一些处理，如和其它应用程序的音频数据混音，可能再通过另一条跨进程的数据通路，将音频数据送给 Audio HAL 服务进程，之后再由 Audio HAL 服务进程送给音频输出设备播放出来。AudioTrack 实现的整体结构如下图：
这里的代码分析基于 android-security-12.0.0_r43 版进行。
AudioTrack 对象构造过程
AudioTrack Java 类继承层次结构如下图：
AudioTrack Java 类构造函数定义 (位于 frameworks/base/media/java/android/media/AudioTrack.java) 如下：
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());
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.
}
AudioTrack Java 类对象构造时，需要传入这样一些关于音频流的配置参数：
- Usage
- ContentType
- PCM 编码格式
- 以赫兹为单位的采样率
- 音频通道配置格式
- 以字节为单位的数据缓冲区大小
- 内存模式，可选值为 MODE_STATIC 或 MODE_STREAM，一般是 MODE_STREAM
- 会话 ID
AudioTrack Java 类对象构造过程如下：
- 检查是否启用 deep buffer 模式；
- 检查音频流的配置参数，如采样率，通道配置格式，PCM 编码格式，缓冲区大小等；
- 初始化部分状态，如 offload 模式，stream type 等；
- 创建一些缓冲区，以便于本地层代码返回数据，如会话 ID 和采样率；
- 执行本底层方法
native_setup()创建本地层资源； - 根据本底层返回的值初始化一些状态，如会话 ID 和采样率；
- 更新状态。
本底层方法 native_setup() 定义 (位于 frameworks/base/core/jni/android_media_AudioTrack.cpp) 如下：
static sp<AudioTrack> setAudioTrack(JNIEnv* env, jobject thiz, const sp<AudioTrack>& at)
{
Mutex::Autolock l(sLock);
sp<AudioTrack> old =
(AudioTrack*)env->GetLongField(thiz, javaAudioTrackFields.nativeTrackInJavaObj);
if (at.get()) {
at->incStrong((void*)setAudioTrack);
}
if (old != 0) {
old->decStrong((void*)setAudioTrack);
}
env->SetLongField(thiz, javaAudioTrackFields.nativeTrackInJavaObj, (jlong)at.get());
return old;
}
// ----------------------------------------------------------------------------
sp<AudioTrack> android_media_AudioTrack_getAudioTrack(JNIEnv* env, jobject audioTrackObj) {
return getAudioTrack(env, audioTrackObj);
}
// ----------------------------------------------------------------------------
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;
AudioTrackJniStorage* lpJniStorage = 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;
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 = new AudioTrack(attributionSource);
// 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
lpJniStorage = new AudioTrackJniStorage();
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.isOffload = offload;
lpJniStorage->mCallbackData.busy = false;
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:
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,
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
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
if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {
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,
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
AudioTrack::TRANSFER_SHARED,
NULL, // 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 = (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
lpJniStorage = new AudioTrackJniStorage();
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 =
new JNIAudioTrackCallback(env, thiz, lpJniStorage->mCallbackData.audioTrack_ref,
javaAudioTrackFields.postNativeEventInJava);
lpTrack->setAudioTrackCallback(lpJniStorage->mAudioTrackCallback);
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);
}
{ // 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);
// 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);
}
env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_class);
env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_ref);
delete lpJniStorage;
env->SetLongField(thiz, javaAudioTrackFields.jniData, 0);
// lpTrack goes out of scope, so reference count drops to zero
return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
}
AudioTrack Java 类提供了一个以指向本底层 AudioTrack 对象的指针为参数的构造函数，如：
/*package*/ AudioTrack(long nativeTrackInJavaObj) {
super(new AudioAttributes.Builder().build(),
AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
// "final"s
mNativeTrackInJavaObj = 0;
mJniData = 0;
// remember which looper is associated with the AudioTrack instantiation
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
mInitializationLooper = looper;
// other initialization...
if (nativeTrackInJavaObj != 0) {
baseRegisterPlayer(AudioSystem.AUDIO_SESSION_ALLOCATE);
deferred_connect(nativeTrackInJavaObj);
} else {
mState = STATE_UNINITIALIZED;
}
}
这个构造函数也将调用本底层方法 native_setup() 创建一些本地层资源，因而可以看到 native_setup() 中指向本底层 AudioTrack 对象的指针非空的分支，这里忽略这个分支，则 native_setup() 方法执行过程如下：
- 检查音频配置参数，如采样率，通道掩码和通道数，采样格式等；
- 根据缓冲区的大小和音频格式计算帧数；
- 创建
android::AudioTrack对象； - 创建
AudioTrackJniStorage对象； - 构造 offload 配置信息；
- 调用
android::AudioTrack对象的set()函数执行初始化； - 将本地层创建的资源等返回给 Java 层，通过设置 Java 对象的字段等方法实现，如采样率、会话 ID、
android::AudioTrack对象、AudioTrackJniStorage对象和 stream type 等。
由于 native_setup() 方法总是在 Java 类 AudioTrack 的构造函数中调用的，因而不用担心直接返回本底层创建的资源，而没有销毁之前的会造成本底层资源泄漏。
AudioTrackJniStorage 用于保存一些与 Java 类 AudioTrack 对象关联的本地层资源，主要是用来让本底层向 Java 层回调事件的对象，以及用来跨进程传输音频数据的对象，这个类的定义 (位于 frameworks/base/core/jni/android_media_AudioTrack.cpp) 如下：
struct audiotrack_callback_cookie {
jclass audioTrack_class;
jobject audioTrack_ref;
bool busy;
Condition cond;
bool isOffload;
};
// keep these values in sync with AudioTrack.java
#define MODE_STATIC 0
#define MODE_STREAM 1
// ----------------------------------------------------------------------------
class AudioTrackJniStorage {
public:
sp<MemoryHeapBase> mMemHeap;
sp<MemoryBase> mMemBase;
audiotrack_callback_cookie mCallbackData{};
sp<JNIDeviceCallback> mDeviceCallback;
sp<JNIAudioTrackCallback> mAudioTrackCallback;
bool allocSharedMem(int sizeInBytes) {
mMemHeap = new MemoryHeapBase(sizeInBytes, 0, "AudioTrack Heap Base");
if (mMemHeap->getHeapID() < 0) {
return false;
}
mMemBase = new MemoryBase(mMemHeap, 0, sizeInBytes);
return true;
}
};
JNIDeviceCallback 和 JNIAudioTrackCallback 实现本底层定义的回调接口，并封装 Java 对象，它们的对象被注册给 android::AudioTrack 对象，在回调接口被调用时，它们通过 JNI 将事件传至 Java 层。如 JNIAudioTrackCallback，其类定义 (位于 frameworks/base/core/jni/android_media_AudioTrackCallback.h) 如下：
namespace android {
#define AUDIO_NATIVE_EVENT_CODEC_FORMAT_CHANGE 100
// TODO(b/149870866) : Extract common part for JNIAudioTrackCallback and JNIDeviceCallback
class JNIAudioTrackCallback : public media::BnAudioTrackCallback {
public:
JNIAudioTrackCallback(JNIEnv* env, jobject thiz, jobject weak_thiz,
jmethodID postEventFromNative);
~JNIAudioTrackCallback() override;
binder::Status onCodecFormatChanged(const std::vector<uint8_t>& audioMetadata) override;
private:
jclass mClass; // Reference to AudioTrack class
jobject mObject; // Weak ref to AudioTrack Java object to call on
jmethodID mPostEventFromNative; // postEventFromNative method ID
jclass mByteBufferClass; // Reference to ByteBuffer class
jmethodID mAllocateDirectMethod; // ByteBuffer.allocateDirect method ID
};
}; // namespace android
它实现 media::BnAudioTrackCallback 接口，其接口实现中，通过 JNI 将事件传给 Java 层，相关代码 (位于 frameworks/base/core/jni/android_media_AudioTrackCallback.cpp) 如下：
using namespace android;
#define BYTE_BUFFER_NAME "java/nio/ByteBuffer"
#define BYTE_BUFFER_ALLOCATE_DIRECT_NAME "allocateDirect"
JNIAudioTrackCallback::JNIAudioTrackCallback(JNIEnv* env, jobject thiz, jobject weak_thiz,
jmethodID postEventFromNative) {
// Hold onto the AudioTrack class for use in calling the static method
// that posts events to the application thread.
jclass clazz = env->GetObjectClass(thiz);
if (clazz == nullptr) {
return;
}
mClass = (jclass)env->NewGlobalRef(clazz);
// We use a weak reference so the AudioTrack object can be garbage collected.
// The reference is only used as a proxy for callbacks.
mObject = env->NewGlobalRef(weak_thiz);
mPostEventFromNative = postEventFromNative;
jclass byteBufferClass = FindClassOrDie(env, BYTE_BUFFER_NAME);
mByteBufferClass = (jclass)env->NewGlobalRef(byteBufferClass);
mAllocateDirectMethod =
GetStaticMethodIDOrDie(env, mByteBufferClass, BYTE_BUFFER_ALLOCATE_DIRECT_NAME,
"(I)Ljava/nio/ByteBuffer;");
}
JNIAudioTrackCallback::~JNIAudioTrackCallback() {
// remove global references
JNIEnv* env = AndroidRuntime::getJNIEnv();
if (env == nullptr) {
return;
}
env->DeleteGlobalRef(mObject);
env->DeleteGlobalRef(mClass);
env->DeleteGlobalRef(mByteBufferClass);
}
binder::Status JNIAudioTrackCallback::onCodecFormatChanged(
const std::vector<uint8_t>& audioMetadata) {
JNIEnv* env = AndroidRuntime::getJNIEnv();
if (env == nullptr) {
return binder::Status::ok();
}
jobject byteBuffer = env->CallStaticObjectMethod(mByteBufferClass, mAllocateDirectMethod,
(jint)audioMetadata.size());
if (env->ExceptionCheck()) {
ALOGW("An exception occurred while allocating direct buffer");
env->ExceptionDescribe();
env->ExceptionClear();
}
if (byteBuffer == nullptr) {
ALOGE("Failed allocating a direct ByteBuffer");
return binder::Status::fromStatusT(NO_MEMORY);
}
uint8_t* byteBufferAddr = (uint8_t*)env->GetDirectBufferAddress(byteBuffer);
std::copy(audioMetadata.begin(), audioMetadata.end(), byteBufferAddr);
env->CallStaticVoidMethod(mClass, mPostEventFromNative, mObject,
AUDIO_NATIVE_EVENT_CODEC_FORMAT_CHANGE, 0, 0, byteBuffer);
if (env->ExceptionCheck()) {
ALOGW("An exception occurred while notifying codec format changed.");
env->ExceptionDescribe();
env->ExceptionClear();
}
return binder::Status::ok();
}
android::AudioTrack 对象构造过程 (位于 frameworks/av/media/libaudioclient/AudioTrack.cpp) 如下：
AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
: mStatus(NO_INIT),
mState(STATE_STOPPED),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
mClientAttributionSource(attributionSource),
mAudioTrackCallback(new AudioTrackCallback())
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
mAttributes.flags = AUDIO_FLAG_NONE;
strcpy(mAttributes.tags, "");
}
android::AudioTrack 的 set() 函数用于执行初始化，这个函数定义 (位于 frameworks/av/media/libaudioclient/AudioTrack.cpp) 如下：
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,
callback_t cbf,
void* user,
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)
{
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));
// 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;
mSelectedDeviceId = selectedDeviceId;
mSessionId = sessionId;
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:
case TRANSFER_SYNC_NOTIF_CALLBACK:
if (cbf == NULL || sharedBuffer != 0) {
ALOGE("%s(): Transfer type %s but cbf == NULL || sharedBuffer != 0",
convertTransferToText(transferType), __func__);
status = BAD_VALUE;
goto exit;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
if (sharedBuffer != 0) {
ALOGE("%s(): Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
status = BAD_VALUE;
goto exit;
}
break;
case TRANSFER_SHARED:
if (sharedBuffer == 0) {
ALOGE("%s(): Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
status = BAD_VALUE;
goto exit;
}
break;
default:
ALOGE("%s(): Invalid transfer type %d",
__func__, transferType);
status = BAD_VALUE;
goto exit;
}
mSharedBuffer = sharedBuffer;
mTransfer = transferType;
mDoNotReconnect = doNotReconnect;
ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
__func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
ALOGV("%s(): streamType %d frameCount %zu flags %04x",
__func__, streamType, frameCount, flags);
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("%s(): Track already in use", __func__);
status = INVALID_OPERATION;
goto exit;
}
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
if (pAttributes == NULL) {
if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("%s(): Invalid stream type %d", __func__, streamType);
status = BAD_VALUE;
goto exit;
}
mOriginalStreamType = streamType;
} else {
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
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);
mOriginalStreamType = AUDIO_STREAM_DEFAULT;
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);
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("%s(): Invalid format %#x", __func__, format);
status = BAD_VALUE;
goto exit;
}
mFormat = format;
if (!audio_is_output_channel(channelMask)) {
ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
status = BAD_VALUE;
goto exit;
}
mChannelMask = channelMask;
channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
// 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);
}
if (flags & 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 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
status = BAD_VALUE;
goto exit;
}
mSampleRate = sampleRate;
mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// 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;
mOffloadInfo = &mOffloadInfoCopy;
} else {
mOffloadInfo = NULL;
memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
}
mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
mSendLevel = 0.0f;
// mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
if (notificationFrames >= 0) {
mNotificationFramesReq = notificationFrames;
mNotificationsPerBufferReq = 0;
} else {
if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
ALOGE("%s(): notificationFrames=%d not permitted for non-fast track",
__func__, notificationFrames);
status = BAD_VALUE;
goto exit;
}
if (frameCount > 0) {
ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
__func__, notificationFrames, frameCount);
status = BAD_VALUE;
goto exit;
}
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;
mOrigFlags = mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
{
AutoMutex lock(mLock);
status = createTrack_l();
}
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
goto exit;
}
mUserData = user;
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();
exit:
mStatus = status;
return status;
}
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
size_t frameCount,
audio_output_flags_t flags,
callback_t cbf,
void* user,
int32_t notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
uid_t uid,
pid_t pid,
const audio_attributes_t* pAttributes,
bool doNotReconnect,
float maxRequiredSpeed,
audio_port_handle_t selectedDeviceId)
{
AttributionSourceState attributionSource;
attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
attributionSource.token = sp<BBinder>::make();
return set(streamType, sampleRate, format,
static_cast<audio_channel_mask_t>(channelMask),
frameCount, flags, cbf, user, notificationFrames, sharedBuffer,
threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
}
这个函数的执行过程如下：
- 检查并处理参数，如共享缓冲区，传输类型，stream type，采样格式，标记，通道掩码和通道数，帧大小，采样率，播放速率，offload 信息等；
- 传入的回调非空时，创建
AudioTrackThread线程； - 请求 AudioFlinger 创建 Track，Track 用于管理应用程序与 AudioFlinger 之间的通信；
- 初始化其它状态。
AudioTrack::createTrack_l() 函数用于请求 AudioFlinger 创建 Track，该函数定义 (位于 frameworks/av/media/libaudioclient/AudioTrack.cpp) 如下：
status_t AudioTrack::createTrack_l()
{
status_t status;
bool callbackAdded = false;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("%s(%d): Could not get audioflinger",
__func__, mPortId);
status = NO_INIT;
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;
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);
IAudioFlinger::CreateTrackOutput output{};
if (status == NO_ERROR) {
output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
}
if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("%s(%d): AudioFlinger could not create track, status: %d output %d",
__func__, mPortId, status, output.outputId);
if (status == NO_ERROR) {
status = NO_INIT;
}
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) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
status = NO_INIT;
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) {
ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
status = NO_INIT;
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 {
ALOGD("%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) {
ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
status = NO_INIT;
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_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 && callbackAdded) {
// note: mOutput is always valid is callbackAdded is true
AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
}
mStatus = status;
// sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
return status;
}
AudioTrack::createTrack_l() 函数的执行过程如下：
- 根据传入的参数构造创建 Track 的请求
IAudioFlinger::CreateTrackInput； - 请求 AudioFlinger 创建 Track，获得输出
IAudioFlinger::CreateTrackOutput，请求 AudioFlinger 创建 Track 时传入的参数表示应用程序的请求配置，部分配置参数可以为默认值或缺省值，AudioFlinger 可能会根据最终选择的设备的实际配置确定一些配置，这些配置由IAudioFlinger::CreateTrackOutput返回； - 根据 AudioFlinger 返回的值重新设置一些配置参数，如会话 ID，帧数，stream type，采样率等；
- AudioFlinger 会返回一个
android::media::IAudioTrack对象，AudioTrack从该对象获得一块用于应用程序和 AudioFlinger 跨进程通信的共享内存块，并基于相关信息创建IMemory对象； - 保存 AudioFlinger 返回的
android::media::IAudioTrack对象； - 应用程序和 AudioFlinger 通过共享内存块上建立的
audio_track_cblk_t结构对象进行通信和同步，将共享内存块在当前进程中映射的内存地址转换为audio_track_cblk_t结构对象指针； - 根据 AudioFlinger 返回的值设置 flags，端口 ID，输出 ID 等；
- 应用程序与 AudioFlinger 通过共享内存块进行数据交换。当执行
AudioTrack::set()函数时，传入的共享缓冲区非空，也就是内存模式为MODE_STATIC的情况，则用于应用程序与 AudioFlinger 之间数据交换的缓冲区buffers，将被初始化为传入的共享缓冲区；否则，初始化为android::media::IAudioTrack对象返回的共享内存块中，大小为audio_track_cblk_t结构体之后的内存地址。也就是说，内存模式为MODE_STATIC时，应用程序与 AudioFlinger 进行通信及同步和数据传递时，用的是两块不同的共享内存块，这两个共享内存块分别由 AudioFlinger 和应用程序创建，其中应用程序创建的共享内存块用于传递数据，AudioFlinger 创建的共享内存块用于同步；内存模式不为MODE_STATIC时，应用程序与 AudioFlinger 之间通过一个内存块，既进行通信和同步，又进行数据传递，这个共享内存块由 AudioFlinger 创建； - 添加 Aux 音效；
- 基于前面获得的
audio_track_cblk_t结构对象和用于应用程序与 AudioFlinger 之间数据交换的缓冲区buffers，以及帧数和帧大小等信息构造代理对象； - 通过代理对象初始化共享的
audio_track_cblk_t结构体； - 更新状态并返回。
AudioFlinger 创建 Track
AudioTrack 请求 AudioFlinger 创建 Track，请求由 AudioFlinger::createTrack() 函数处理，该函数定义 (位于 frameworks/av/services/audioflinger/AudioFlinger.cpp) 如下：
status_t AudioFlinger::createTrack(const media::CreateTrackRequest& _input,
media::CreateTrackResponse& _output)
{
// Local version of VALUE_OR_RETURN, specific to this method's calling conventions.
CreateTrackInput input = VALUE_OR_RETURN_STATUS(CreateTrackInput::fromAidl(_input));
CreateTrackOutput output;
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
status_t lStatus;
audio_stream_type_t streamType;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
std::vector<audio_io_handle_t> secondaryOutputs;
// TODO b/182392553: refactor or make clearer
pid_t clientPid =
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(input.clientInfo.attributionSource.pid));
bool updatePid = (clientPid == (pid_t)-1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
uid_t clientUid =
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(input.clientInfo.attributionSource.uid));
audio_io_handle_t effectThreadId = AUDIO_IO_HANDLE_NONE;
std::vector<int> effectIds;
audio_attributes_t localAttr = input.attr;
AttributionSourceState adjAttributionSource = input.clientInfo.attributionSource;
if (!isAudioServerOrMediaServerUid(callingUid)) {
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
clientUid = callingUid;
updatePid = true;
}
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
if (updatePid) {
ALOGW_IF(clientPid != (pid_t)-1 && clientPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, clientPid);
clientPid = callingPid;
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
}
audio_session_t sessionId = input.sessionId;
if (sessionId == AUDIO_SESSION_ALLOCATE) {
sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
} else if (audio_unique_id_get_use(sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
lStatus = BAD_VALUE;
goto Exit;
}
output.sessionId = sessionId;
output.outputId = AUDIO_IO_HANDLE_NONE;
output.selectedDeviceId = input.selectedDeviceId;
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
adjAttributionSource, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs);
if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
goto Exit;
}
// client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
// but if someone uses binder directly they could bypass that and cause us to crash
if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
// further channel mask checks are performed by createTrack_l() depending on the thread type
if (!audio_is_output_channel(input.config.channel_mask)) {
ALOGE("createTrack() invalid channel mask %#x", input.config.channel_mask);
lStatus = BAD_VALUE;
goto Exit;
}
// further format checks are performed by createTrack_l() depending on the thread type
if (!audio_is_valid_format(input.config.format)) {
ALOGE("createTrack() invalid format %#x", input.config.format);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output.outputId);
lStatus = BAD_VALUE;
goto Exit;
}
client = registerPid(clientPid);
PlaybackThread *effectThread = NULL;
// check if an effect chain with the same session ID is present on another
// output thread and move it here.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output.outputId) {
uint32_t sessions = t->hasAudioSession(sessionId);
if (sessions & ThreadBase::EFFECT_SESSION) {
effectThread = t.get();
break;
}
}
}
ALOGV("createTrack() sessionId: %d", sessionId);
output.sampleRate = input.config.sample_rate;
output.frameCount = input.frameCount;
output.notificationFrameCount = input.notificationFrameCount;
output.flags = input.flags;
output.streamType = streamType;
track = thread->createTrack_l(client, streamType, localAttr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, &output.notificationFrameCount,
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
callingPid, adjAttributionSource, input.clientInfo.clientTid,
&lStatus, portId, input.audioTrackCallback);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
output.afFrameCount = thread->frameCount();
output.afSampleRate = thread->sampleRate();
output.afLatencyMs = thread->latency();
output.portId = portId;
if (lStatus == NO_ERROR) {
// Connect secondary outputs. Failure on a secondary output must not imped the primary
// Any secondary output setup failure will lead to a desync between the AP and AF until
// the track is destroyed.
updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
}
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
// no risk of deadlock because AudioFlinger::mLock is held
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
if (moveEffectChain_l(sessionId, effectThread, thread) == NO_ERROR) {
effectThreadId = thread->id();
effectIds = thread->getEffectIds_l(sessionId);
}
}
// Look for sync events awaiting for a session to be used.
for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
if (mPendingSyncEvents[i]->triggerSession() == sessionId) {
if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
if (lStatus == NO_ERROR) {
(void) track->setSyncEvent(mPendingSyncEvents[i]);
} else {
mPendingSyncEvents[i]->cancel();
}
mPendingSyncEvents.removeAt(i);
i--;
}
}
}
setAudioHwSyncForSession_l(thread, sessionId);
}
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the Track so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
// Don't hold mClientLock when releasing the reference on the track as the
// destructor will acquire it.
{
Mutex::Autolock _cl(mClientLock);
client.clear();
}
track.clear();
goto Exit;
}
// effectThreadId is not NONE if an effect chain corresponding to the track session
// was found on another thread and must be moved on this thread
if (effectThreadId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
}
output.audioTrack = new TrackHandle(track);
_output = VALUE_OR_FATAL(output.toAidl());
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(portId);
}
return lStatus;
}
AudioFlinger 创建 Track 的过程如下：
- 将传入的
android::media::CreateTrackRequest对象转为android::IAudioFlinger::CreateTrackInput对象，如果传入的共享内存块非空，则此时会将其android::media::SharedFileRegion的描述，转换为MemoryBase/MemoryHeapBase的描述； - 根据需要更新输入的调用者 UID 和 PID 等
AttributionSourceState信息，获得输入的音频属性信息audio_attributes_t，根据需要生成会话 ID； - 通过
AudioSystem::getOutputForAttr()请求AudioPolicyService选择音频设备，stream type，port id等； - 检查
AudioPolicyService选择的 stream type、IO handle 等，以及输入的音频配置； - 根据找到的 IO handle 获得对应的
PlaybackThread，AudioPolicyService初始化时请求 AudioFlinger 打开音频输出设备并给它们创建PlaybackThread，这些PlaybackThread对象保存在一个以 IO handle 为键的映射中； - 注册 PID，根据需要创建每个进程一个的
AudioFlinger::Client对象； - 找到具有相同会话 ID 的音效链的其它输出线程；
- 请求
PlaybackThread创建 Track，PlaybackThread返回一个AudioFlinger::PlaybackThread::Track对象； - 设置输出的帧数，采样率，port ID 等；
- 将其它输出线程上，具有相同会话 ID 的音效链移动到当前输出线程上；
- 包装
AudioFlinger::PlaybackThread::Track对象创建TrackHandle对象，并设置给返回值； - 将
android::IAudioFlinger::CreateTrackOutput描述的输出转为android::media::CreateTrackResponse描述的输出并返回。
AudioFlinger 创建 Track 的过程更精简的描述大概如下面这样：
- 根据请求配置信息请求
AudioPolicyService选择音频设备，stream type，port id 等； - 请求
PlaybackThread创建 Track，创建一个AudioFlinger::PlaybackThread::Track对象； - 包装
AudioFlinger::PlaybackThread::Track对象创建TrackHandle对象，并返回给应用程序。
TrackHandle 对象是 binder 对象，这个类的继承层次结构如下：
TrackHandle 对象是一个 binder 服务，应用程序中的 android::AudioTrack 拿到它的客户端代理 android::media::BpAudioTrack 对象，并通过该代理对象控制 AudioFlinger 中 TrackHandle 包装的 AudioFlinger::PlaybackThread::Track 对象。TrackHandle 与 AudioFlinger::PlaybackThread::Track 对象的关系类似于 android::AudioFlingerServerAdapter 与 android::AudioFlinger 对象的关系。
AudioFlinger 通过 AudioSystem 请求 AudioPolicyService 选择音频设备等配置，具体的 AudioSystem::getOutputForAttr() 函数定义 (位于 frameworks/av/media/libaudioclient/AudioSystem.cpp) 如下：
status_t AudioSystem::getOutputForAttr(audio_attributes_t* attr,
audio_io_handle_t* output,
audio_session_t session,
audio_stream_type_t* stream,
const AttributionSourceState& attributionSource,
const audio_config_t* config,
audio_output_flags_t flags,
audio_port_handle_t* selectedDeviceId,
audio_port_handle_t* portId,
std::vector<audio_io_handle_t>* secondaryOutputs) {
if (attr == nullptr) {
ALOGE("%s NULL audio attributes", __func__);
return BAD_VALUE;
}
if (output == nullptr) {
ALOGE("%s NULL output - shouldn't happen", __func__);
return BAD_VALUE;
}
if (selectedDeviceId == nullptr) {
ALOGE("%s NULL selectedDeviceId - shouldn't happen", __func__);
return BAD_VALUE;
}
if (portId == nullptr) {
ALOGE("%s NULL portId - shouldn't happen", __func__);
return BAD_VALUE;
}
if (secondaryOutputs == nullptr) {
ALOGE("%s NULL secondaryOutputs - shouldn't happen", __func__);
return BAD_VALUE;
}
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return NO_INIT;
media::AudioAttributesInternal attrAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_attributes_t_AudioAttributesInternal(*attr));
int32_t sessionAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(session));
media::AudioConfig configAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(*config));
int32_t flagsAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
int32_t selectedDeviceIdAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_port_handle_t_int32_t(*selectedDeviceId));
media::GetOutputForAttrResponse responseAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->getOutputForAttr(attrAidl, sessionAidl, attributionSource, configAidl, flagsAidl,
selectedDeviceIdAidl, &responseAidl)));
*output = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_io_handle_t(responseAidl.output));
if (stream != nullptr) {
*stream = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(responseAidl.stream));
}
*selectedDeviceId = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_port_handle_t(responseAidl.selectedDeviceId));
*portId = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_port_handle_t(responseAidl.portId));
*secondaryOutputs = VALUE_OR_RETURN_STATUS(convertContainer<std::vector<audio_io_handle_t>>(
responseAidl.secondaryOutputs, aidl2legacy_int32_t_audio_io_handle_t));
return OK;
}
注册 PID 时，AudioFlinger::registerPid(pid_t pid) 函数根据需要创建每个进程一个的 AudioFlinger::Client 对象，该函数定义 (位于 frameworks/av/services/audioflinger/AudioFlinger.cpp) 如下：
sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
{
Mutex::Autolock _cl(mClientLock);
// If pid is already in the mClients wp<> map, then use that entry
// (for which promote() is always != 0), otherwise create a new entry and Client.
sp<Client> client = mClients.valueFor(pid).promote();
if (client == 0) {
client = new Client(this, pid);
mClients.add(pid, client);
}
return client;
}
AudioFlinger 使用了一个以 PID 为键的映射来保存 AudioFlinger::Client 对象，当映射中不存在对应 PID 的 AudioFlinger::Client 对象时，则会创建一个。
AudioFlinger::Client 类定义 (位于 frameworks/av/services/audioflinger/AudioFlinger.h) 如下：
class Client : public RefBase {
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
sp<MemoryDealer> heap() const;
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
private:
DISALLOW_COPY_AND_ASSIGN(Client);
const sp<AudioFlinger> mAudioFlinger;
sp<MemoryDealer> mMemoryDealer;
const pid_t mPid;
};
AudioFlinger::Client 类各成员函数实现如下：
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mPid(pid)
{
mMemoryDealer = new MemoryDealer(
audioFlinger->getClientSharedHeapSize(),
(std::string("AudioFlinger::Client(") + std::to_string(pid) + ")").c_str());
}
// Client destructor must be called with AudioFlinger::mClientLock held
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient_l(mPid);
}
sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
AudioFlinger::Client 对象主要维护了一个 MemoryDealer 对象和对 AudioFlinger 的引用。MemoryDealer 对象管理一块用于 AudioFlinger 与应用程序跨进程通信的共享内存，共享内存块的大小从 AudioFlinger::getClientSharedHeapSize() 函数获得。AudioFlinger::getClientSharedHeapSize() 函数定义如下：
status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice, int64_t totalMemory)
{
uid_t uid = IPCThreadState::self()->getCallingUid();
if (!isAudioServerOrSystemServerUid(uid)) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
if (mIsDeviceTypeKnown) {
return INVALID_OPERATION;
}
mIsLowRamDevice = isLowRamDevice;
mTotalMemory = totalMemory;
// mIsLowRamDevice and mTotalMemory are obtained through ActivityManager;
// see ActivityManager.isLowRamDevice() and ActivityManager.getMemoryInfo().
// mIsLowRamDevice generally represent devices with less than 1GB of memory,
// though actual setting is determined through device configuration.
constexpr int64_t GB = 1024 * 1024 * 1024;
mClientSharedHeapSize =
isLowRamDevice ? kMinimumClientSharedHeapSizeBytes
: mTotalMemory < 2 * GB ? 4 * kMinimumClientSharedHeapSizeBytes
: mTotalMemory < 3 * GB ? 8 * kMinimumClientSharedHeapSizeBytes
: mTotalMemory < 4 * GB ? 16 * kMinimumClientSharedHeapSizeBytes
: 32 * kMinimumClientSharedHeapSizeBytes;
mIsDeviceTypeKnown = true;
// TODO: Cache the client shared heap size in a persistent property.
// It's possible that a native process or Java service or app accesses audioserver
// after it is registered by system server, but before AudioService updates
// the memory info. This would occur immediately after boot or an audioserver
// crash and restore. Before update from AudioService, the client would get the
// minimum heap size.
ALOGD("isLowRamDevice:%s totalMemory:%lld mClientSharedHeapSize:%zu",
(isLowRamDevice ? "true" : "false"),
(long long)mTotalMemory,
mClientSharedHeapSize.load());
return NO_ERROR;
}
size_t AudioFlinger::getClientSharedHeapSize() const
{
size_t heapSizeInBytes = property_get_int32("ro.af.client_heap_size_kbyte", 0) * 1024;
if (heapSizeInBytes != 0) { // read-only property overrides all.
return heapSizeInBytes;
}
return mClientSharedHeapSize;
}
AudioFlinger::getClientSharedHeapSize() 函数首先尝试从系统属性 ro.af.client_heap_size_kbyte 获得共享内存块大小，失败时则返回成员变量 mClientSharedHeapSize 的值。在 AudioFlinger 对象构造时，mClientSharedHeapSize 被初始化为 1MB；当 AudioFlinger::setLowRamDevice() 被调用时，mClientSharedHeapSize 的值会根据系统的内存大小来调整，具体来说，规则如下：
- 设备为低内存设备，为 1MB；
- 设备内存小于 2GB，为 4MB；
- 设备内存小于 3GB，为 8MB；
- 设备内存小于 4GB，为 16MB；
- 设备内存大于等于 4GB，为 32MB。
AudioFlinger 中，用于与应用程序进行跨进程通信的共享内存块，是针对进程一次性创建的，而不是针对 AudioTrack 和 AudioRecord 来创建的。特定应用进程的所有 AudioTrack 和 AudioRecord 都从这块共享内存中分配一块内存，用以进行与 AudioFlinger 的同步和数据传递。
AudioFlinger 请求 PlaybackThread 创建 Track，创建一个 AudioFlinger::PlaybackThread::Track 对象通过 AudioFlinger::PlaybackThread::createTrack_l() 函数完成，该函数定义 (位于 frameworks/av/services/audioflinger/Threads.cpp) 如下：
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
size_t *pNotificationFrameCount,
uint32_t notificationsPerBuffer,
float speed,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
audio_output_flags_t *flags,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
pid_t tid,
status_t *status,
audio_port_handle_t portId,
const sp<media::IAudioTrackCallback>& callback)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
sp<Track> track;
status_t lStatus;
audio_output_flags_t outputFlags = mOutput->flags;
audio_output_flags_t requestedFlags = *flags;
uint32_t sampleRate;
if (sharedBuffer != 0 && checkIMemory(sharedBuffer) != NO_ERROR) {
lStatus = BAD_VALUE;
goto Exit;
}
if (*pSampleRate == 0) {
*pSampleRate = mSampleRate;
}
sampleRate = *pSampleRate;
// special case for FAST flag considered OK if fast mixer is present
if (hasFastMixer()) {
outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST);
}
// Check if requested flags are compatible with output stream flags
if ((*flags & outputFlags) != *flags) {
ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)",
*flags, outputFlags);
*flags = (audio_output_flags_t)(*flags & outputFlags);
}
// client expresses a preference for FAST, but we get the final say
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (
// PCM data
audio_is_linear_pcm(format) &&
// TODO: extract as a data library function that checks that a computationally
// expensive downmixer is not required: isFastOutputChannelConversion()
(channelMask == (mChannelMask | mHapticChannelMask) ||
mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
(channelMask == AUDIO_CHANNEL_OUT_MONO
/* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// normal mixer has an associated fast mixer
hasFastMixer() &&
// there are sufficient fast track slots available
(mFastTrackAvailMask != 0)
// FIXME test that MixerThread for this fast track has a capable output HAL
// FIXME add a permission test also?
) {
// static tracks can have any nonzero framecount, streaming tracks check against minimum.
if (sharedBuffer == 0) {
// read the fast track multiplier property the first time it is needed
int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
if (ok != 0) {
ALOGE("%s pthread_once failed: %d", __func__, ok);
}
frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
}
// check compatibility with audio effects.
{ // scope for mLock
Mutex::Autolock _l(mLock);
for (audio_session_t session : {
AUDIO_SESSION_DEVICE,
AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_OUTPUT_MIX,
sessionId,
}) {
sp<EffectChain> chain = getEffectChain_l(session);
if (chain.get() != nullptr) {
audio_output_flags_t old = *flags;
chain->checkOutputFlagCompatibility(flags);
if (old != *flags) {
ALOGV("AUDIO_OUTPUT_FLAGS denied by effect, session=%d old=%#x new=%#x",
(int)session, (int)old, (int)*flags);
}
}
}
}
ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0,
"AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
"mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
"sampleRate=%u mSampleRate=%u "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
audio_is_linear_pcm(format), channelMask, sampleRate,
mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
if (!audio_has_proportional_frames(format)) {
if (sharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mNormalFrameCount;
}
if (notificationFrameCount != frameCount) {
notificationFrameCount = frameCount;
}
} else if (sharedBuffer != 0) {
// FIXME: Ensure client side memory buffers need
// not have additional alignment beyond sample
// (e.g. 16 bit stereo accessed as 32 bit frame).
size_t alignment = audio_bytes_per_sample(format);
if (alignment & 1) {
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
size_t frameSize = channelCount * audio_bytes_per_sample(format);
if (channelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)sharedBuffer->unsecurePointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
sharedBuffer->unsecurePointer(), channelCount);
lStatus = BAD_VALUE;
goto Exit;
}
// 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.
frameCount = sharedBuffer->size() / frameSize;
} else {
size_t minFrameCount = 0;
// For fast tracks we try to respect the application's request for notifications per buffer.
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (notificationsPerBuffer > 0) {
// Avoid possible arithmetic overflow during multiplication.
if (notificationsPerBuffer > SIZE_MAX / mFrameCount) {
ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",
notificationsPerBuffer, mFrameCount);
} else {
minFrameCount = mFrameCount * notificationsPerBuffer;
}
}
} else {
// For normal PCM streaming tracks, update minimum frame count.
// Buffer depth is forced to be at least 2 x the normal mixer frame count and
// cover audio hardware latency.
// This is probably too conservative, but legacy application code may depend on it.
// If you change this calculation, also review the start threshold which is related.
uint32_t latencyMs = latency_l();
if (latencyMs == 0) {
ALOGE("Error when retrieving output stream latency");
lStatus = UNKNOWN_ERROR;
goto Exit;
}
minFrameCount = AudioSystem::calculateMinFrameCount(latencyMs, mNormalFrameCount,
mSampleRate, sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
}
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
// Make sure that application is notified with sufficient margin before underrun.
// The client can divide the AudioTrack buffer into sub-buffers,
// and expresses its desire to server as the notification frame count.
if (sharedBuffer == 0 && audio_is_linear_pcm(format)) {
size_t maxNotificationFrames;
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
// notify every HAL buffer, regardless of the size of the track buffer
maxNotificationFrames = mFrameCount;
} else {
// Triple buffer the notification period for a triple buffered mixer period;
// otherwise, double buffering for the notification period is fine.
//
// TODO: This should be moved to AudioTrack to modify the notification period
// on AudioTrack::setBufferSizeInFrames() changes.
const int nBuffering =
(uint64_t{frameCount} * mSampleRate)
/ (uint64_t{mNormalFrameCount} * sampleRate) == 3 ? 3 : 2;
maxNotificationFrames = frameCount / nBuffering;
// If client requested a fast track but this was denied, then use the smaller maximum.
if (requestedFlags & AUDIO_OUTPUT_FLAG_FAST) {
size_t maxNotificationFramesFastDenied = FMS_20 * sampleRate / 1000;
if (maxNotificationFrames > maxNotificationFramesFastDenied) {
maxNotificationFrames = maxNotificationFramesFastDenied;
}
}
}
if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) {
if (notificationFrameCount == 0) {
ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",
maxNotificationFrames, frameCount);
} else {
ALOGW("Client adjusted notificationFrames from %zu to %zu for frameCount %zu",
notificationFrameCount, maxNotificationFrames, frameCount);
}
notificationFrameCount = maxNotificationFrames;
}
}
*pFrameCount = frameCount;
*pNotificationFrameCount = notificationFrameCount;
switch (mType) {
case DIRECT:
if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
"for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
}
break;
case OFFLOAD:
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
"for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
break;
default:
if (!audio_is_linear_pcm(format)) {
ALOGE("createTrack_l() Bad parameter: format %#x \""
"for output %p with format %#x",
format, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
break;
}
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() audio driver not initialized");
goto Exit;
}
{ // scope for mLock
Mutex::Autolock _l(mLock);
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
product_strategy_t strategy = AudioSystem::getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0 && t->isExternalTrack()) {
product_strategy_t actual = AudioSystem::getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
strategy, actual);
lStatus = BAD_VALUE;
goto Exit;
}
}
}
// Set DIRECT flag if current thread is DirectOutputThread. This can
// happen when the playback is rerouted to direct output thread by
// dynamic audio policy.
// Do NOT report the flag changes back to client, since the client
// doesn't explicitly request a direct flag.
audio_output_flags_t trackFlags = *flags;
if (mType == DIRECT) {
trackFlags = static_cast<audio_output_flags_t>(trackFlags | AUDIO_OUTPUT_FLAG_DIRECT);
}
track = new Track(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, creatorPid, attributionSource, trackFlags,
TrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/, speed);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
// track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
{
Mutex::Autolock _atCbL(mAudioTrackCbLock);
if (callback.get() != nullptr) {
mAudioTrackCallbacks.emplace(track, callback);
}
}
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) {
pid_t callingPid = IPCThreadState::self()->getCallingPid();
// we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
// so ask activity manager to do this on our behalf
sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/);
}
}
lStatus = NO_ERROR;
Exit:
*status = lStatus;
return track;
}
AudioFlinger::PlaybackThread::createTrack_l() 函数执行过程如下：
- 返回采样率；
- 计算并返回 flags；
- 计算并返回 frameCount；
- 计算并返回 notificationFrameCount；
- 针对
DIRECT和OFFLOAD两种设备类型检查采样率、采样格式和通道掩码格式等； - 检查 stream type 的 strategy；
- 创建
AudioFlinger::PlaybackThread::Track对象，初始化并保存它； - 处理音效链。
AudioFlinger::PlaybackThread::Track 类的继承层次结构如下图：
AudioFlinger::PlaybackThread::Track 对象构造过程 (位于 frameworks/av/services/audioflinger/Tracks.cpp) 如下：
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
ThreadBase *thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_session_t sessionId,
pid_t creatorPid,
uid_t clientUid,
bool isOut,
alloc_type alloc,
track_type type,
audio_port_handle_t portId,
std::string metricsId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(NULL),
// mBuffer, mBufferSize
mState(IDLE),
mAttr(attr),
mSampleRate(sampleRate),
mFormat(format),
mChannelMask(channelMask),
mChannelCount(isOut ?
audio_channel_count_from_out_mask(channelMask) :
audio_channel_count_from_in_mask(channelMask)),
mFrameSize(audio_has_proportional_frames(format) ?
mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mFrameCount(frameCount),
mSessionId(sessionId),
mIsOut(isOut),
mId(android_atomic_inc(&nextTrackId)),
mTerminated(false),
mType(type),
mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE),
mPortId(portId),
mIsInvalid(false),
mTrackMetrics(std::move(metricsId), isOut),
mCreatorPid(creatorPid)
{
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (!isAudioServerOrMediaServerUid(callingUid) || clientUid == AUDIO_UID_INVALID) {
ALOGW_IF(clientUid != AUDIO_UID_INVALID && clientUid != callingUid,
"%s(%d): uid %d tried to pass itself off as %d",
__func__, mId, callingUid, clientUid);
clientUid = callingUid;
}
// clientUid contains the uid of the app that is responsible for this track, so we can blame
// battery usage on it.
mUid = clientUid;
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount;
// check overflow when computing bufferSize due to multiplication by mFrameSize.
if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2
|| mFrameSize == 0 // format needs to be correct
|| minBufferSize > SIZE_MAX / mFrameSize) {
android_errorWriteLog(0x534e4554, "34749571");
return;
}
minBufferSize *= mFrameSize;
if (buffer == nullptr) {
bufferSize = minBufferSize; // allocated here.
} else if (minBufferSize > bufferSize) {
android_errorWriteLog(0x534e4554, "38340117");
return;
}
size_t size = sizeof(audio_track_cblk_t);
if (buffer == NULL && alloc == ALLOC_CBLK) {
// check overflow when computing allocation size for streaming tracks.
if (size > SIZE_MAX - bufferSize) {
android_errorWriteLog(0x534e4554, "34749571");
return;
}
size += bufferSize;
}
if (client != 0) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory == 0 ||
(mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->unsecurePointer())) == NULL) {
ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
client->heap()->dump("AudioTrack");
mCblkMemory.clear();
return;
}
} else {
mCblk = (audio_track_cblk_t *) malloc(size);
if (mCblk == NULL) {
ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
return;
}
}
// construct the shared structure in-place.
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
switch (alloc) {
case ALLOC_READONLY: {
const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
if (roHeap == 0 ||
(mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
(mBuffer = mBufferMemory->unsecurePointer()) == NULL) {
ALOGE("%s(%d): not enough memory for read-only buffer size=%zu",
__func__, mId, bufferSize);
if (roHeap != 0) {
roHeap->dump("buffer");
}
mCblkMemory.clear();
mBufferMemory.clear();
return;
}
memset(mBuffer, 0, bufferSize);
} break;
case ALLOC_PIPE:
mBufferMemory = thread->pipeMemory();
// mBuffer is the virtual address as seen from current process (mediaserver),
// and should normally be coming from mBufferMemory->unsecurePointer().
// However in this case the TrackBase does not reference the buffer directly.
// It should references the buffer via the pipe.
// Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
mBuffer = NULL;
bufferSize = 0;
break;
case ALLOC_CBLK:
// clear all buffers
if (buffer == NULL) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
} else {
mBuffer = buffer;
#if 0
mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
#endif
}
break;
case ALLOC_LOCAL:
mBuffer = calloc(1, bufferSize);
break;
case ALLOC_NONE:
mBuffer = buffer;
break;
default:
LOG_ALWAYS_FATAL("%s(%d): invalid allocation type: %d", __func__, mId, (int)alloc);
}
mBufferSize = bufferSize;
#ifdef TEE_SINK
mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
#endif
}
}
. . . . . .
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
audio_output_flags_t flags,
track_type type,
audio_port_handle_t portId,
size_t frameCountToBeReady,
float speed)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
// 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).
(sharedBuffer != 0) ? sharedBuffer->unsecurePointer() : buffer,
(sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
sessionId, creatorPid,
VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid)), true /*isOut*/,
(type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
type,
portId,
std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(portId)),
mFillingUpStatus(FS_INVALID),
// mRetryCount initialized later when needed
mSharedBuffer(sharedBuffer),
mStreamType(streamType),
mMainBuffer(thread->sinkBuffer()),
mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false),
mFrameMap(16 /* sink-frame-to-track-frame map memory */),
mVolumeHandler(new media::VolumeHandler(sampleRate)),
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(attributionSource, attr, id(),
streamType)),
// mSinkTimestamp
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
* When the track starts playing, its volume will be computed. */
mFinalVolume(0.f),
mResumeToStopping(false),
mFlushHwPending(false),
mFlags(flags),
mSpeed(speed),
mResetDone(false),
mRetryCount(0) {
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
ALOGV_IF(sharedBuffer != 0, "%s(%d): sharedBuffer: %p, size: %zu",
__func__, mId, sharedBuffer->unsecurePointer(), sharedBuffer->size());
if (mCblk == NULL) {
return;
}
uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
ALOGE("%s(%d): no more tracks available", __func__, mId);
releaseCblk(); // this makes the track invalid.
return;
}
if (sharedBuffer == 0) {
mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack(), sampleRate);
} else {
mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, sampleRate);
}
mServerProxy = mAudioTrackServerProxy;
mServerProxy->setStartThresholdInFrames(frameCountToBeReady); // update the Cblk value
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
// FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
// race with setSyncEvent(). However, if we call it, we cannot properly start
// static fast tracks (SoundPool) immediately after stopping.
//mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
int i = __builtin_ctz(thread->mFastTrackAvailMask);
ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
// FIXME This is too eager. We allocate a fast track index before the
// fast track becomes active. Since fast tracks are a scarce resource,
// this means we are potentially denying other more important fast tracks from
// being created. It would be better to allocate the index dynamically.
mFastIndex = i;
thread->mFastTrackAvailMask &= ~(1 << i);
}
mServerLatencySupported = thread->type() == ThreadBase::MIXER
|| thread->type() == ThreadBase::DUPLICATING;
#ifdef TEE_SINK
mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+ "_" + std::to_string(mId) + "_T");
#endif
if (thread->supportsHapticPlayback()) {
// If the track is attached to haptic playback thread, it is potentially to have
// HapticGenerator effect, which will generate haptic data, on the track. In that case,
// external vibration is always created for all tracks attached to haptic playback thread.
mAudioVibrationController = new AudioVibrationController(this);
std::string packageName = attributionSource.packageName.has_value() ?
attributionSource.packageName.value() : "";
mExternalVibration = new os::ExternalVibration(
mUid, packageName, mAttr, mAudioVibrationController);
}
// Once this item is logged by the server, the client can add properties.
const char * const traits = sharedBuffer == 0 ? "" : "static";
mTrackMetrics.logConstructor(creatorPid, uid, id(), traits, streamType);
}
在 AudioFlinger::PlaybackThread::Track 父类 AudioFlinger::ThreadBase::TrackBase 的对象构造过程中，会从 AudioFlinger::Client 管理的共享内存块中分配一块内存，具体的大小，将根据用于数据交换的共享内存块是否由应用程序分配来定；会在这块内存上构造 audio_track_cblk_t 对象；会初始化用于数据交换的共享内存块 mBuffer。
AudioFlinger::PlaybackThread::Track 类的构造函数主要是基于 audio_track_cblk_t 和用于数据交换的共享内存块 mBuffer 创建了 ServerProxy 对象，当内存模式为 MODE_STREAM 时，创建 AudioTrackServerProxy 对象，否则创建 StaticAudioTrackServerProxy 对象。
应用程序和 AudioFlinger 之间的音频数据传递
应用程序和 AudioFlinger 通过 Proxy 进行音频数据的传递及同步。相关类的继承层次结构如下图：
Proxy 封装在共享内存块上创建的 audio_track_cblk_t 对象和数据缓冲区。应用程序通过 AudioTrackClientProxy/StaticAudioTrackClientProxy/AudioRecordClientProxy 传递数据，AudioFlinger 则通过 AudioTrackServerProxy/StaticAudioTrackServerProxy/AudioRecordServerProxy 传递数据。
对于 audio_track_cblk_t 对象，android::AudioTrack 在请求 AudioFlinger 创建 Track 之后，会通过 android::media::IAudioTrack 获得 AudioFlinger 创建的共享内存块：
std::optional<media::SharedFileRegion> sfr;
output.audioTrack->getCblk(&sfr);
sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
if (iMem == 0) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
status = NO_INIT;
goto exit;
}
这个调用将由 AudioFlinger::TrackHandle::getCblk() 函数处理，该函数定义 (位于 frameworks/av/services/audioflinger/Tracks.cpp) 如下：
Status AudioFlinger::TrackHandle::getCblk(
std::optional<media::SharedFileRegion>* _aidl_return) {
*_aidl_return = legacy2aidl_NullableIMemory_SharedFileRegion(mTrack->getCblk()).value();
return Status::ok();
}
如我们前面看到的，AudioFlinger::TrackHandle 是 AudioFlinger::PlaybackThread::Track 的壳。这里 AudioFlinger::TrackHandle 从 AudioFlinger::PlaybackThread::Track 获得共享内存块，并将其从 IMemory/IMemoryHeap 的形式转换为 android::media::SharedFileRegion 的形式。这里实际调用的 AudioFlinger::ThreadBase::TrackBase::getCblk() 函数定义 (位于 frameworks/av/services/audioflinger/TrackBase.h) 如下：
sp<IMemory> getCblk() const { return mCblkMemory; }
如我们在前面看到的，mCblkMemory 在 AudioFlinger::PlaybackThread::Track 对象构造过程中创建。
Proxy 基于共享内存块实现了环形缓冲区的语义。对于音频播放，android::AudioTrack 通过 Proxy 向共享内存块中写入数据。应用程序通过 Java 的 AudioTrack 写入音频数据，并继而调用 android::AudioTrack::write() 函数写入数据，该函数定义 (位于 frameworks/av/media/libaudioclient/AudioTrack.cpp) 如下：
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed, size_t *nonContig)
{
// previous and new IAudioTrack sequence numbers are used to detect track re-creation
uint32_t oldSequence = 0;
Proxy::Buffer buffer;
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
do {
// obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
// keep them from going away if another thread re-creates the track during obtainBuffer()
sp<AudioTrackClientProxy> proxy;
sp<IMemory> iMem;
{ // start of lock scope
AutoMutex lock(mLock);
uint32_t newSequence = mSequence;
// did previous obtainBuffer() fail due to media server death or voluntary invalidation?
if (status == DEAD_OBJECT) {
// re-create track, unless someone else has already done so
if (newSequence == oldSequence) {
status = restoreTrack_l("obtainBuffer");
if (status != NO_ERROR) {
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
}
}
oldSequence = newSequence;
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
}
// Keep the extra references
proxy = mProxy;
iMem = mCblkMemory;
if (mState == STATE_STOPPING) {
status = -EINTR;
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
// Non-blocking if track is stopped or paused
if (mState != STATE_ACTIVE) {
requested = &ClientProxy::kNonBlocking;
}
} // end of lock scope
buffer.mFrameCount = audioBuffer->frameCount;
// FIXME starts the requested timeout and elapsed over from scratch
status = proxy->obtainBuffer(&buffer, requested, elapsed);
} while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
audioBuffer->size = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
audioBuffer->sequence = oldSequence;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
}
return status;
}
void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
{
// FIXME add error checking on mode, by adding an internal version
if (mTransfer == TRANSFER_SHARED) {
return;
}
size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
return;
}
Proxy::Buffer buffer;
buffer.mFrameCount = stepCount;
buffer.mRaw = audioBuffer->raw;
AutoMutex lock(mLock);
if (audioBuffer->sequence != mSequence) {
// This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
__func__, audioBuffer->sequence, mSequence);
return;
}
mReleased += stepCount;
mInUnderrun = false;
mProxy->releaseBuffer(&buffer);
// restart track if it was disabled by audioflinger due to previous underrun
restartIfDisabled();
}
. . . . . .
ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
return INVALID_OPERATION;
}
if (isDirect()) {
AutoMutex lock(mLock);
int32_t flags = android_atomic_and(
~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
&mCblk->mFlags);
if (flags & CBLK_INVALID) {
return DEAD_OBJECT;
}
}
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Validation: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
__func__, mPortId, buffer, userSize, userSize);
return BAD_VALUE;
}
size_t written = 0;
Buffer audioBuffer;
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
status_t err = obtainBuffer(&audioBuffer,
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
if (err < 0) {
if (written > 0) {
break;
}
if (err == TIMED_OUT || err == -EINTR) {
err = WOULD_BLOCK;
}
return ssize_t(err);
}
size_t toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
}
if (written > 0) {
mFramesWritten += written / mFrameSize;
if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
const sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
// causes wake up of the playback thread, that will callback the client for
// more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
t->wake();
}
}
}
return written;
}
android::AudioTrack 循环从 AudioTrackClientProxy 获得一块数据缓冲区，向其中写入音频数据，并释放缓冲区，直到写入了所有音频数据。
在 AudioFlinger 中，从共享内存块中获取音频数据。AudioFlinger 以 AudioBufferProvider 的形式给 AudioMixer 的一路音频源 AudioMixer::Track 设置音频数据来源。AudioFlinger::PlaybackThread::Track 继承自 AudioBufferProvider，它会被设置给 AudioMixer。当混音器 AudioMixer 执行混音时，它通过 AudioBufferProvider 的 getNextBuffer(AudioBufferProvider::Buffer* buffer) 函数获得音频数据，以用于混音。AudioFlinger::PlaybackThread::Track 的 getNextBuffer(AudioBufferProvider::Buffer* buffer) 函数定义 (位于 frameworks/av/services/audioflinger/Tracks.cpp) 如下：
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
buf.mFrameCount = desiredFrames;
status_t status = mServerProxy->obtainBuffer(&buf);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
if (buf.mFrameCount == 0 && !isStopping() && !isStopped() && !isPaused() && !isOffloaded()) {
ALOGV("%s(%d): underrun, framesReady(%zu) < framesDesired(%zd), state: %d",
__func__, mId, buf.mFrameCount, desiredFrames, mState);
mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
} else {
mAudioTrackServerProxy->tallyUnderrunFrames(0);
}
return status;
}
AudioFlinger::PlaybackThread::Track 通过 AudioTrackServerProxy 获得音频数据缓冲区。当 AudioMixer 执行混音处理结束后，它会释放缓冲区。相应的函数定义 (位于 frameworks/av/services/audioflinger/Tracks.cpp) 如下：
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
#ifdef TEE_SINK
mTee.write(buffer->raw, buffer->frameCount);
#endif
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
buffer->frameCount = 0;
buffer->raw = NULL;
mServerProxy->releaseBuffer(&buf);
}
. . . . . .
void AudioFlinger::PlaybackThread::Track::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
interceptBuffer(*buffer);
TrackBase::releaseBuffer(buffer);
}
AudioFlinger::PlaybackThread::Track 通过 AudioTrackServerProxy 释放音频数据缓冲区。
在各个 Proxy 类实现内部，基于共享内存上创建的 audio_track_cblk_t 对象的字段创建 futex 来实现 AudioFlinger 和应用程序之间的同步，包括等待和唤醒等。
参考文档：
Android系统Audio框架介绍
深入剖析Android音频之AudioTrack
Done.
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