Android音频之AudioTrack简析
AudioTrack 是 Android 系统中用于音频播放的一个核心类，它提供了对音频数据的高效播放，特别适用于播放原始 PCM 数据。与 MediaPlayer 相比，AudioTrack 更加灵活，能够提供低延迟的音频播放，广泛应用于实时音频播放、音效处理以及游戏开发等场景。
1. AudioTrack 的作用与功能
AudioTrack 用于播放 PCM（脉冲编码调制）格式的音频数据。它直接将音频数据从内存写入音频硬件设备，并通过缓冲区将音频流送出。其核心功能包括：
- 音频数据播放：可以播放从应用程序传输过来的 PCM 音频数据。
- 低延迟音频播放：适合需要实时处理音频的场景，比如语音通信、游戏音效等。
- 音频流管理：提供了对多个音频流的管理，包括播放、停止、暂停等。
2. AudioTrack 的工作模式
AudioTrack 支持两种工作模式：MODE_STREAM 和 MODE_STATIC。
MODE_STREAM：适用于动态、流式的音频播放。数据可以逐步写入缓冲区并播放，常用于长时间播放的音频（例如音乐、语音通话等）。MODE_STATIC：适用于短小的音频文件播放，如音效。所有数据在播放前一次性加载到缓冲区。
3. AudioTrack 的主要方法
play()：开始播放音频。stop()：停止播放音频。write()：将音频数据写入缓冲区并播放。flush()：清空缓冲区。release()：释放AudioTrack实例资源。
4. AudioTrack 的构造
创建 AudioTrack 时，常用构造函数如下：
AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, int bufferSizeInBytes, int mode)
streamType: 音频流类型，如AudioManager.STREAM_MUSIC。sampleRateInHz: 音频采样率，常见值如 44100Hz 或 48000Hz。channelConfig: 音频通道配置，如单声道或立体声。audioFormat: 音频格式，如AudioFormat.ENCODING_PCM_16BIT。bufferSizeInBytes: 缓冲区大小，决定了音频播放的流畅性。mode: 工作模式，MODE_STREAM或MODE_STATIC。
5. 使用示例
播放简单的 PCM 数据
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
public class AudioTrackExample {
public void playAudio() {
// 设置音频参数
int sampleRate = 44100; // 采样率
int channelConfig = AudioFormat.CHANNEL_OUT_MONO; // 单声道
int audioFormat = AudioFormat.ENCODING_PCM_16BIT; // PCM 16 位
int bufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat);
// 创建 AudioTrack 实例
AudioTrack audioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC, // 音频流类型
sampleRate, // 采样率
channelConfig, // 通道配置
audioFormat, // 音频格式
bufferSize, // 缓冲区大小
AudioTrack.MODE_STREAM // 流式播放
);
// 启动音频播放
audioTrack.play();
// 播放音频数据
byte[] audioData = generateSampleData(); // 生成 PCM 数据
audioTrack.write(audioData, 0, audioData.length);
}
// 生成简单的音频数据（如正弦波）
private byte[] generateSampleData() {
int duration = 3; // 持续时间 3 秒
int numSamples = duration * 44100; // 采样数
byte[] buffer = new byte[numSamples * 2]; // 每个样本 2 字节（16 位 PCM）
double frequency = 440.0; // 440Hz，A4 音符
for (int i = 0; i < numSamples; i++) {
short sample = (short) (Math.sin(2 * Math.PI * frequency * i / 44100) * Short.MAX_VALUE);
buffer[i * 2] = (byte) (sample & 0xFF); // 低位字节
buffer[i * 2 + 1] = (byte) ((sample >> 8) & 0xFF); // 高位字节
}
return buffer;
}
}
代码解释：
- 创建
AudioTrack实例：定义音频参数（如采样率、通道配置、音频格式）并创建AudioTrack对象。选择MODE_STREAM模式用于动态播放数据。 - 生成音频数据：使用正弦波公式生成 PCM 音频数据（440Hz A4 音符）。
- 播放音频：调用
audioTrack.play()启动播放，并使用audioTrack.write()写入音频数据。
6. 适用场景
- 音效播放：适用于游戏中的短音效播放（如按钮点击音、得分音效等），尤其在
MODE_STATIC模式下。 - 实时音频播放：适用于需要实时音频流的应用（如语音通话、在线广播、音乐流媒体等）。
- 音频处理：用于开发音频处理应用程序，如音频合成、滤波、效果处理等。
7. 源码流程
构造
//frameworks/base/media/java/android/media/AudioTrack.java
--> AudioTrack(AudioAttributes, AudioFormat, int,int, int, boolean, int,TunerConfiguration)
----> mConfiguredAudioAttributes = attributes;
----> Looper looper;
----> int channelIndexMask = 0;
----> channelIndexMask = format.getChannelIndexMask();
----> int channelMask = 0;
----> channelMask = format.getChannelMask();
----> int encoding = AudioFormat.ENCODING_DEFAULT;
----> encoding = format.getEncoding();
----> mOffloaded = offload;
----> mStreamType = AudioSystem.STREAM_DEFAULT;
----> audioBuffSizeCheck(bufferSizeInBytes);
----> mInitializationLooper = looper;
----> 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());
----> // frameworks/base/core/jni/android_media_AudioTrack.cpp
----> // {"native_setup","(Ljava/lang/Object;Ljava/lang/Object;[IIIIII[IJZILjava/lang/Object;Ljava/lang/String;)I",
----> // (void *)android_media_AudioTrack_setup}
----> mSampleRate = sampleRate[0];
----> mSessionId = session[0];
// frameworks/base/core/jni/android_media_AudioTrack.cpp
--> android_media_AudioTrack_setup(...)
----> sp<AudioTrack> lpTrack;
----> audio_channel_mask_t nativeChannelMask = nativeChannelMaskFromJavaChannelMasks(channelPositionMask, channelIndexMask);
----> uint32_t channelCount = audio_channel_count_from_out_mask(nativeChannelMask);
----> audio_format_t format = audioFormatToNative(audioFormat);
----> lpTrack = new AudioTrack(opPackageNameStr.c_str());
// frameworks/av/media/libaudioclient/AudioTrack.cpp
------> AudioTrack::AudioTrack(const std::string& opPackageName);
----> status = lpTrack->set(AUDIO_STREAM_DEFAULT...);
------> AudioTrack::set(audio_stream_type_t streamType...);
--------> mAudioTrackThread = new AudioTrackThread(*this);
--------> mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
--------> createTrack_l();
----------> const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
----------> IAudioFlinger::CreateTrackInput input;
----------> 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.clientUid = mClientUid;
----------> input.clientInfo.clientPid = mClientPid;
----------> input.clientInfo.clientTid = -1;
----------> input.sharedBuffer = mSharedBuffer;
----------> input.notificationsPerBuffer = mNotificationsPerBufferReq;
----------> input.flags = mFlags;
----------> input.frameCount = mReqFrameCount;
----------> input.notificationFrameCount = mNotificationFramesReq;
----------> input.selectedDeviceId = mSelectedDeviceId;
----------> input.sessionId = mSessionId;
----------> input.audioTrackCallback = mAudioTrackCallback;
----------> input.opPackageName = mOpPackageName;
----------> IAudioFlinger::CreateTrackOutput output;
----------> sp<IAudioTrack> track = audioFlinger->createTrack(input,output,&status);
----------> sp<IMemory> iMem = track->getCblk();
----------> void *iMemPointer = iMem->unsecurePointer();
----------> mAudioTrack = track;
----------> mCblkMemory = iMem;
----------> IPCThreadState::self()->flushCommands();
----------> audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
----------> mCblk = cblk;
----------> void* buffers;
----------> if (mSharedBuffer == 0) { buffers = cblk + 1;}
----------> else { buffers = mSharedBuffer->unsecurePointer();}
----------> mAudioTrack->attachAuxEffect(mAuxEffectId);
----------> if (mSharedBuffer == 0) {
----------> mStaticProxy.clear();
----------> mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
----------> } else {
----------> mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
----------> mProxy = mStaticProxy;
----------> }
// frameworks/av/services/audioflinger/AudioFlinger.cpp
sp<IAudioTrack> AudioFlinger::createTrack(const CreateTrackInput& input,CreateTrackOutput& output,status_t *status)
--> sp<PlaybackThread::Track> track;
--> sp<TrackHandle> trackHandle;
--> output.sessionId = sessionId;
--> output.outputId = AUDIO_IO_HANDLE_NONE;
--> output.selectedDeviceId = input.selectedDeviceId;
--> lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
--> clientPid, clientUid, &input.config, input.flags,
--> &output.selectedDeviceId, &portId, &secondaryOutputs);
// frameworks/av/media/libaudioclient/AudioSystem.cpp
----> status_t AudioSystem::getOutputForAttr(audio_attributes_t *attr..);
----> const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
----> return aps->getOutputForAttr(attr, output, session, stream, pid, uid,config,flags, selectedDeviceId, portId, secondaryOutputs);
// frameworks/av/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
------> AudioPolicyService::getOutputForAttr(audio_attributes_t *attr...);
--------> mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, uid,config...);
----------> getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid...);
------------> mPolicyMixes.getOutputForAttr(*resultAttr, uid, *flags, primaryMix, secondaryMixes);
------------> bool usePrimaryOutputFromPolicyMixes = requestedDevice == nullptr && primaryMix != nullptr;
------------> if (usePrimaryOutputFromPolicyMixes) {
------------> sp<DeviceDescriptor> deviceDesc = mAvailableOutputDevices.getDevice(primaryMix->mDeviceType,primaryMix->mDeviceAddress...);
------------> sp<SwAudioOutputDescriptor> policyDesc = primaryMix->getOutput();
------------> sp<IOProfile> profile = getProfileForOutput(DeviceVector(deviceDesc),config->sample_rate,config->format,config->channel_mask,...);
------------> policyDesc->mPolicyMix = primaryMix;
------------> *output = policyDesc->mIoHandle;
------------> *selectedDeviceId = deviceDesc != 0 ? deviceDesc->getId() : AUDIO_PORT_HANDLE_NONE;
------------> *outputType = API_OUTPUT_LEGACY;
------------> return NO_ERROR;
------------> }
------------> outputDevices = mEngine->getOutputDevicesForAttributes(*resultAttr, requestedDevice, false);
------------> *output = getOutputForDevices(outputDevices, session, *stream, config,flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
------------> *selectedDeviceId = getFirstDeviceId(outputDevices);
------------> return NO_ERROR;
----------> *portId = PolicyAudioPort::getNextUniqueId();
----------> sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
----------> sp<TrackClientDescriptor> clientDesc =new TrackClientDescriptor(*portId, uid, session...);
----------> outputDesc->addClient(clientDesc);
--------> sp <AudioPlaybackClient> client = new AudioPlaybackClient(*attr, *output, uid, pid, session, *portId, *selectedDeviceId, *stream);
--------> mAudioPlaybackClients.add(*portId, client);
--> PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
--> output.sampleRate = input.config.sample_rate;
--> output.frameCount = input.frameCount;
--> output.notificationFrameCount = input.notificationFrameCount;
--> output.flags = input.flags;
-->
--> 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, input.clientInfo.clientTid, clientUid,
--> &lStatus, portId, input.audioTrackCallback,
--> input.opPackageName);
--> trackHandle = new TrackHandle(track);
--> return trackHandle;
start
stop
write
setPreferredDevice
public boolean setPreferredDevice(AudioDeviceInfo deviceInfo)
--> native_setOutputDevice(preferredDeviceId);
// {"native_setOutputDevice", "(I)Z", (void *)android_media_AudioTrack_setOutputDevice},
--> mPreferredDevice = deviceInfo;
jboolean android_media_AudioTrack_setOutputDevice(JNIEnv *env, jobject thiz, jint device_id)
--> sp<AudioTrack> lpTrack = getAudioTrack(env, thiz);
--> lpTrack->setOutputDevice(device_id)
----> mSelectedDeviceId = deviceId;
----> mProxy->interrupt();