SDL2库(4)-Android(安卓)端源码简要分析（AudioSubSystem）
16lz
2021-01-25
SDL.png 初始化
打开音频播放器
SDL_OpenAudio.png 音频线程
音频线程SDL_RunAudio.png 开始或者暂停音频播放器
项目位置 https://github.com/deepsadness/SDLCmakeDemo
系列内容导读
- SDL2-移植Android Studio+CMakeList集成
- Android端FFmpeg +SDL2的简单播放器
- SDL2 Android端的简要分析(VideoSubSystem)
- SDL2 Android端的简要分析(AudioSubSystem)
Android 部分源码分析
Android部分的初始化和视频部分基本相同。
这里简单看一下。
在SDLActivity中调用了
SDL.setupJNI()。SDL.setupJNI()中SDLAudioManager.nativeSetupJNI()开始对JNI方法进行初始化。在
SDL_android.c中,nativeSetupJNI初始化JNI回调java方法的指针。
/* Audio initialization -- called before SDL_main() to initialize JNI bindings */JNIEXPORT void JNICALL SDL_JAVA_AUDIO_INTERFACE(nativeSetupJNI)(JNIEnv* mEnv, jclass cls){ __android_log_print(ANDROID_LOG_VERBOSE, "SDL", "AUDIO nativeSetupJNI()"); Android_JNI_SetupThread(); mAudioManagerClass = (jclass)((*mEnv)->NewGlobalRef(mEnv, cls)); midAudioOpen = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "audioOpen", "(IIII)[I"); midAudioWriteByteBuffer = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "audioWriteByteBuffer", "([B)V"); midAudioWriteShortBuffer = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "audioWriteShortBuffer", "([S)V"); midAudioWriteFloatBuffer = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "audioWriteFloatBuffer", "([F)V"); midAudioClose = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "audioClose", "()V"); midCaptureOpen = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "captureOpen", "(IIII)[I"); midCaptureReadByteBuffer = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "captureReadByteBuffer", "([BZ)I"); midCaptureReadShortBuffer = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "captureReadShortBuffer", "([SZ)I"); midCaptureReadFloatBuffer = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "captureReadFloatBuffer", "([FZ)I"); midCaptureClose = (*mEnv)->GetStaticMethodID(mEnv, mAudioManagerClass, "captureClose", "()V"); if (!midAudioOpen || !midAudioWriteByteBuffer || !midAudioWriteShortBuffer || !midAudioWriteFloatBuffer || !midAudioClose || !midCaptureOpen || !midCaptureReadByteBuffer || !midCaptureReadShortBuffer || !midCaptureReadFloatBuffer || !midCaptureClose) { __android_log_print(ANDROID_LOG_WARN, "SDL", "Missing some Java callbacks, do you have the latest version of SDLAudioManager.java?"); } checkJNIReady();}简单的看一下播放的几个方法，都做了什么
audioOpen
SDLAudioManager->audioOpen
传入的参数
- sampleRate
采样率。表示每秒需要的采样字节数 - is16Bit
表示是否采用16位的深度进行采样 - isStereo
表示是否使用双声道进行采样 - desiredFrames
预期的每次采样的音频帧数
public static int audioOpen(int sampleRate, boolean is16Bit, boolean isStereo, int desiredFrames) { int channelConfig = isStereo ? AudioFormat.CHANNEL_CONFIGURATION_STEREO : AudioFormat.CHANNEL_CONFIGURATION_MONO; int audioFormat = is16Bit ? AudioFormat.ENCODING_PCM_16BIT : AudioFormat.ENCODING_PCM_8BIT; //计算每一帧的大小 int frameSize = (isStereo ? 2 : 1) * (is16Bit ? 2 : 1); Log.v(TAG, "SDL audio: wanted " + (isStereo ? "stereo" : "mono") + " " + (is16Bit ? "16-bit" : "8-bit") + " " + (sampleRate / 1000f) + "kHz, " + desiredFrames + " frames buffer"); //得到预期的帧数，来计算buffer。 //我们预期的帧数，也不能小于getMinBufferSize得到的帧数 // Let the user pick a larger buffer if they really want -- but ye // gods they probably shouldn't, the minimums are horrifyingly high // latency already desiredFrames = Math.max(desiredFrames, (AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat) + frameSize - 1) / frameSize); if (mAudioTrack == null) { //打开AudioTrack mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig, audioFormat, desiredFrames * frameSize, AudioTrack.MODE_STREAM); // Instantiating AudioTrack can "succeed" without an exception and the track may still be invalid // Ref: https://android.googlesource.com/platform/frameworks/base/+/refs/heads/master/media/java/android/media/AudioTrack.java // Ref: http://developer.android.com/reference/android/media/AudioTrack.html#getState() if (mAudioTrack.getState() != AudioTrack.STATE_INITIALIZED) { Log.e(TAG, "Failed during initialization of Audio Track"); mAudioTrack = null; return -1; } mAudioTrack.play(); } Log.v(TAG, "SDL audio: got " + ((mAudioTrack.getChannelCount() >= 2) ? "stereo" : "mono") + " " + ((mAudioTrack.getAudioFormat() == AudioFormat.ENCODING_PCM_16BIT) ? "16-bit" : "8-bit") + " " + (mAudioTrack.getSampleRate() / 1000f) + "kHz, " + desiredFrames + " frames buffer"); return 0; }audioWriteXXXBuffer
写入不同的Buffer格式。这方法比较简单。我们就看一种
public static void audioWriteByteBuffer(byte[] buffer) { if (mAudioTrack == null) { Log.e(TAG, "Attempted to make audio call with uninitialized audio!"); return; } for (int i = 0; i < buffer.length; ) { //把buffer写入，进行播放 int result = mAudioTrack.write(buffer, i, buffer.length - i); if (result > 0) { i += result; } else if (result == 0) { try { Thread.sleep(1); } catch(InterruptedException e) { // Nom nom } } else { Log.w(TAG, "SDL audio: error return from write(byte)"); return; } } }audioClose
进行关闭和释放
public static void audioClose() { if (mAudioTrack != null) { mAudioTrack.stop(); mAudioTrack.release(); mAudioTrack = null; } }SDL流程
SDL初始化
SDL_Init(): 初始化SDL。
SDL_OpenAudio(): 打开音频播放器。
SDL_PauseAudio(): 开始播放。
SDL循环渲染数据
调用callback，将正确的数据喂入
初始化SDL_AudioInit
在视频初始化的过程，我们就看到了。SDL_Init方法，传入SDL_INIT_AUDIO标志位，就会走到SDL_AudioInit方法，对音频系统进行初始化。
SDL_AudioInit方法比较简单,就是将JNI的方法指针给audio.impl。同时设置变量的标志位。
static intANDROIDAUDIO_Init(SDL_AudioDriverImpl * impl){ /* Set the function pointers */ impl->OpenDevice = ANDROIDAUDIO_OpenDevice; impl->PlayDevice = ANDROIDAUDIO_PlayDevice; impl->GetDeviceBuf = ANDROIDAUDIO_GetDeviceBuf; impl->CloseDevice = ANDROIDAUDIO_CloseDevice; impl->CaptureFromDevice = ANDROIDAUDIO_CaptureFromDevice; impl->FlushCapture = ANDROIDAUDIO_FlushCapture; /* and the capabilities */ impl->HasCaptureSupport = SDL_TRUE; impl->OnlyHasDefaultOutputDevice = 1; impl->OnlyHasDefaultCaptureDevice = 1; return 1; /* this audio target is available. */}在上面Android方法的初始化中，可以看到这些JNI回调java 的方法的实现，都在SDLAudioManager里面。
打开音频播放器SDL_OpenAudio
SDL_OpenAudio.png - 方法签名
extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained);我们可以看到，SDL_OpenAudio需要传入两个参数，一个是我们想要的音频格式。一个是最后实际的音频格式。
这里的SDL_AudioSpec，是SDL中记录音频格式的结构体。
typedef struct SDL_AudioSpec{ int freq; /**< DSP frequency -- samples per second */ SDL_AudioFormat format; /**< Audio data format */ Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ Uint8 silence; /**< Audio buffer silence value (calculated) */ Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */ Uint16 padding; /**< Necessary for some compile environments */ Uint32 size; /**< Audio buffer size in bytes (calculated) */ SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */ void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */} SDL_AudioSpec;对照函数调用图。
- 对结果的音频格式参数，先进行初步的初始化。
- 分配
SDL_AudioDevice,并初始化 - 对音频的状态进行初始化
//目前是非关闭 SDL_AtomicSet(&device->shutdown, 0); /* just in case. */ //暂停 SDL_AtomicSet(&device->paused, 1); //可用状态 SDL_AtomicSet(&device->enabled, 1);- 调用
current_audio.impl.OpenDevice，开启打开音频设备。
这个方法最终会调用到SDLAudioManager的open方法。
public static int[] audioOpen(int sampleRate, int audioFormat, int desiredChannels, int desiredFrames) { return open(false, sampleRate, audioFormat, desiredChannels, desiredFrames); } protected static int[] open(boolean isCapture, int sampleRate, int audioFormat, int desiredChannels, int desiredFrames) { int channelConfig; int sampleSize; int frameSize; Log.v(TAG, "Opening " + (isCapture ? "capture" : "playback") + ", requested " + desiredFrames + " frames of " + desiredChannels + " channel " + getAudioFormatString(audioFormat) + " audio at " + sampleRate + " Hz"); /* On older devices let's use known good settings */ if (Build.VERSION.SDK_INT < 21) { if (desiredChannels > 2) { desiredChannels = 2; } if (sampleRate < 8000) { sampleRate = 8000; } else if (sampleRate > 48000) { sampleRate = 48000; } } if (audioFormat == AudioFormat.ENCODING_PCM_FLOAT) { int minSDKVersion = (isCapture ? 23 : 21); if (Build.VERSION.SDK_INT < minSDKVersion) { audioFormat = AudioFormat.ENCODING_PCM_16BIT; } } switch (audioFormat) { case AudioFormat.ENCODING_PCM_8BIT: sampleSize = 1; break; case AudioFormat.ENCODING_PCM_16BIT: sampleSize = 2; break; case AudioFormat.ENCODING_PCM_FLOAT: sampleSize = 4; break; default: Log.v(TAG, "Requested format " + audioFormat + ", getting ENCODING_PCM_16BIT"); audioFormat = AudioFormat.ENCODING_PCM_16BIT; sampleSize = 2; break; } if (isCapture) { switch (desiredChannels) { case 1: channelConfig = AudioFormat.CHANNEL_IN_MONO; break; case 2: channelConfig = AudioFormat.CHANNEL_IN_STEREO; break; default: Log.v(TAG, "Requested " + desiredChannels + " channels, getting stereo"); desiredChannels = 2; channelConfig = AudioFormat.CHANNEL_IN_STEREO; break; } } else { switch (desiredChannels) { case 1: channelConfig = AudioFormat.CHANNEL_OUT_MONO; break; case 2: channelConfig = AudioFormat.CHANNEL_OUT_STEREO; break; case 3: channelConfig = AudioFormat.CHANNEL_OUT_STEREO | AudioFormat.CHANNEL_OUT_FRONT_CENTER; break; case 4: channelConfig = AudioFormat.CHANNEL_OUT_QUAD; break; case 5: channelConfig = AudioFormat.CHANNEL_OUT_QUAD | AudioFormat.CHANNEL_OUT_FRONT_CENTER; break; case 6: channelConfig = AudioFormat.CHANNEL_OUT_5POINT1; break; case 7: channelConfig = AudioFormat.CHANNEL_OUT_5POINT1 | AudioFormat.CHANNEL_OUT_BACK_CENTER; break; case 8: if (Build.VERSION.SDK_INT >= 23) { channelConfig = AudioFormat.CHANNEL_OUT_7POINT1_SURROUND; } else { Log.v(TAG, "Requested " + desiredChannels + " channels, getting 5.1 surround"); desiredChannels = 6; channelConfig = AudioFormat.CHANNEL_OUT_5POINT1; } break; default: Log.v(TAG, "Requested " + desiredChannels + " channels, getting stereo"); desiredChannels = 2; channelConfig = AudioFormat.CHANNEL_OUT_STEREO; break; }/* Log.v(TAG, "Speaker configuration (and order of channels):"); if ((channelConfig & 0x00000004) != 0) { Log.v(TAG, " CHANNEL_OUT_FRONT_LEFT"); } if ((channelConfig & 0x00000008) != 0) { Log.v(TAG, " CHANNEL_OUT_FRONT_RIGHT"); } if ((channelConfig & 0x00000010) != 0) { Log.v(TAG, " CHANNEL_OUT_FRONT_CENTER"); } if ((channelConfig & 0x00000020) != 0) { Log.v(TAG, " CHANNEL_OUT_LOW_FREQUENCY"); } if ((channelConfig & 0x00000040) != 0) { Log.v(TAG, " CHANNEL_OUT_BACK_LEFT"); } if ((channelConfig & 0x00000080) != 0) { Log.v(TAG, " CHANNEL_OUT_BACK_RIGHT"); } if ((channelConfig & 0x00000100) != 0) { Log.v(TAG, " CHANNEL_OUT_FRONT_LEFT_OF_CENTER"); } if ((channelConfig & 0x00000200) != 0) { Log.v(TAG, " CHANNEL_OUT_FRONT_RIGHT_OF_CENTER"); } if ((channelConfig & 0x00000400) != 0) { Log.v(TAG, " CHANNEL_OUT_BACK_CENTER"); } if ((channelConfig & 0x00000800) != 0) { Log.v(TAG, " CHANNEL_OUT_SIDE_LEFT"); } if ((channelConfig & 0x00001000) != 0) { Log.v(TAG, " CHANNEL_OUT_SIDE_RIGHT"); }*/ } frameSize = (sampleSize * desiredChannels); // Let the user pick a larger buffer if they really want -- but ye // gods they probably shouldn't, the minimums are horrifyingly high // latency already int minBufferSize; if (isCapture) { minBufferSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat); } else { minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat); } desiredFrames = Math.max(desiredFrames, (minBufferSize + frameSize - 1) / frameSize); int[] results = new int[4]; if (isCapture) { if (mAudioRecord == null) { mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.DEFAULT, sampleRate, channelConfig, audioFormat, desiredFrames * frameSize); // see notes about AudioTrack state in audioOpen(), above. Probably also applies here. if (mAudioRecord.getState() != AudioRecord.STATE_INITIALIZED) { Log.e(TAG, "Failed during initialization of AudioRecord"); mAudioRecord.release(); mAudioRecord = null; return null; } mAudioRecord.startRecording(); } results[0] = mAudioRecord.getSampleRate(); results[1] = mAudioRecord.getAudioFormat(); results[2] = mAudioRecord.getChannelCount(); results[3] = desiredFrames; } else { if (mAudioTrack == null) { mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig, audioFormat, desiredFrames * frameSize, AudioTrack.MODE_STREAM); // Instantiating AudioTrack can "succeed" without an exception and the track may still be invalid // Ref: https://android.googlesource.com/platform/frameworks/base/+/refs/heads/master/media/java/android/media/AudioTrack.java // Ref: http://developer.android.com/reference/android/media/AudioTrack.html#getState() if (mAudioTrack.getState() != AudioTrack.STATE_INITIALIZED) { /* Try again, with safer values */ Log.e(TAG, "Failed during initialization of Audio Track"); mAudioTrack.release(); mAudioTrack = null; return null; } mAudioTrack.play(); } results[0] = mAudioTrack.getSampleRate(); results[1] = mAudioTrack.getAudioFormat(); results[2] = mAudioTrack.getChannelCount(); results[3] = desiredFrames; } Log.v(TAG, "Opening " + (isCapture ? "capture" : "playback") + ", got " + results[3] + " frames of " + results[2] + " channel " + getAudioFormatString(results[1]) + " audio at " + results[0] + " Hz"); return results; }这个方法，对应SDL传递过来的参数。初始化播放使用时对应使用的AudioTrack。并将最后AudioTrack配置的后的参数，返回给SDL的desireSpec。
- 接着使用返回的
audioSpec和当前的进行对比，重新复制，并且如果发生了改变，则重新创建SDL_AudioStream。
if (build_stream) { if (iscapture) { device->stream = SDL_NewAudioStream(device->spec.format, device->spec.channels, device->spec.freq, obtained->format, obtained->channels, obtained->freq); } else { device->stream = SDL_NewAudioStream(obtained->format, obtained->channels, obtained->freq, device->spec.format, device->spec.channels, device->spec.freq); } if (!device->stream) { close_audio_device(device); return 0; } }结构体SDL_AudioStream
struct _SDL_AudioStream{ SDL_AudioCVT cvt_before_resampling; SDL_AudioCVT cvt_after_resampling; SDL_DataQueue *queue; SDL_bool first_run; Uint8 *staging_buffer; int staging_buffer_size; int staging_buffer_filled; Uint8 *work_buffer_base; /* maybe unaligned pointer from SDL_realloc(). */ int work_buffer_len; int src_sample_frame_size; SDL_AudioFormat src_format; Uint8 src_channels; int src_rate; int dst_sample_frame_size; SDL_AudioFormat dst_format; Uint8 dst_channels; int dst_rate; double rate_incr; Uint8 pre_resample_channels; int packetlen; int resampler_padding_samples; float *resampler_padding; void *resampler_state; SDL_ResampleAudioStreamFunc resampler_func; SDL_ResetAudioStreamResamplerFunc reset_resampler_func; SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;};这个结构体。保存src和dst 的对应的参数，并通过保存的CVT方法，可以进行方便的转换。
- 设置callback
如果我们不能设置callback(音频数据的回调)的话，SDL会默认给设置一个数据队列的管理。
因为通常，我们不会直接播放 PCM的数据，所以都会自己设置callback，在callback当中进行音频数据的格式转换和数据设置。
if (device->spec.callback == NULL) { /* use buffer queueing? */ /* pool a few packets to start. Enough for two callbacks. */ device->buffer_queue = SDL_NewDataQueue(SDL_AUDIOBUFFERQUEUE_PACKETLEN, obtained->size * 2); if (!device->buffer_queue) { close_audio_device(device); SDL_SetError("Couldn't create audio buffer queue"); return 0; } device->callbackspec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback; device->callbackspec.userdata = device; }- 最后通过一些配置，然后开启
SDL_RunAudio线程。（因为是播放，如果是录制，就走另外一个线程SDL_CaptureAudio）
device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, threadname, stacksize, device);音频线程SDL_RunAudio
音频线程SDL_RunAudio.png 设置线程的优先级
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_TIME_CRITICAL)音频的线程优先级必须是高。判断是否关闭了device
如果关闭了，就推出循环，否则进入循环。
SDL_AtomicGet(&device->shutdown)可以看到SDL这里的音频播放的几个参数shutdown,pause,enable都是用了原子性的变量参数，保持其原子性和一致性。
- 确定数据的buff大小。
if (!device->stream && SDL_AtomicGet(&device->enabled)) { SDL_assert(data_len == device->spec.size); data = current_audio.impl.GetDeviceBuf(device); } else { /* if the device isn't enabled, we still write to the work_buffer, so the app's callback will fire with a regular frequency, in case they depend on that for timing or progress. They can use hotplug now to know if the device failed. Streaming playback uses work_buffer, too. */ data = NULL; } if (data == NULL) { data = device->work_buffer; }如果没有转换流而且有设备的话，就去取设备的许可的buf。这个变量在打开设备的时候，进行初始化。值为 samples*channels
不是的话，就用我们初始化时，传入的大小。作为buf.
- 判断是否callback数据
SDL_LockMutex(device->mixer_lock); if (SDL_AtomicGet(&device->paused)) { SDL_memset(data, device->spec.silence, data_len); } else { callback(udata, data, data_len); } SDL_UnlockMutex(device->mixer_lock);如果是暂停的情况下，就是简单设置数据，就结束了。
如果不是暂停的话，就会进入callback（callback中，我们对音频数据进行读取，解码和设置）
- 播放
if (device->stream) { /* Stream available audio to device, converting/resampling. */ /* if this fails...oh well. We'll play silence here. */ SDL_AudioStreamPut(device->stream, data, data_len); while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->spec.size)) { int got; data = SDL_AtomicGet(&device->enabled) ? current_audio.impl.GetDeviceBuf(device) : NULL; got = SDL_AudioStreamGet(device->stream, data ? data : device->work_buffer, device->spec.size); SDL_assert((got < 0) || (got == device->spec.size)); if (data == NULL) { /* device is having issues... */ const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq); SDL_Delay(delay); /* wait for as long as this buffer would have played. Maybe device recovers later? */ } else { if (got != device->spec.size) { SDL_memset(data, device->spec.silence, device->spec.size); } current_audio.impl.PlayDevice(device); current_audio.impl.WaitDevice(device); } } } else if (data == device->work_buffer) { /* nothing to do; pause like we queued a buffer to play. */ const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq); SDL_Delay(delay); } else { /* writing directly to the device. */ /* queue this buffer and wait for it to finish playing. */ current_audio.impl.PlayDevice(device); current_audio.impl.WaitDevice(device); }最后就是进行播放。如果需要转换的话，就会先进行转换，再播放。转换失败的话，就不会播放声音。
最后是通过current_audio.impl.PlayDevice(device)方法播放
该方法，实际上是调用了Android_JNI_WriteAudioBuffer(SDL_android.c)方法。
因为是在子线程中，所以需要先通过Android_JNI_GetEnv，来调用
int status = (*mJavaVM)->AttachCurrentThread(mJavaVM, &env, NULL);将当前的线程和JVM进行绑定，才可以调用JNI方法。
然后最后调用的是SDLAudioManager中的对应的 audioWriteXXXBuffer方法。使用AudioTrack,将数据进行write（实际上就是播放）
开始或者暂停音频播放器SDL_PauseAudio
voidSDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on){ SDL_AudioDevice *device = get_audio_device(devid); if (device) { current_audio.impl.LockDevice(device); SDL_AtomicSet(&device->paused, pause_on ? 1 : 0); current_audio.impl.UnlockDevice(device); }}通过上面对RunAudio线程的分析，我们知道其是改变device->paused标志位。来回调callback。
callback
我们来关注一下我们如何进行callback的操作
- 传递自己的callback
//通过desired_spec 的callback来传递我们自己的callback wanted_spec.callback = audio_callback; if (SDL_OpenAudio(&wanted_spec, &spec) < 0) { ALOGE("SDL_OpenAudio: %s \n", SDL_GetError()); return -1; }- 定义callback
void audio_callback(void *userdata, Uint8 *stream, int len) { AVCodecContext *aCodecCtx = (AVCodecContext *) userdata; int len1, audio_size; static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2]; static unsigned int audio_buf_size = 0; static unsigned int audio_buf_index = 0;// 这里把得到的数据给重置了 SDL_memset(stream, 0, len); ALOGI("audio_callback len=%d \n", len); //向设备发送长度为len的数据 while (len > 0) { //缓冲区中无数据 if (audio_buf_index >= audio_buf_size) { //从packet中解码数据 audio_size = audio_decode_frame(aCodecCtx, audio_buf, audio_buf_size); //ALOGI("audio_decode_frame finish audio_size=%d \n", audio_size); if (audio_size < 0) //没有解码到数据或者出错，填充0 { audio_buf_size = 1024; memset(audio_buf, 0, audio_buf_size); } else { audio_buf_size = audio_size; } audio_buf_index = 0; } len1 = audio_buf_size - audio_buf_index; if (len1 > len) len1 = len;//这种方式是可以直接把数据复制过去 memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1); //通过SDL_MixAudio方法，可以控制音量，如果直接使用memcpy是无法控制音量的// SDL_MixAudio(stream, audio_buf + audio_buf_index, len1, SDL_MIX_MAXVOLUME);//SDL_MixAudioFormat() len -= len1; stream += len1; audio_buf_index += len1; }}关闭
- 关闭
/** This method is called by SDL using JNI. */ public static void audioClose() { if (mAudioTrack != null) { mAudioTrack.stop(); mAudioTrack.release(); mAudioTrack = null; } }最后关闭音频，就是将其stop和release
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