android 音频采集、FLTP重采样与AAC编码推流

本文主要是介绍android 音频采集、FLTP重采样与AAC编码推流，希望对大家解决编程问题提供一定的参考价值，需要的开发者们随着小编来一起学习吧！

相比较视频编码，音频编码要简单很多，主要就是将采集到的音频源数据PCM编码AAC.
MediaPlus中FFmpeg使用的是libfdk-aac编码器，这里有个问题需要注意下:FFmpeg已经废弃了AV_SAMPLE_FMT_S16格式PCM编码AAC，也就是说如果使用FFmpeg自带的AAC编码器，必须做音频的重采样（重采样为:AV_SAMPLE_FMT_FLTP），否则AAC编码是失败的。

接下来，看下MediaPlus中是如何采集音频与AAC编码的.

在app.mobile.nativeapp.com.libmedia.core.streamer.RtmpPushStreamer的AudioThread获取AudioRecord采集到的音频数据并传入底层：

  class AudioThread extends Thread {public volatile boolean m_bExit = false;@Overridepublic void run() {// TODO Auto-generated method stubsuper.run();int[] dataLength;byte[] audioBuffer;AudioCaptureInterface.GetAudioDataReturn ret;dataLength = new int[1];audioBuffer = new byte[m_aiBufferLength[0]];while (!m_bExit) {try {Thread.sleep(1, 10);if (m_bExit) {break;}} catch (InterruptedException e) {e.printStackTrace();}try {ret = mAudioCapture.GetAudioData(audioBuffer,m_aiBufferLength[0], dataLength);if (ret == AudioCaptureInterface.GetAudioDataReturn.RET_SUCCESS) {encodeAudio(audioBuffer, dataLength[0]);}} catch (Exception e) {e.printStackTrace();stopThread();}}}

具体AudioRecord采集具体实现是avcapture.jar包中，代码比较简单，相关android音视频采集初始化及API调用网上都有相关Demo，这里不再赘述！

encodeAudio(audioBuffer, dataLength[0]);将音频数据传入底层。

  /*** 采集的PCM音频数据** @param audioBuffer* @param length*/public void encodeAudio(byte[] audioBuffer, int length) {try {LiveJniMediaManager.EncodeAAC(audioBuffer, length);} catch (Exception e) {e.printStackTrace();}}

JNI层接收到PCM音频数据，添加到AudioCapture同步队列中：

JNIEXPORT jint JNICALL
Java_app_mobile_nativeapp_com_libmedia_core_jni_LiveJniMediaManager_EncodeAAC(JNIEnv *env,jclass type,jbyteArray audioBuffer_,jint length) {if (audioCaptureInit && !isClose) {jbyte *audioSrc = env->GetByteArrayElements(audioBuffer_, 0);uint8_t *audioDstData = (uint8_t *) malloc(length);memcpy(audioDstData, audioSrc, length);OriginData *audioOriginData = new OriginData();audioOriginData->size = length;audioOriginData->data = audioDstData;audioCapture->PushAudioData(audioOriginData);env->ReleaseByteArrayElements(audioBuffer_, audioSrc, 0);}return 0;
}

以上代码，是在调用app.mobile.nativeapp.com.libmedia.core.streamer.RtmpPushStreamer>>startPushStream()开启推流前的相关调用：主要就是初始化音频采集，并将数据传入底层。

startPushStream的调用，会重置AudioCapture::ExitCapture=false;
数据才会被加入到audioCaputureframeQueue对列中.

/*** 开启推流* @param pushUrl* @return*/private boolean startPushStream(String pushUrl) {if (nativeInt) {int ret = 0;ret = LiveJniMediaManager.StartPush(pushUrl);if (ret < 0) {Log.d("initNative", "native push failed!");return false;}return true;}return false;}

如下图：

重置标记后，audioCaputureframeQueue.push将数据添中到队列中.


int AudioCapture::PushAudioData(OriginData *originData) {if (ExitCapture) {return 0;}originData->pts = av_gettime();LOG_D(DEBUG,"audio capture pts :%lld",originData->pts);audioCaputureframeQueue.push(originData);return 0;
}

上面这些代码与视频的处理方式都是一样的流程，在调用app.mobile.nativeapp.com.libmedia.core.streamer.RtmpPushStreamer>>startPushStream()，已经开始往音频队列中添加数据，紧接着调用rtmpStreamer->StartPushStream() ，实际也就是开启了音视频的两个编码线程及推流,推流相关代码与视频一致.


int RtmpStreamer::StartPushStream() {videoStreamIndex = AddStream(videoEncoder->videoCodecContext);audioStreamIndex = AddStream(audioEncoder->audioCodecContext);pthread_create(&t3, NULL, RtmpStreamer::WriteHead, this);pthread_join(t3, NULL);VideoCapture *pVideoCapture = videoEncoder->GetVideoCapture();AudioCapture *pAudioCapture = audioEncoder->GetAudioCapture();pVideoCapture->videoCaputureframeQueue.clear();pAudioCapture->audioCaputureframeQueue.clear();if(writeHeadFinish) {pthread_create(&t1, NULL, RtmpStreamer::PushAudioStreamTask, this);pthread_create(&t2, NULL, RtmpStreamer::PushVideoStreamTask, this);}else{return -1;}return 0;
}

PushAudioStreamTask中从队列中获取数据编码、推流.
rtmpStreamer->audioEncoder->EncodeAAC(&pAudioData);AAC编码.

rtmpStreamer->SendFrame(pAudioData, rtmpStreamer->audioStreamIndex);推流(与视频推流一致)

这里说明下，音频编码前获取编码器及一些参数的指定：
libmedia/src/main/cpp/AudioEncoder.cpp是音频编码的核心类，int AudioEncoder::InitEncode() 方法封装了音频编码器的初始化。

int AudioEncoder::InitEncode() {std::lock_guard<std::mutex> lk(mut);avCodec = avcodec_find_encoder_by_name("libfdk_aac");int ret = 0;if (!avCodec) {LOG_D(DEBUG, "aac encoder not found!")return -1;}audioCodecContext = avcodec_alloc_context3(avCodec);if (!audioCodecContext) {LOG_D(DEBUG, "avcodec alloc context3 failed!");return -1;}audioCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;audioCodecContext->sample_fmt = AV_SAMPLE_FMT_S16;audioCodecContext->sample_rate = audioCapture->GetAudioEncodeArgs()->sampleRate;audioCodecContext->thread_count = 8;audioCodecContext->bit_rate = 50*1024*8;audioCodecContext->channels = audioCapture->GetAudioEncodeArgs()->channels;audioCodecContext->frame_size = audioCapture->GetAudioEncodeArgs()->nb_samples;audioCodecContext->time_base = {1, 1000000};//AUDIO VIDEO 两边时间基数要相同audioCodecContext->channel_layout = av_get_default_channel_layout(audioCodecContext->channels);outputFrame = av_frame_alloc();outputFrame->channels = audioCodecContext->channels;outputFrame->channel_layout = av_get_default_channel_layout(outputFrame->channels);outputFrame->format = audioCodecContext->sample_fmt;outputFrame->nb_samples = 1024;ret = av_frame_get_buffer(outputFrame, 0);if (ret != 0) {LOG_D(DEBUG, "av_frame_get_buffer failed!");return -1;}LOG_D(DEBUG, "av_frame_get_buffer success!");ret = avcodec_open2(audioCodecContext, NULL, NULL);if (ret != 0) {char buf[1024] = {0};av_strerror(ret, buf, sizeof(buf));LOG_D(DEBUG, "avcodec open failed! info:%s", buf);return -1;}LOG_D(DEBUG, "open audio codec success!");LOG_D(DEBUG, "Complete init Audio Encode!")return 0;
}

指定获取libfdk_aac编码器

avCodec = avcodec_find_encoder_by_name("libfdk_aac");

初始化编码器上下文

audioCodecContext = avcodec_alloc_context3(avCodec);

创建AVFrame，并分配内存负责封装PCM源数据

outputFrame = av_frame_alloc();outputFrame->channels = audioCodecContext->channels;//通道数outputFrame->channel_layout = av_get_default_channel_layout(outputFrame->channels);outputFrame->format = audioCodecContext->sample_fmt;outputFrame->nb_samples = 1024;//默认值ret = av_frame_get_buffer(outputFrame, 0);if (ret != 0) {LOG_D(DEBUG, "av_frame_get_buffer failed!");return -1;}LOG_D(DEBUG, "av_frame_get_buffer success!");

打开编码器
```
ret = avcodec_open2(audioCodecContext, NULL, NULL);
```
以上是编码前必须要完成的初始化.

int AudioEncoder::EncodeAAC 方法封装了AAC编码：

int AudioEncoder::EncodeAAC(OriginData **originData) {int ret = 0;ret = avcodec_fill_audio_frame(outputFrame,audioCodecContext->channels,audioCodecContext->sample_fmt, (*originData)->data,8192, 0);outputFrame->pts = (*originData)->pts;ret = avcodec_send_frame(audioCodecContext, outputFrame);if (ret != 0) {
#ifdef SHOW_DEBUG_INFOLOG_D(DEBUG, "send frame failed!");
#endif}av_packet_unref(&audioPacket);ret = avcodec_receive_packet(audioCodecContext, &audioPacket);if (ret != 0) {
#ifdef SHOW_DEBUG_INFOLOG_D(DEBUG, "receive packet failed!");
#endif}(*originData)->Drop();(*originData)->avPacket = &audioPacket;#ifdef SHOW_DEBUG_INFOLOG_D(DEBUG, "encode audio packet size:%d pts:%lld", (*originData)->avPacket->size,(*originData)->avPacket->pts);LOG_D(DEBUG, "Audio frame encode success!");
#endif(*originData)->avPacket->size;return audioPacket.size;
}

*originData->data填充到AVFrame中，
```
ret = avcodec_send_frame(audioCodecContext, outputFrame);
ret = avcodec_receive_packet(audioCodecContext, &audioPacket);
```
audioPacket就是编码后的数据了，data是编码后的数据，size是大小，这样就完成了编码.

注意：在int AudioEncoder::InitEncode()方法中

avcodec_find_encoder_by_name("libfdk_aac");

这里使用了fdk-aac编码器，前提是你必须要将libfdk-aac库，链接到ffmpeg动态库中，否则是找不到此编码器的。FFmpeg自带有AAC编码器，可以通过：

 avcodec_find_encoder(AV_CODEC_ID_AAC);

获取到AAC编码器,当然如果使用FFmpeg的AAC编码器，就会涉及到一个问题，就是刚开始文中提到了，AV_SAMPLE_FMT_S16需要重采样为：AV_SAMPLE_FMT_FLTP的问题，由于FFmpeg废弃了AV_SAMPLE_FMT_S16格式PCM编码AAC，那么在编码前就需要多一步重采样的处理.

以下AV_SAMPLE_FMT_S16 PCM音频数据重采样相关代码仅供参考：

初始化SwrContext,指定输入输出参数

  swrContext = swr_alloc_set_opts(swrContext, av_get_default_channel_layout(CHANNELS),//输出通道LayoutAV_SAMPLE_FMT_FLTP,//输出格式48000,//输出采样率av_get_default_channel_layout(CHANNELS),//输入通道LayoutAV_SAMPLE_FMT_S16,//输入格式48000,//输入采样率NULL,//NULLNULL);//NULLret = swr_init(swrContext);//初始化SwrContextif (ret != 0) {LOG_D(DEBUG, "swr_init failed!");return -1;}

AAC编码前，将源数据重采样

 for (; ;) {if (encodeAAC->exit) {break;}if (encodeAAC->frame_queue.empty()) {continue;}const uint8_t *indata[AV_NUM_DATA_POINTERS] = {0};//PCM s16uint8_t *buf = *encodeAAC->frame_queue.wait_and_pop().get();//PCM 16bit
#ifdef FDK_CODEC//fdk-aac无需重采样ret = avcodec_fill_audio_frame(encodeAAC->outputFrame, encodeAAC->avCodecContext->channels,encodeAAC->avCodecContext->sample_fmt, buf, BUFFER_SIZE, 0);if (ret < 0) {LOG_D(DEBUG, "fill frame failed!");continue;}
#else//重采样AM_SAMPLE_FMT_FLTPindata[0] = buf;swr_convert(encodeAAC->swrContext, encodeAAC->outputFrame->data,encodeAAC->outputFrame->nb_samples, indata,encodeAAC->outputFrame->nb_samples);
#endif