android将pcm byte[]通过Librtmp进行rtmp推流
需求
我们这边做的功能是智能戒指，戒指可以录音，然后app通过蓝牙连接，将音频的byte[]进行rtmp推流
技术
因为我们不涉及直播，也不涉及视频，工期也比较短，只是音频推流，所以没用更复杂的ffmpeg，使用了更简单的Librtmp，灵感来源于这篇文章
使用MediaCodec和RTMP做直播推流
但是这个的坏处是封装太深了，方便确实方便，不太好提取，我这边只是作为参考
思路
戒指上传的音频，或者手机录音的音频，都是pcm的，需要将先转换为aac格式的，所以必现先实现转换，然后再写入aac文件的时候，进行推流。
Librtmp Client for Android
PCMToAAC：
public class PCMToAAC {
private String encodeType = MediaFormat.MIMETYPE_AUDIO_AAC;
private static final int samples_per_frame = 2048;
private MediaCodec mediaEncode;
private MediaCodec.BufferInfo encodeBufferInfo;
private ByteBuffer[] encodeInputBuffers;
private ByteBuffer[] encodeOutputBuffers;
private byte[] chunkAudio = new byte[0];
private BufferedOutputStream out;
File aacFile;
File pcmFile;
private RTMPMuxer rtmpMuxer=new RTMPMuxer();
public PCMToAAC(String aacPath, String pcmPath) {
aacFile = new File(aacPath);
pcmFile = new File(pcmPath);
if (!aacFile.exists()) {
try {
aacFile.getParentFile().mkdirs();
aacFile.createNewFile();
} catch (Exception e) {
e.printStackTrace();
}
}
try {
out = new BufferedOutputStream(new FileOutputStream(aacFile, false));
} catch (FileNotFoundException e) {
e.printStackTrace();
}
initAACMediaEncode();
rtmpMuxer.open("rtmp://4xxxx/live/stream",100,100);
}
public PCMToAAC() {
aacFile = new File(com.smart.bing.utils.FileUtil.getSDPath(App.getInstance(), "test.aac"));
try{
aacFile.delete();
}catch (Exception e){
e.printStackTrace();
}
if (!aacFile.exists()) {
try {
aacFile.getParentFile().mkdirs();
aacFile.createNewFile();
} catch (Exception e) {
e.printStackTrace();
}
}
try {
out = new BufferedOutputStream(new FileOutputStream(aacFile, false));
} catch (FileNotFoundException e) {
e.printStackTrace();
}
initAACMediaEncode();
rtmpMuxer.open("rtmp://xxxx/live/stream",100,100);
}
/**
* 初始化AAC编码器
*/
private void initAACMediaEncode() {
try {
//参数对应-> mime type、采样率、声道数
MediaFormat encodeFormat = MediaFormat.createAudioFormat(MediaFormat.MIMETYPE_AUDIO_AAC, 16000, 1);
encodeFormat.setInteger(MediaFormat.KEY_BIT_RATE, 64000);//比特率
encodeFormat.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1);
encodeFormat.setInteger(MediaFormat.KEY_CHANNEL_MASK, AudioFormat.CHANNEL_IN_MONO);
encodeFormat.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
encodeFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, samples_per_frame);//作用于inputBuffer的大小
mediaEncode = MediaCodec.createEncoderByType(encodeType);
mediaEncode.configure(encodeFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
} catch (IOException e) {
e.printStackTrace();
}
if (mediaEncode == null) {
return;
}
mediaEncode.start();
encodeInputBuffers = mediaEncode.getInputBuffers();
encodeOutputBuffers = mediaEncode.getOutputBuffers();
encodeBufferInfo = new MediaCodec.BufferInfo();
}
/**
* 编码PCM数据 得到AAC格式的音频文件
*/
public void dstAudioFormatFromPCM(byte[] pcmData) {
int inputIndex;
ByteBuffer inputBuffer;
int outputIndex;
ByteBuffer outputBuffer;
int outBitSize;
int outPacketSize;
byte[] PCMAudio;
PCMAudio = pcmData;
encodeInputBuffers = mediaEncode.getInputBuffers();
encodeOutputBuffers = mediaEncode.getOutputBuffers();
encodeBufferInfo = new MediaCodec.BufferInfo();
inputIndex = mediaEncode.dequeueInputBuffer(0);
if (inputIndex != -1) {
inputBuffer = encodeInputBuffers[inputIndex];
inputBuffer.clear();
inputBuffer.limit(PCMAudio.length);
inputBuffer.put(PCMAudio);//PCM数据填充给inputBuffer
mediaEncode.queueInputBuffer(inputIndex, 0, PCMAudio.length, 0, 0);//通知编码器 编码
outputIndex = mediaEncode.dequeueOutputBuffer(encodeBufferInfo, 0);
while (outputIndex > 0) {
outBitSize = encodeBufferInfo.size;
outPacketSize = outBitSize + 7;//7为ADT头部的大小
outputBuffer = encodeOutputBuffers[outputIndex];//拿到输出Buffer
outputBuffer.position(encodeBufferInfo.offset);
outputBuffer.limit(encodeBufferInfo.offset + outBitSize);
chunkAudio = new byte[outPacketSize];
addADTStoPacket(chunkAudio, outPacketSize);//添加ADTS
outputBuffer.get(chunkAudio, 7, outBitSize);//将编码得到的AAC数据 取出到byte[]中
try {
//录制aac音频文件，保存在手机内存中
out.write(chunkAudio, 0, chunkAudio.length);
rtmpMuxer.writeAudio(chunkAudio,0,chunkAudio.length,encodeBufferInfo.presentationTimeUs);
Log.d("chunkAudio", Arrays.toString(chunkAudio));
out.flush();
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
outputBuffer.position(encodeBufferInfo.offset);
mediaEncode.releaseOutputBuffer(outputIndex, false);
outputIndex = mediaEncode.dequeueOutputBuffer(encodeBufferInfo, 0);
}
}
}
public void closeRtmp(){
rtmpMuxer.close();
}
/**
* 添加ADTS头
*
* @param packet
* @param packetLen
*/
private void addADTStoPacket(byte[] packet, int packetLen) {
int profile = 2; // AAC LC
int freqIdx = 8; // 16KHz
int chanCfg = 1; // CPE
// fill in ADTS data
packet[0] = (byte) 0xFF;
packet[1] = (byte) 0xF1;
packet[2] = (byte) (((profile - 1) << 6) + (freqIdx << 2) + (chanCfg >> 2));
packet[3] = (byte) (((chanCfg & 3) << 6) + (packetLen >> 11));
packet[4] = (byte) ((packetLen & 0x7FF) >> 3);
packet[5] = (byte) (((packetLen & 7) << 5) + 0x1F);
packet[6] = (byte) 0xFC;
}
public byte[] readInputStream() {
InputStream inputStream = null;
try {
inputStream = new FileInputStream(pcmFile);
} catch (FileNotFoundException e) {
e.printStackTrace();
}
// 1.建立通道对象
ByteArrayOutputStream outputStream = new ByteArrayOutputStream();
// 2.定义存储空间
byte[] buffer = new byte[1024];
// 3.开始读文件
int len = -1;
try {
if (inputStream != null) {
while ((len = inputStream.read(buffer)) != -1) {
// 将Buffer中的数据写到outputStream对象中
// outputStream.write(buffer, 0, len);
dstAudioFormatFromPCM(buffer);
Log.e("wqs+readInputStream", "readInputStream: " + buffer);
}
}
// 4.关闭流
outputStream.close();
inputStream.close();
} catch (IOException e) {
e.printStackTrace();
}
return outputStream.toByteArray();
}
}
坑1：
rtmpMuxer.open(“rtmp://xxxx/live/stream”,100,100);
要放在外层，千万不要和rtmpMuxer.writeAudio放一起，要不然会卡死
坑2：
推流的地址，后边要加上音频流的名称，名字可以随便取，我这边叫stream
坑3：
这个PCMToAAC 涉及的操作较多，所以进入页面会有卡顿，可以在子线程初始化
使用
PCMToAAC pcmToAAC;
.....
new Thread(new Runnable() {
@Override
public void run() {
pcmToAAC = new PCMToAAC();
}
}).start();
@Override
protected void onDestroy() {
super.onDestroy();
pcmToAAC.closeRtmp();
}
/**
戒指录音回传
**/
@Override
public void CONTROL_AUDIO(byte[] bytes) {
new Thread(new Runnable() {
@Override
public void run() {
try {
pcmToAAC.dstAudioFormatFromPCM(bytes);
} catch (Exception e) {
e.printStackTrace();
}
}
}).start();
}
原文地址：https://blog.csdn.net/jifashihan/article/details/144130092
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