Android使用libRtmp直播推流

2020-01-20 16:47:52 浏览数 (2)
  1. 初始化rtmp
代码语言：javascript复制
//分配空间
RTMP *rtmp = RTMP_Alloc();
//初始化
RTMP_Init(rtmp);
//设置推流URL
RTMP_SetupURL(rtmp, url);
//设置可写状态
RTMP_EnableWrite(rtmp);
//链接服务器
RTMP_Connect(rtmp, NULL);
//链接流
RTMP_ConnectStream(rtmp, 0);

//循环推流（AAC、H264）	//开始推流
while(1){
     int result = RTMP_SendPacket(rtmp, packet, 1);
     RTMPPacket_Free(packet);
     free(packet);
     packet = NULL;
}

//关闭链接
RTMP_Close(rtmp);
//释放资源
RTMP_Free(rtmp);
rtmp=NULL;
  1. H264包封装。在发送每一帧关键帧之前得先发送SPS、PPS帧信息，发送的每一帧（I、P、SPS、PPS）数据得添加头部信息。

获取摄像头预览数据并编码为H264,pcm数据编码AAC

2.1 SPS PPS数据

代码语言：javascript复制

void RtmpPush::pushSPSPPS(char *sps, int spsLen, char *pps, int ppsLen) {
    if (!this->queue) return;
    int bodySize = spsLen   ppsLen   16;
    RTMPPacket *rtmpPacket = static_cast<RTMPPacket *>(malloc(sizeof(RTMPPacket)));
    RTMPPacket_Alloc(rtmpPacket, bodySize);
    RTMPPacket_Reset(rtmpPacket);

    char *body = rtmpPacket->m_body;

    int i = 0;
    //frame type(4bit)和CodecId(4bit)合成一个字节(byte)
    //frame type 关键帧1  非关键帧2
    //CodecId  7表示avc
    body[i  ] = 0x17;

    //fixed 4byte
    body[i  ] = 0x00;
    body[i  ] = 0x00;
    body[i  ] = 0x00;
    body[i  ] = 0x00;

    //configurationVersion： 版本 1byte
    body[i  ] = 0x01;

    //AVCProfileIndication：Profile 1byte  sps[1]
    body[i  ] = sps[1];

    //compatibility：  兼容性 1byte  sps[2]
    body[i  ] = sps[2];

    //AVCLevelIndication： ProfileLevel 1byte  sps[3]
    body[i  ] = sps[3];

    //lengthSizeMinusOne： 包长数据所使用的字节数  1byte
    body[i  ] = 0xff;

    //sps个数 1byte
    body[i  ] = 0xe1;
    //sps长度 2byte
    body[i  ] = (spsLen >> 8) & 0xff;
    body[i  ] = spsLen & 0xff;

    //sps data 内容
    memcpy(&body[i], sps, spsLen);
    i  = spsLen;
    //pps个数 1byte
    body[i  ] = 0x01;
    //pps长度 2byte
    body[i  ] = (ppsLen >> 8) & 0xff;
    body[i  ] = ppsLen & 0xff;
    //pps data 内容
    memcpy(&body[i], pps, ppsLen);


    rtmpPacket->m_packetType = RTMP_PACKET_TYPE_VIDEO;
    rtmpPacket->m_nBodySize = bodySize;
    rtmpPacket->m_nTimeStamp = 0;
    rtmpPacket->m_hasAbsTimestamp = 0;
    rtmpPacket->m_nChannel = 0x04;//音频或者视频
    rtmpPacket->m_headerType = RTMP_PACKET_SIZE_MEDIUM;
    rtmpPacket->m_nInfoField2 = this->rtmp->m_stream_id;

    queue->putRtmpPacket(rtmpPacket);

}

2.2 H264数据

代码语言：javascript复制
void RtmpPush::pushVideoData(char *data, int dataLen, bool keyFrame) {
    if (!this->queue) return;
    int bodySize = dataLen   9;
    RTMPPacket *rtmpPacket = static_cast<RTMPPacket *>(malloc(sizeof(RTMPPacket)));
    RTMPPacket_Alloc(rtmpPacket, bodySize);
    RTMPPacket_Reset(rtmpPacket);

    char *body = rtmpPacket->m_body;

    int i = 0;
    //frame type(4bit)和CodecId(4bit)合成一个字节(byte)
    //frame type 关键帧1  非关键帧2
    //CodecId  7表示avc
    if (keyFrame) {
        body[i  ] = 0x17;
    } else {
        body[i  ] = 0x27;
    }

    //fixed 4byte   0x01表示NALU单元
    body[i  ] = 0x01;
    body[i  ] = 0x00;
    body[i  ] = 0x00;
    body[i  ] = 0x00;

    //dataLen  4byte
    body[i  ] = (dataLen >> 24) & 0xff;
    body[i  ] = (dataLen >> 16) & 0xff;
    body[i  ] = (dataLen >> 8) & 0xff;
    body[i  ] = dataLen & 0xff;

    //data
    memcpy(&body[i], data, dataLen);

    rtmpPacket->m_packetType = RTMP_PACKET_TYPE_VIDEO;
    rtmpPacket->m_nBodySize = bodySize;
    //持续播放时间
    rtmpPacket->m_nTimeStamp = RTMP_GetTime() - this->startTime;
    //进入直播播放开始时间
    rtmpPacket->m_hasAbsTimestamp = 0;
    rtmpPacket->m_nChannel = 0x04;//音频或者视频
    rtmpPacket->m_headerType = RTMP_PACKET_SIZE_LARGE;
    rtmpPacket->m_nInfoField2 = this->rtmp->m_stream_id;

    queue->putRtmpPacket(rtmpPacket);


}
  1. AAC包封装 需要添加头部
代码语言：javascript复制

void RtmpPush::pushAudioData(char *data, int dataLen) {
    if (!this->queue) return;
    int bodySize = dataLen   2;
    RTMPPacket *rtmpPacket = static_cast<RTMPPacket *>(malloc(sizeof(RTMPPacket)));
    RTMPPacket_Alloc(rtmpPacket, bodySize);
    RTMPPacket_Reset(rtmpPacket);

    char *body = rtmpPacket->m_body;
    //前四位表示音频数据格式  10（十进制）表示AAC，16进制就是A
    //第5-6位的数值表示采样率，0 = 5.5 kHz，1 = 11 kHz，2 = 22 kHz，3(11) = 44 kHz。
    //第7位表示采样精度，0 = 8bits，1 = 16bits。
    //第8位表示音频类型，0 = mono，1 = stereo
    //这里是44100 立体声 16bit 二进制就是1111   16进制就是F
    body[0] = 0xAF;

    //0x00 aac头信息,  0x01 aac 原始数据
    //这里都用0x01都可以
    body[1] = 0x01;

    //data
    memcpy(&body[2], data, dataLen);

    rtmpPacket->m_packetType = RTMP_PACKET_TYPE_AUDIO;
    rtmpPacket->m_nBodySize = bodySize;
    //持续播放时间
    rtmpPacket->m_nTimeStamp = RTMP_GetTime() - this->startTime;
    //进入直播播放开始时间
    rtmpPacket->m_hasAbsTimestamp = 0;
    rtmpPacket->m_nChannel = 0x04;//音频或者视频
    rtmpPacket->m_headerType = RTMP_PACKET_SIZE_LARGE;
    rtmpPacket->m_nInfoField2 = this->rtmp->m_stream_id;

    queue->putRtmpPacket(rtmpPacket);
}
  1. Android MediaCodec获取PPS和SPS
代码语言：javascript复制
int outputBufferIndex = videoEncodec.dequeueOutputBuffer(videoBufferinfo, 0);
if (outputBufferIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
    ByteBuffer spsb = videoEncodec.getOutputFormat().getByteBuffer("csd-0");
    byte[] sps = new byte[spsb.remaining()];
    spsb.get(sps, 0,sps.length);
    Log.e("zzz", "sps: "   ByteUtil.bytesToHexSpaceString(sps));

    ByteBuffer ppsb = videoEncodec.getOutputFormat().getByteBuffer("csd-1");
    byte[] pps = new byte[ppsb.remaining()];
    ppsb.get(pps, 0,pps.length);
    Log.e("zzz", "pps: "   ByteUtil.bytesToHexSpaceString(pps));

}

具体查看demo: https://github.com/ChinaZeng/RtmpLivePushDemo

android media rtmp 编码 摄像头

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