1. 初始化rtmp

//分配空間
RTMP *rtmp = RTMP_Alloc();
//初始化
RTMP_Init(rtmp);
//設(shè)置推流URL
RTMP_SetupURL(rtmp, url);
//設(shè)置可寫狀態(tài)
RTMP_EnableWrite(rtmp);
//鏈接服務(wù)器
RTMP_Connect(rtmp, NULL);
//鏈接流
RTMP_ConnectStream(rtmp, 0);

//循環(huán)推流（AAC、H264）    //開始推流
while(1){
     int result = RTMP_SendPacket(rtmp, packet, 1);
     RTMPPacket_Free(packet);
     free(packet);
     packet = NULL;
}

//關(guān)閉鏈接
RTMP_Close(rtmp);
//釋放資源
RTMP_Free(rtmp);
rtmp=NULL;

2. H264包封裝。在發(fā)送每一幀關(guān)鍵幀之前得先發(fā)送SPS、PPS幀信息，發(fā)送的每一幀（I、P、SPS、PPS）數(shù)據(jù)得添加頭部信息。

獲取攝像頭預(yù)覽數(shù)據(jù)并編碼為H264,pcm數(shù)據(jù)編碼AAC

2.1 SPS PPS數(shù)據(jù)

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)合成一個字節(jié)(byte)
    //frame type 關(guān)鍵幀1  非關(guān)鍵幀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： 包長數(shù)據(jù)所使用的字節(jié)數(shù)  1byte
    body[i++] = 0xff;

    //sps個數(shù) 1byte
    body[i++] = 0xe1;
    //sps長度 2byte
    body[i++] = (spsLen >> 8) & 0xff;
    body[i++] = spsLen & 0xff;

    //sps data 內(nèi)容
    memcpy(&body[i], sps, spsLen);
    i += spsLen;
    //pps個數(shù) 1byte
    body[i++] = 0x01;
    //pps長度 2byte
    body[i++] = (ppsLen >> 8) & 0xff;
    body[i++] = ppsLen & 0xff;
    //pps data 內(nèi)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;//音頻或者視頻
    rtmpPacket->m_headerType = RTMP_PACKET_SIZE_MEDIUM;
    rtmpPacket->m_nInfoField2 = this->rtmp->m_stream_id;

    queue->putRtmpPacket(rtmpPacket);

}

2.2 H264數(shù)據(jù)

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)合成一個字節(jié)(byte)
    //frame type 關(guān)鍵幀1  非關(guān)鍵幀2
    //CodecId  7表示avc
    if (keyFrame) {
        body[i++] = 0x17;
    } else {
        body[i++] = 0x27;
    }

    //fixed 4byte   0x01表示NALU單元
    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;
    //持續(xù)播放時間
    rtmpPacket->m_nTimeStamp = RTMP_GetTime() - this->startTime;
    //進(jìn)入直播播放開始時間
    rtmpPacket->m_hasAbsTimestamp = 0;
    rtmpPacket->m_nChannel = 0x04;//音頻或者視頻
    rtmpPacket->m_headerType = RTMP_PACKET_SIZE_LARGE;
    rtmpPacket->m_nInfoField2 = this->rtmp->m_stream_id;

    queue->putRtmpPacket(rtmpPacket);


}

3. AAC包封裝 需要添加頭部

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;
    //前四位表示音頻數(shù)據(jù)格式  10（十進(jìn)制）表示AAC，16進(jìn)制就是A
    //第5-6位的數(shù)值表示采樣率，0 = 5.5 kHz，1 = 11 kHz，2 = 22 kHz，3(11) = 44 kHz。
    //第7位表示采樣精度，0 = 8bits，1 = 16bits。
    //第8位表示音頻類型，0 = mono，1 = stereo
    //這里是44100 立體聲 16bit 二進(jìn)制就是1111   16進(jìn)制就是F
    body[0] = 0xAF;

    //0x00 aac頭信息,  0x01 aac 原始數(shù)據(jù)
    //這里都用0x01都可以
    body[1] = 0x01;

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

    rtmpPacket->m_packetType = RTMP_PACKET_TYPE_AUDIO;
    rtmpPacket->m_nBodySize = bodySize;
    //持續(xù)播放時間
    rtmpPacket->m_nTimeStamp = RTMP_GetTime() - this->startTime;
    //進(jìn)入直播播放開始時間
    rtmpPacket->m_hasAbsTimestamp = 0;
    rtmpPacket->m_nChannel = 0x04;//音頻或者視頻
    rtmpPacket->m_headerType = RTMP_PACKET_SIZE_LARGE;
    rtmpPacket->m_nInfoField2 = this->rtmp->m_stream_id;

    queue->putRtmpPacket(rtmpPacket);
}

4. Android MediaCodec獲取PPS和SPS

    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));

    }

image

具體查看demo: https://github.com/ChinaZeng/RtmpLivePushDemo