AACSource packet size over MAX_RTP_PAYLOAD_SIZE-AU_SIZE will abandon

[fatalfeel]

when aac file some packet is over than MAX_RTP_PAYLOAD_SIZE-AU_SIZE
its need like H264Source H265Source piece by piece send

[fatalfeel]

fixed way

bool AACSource::HandleFrame(MediaChannelId channel_id, AVFrame frame)
{
uint8_t* frame_buf = frame.buffer.get();
uint32_t frame_size = frame.size;

if (frame.timestamp == 0) {
    GetTimestamp(&frame.timeNow, &frame.timestamp);
}

/*printf("sendto: %02x%02x%02x%02x %02x%02x%02x%02x%02x%02x%02x%02x\n\n",
            *((unsigned char*)(frame_buf+0)),
            *((unsigned char*)(frame_buf+1)),
            *((unsigned char*)(frame_buf+2)),
            *((unsigned char*)(frame_buf+3)),
            *((unsigned char*)(frame_buf+4)),
            *((unsigned char*)(frame_buf+5)),
            *((unsigned char*)(frame_buf+6)),
            *((unsigned char*)(frame_buf+7)),
            *((unsigned char*)(frame_buf+8)),
            *((unsigned char*)(frame_buf+9)),
            *((unsigned char*)(frame_buf+10)),
            *((unsigned char*)(frame_buf+11)));*/

/*if (frame.size > (MAX_RTP_PAYLOAD_SIZE-AU_SIZE)) {
	return false;
}*/

int adts_size = 0;
if (has_adts_) {
	adts_size = ADTS_SIZE;
}

/*struct AdtsHeader
{
    unsigned int syncword;          //12 bit 同步字 '1111 1111 1111'，说明一个ADTS帧的开始
    unsigned int id;                //1 bit MPEG 标示符， 0 for MPEG-4，1 for MPEG-2
    unsigned int layer;             //2 bit 总是'00'
    unsigned int protectionAbsent;  //1 bit 1表示没有crc，0表示有crc
    unsigned int profile;           //2 bit 表示使用哪个级别的AAC
    unsigned int samplingFreqIndex; //4 bit 表示使用的采样频率
    unsigned int privateBit;        //1 bit
    unsigned int channelCfg;        //3 bit 表示声道数
    unsigned int originalCopy;      //1 bit
    unsigned int home;              //1 bit
   //下面的为改变的参数即每一帧都不同
    unsigned int copyrightIdentificationBit;    //1 bit
    unsigned int copyrightIdentificationStart;  //1 bit
    unsigned int aacFrameLength;                //13 bit 一个ADTS帧的长度包括ADTS头和AAC原始流
    unsigned int adtsBufferFullness;            //11 bit 0x7FF 说明是码率可变的码流
    unsigned int numberOfRawDataBlockInFrame;   //2 bit 有number_of_raw_data_blocks_in_frame + 1个AAC原始帧
};*/

frame_buf  +=  adts_size;
frame_size -=  adts_size;

char AU[AU_SIZE] = { 0 };
AU[0] = 0x00;
AU[1] = 0x10;
AU[2] = (frame_size & 0x1fe0) >> 5;
AU[3] = (frame_size & 0x1f) << 3;

/*RtpPacket rtp_pkt;
rtp_pkt.type        = frame.type;
rtp_pkt.timestamp   = frame.timestamp;
rtp_pkt.timeNow     = frame.timeNow;
rtp_pkt.size        = frame_size + RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + AU_SIZE;
rtp_pkt.last        = 1;

rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 0] = AU[0];
rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 1] = AU[1];
rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 2] = AU[2];
rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 3] = AU[3];

memcpy(rtp_pkt.data.get()+RTP_TCP_HEAD_SIZE+RTP_HEADER_SIZE+AU_SIZE, frame_buf, frame_size);

if (send_frame_callback_) {
	send_frame_callback_(channel_id, rtp_pkt);
}*/

while (frame_size + AU_SIZE > MAX_RTP_PAYLOAD_SIZE)
{
    RtpPacket rtp_pkt;
    rtp_pkt.type        = frame.type;
    rtp_pkt.timestamp   = frame.timestamp;
    rtp_pkt.timeNow     = frame.timeNow;
    rtp_pkt.size        = RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + MAX_RTP_PAYLOAD_SIZE; //3 FU in
    rtp_pkt.last        = 0;

    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 0] = AU[0];
    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 1] = AU[1];
    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 2] = AU[2];
    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 3] = AU[3];

    memcpy(rtp_pkt.data.get()+RTP_TCP_HEAD_SIZE+RTP_HEADER_SIZE+AU_SIZE, frame_buf, MAX_RTP_PAYLOAD_SIZE-AU_SIZE);

    if (send_frame_callback_)
    {
        /*uint8_t* buff = rtp_pkt.data.get();
        printf("case 2:");
        for(int i=0; i<4; i++)
        {
            printf("%02x ", buff[i]);
        }
        printf("\n");*/

        if (!send_frame_callback_(channel_id, rtp_pkt))
        {
            return false;
        }
    }

    frame_buf  += (MAX_RTP_PAYLOAD_SIZE - AU_SIZE);
    frame_size -= (MAX_RTP_PAYLOAD_SIZE - AU_SIZE);
}

//final packet
{
    RtpPacket rtp_pkt;
    rtp_pkt.type        = frame.type;
    rtp_pkt.timestamp   = frame.timestamp;
    rtp_pkt.timeNow     = frame.timeNow;
    rtp_pkt.size        = RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + AU_SIZE + frame_size;
    rtp_pkt.last        = 1;

    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 0] = AU[0];
    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 1] = AU[1];
    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 2] = AU[2];
    rtp_pkt.data.get()[RTP_TCP_HEAD_SIZE + RTP_HEADER_SIZE + 3] = AU[3];

    memcpy(rtp_pkt.data.get()+RTP_TCP_HEAD_SIZE+RTP_HEADER_SIZE+AU_SIZE, frame_buf, frame_size);

    if (send_frame_callback_)
    {
        if (!send_frame_callback_(channel_id, rtp_pkt))
        {
            return false;
        }
    }
}

return true;

}

[fatalfeel]

here is are links you can download it
http://fatalfeel.blogspot.com/2013/12/rtsp-server-for-h264-h265-aac.html