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修复rtsp流播放声音不连续情况, 修改AAC数据包解析bug

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This commit is contained in:
droid.chow 2019-02-26 10:11:42 +08:00
parent cce454d78c
commit af194677de
1 changed files with 58 additions and 13 deletions
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51
src/RtspMuxer/AACRtpCodec.cpp
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@ -104,20 +104,65 @@ AACFrame::Ptr AACRtpDecoder::obtainFrame() {
bool AACRtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
RtpCodec::inputRtp(rtppack, false);
// 获取rtp数据长度
int length = rtppack->length - rtppack->offset;
if (_adts->aac_frame_length + length - 4 > sizeof(AACFrame::buffer)) {
// 获取rtp数据
const uint8_t *rtp_packet_buf = (uint8_t *)rtppack->payload + rtppack->offset;
do
{
// 查询头部的偏移，每次2字节
uint32_t au_header_offset = 0;
//首2字节表示Au-Header的长度，单位bit，所以除以16得到Au-Header字节数
const uint16_t au_header_length = (((rtp_packet_buf[au_header_offset] << 8) | rtp_packet_buf[au_header_offset + 1]) >> 4);
au_header_offset += 2;
//assert(length > (2 + au_header_length * 2));
if (length < (2 + au_header_length * 2))
break;
// 存放每一个aac帧长度
std::vector<uint32_t > vec_aac_len;
for (int i = 0; i < au_header_length; ++i)
{
// 之后的2字节是AU_HEADER
const uint16_t au_header = ((rtp_packet_buf[au_header_offset] << 8) | rtp_packet_buf[au_header_offset + 1]);
// 其中高13位表示一帧AAC负载的字节长度，低3位无用
uint32_t nAac = (au_header >> 3);
vec_aac_len.push_back(nAac);
au_header_offset += 2;
}
// 真正aac负载开始处
const uint8_t *rtp_packet_payload = rtp_packet_buf + au_header_offset;
// 载荷查找
uint32_t next_aac_payload_offset = 0;
for (int j = 0; j < au_header_length; ++j)
{
// 当前aac包长度
const uint32_t cur_aac_payload_len = vec_aac_len.at(j);
if (_adts->aac_frame_length + cur_aac_payload_len > sizeof(AACFrame::buffer)) {
_adts->aac_frame_length = 7;
WarnL << "aac负载数据太长";
return false;
}
memcpy(_adts->buffer + _adts->aac_frame_length, rtppack->payload + rtppack->offset + 4, length - 4);
_adts->aac_frame_length += (length - 4);
// 提取每一包aac载荷数据
memcpy(_adts->buffer + _adts->aac_frame_length, rtp_packet_payload + next_aac_payload_offset, cur_aac_payload_len);
_adts->aac_frame_length += (cur_aac_payload_len);
if (rtppack->mark == true) {
_adts->sequence = rtppack->sequence;
_adts->timeStamp = rtppack->timeStamp;
writeAdtsHeader(*_adts, _adts->buffer);
onGetAAC(_adts);
}
next_aac_payload_offset += cur_aac_payload_len;
}
} while (0);
return false;
}