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解决rtsp推流,aac时间戳增量太大导致flv.js播放异常的问题:#392
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@ -9,7 +9,6 @@
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*/
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#include "AACRtp.h"
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#define AAC_MAX_FRAME_SIZE (2 * 1024)
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namespace mediakit{
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@ -87,32 +86,42 @@ bool AACRtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
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//rtp数据开始部分
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uint8_t *ptr = (uint8_t *) rtppack->data() + rtppack->offset;
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//rtp数据末尾
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const uint8_t *end = (uint8_t *) rtppack->data() + rtppack->size();
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uint8_t *end = (uint8_t *) rtppack->data() + rtppack->size();
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//首2字节表示Au-Header的个数,单位bit,所以除以16得到Au-Header个数
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const uint16_t au_header_count = ((ptr[0] << 8) | ptr[1]) >> 4;
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//忽略Au-Header区
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ptr += 2 + au_header_count * 2;
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uint16_t au_header_count = ((ptr[0] << 8) | ptr[1]) >> 4;
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//记录au_header起始指针
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uint8_t *au_header_ptr = ptr + 2;
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ptr = au_header_ptr + au_header_count * 2;
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while (ptr < end) {
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auto size = (uint32_t) (end - ptr);
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if (size > AAC_MAX_FRAME_SIZE) {
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size = AAC_MAX_FRAME_SIZE;
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}
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if (_frame->size() + size > AAC_MAX_FRAME_SIZE) {
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//数据太多了,先清空
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flushData();
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}
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//追加aac数据
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_frame->_buffer.append((char *) ptr, size);
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_frame->_dts = rtppack->timeStamp;
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ptr += size;
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if (end < ptr) {
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//数据不够
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return false;
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}
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if (rtppack->mark) {
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//最后一个rtp分片
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//每个audio unit时间戳增量
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auto dts_inc = (rtppack->timeStamp - _last_dts) / au_header_count;
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if(dts_inc < 0 && dts_inc > 100){
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//时间戳增量异常
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dts_inc = 0;
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}
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for (int i = 0; i < au_header_count; ++i) {
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// 之后的2字节是AU_HEADER,其中高13位表示一帧AAC负载的字节长度,低3位无用
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uint16_t size = ((au_header_ptr[0] << 8) | au_header_ptr[1]) >> 3;
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if (ptr + size > end) {
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//数据不够
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break;
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}
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//设置aac数据
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_frame->_buffer.assign((char *) ptr, size);
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//设置当前audio unit时间戳
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_frame->_dts = _last_dts + i * dts_inc;
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ptr += size;
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au_header_ptr += 2;
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flushData();
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}
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//记录上次时间戳
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_last_dts = rtppack->timeStamp;
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return false;
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}
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@ -45,6 +45,7 @@ private:
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private:
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AACFrame::Ptr _frame;
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string _aac_cfg;
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uint32_t _last_dts = 0;
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};
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