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Squash: Merge v4.0.203

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This commit is contained in:
winlin 2021-12-04 11:21:35 +08:00
parent e16830e989
commit 8576fa7052
12 changed files with 137 additions and 62 deletions
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98
trunk/src/app/srs_app_rtc_source.cpp
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@ -1278,7 +1278,7 @@ SrsRtmpFromRtcBridger::SrsRtmpFromRtcBridger(SrsLiveSource *src)
is_first_audio = true;
is_first_video = true;
format = NULL;
key_frame_ts_ = -1;
rtp_key_frame_ts_ = -1;
header_sn_ = 0;
memset(cache_video_pkts_, 0, sizeof(cache_video_pkts_));
}
@ -1504,24 +1504,24 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_key_frame(SrsRtpPacket* pkt)
}
}
if (-1 == key_frame_ts_) {
key_frame_ts_ = pkt->get_avsync_time();
if (-1 == rtp_key_frame_ts_) {
rtp_key_frame_ts_ = pkt->header.get_timestamp();
header_sn_ = pkt->header.get_sequence();
lost_sn_ = header_sn_ + 1;
// Received key frame and clean cache of old p frame pkts
clear_cached_video();
srs_trace("set ts=%lld, header=%hu, lost=%hu", key_frame_ts_, header_sn_, lost_sn_);
} else if (key_frame_ts_ != pkt->get_avsync_time()) {
srs_trace("set ts=%u, header=%hu, lost=%hu", (uint32_t)rtp_key_frame_ts_, header_sn_, lost_sn_);
} else if (rtp_key_frame_ts_ != pkt->header.get_timestamp()) {
//new key frame, clean cache
int64_t old_ts = key_frame_ts_;
int64_t old_ts = rtp_key_frame_ts_;
uint16_t old_header_sn = header_sn_;
uint16_t old_lost_sn = lost_sn_;
key_frame_ts_ = pkt->get_avsync_time();
rtp_key_frame_ts_ = pkt->header.get_timestamp();
header_sn_ = pkt->header.get_sequence();
lost_sn_ = header_sn_ + 1;
clear_cached_video();
srs_trace("drop old ts=%lld, header=%hu, lost=%hu, set new ts=%lld, header=%hu, lost=%hu",
old_ts, old_header_sn, old_lost_sn, key_frame_ts_, header_sn_, lost_sn_);
srs_warn("drop old ts=%u, header=%hu, lost=%hu, set new ts=%u, header=%hu, lost=%hu",
(uint32_t)old_ts, old_header_sn, old_lost_sn, (uint32_t)rtp_key_frame_ts_, header_sn_, lost_sn_);
}
uint16_t index = cache_index(pkt->header.get_sequence());
@ -1561,9 +1561,10 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
srs_error_t err = srs_success;
int nb_payload = 0;
uint16_t cnt = end - start + 1;
int16_t cnt = srs_rtp_seq_distance(start, end) + 1;
srs_assert(cnt >= 1);
for (uint16_t i = 0; i < cnt; ++i) {
for (uint16_t i = 0; i < (uint16_t)cnt; ++i) {
uint16_t sn = start + i;
uint16_t index = cache_index(sn);
SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
@ -1615,7 +1616,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
SrsBuffer payload(rtmp.payload, rtmp.size);
if (pkt->is_keyframe()) {
payload.write_1bytes(0x17); // type(4 bits): key frame; code(4bits): avc
key_frame_ts_ = -1;
rtp_key_frame_ts_ = -1;
} else {
payload.write_1bytes(0x27); // type(4 bits): inter frame; code(4bits): avc
}
@ -1625,7 +1626,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
payload.write_1bytes(0x0);
int nalu_len = 0;
for (uint16_t i = 0; i < cnt; ++i) {
for (uint16_t i = 0; i < (uint16_t)cnt; ++i) {
uint16_t index = cache_index((start + i));
SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
@ -1664,10 +1665,10 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
if (stap_payload) {
for (int j = 0; j < (int)stap_payload->nalus.size(); ++j) {
SrsSample* sample = stap_payload->nalus.at(j);
if (sample->size > 0) {
payload.write_4bytes(sample->size);
if (sample->size > 0) {
payload.write_4bytes(sample->size);
payload.write_bytes(sample->bytes, sample->size);
}
}
}
srs_freep(pkt);
continue;
@ -1726,7 +1727,7 @@ int32_t SrsRtmpFromRtcBridger::find_next_lost_sn(uint16_t current_sn, uint16_t&
}
}
srs_error("the cache is mess. the packet count of video frame is more than %u", s_cache_size);
srs_error("cache overflow. the packet count of video frame is more than %u", s_cache_size);
return -2;
}
@ -1746,10 +1747,12 @@ void SrsRtmpFromRtcBridger::clear_cached_video()
bool SrsRtmpFromRtcBridger::check_frame_complete(const uint16_t start, const uint16_t end)
{
uint16_t cnt = (end - start + 1);
int16_t cnt = srs_rtp_seq_distance(start, end) + 1;
srs_assert(cnt >= 1);
uint16_t fu_s_c = 0;
uint16_t fu_e_c = 0;
for (uint16_t i = 0; i < cnt; ++i) {
for (uint16_t i = 0; i < (uint16_t)cnt; ++i) {
int index = cache_index((start + i));
SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
@ -2264,6 +2267,8 @@ SrsRtcRecvTrack::SrsRtcRecvTrack(SrsRtcConnection* session, SrsRtcTrackDescripti
last_sender_report_rtp_time_ = 0;
last_sender_report_rtp_time1_ = 0;
rate_ = 0.0;
last_sender_report_sys_time_ = 0;
}
@ -2299,40 +2304,41 @@ void SrsRtcRecvTrack::update_send_report_time(const SrsNtp& ntp, uint32_t rtp_ti
// TODO: FIXME: Use system wall clock.
last_sender_report_sys_time_ = srs_update_system_time();
if (last_sender_report_rtp_time1_ > 0) {
// WebRTC using sender report to sync audio/video timestamp, because audio video have different timebase,
// typical audio opus is 48000Hz, video is 90000Hz.
// We using two sender report point to calculate avsync timestamp(clock time) with any given rtp timestamp.
// For example, there are two history sender report of audio as below.
// sender_report1: rtp_time1 = 10000, ntp_time1 = 40000
// sender_report : rtp_time = 10960, ntp_time = 40020
// (rtp_time - rtp_time1) / (ntp_time - ntp_time1) = 960 / 20 = 48,
// Now we can calcualte ntp time(ntp_x) of any given rtp timestamp(rtp_x),
// (rtp_x - rtp_time) / (ntp_x - ntp_time) = 48 => ntp_x = (rtp_x - rtp_time) / 48 + ntp_time;
double sys_time_elapsed = static_cast<double>(last_sender_report_ntp_.system_ms_) - static_cast<double>(last_sender_report_ntp1_.system_ms_);
// Check sys_time_elapsed is equal to zero.
if (fpclassify(sys_time_elapsed) == FP_ZERO) {
return;
}
double rtp_time_elpased = static_cast<double>(last_sender_report_rtp_time_) - static_cast<double>(last_sender_report_rtp_time1_);
double rate = round(rtp_time_elpased / sys_time_elapsed);
// TODO: FIXME: use the sample rate from sdp.
if (rate > 0) {
rate_ = rate;
}
}
}
int64_t SrsRtcRecvTrack::cal_avsync_time(uint32_t rtp_time)
{
// Have no recv at least 2 sender reports, can't calculate sync time.
// TODO: FIXME: use the sample rate from sdp.
if (last_sender_report_rtp_time1_ <= 0) {
if (rate_ < 0.001) {
return -1;
}
// WebRTC using sender report to sync audio/video timestamp, because audio video have different timebase,
// typical audio opus is 48000Hz, video is 90000Hz.
// We using two sender report point to calculate avsync timestamp(clock time) with any given rtp timestamp.
// For example, there are two history sender report of audio as below.
// sender_report1: rtp_time1 = 10000, ntp_time1 = 40000
// sender_report : rtp_time = 10960, ntp_time = 40020
// (rtp_time - rtp_time1) / (ntp_time - ntp_time1) = 960 / 20 = 48,
// Now we can calcualte ntp time(ntp_x) of any given rtp timestamp(rtp_x),
// (rtp_x - rtp_time) / (ntp_x - ntp_time) = 48 => ntp_x = (rtp_x - rtp_time) / 48 + ntp_time;
double sys_time_elapsed = static_cast<double>(last_sender_report_ntp_.system_ms_) - static_cast<double>(last_sender_report_ntp1_.system_ms_);
// Check sys_time_elapsed is equal to zero.
if (fpclassify(sys_time_elapsed) == FP_ZERO) {
return -1;
}
double rtp_time_elpased = static_cast<double>(last_sender_report_rtp_time_) - static_cast<double>(last_sender_report_rtp_time1_);
int rate = round(rtp_time_elpased / sys_time_elapsed);
if (rate <= 0) {
return -1;
}
double delta = round((rtp_time - last_sender_report_rtp_time_) / rate);
double delta = round((rtp_time - last_sender_report_rtp_time_) / rate_);
int64_t avsync_time = delta + last_sender_report_ntp_.system_ms_;