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fix(rtp): improve timestamp accuracy for video (#3883)

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Instead of using now() when the RTP packet is created, use the earlier packet->frame_timestamp that we're already collecting for host latency stats. This timestamp is more accurate to when we captured the frame, and the same timestamp value is shared by all RTP packets that make up the same video frame. Duplicate frames without capture timestamps use the ratecontrol timestamp.
This commit is contained in:
Andy Grundman 2025-05-21 19:56:41 -04:00 committed by GitHub
parent 9c08c75a44
commit 3092471be5
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1 changed files with 18 additions and 11 deletions
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29
src/stream.cpp
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@ -1263,7 +1263,7 @@ namespace stream {
void videoBroadcastThread(udp::socket &sock) {
auto shutdown_event = mail::man->event<bool>(mail::broadcast_shutdown);
auto packets = mail::man->queue<video::packet_t>(mail::video_packets);
auto timebase = boost::posix_time::microsec_clock::universal_time();
auto video_epoch = std::chrono::steady_clock::now();
// Video traffic is sent on this thread
platf::adjust_thread_priority(platf::thread_priority_e::high);
@ -1459,14 +1459,20 @@ namespace stream {
size_t next_shard_to_send = 0;
// RTP video timestamps use a 90 KHz clock and the frame_timestamp from when the frame was captured
// When a timestamp isn't available (duplicate frames), the timestamp from rate control is used instead.
bool frame_is_dupe = false;
if (!packet->frame_timestamp) {
packet->frame_timestamp = ratecontrol_next_frame_start;
frame_is_dupe = true;
}
using rtp_tick = std::chrono::duration<uint32_t, std::ratio<1, 90000>>;
uint32_t timestamp = std::chrono::round<rtp_tick>(*packet->frame_timestamp - video_epoch).count();
// set FEC info now that we know for sure what our percentage will be for this frame
for (auto x = 0; x < shards.size(); ++x) {
auto *inspect = (video_packet_raw_t *) shards.data(x);
// RTP video timestamps use a 90 KHz clock
auto now = boost::posix_time::microsec_clock::universal_time();
auto timestamp = (now - timebase).total_microseconds() / (1000 / 90);
inspect->packet.fecInfo =
(x << 12 |
shards.data_shards << 22 |
@ -1558,11 +1564,11 @@ namespace stream {
frame_network_latency_logger.second_point_now_and_log();
if (packet->is_idr()) {
BOOST_LOG(verbose) << "Key Frame ["sv << packet->frame_index() << "] :: send ["sv << shards.size() << "] shards..."sv;
} else {
BOOST_LOG(verbose) << "Frame ["sv << packet->frame_index() << "] :: send ["sv << shards.size() << "] shards..."sv << std::endl;
}
BOOST_LOG(verbose) << "Sent Frame seq ["sv << packet->frame_index() << "] pts ["sv << timestamp
<< "] shards ["sv << shards.size() << "/"sv << shards.percentage << "%]"sv
<< (frame_is_dupe ? " Dupe" : "")
<< (packet->is_idr() ? " Key" : "")
<< (packet->after_ref_frame_invalidation ? " RFI" : "");
++blockIndex;
lowseq += shards.size();
@ -1622,6 +1628,8 @@ namespace stream {
break;
}
BOOST_LOG(verbose) << "Audio [seq "sv << sequenceNumber << ", pts "sv << timestamp << "] :: send..."sv;
audio_packet.rtp.sequenceNumber = util::endian::big(sequenceNumber);
audio_packet.rtp.timestamp = util::endian::big(timestamp);
@ -1641,7 +1649,6 @@ namespace stream {
session->localAddress,
};
platf::send(send_info);
BOOST_LOG(verbose) << "Audio ["sv << sequenceNumber << "] :: send..."sv;
auto &fec_packet = session->audio.fec_packet;
// initialize the FEC header at the beginning of the FEC block