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1：webrtc_video_capture线程
2：EncoderQueue线程
源数据入口函数：
void VideoStreamEncoder::OnFrame(const VideoFrame& video_frame) {
RTC_DCHECK_RUNS_SERIALIZED(&incoming_frame_race_checker_);VideoFrame incoming_frame = video_frame;
// Local time in webrtc time base.
int64_t current_time_us = clock_->TimeInMicroseconds();
int64_t current_time_ms = current_time_us / rtc::kNumMicrosecsPerMillisec;
// In some cases, e.g., when the frame from decoder is fed to encoder,
// the timestamp may be set to the future. As the encoding pipeline assumes
// capture time to be less than present time, we should reset the capture
// timestamps here. Otherwise there may be issues with RTP send stream.
if (incoming_frame.timestamp_us() > current_time_us)
incoming_frame.set_timestamp_us(current_time_us);
// Capture time may come from clock with an offset and drift from clock_.
int64_t capture_ntp_time_ms;
if (video_frame.ntp_time_ms() > 0) {
capture_ntp_time_ms = video_frame.ntp_time_ms();
} else if (video_frame.render_time_ms() != 0) {
capture_ntp_time_ms = video_frame.render_time_ms() + delta_ntp_internal_ms_;
} else {
capture_ntp_time_ms = current_time_ms + delta_ntp_internal_ms_;
}
incoming_frame.set_ntp_time_ms(capture_ntp_time_ms);
// Convert NTP time, in ms, to RTP timestamp.
const int kMsToRtpTimestamp = 90;
incoming_frame.set_timestamp(
kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
// We don't allow the same capture time for two frames, drop this one.
RTC_LOG(LS_WARNING) << "Same/old NTP timestamp ("
<< incoming_frame.ntp_time_ms()
<< " <= " << last_captured_timestamp_
<< ") for incoming frame. Dropping.";
return;
}
bool log_stats = false;
if (current_time_ms - last_frame_log_ms_ > kFrameLogIntervalMs) {
last_frame_log_ms_ = current_time_ms;
log_stats = true;
}
last_captured_timestamp_ = incoming_frame.ntp_time_ms();
encoder_queue_.PostTask(std::unique_ptr<rtc::QueuedTask>(new EncodeTask(
incoming_frame, this, rtc::TimeMicros(), log_stats)));
}
3：PacedSender
const std::unique_ptr<PacketQueue> packets_
这个类中有一个成员变量：packets_：将打包好的rtp数据缓存到packets_中；
void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) {
rtc::CritScope cs(&critsect_);
RTC_DCHECK(estimated_bitrate_bps_ > 0)
<< "SetEstimatedBitrate must be called before InsertPacket.";
int64_t now_ms = clock_->TimeInMilliseconds();
prober_->OnIncomingPacket(bytes);
if (capture_time_ms < 0)
capture_time_ms = now_ms;
packets_->Push(PacketQueue::Packet(priority, ssrc, sequence_number,
capture_time_ms, now_ms, bytes,
retransmission, packet_counter_++));
}