【WebRTC---源码篇】（十：一）WEBRTC 发送视频RTP包

RTPSenderVideo在整个框架中起到重要的作用，它把采集的数据进行编码，并且在流程中会进行将编码后的数据进行RTP打包，最后发送到网络层

RTPSenderVideo::SendVideo

//对编码数据打包
bool RTPSenderVideo::SendVideo(
    int payload_type,
    absl::optional codec_type,
    uint32_t rtp_timestamp,
    int64_t capture_time_ms,
    rtc::ArrayView<const uint8_t> payload,
    const RTPFragmentationHeader* fragmentation,
    RTPVideoHeader video_header,
    absl::optional<int64_t> expected_retransmission_time_ms) {
#if RTC_TRACE_EVENTS_ENABLED
  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
                          FrameTypeToString(video_header.frame_type));
#endif
  RTC_CHECK_RUNS_SERIALIZED(&send_checker_);
  //如果是空帧直接返回
  if (video_header.frame_type == VideoFrameType::kEmptyFrame)
    return true;
  //如果有效荷载没有内容，直接返回false
  if (payload.empty())
    return false;
 
  int32_t retra