webrtc BWE 判断流程
这里主要说的是接收端的判断流程：
// Send-side BWE. rtp 扩展 transport sequence number
// Receive-side BWE. Rtp 扩展 abs-send-time
1：
PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us)
2：
//每一个 rtp packet 都判断；
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type) {
auto it = receive_rtp_config_.find(packet.Ssrc());
bool use_send_side_bwe =
(it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
RTPHeader header;
packet.GetHeader(&header);
if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
// Inconsistent configuration of send side BWE. Do nothing.
// TODO(nisse): Without this check, we may produce RTCP feedback
// packets even when not negotiated. But it would be cleaner to
// move the check down to RTCPSender::SendFeedbackPacket, which
// would also help the PacketRouter to select an appropriate rtp
// module in the case that some, but not all, have RTCP feedback
// enabled.
return;
}
// For audio, we only support send side BWE.
//注意这里，audio 仅支持 send side BWE；
if (media_type == MediaType::VIDEO ||
(use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
receive_side_cc_.OnReceivedPacket(
packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
header);
}
}
3：
//就是在这个函数里判断；
void ReceiveSideCongestionController::OnReceivedPacket(
int64_t arrival_time_ms,
size_t payload_size,
const RTPHeader& header) {
// Send-side BWE.
if (header.extension.hasTransportSequenceNumber) {
remote_estimator_proxy_.IncomingPacket(arrival_time_ms, payload_size,
header);
} else {
// Receive-side BWE.
remote_bitrate_estimator_.IncomingPacket(arrival_time_ms, payload_size,
header);
}
}