webrtc可以对网络带宽进行主动探测。视频流的码率相对稳定，因此在带宽探测的时候，因为要向网络中发送多余的数据包，可能会面临着没有新的数据包向外发送的情况。对于这个问题，webrtc采取冗余发送数据包的方式，对带宽进行探测。数据包在发送到网络中，webrtc/modules/rtp_rtcp/source/rtp_sender.cc会将数据包缓存到 packet_history_。在带宽探测是的时候，可以拿出一个与探测包最合适的历史包，冗余发送到网络中。
size_t PacedSender::SendPadding(size_t padding_needed, const PacedPacketInfo& pacing_info) { RTC_DCHECK_GT(packet_counter_, 0); critsect_.Leave(); size_t bytes_sent = packet_sender_->TimeToSendPadding(padding_needed, pacing_info);//PacketRouter critsect_.Enter(); if (bytes_sent > 0) { UpdateBudgetWithBytesSent(bytes_sent); } return bytes_sent; }PacketRouter
size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send, const PacedPacketInfo& pacing_info) { size_t total_bytes_sent = 0; rtc::CritScope cs(&modules_crit_); // First try on the last rtp module to have sent media. This increases the // the chance that any payload based padding will be useful as it will be // somewhat distributed over modules according the packet rate, even if it // will be more skewed towards the highest bitrate stream. At the very least // this prevents sending payload padding on a disabled stream where it's // guaranteed not to be useful. if (last_send_module_ != nullptr) { RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), last_send_module_) != rtp_send_modules_.end()); RTC_DCHECK(last_send_module_->HasBweExtensions()); total_bytes_sent += last_send_module_->TimeToSendPadding( bytes_to_send - total_bytes_sent, pacing_info); if (total_bytes_sent >= bytes_to_send) { return total_bytes_sent; } } // Rtp modules are ordered by which stream can most benefit from padding. for (RtpRtcp* module : rtp_send_modules_) { if (module->SendingMedia() && module->HasBweExtensions()) { size_t bytes_sent = module->TimeToSendPadding( bytes_to_send - total_bytes_sent, pacing_info);//ModuleRtpRtcpImpl total_bytes_sent += bytes_sent; if (total_bytes_sent >= bytes_to_send) break; } } return total_bytes_sent; }ModuleRtpRtcpImpl
size_t ModuleRtpRtcpImpl::TimeToSendPadding( size_t bytes, const PacedPacketInfo& pacing_info) { return rtp_sender_->TimeToSendPadding(bytes, pacing_info);//RTPSender }RTPSender
size_t RTPSender::TimeToSendPadding(size_t bytes, const PacedPacketInfo& pacing_info) { if (bytes == 0) return 0; size_t bytes_sent = TrySendRedundantPayloads(bytes, pacing_info); if (bytes_sent < bytes) bytes_sent += SendPadData(bytes - bytes_sent, pacing_info); return bytes_sent; } size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send, const PacedPacketInfo& pacing_info) { { rtc::CritScope lock(&send_critsect_); if (!sending_media_) return 0; if ((rtx_ & kRtxRedundantPayloads) == 0) return 0; } int bytes_left = static_cast<int>(bytes_to_send); while (bytes_left > 0) { std::unique_ptr<RtpPacketToSend> packet = packet_history_.GetBestFittingPacket(bytes_left);/**if any optimization?**/ if (!packet) break; size_t payload_size = packet->payload_size(); if (!PrepareAndSendPacket(std::move(packet), true, false, pacing_info)) break; bytes_left -= payload_size; } return bytes_to_send - bytes_left; }最终，拿出一个已经发过的历史数据，再次发送出去，packet_history_.GetBestFittingPacket(bytes_left)。 [1]WebRTC 中的带宽侦测
