webrtc_m130/call/rtp_transport_controller_send.cc

/*
 *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "call/rtp_transport_controller_send.h"
#include "rtc_base/logging.h"

namespace webrtc {

RtpTransportControllerSend::RtpTransportControllerSend(
    Clock* clock,
    webrtc::RtcEventLog* event_log,
    const BitrateConstraints& bitrate_config)
    : pacer_(clock, &packet_router_, event_log),
      send_side_cc_(clock, nullptr /* observer */, event_log, &pacer_),
      bitrate_configurator_(bitrate_config) {
  send_side_cc_.SignalNetworkState(kNetworkDown);
  send_side_cc_.SetBweBitrates(bitrate_config.min_bitrate_bps,
                               bitrate_config.start_bitrate_bps,
                               bitrate_config.max_bitrate_bps);
}
RtpTransportControllerSend::~RtpTransportControllerSend() = default;

PacketRouter* RtpTransportControllerSend::packet_router() {
  return &packet_router_;
}

TransportFeedbackObserver*
RtpTransportControllerSend::transport_feedback_observer() {
  return &send_side_cc_;
}

RtpPacketSender* RtpTransportControllerSend::packet_sender() {
  return &pacer_;
}

const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const {
  return keepalive_;
}

void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
    int min_send_bitrate_bps,
    int max_padding_bitrate_bps) {
  pacer_.SetSendBitrateLimits(min_send_bitrate_bps, max_padding_bitrate_bps);
}

void RtpTransportControllerSend::SetKeepAliveConfig(
    const RtpKeepAliveConfig& config) {
  keepalive_ = config;
}
Module* RtpTransportControllerSend::GetPacerModule() {
  return &pacer_;
}
void RtpTransportControllerSend::SetPacingFactor(float pacing_factor) {
  pacer_.SetPacingFactor(pacing_factor);
}
void RtpTransportControllerSend::SetQueueTimeLimit(int limit_ms) {
  pacer_.SetQueueTimeLimit(limit_ms);
}
Module* RtpTransportControllerSend::GetModule() {
  return &send_side_cc_;
}
CallStatsObserver* RtpTransportControllerSend::GetCallStatsObserver() {
  return &send_side_cc_;
}
void RtpTransportControllerSend::RegisterPacketFeedbackObserver(
    PacketFeedbackObserver* observer) {
  send_side_cc_.RegisterPacketFeedbackObserver(observer);
}
void RtpTransportControllerSend::DeRegisterPacketFeedbackObserver(
    PacketFeedbackObserver* observer) {
  send_side_cc_.DeRegisterPacketFeedbackObserver(observer);
}
void RtpTransportControllerSend::RegisterNetworkObserver(
    NetworkChangedObserver* observer) {
  send_side_cc_.RegisterNetworkObserver(observer);
}
void RtpTransportControllerSend::DeRegisterNetworkObserver(
    NetworkChangedObserver* observer) {
  send_side_cc_.DeRegisterNetworkObserver(observer);
}
void RtpTransportControllerSend::OnNetworkRouteChanged(
    const std::string& transport_name,
    const rtc::NetworkRoute& network_route) {
  BitrateConstraints bitrate_config = bitrate_configurator_.GetConfig();
  RTC_LOG(LS_INFO) << "Network route changed on transport " << transport_name
                   << ": new local network id "
                   << network_route.local_network_id
                   << " new remote network id "
                   << network_route.remote_network_id
                   << " Reset bitrates to min: "
                   << bitrate_config.min_bitrate_bps
                   << " bps, start: " << bitrate_config.start_bitrate_bps
                   << " bps,  max: " << bitrate_config.max_bitrate_bps
                   << " bps.";
  RTC_DCHECK_GT(bitrate_config.start_bitrate_bps, 0);
  send_side_cc_.OnNetworkRouteChanged(
      network_route, bitrate_config.start_bitrate_bps,
      bitrate_config.min_bitrate_bps, bitrate_config.max_bitrate_bps);
}
void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) {
  send_side_cc_.SignalNetworkState(network_available ? kNetworkUp
                                                     : kNetworkDown);
}
void RtpTransportControllerSend::SetTransportOverhead(
    size_t transport_overhead_bytes_per_packet) {
  send_side_cc_.SetTransportOverhead(transport_overhead_bytes_per_packet);
}
RtcpBandwidthObserver* RtpTransportControllerSend::GetBandwidthObserver() {
  return send_side_cc_.GetBandwidthObserver();
}
bool RtpTransportControllerSend::AvailableBandwidth(uint32_t* bandwidth) const {
  return send_side_cc_.AvailableBandwidth(bandwidth);
}
int64_t RtpTransportControllerSend::GetPacerQueuingDelayMs() const {
  return send_side_cc_.GetPacerQueuingDelayMs();
}
int64_t RtpTransportControllerSend::GetFirstPacketTimeMs() const {
  return send_side_cc_.GetFirstPacketTimeMs();
}
RateLimiter* RtpTransportControllerSend::GetRetransmissionRateLimiter() {
  return send_side_cc_.GetRetransmissionRateLimiter();
}
void RtpTransportControllerSend::EnablePeriodicAlrProbing(bool enable) {
  send_side_cc_.EnablePeriodicAlrProbing(enable);
}
void RtpTransportControllerSend::OnSentPacket(
    const rtc::SentPacket& sent_packet) {
  send_side_cc_.OnSentPacket(sent_packet);
}

void RtpTransportControllerSend::SetSdpBitrateParameters(
    const BitrateConstraints& constraints) {
  rtc::Optional<BitrateConstraints> updated =
      bitrate_configurator_.UpdateWithSdpParameters(constraints);
  if (updated.has_value()) {
    send_side_cc_.SetBweBitrates(updated->min_bitrate_bps,
                                 updated->start_bitrate_bps,
                                 updated->max_bitrate_bps);
  } else {
    RTC_LOG(LS_VERBOSE)
        << "WebRTC.RtpTransportControllerSend.SetBitrateConfig: "
        << "nothing to update";
  }
}

void RtpTransportControllerSend::SetClientBitratePreferences(
    const BitrateConstraintsMask& preferences) {
  rtc::Optional<BitrateConstraints> updated =
      bitrate_configurator_.UpdateWithClientPreferences(preferences);
  if (updated.has_value()) {
    send_side_cc_.SetBweBitrates(updated->min_bitrate_bps,
                                 updated->start_bitrate_bps,
                                 updated->max_bitrate_bps);
  } else {
    RTC_LOG(LS_VERBOSE)
        << "WebRTC.RtpTransportControllerSend.SetBitrateConfigMask: "
        << "nothing to update";
  }
}
}  // namespace webrtc