
This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
222 lines
7.2 KiB
C++
222 lines
7.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*
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*/
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#include "webrtc/modules/bitrate_controller/bitrate_controller_impl.h"
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#include <algorithm>
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#include <utility>
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class BitrateControllerImpl::RtcpBandwidthObserverImpl
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: public RtcpBandwidthObserver {
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public:
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explicit RtcpBandwidthObserverImpl(BitrateControllerImpl* owner)
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: owner_(owner) {
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}
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virtual ~RtcpBandwidthObserverImpl() {
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}
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// Received RTCP REMB or TMMBR.
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void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
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owner_->OnReceivedEstimatedBitrate(bitrate);
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}
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// Received RTCP receiver block.
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void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
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int64_t rtt,
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int64_t now_ms) override {
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if (report_blocks.empty())
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return;
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int fraction_lost_aggregate = 0;
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int total_number_of_packets = 0;
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// Compute the a weighted average of the fraction loss from all report
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// blocks.
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for (ReportBlockList::const_iterator it = report_blocks.begin();
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it != report_blocks.end(); ++it) {
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std::map<uint32_t, uint32_t>::iterator seq_num_it =
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ssrc_to_last_received_extended_high_seq_num_.find(it->sourceSSRC);
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int number_of_packets = 0;
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if (seq_num_it != ssrc_to_last_received_extended_high_seq_num_.end())
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number_of_packets = it->extendedHighSeqNum -
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seq_num_it->second;
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fraction_lost_aggregate += number_of_packets * it->fractionLost;
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total_number_of_packets += number_of_packets;
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// Update last received for this SSRC.
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ssrc_to_last_received_extended_high_seq_num_[it->sourceSSRC] =
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it->extendedHighSeqNum;
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}
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if (total_number_of_packets == 0)
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fraction_lost_aggregate = 0;
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else
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fraction_lost_aggregate = (fraction_lost_aggregate +
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total_number_of_packets / 2) / total_number_of_packets;
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if (fraction_lost_aggregate > 255)
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return;
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owner_->OnReceivedRtcpReceiverReport(fraction_lost_aggregate, rtt,
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total_number_of_packets, now_ms);
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}
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private:
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std::map<uint32_t, uint32_t> ssrc_to_last_received_extended_high_seq_num_;
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BitrateControllerImpl* owner_;
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};
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BitrateController* BitrateController::CreateBitrateController(
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Clock* clock,
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BitrateObserver* observer) {
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return new BitrateControllerImpl(clock, observer);
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}
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BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
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BitrateObserver* observer)
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: clock_(clock),
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observer_(observer),
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last_bitrate_update_ms_(clock_->TimeInMilliseconds()),
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bandwidth_estimation_(),
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reserved_bitrate_bps_(0),
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last_bitrate_bps_(0),
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last_fraction_loss_(0),
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last_rtt_ms_(0),
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last_reserved_bitrate_bps_(0) {
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// This calls the observer_, which means that the observer provided by the
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// user must be ready to accept a bitrate update when it constructs the
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// controller. We do this to avoid having to keep synchronized initial values
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// in both the controller and the allocator.
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MaybeTriggerOnNetworkChanged();
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}
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RtcpBandwidthObserver* BitrateControllerImpl::CreateRtcpBandwidthObserver() {
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return new RtcpBandwidthObserverImpl(this);
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}
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void BitrateControllerImpl::SetStartBitrate(int start_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.SetSendBitrate(start_bitrate_bps);
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::SetMinMaxBitrate(int min_bitrate_bps,
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int max_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::SetReservedBitrate(uint32_t reserved_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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reserved_bitrate_bps_ = reserved_bitrate_bps;
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::OnReceivedEstimatedBitrate(uint32_t bitrate) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.UpdateReceiverEstimate(clock_->TimeInMilliseconds(),
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bitrate);
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}
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MaybeTriggerOnNetworkChanged();
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}
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int64_t BitrateControllerImpl::TimeUntilNextProcess() {
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const int64_t kBitrateControllerUpdateIntervalMs = 25;
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rtc::CritScope cs(&critsect_);
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int64_t time_since_update_ms =
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clock_->TimeInMilliseconds() - last_bitrate_update_ms_;
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return std::max<int64_t>(
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kBitrateControllerUpdateIntervalMs - time_since_update_ms, 0);
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}
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int32_t BitrateControllerImpl::Process() {
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if (TimeUntilNextProcess() > 0)
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return 0;
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.UpdateEstimate(clock_->TimeInMilliseconds());
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}
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MaybeTriggerOnNetworkChanged();
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last_bitrate_update_ms_ = clock_->TimeInMilliseconds();
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return 0;
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}
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void BitrateControllerImpl::OnReceivedRtcpReceiverReport(
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uint8_t fraction_loss,
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int64_t rtt,
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int number_of_packets,
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int64_t now_ms) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.UpdateReceiverBlock(fraction_loss, rtt,
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number_of_packets, now_ms);
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() {
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uint32_t bitrate;
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uint8_t fraction_loss;
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int64_t rtt;
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if (GetNetworkParameters(&bitrate, &fraction_loss, &rtt))
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observer_->OnNetworkChanged(bitrate, fraction_loss, rtt);
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}
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bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,
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uint8_t* fraction_loss,
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int64_t* rtt) {
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rtc::CritScope cs(&critsect_);
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int current_bitrate;
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bandwidth_estimation_.CurrentEstimate(¤t_bitrate, fraction_loss, rtt);
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*bitrate = current_bitrate;
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*bitrate -= std::min(*bitrate, reserved_bitrate_bps_);
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*bitrate =
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std::max<uint32_t>(*bitrate, bandwidth_estimation_.GetMinBitrate());
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bool new_bitrate = false;
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if (*bitrate != last_bitrate_bps_ || *fraction_loss != last_fraction_loss_ ||
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*rtt != last_rtt_ms_ ||
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last_reserved_bitrate_bps_ != reserved_bitrate_bps_) {
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last_bitrate_bps_ = *bitrate;
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last_fraction_loss_ = *fraction_loss;
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last_rtt_ms_ = *rtt;
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last_reserved_bitrate_bps_ = reserved_bitrate_bps_;
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new_bitrate = true;
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}
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return new_bitrate;
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}
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bool BitrateControllerImpl::AvailableBandwidth(uint32_t* bandwidth) const {
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rtc::CritScope cs(&critsect_);
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int bitrate;
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uint8_t fraction_loss;
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int64_t rtt;
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bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
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if (bitrate > 0) {
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bitrate = bitrate - std::min<int>(bitrate, reserved_bitrate_bps_);
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bitrate = std::max(bitrate, bandwidth_estimation_.GetMinBitrate());
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*bandwidth = bitrate;
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return true;
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}
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return false;
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}
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} // namespace webrtc
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