blob: 46639067c283cf0667e38957aaff29476d2414cb [file] [log] [blame]
/*
* Copyright (c) 2015 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 "webrtc/modules/congestion_controller/transport_feedback_adapter.h"
#include <algorithm>
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/delay_based_bwe.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/utility/include/process_thread.h"
namespace webrtc {
const int64_t kNoTimestamp = -1;
const int64_t kSendTimeHistoryWindowMs = 10000;
const int64_t kBaseTimestampScaleFactor =
rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
class PacketInfoComparator {
public:
inline bool operator()(const PacketInfo& lhs, const PacketInfo& rhs) {
if (lhs.arrival_time_ms != rhs.arrival_time_ms)
return lhs.arrival_time_ms < rhs.arrival_time_ms;
if (lhs.send_time_ms != rhs.send_time_ms)
return lhs.send_time_ms < rhs.send_time_ms;
return lhs.sequence_number < rhs.sequence_number;
}
};
TransportFeedbackAdapter::TransportFeedbackAdapter(
Clock* clock,
BitrateController* bitrate_controller)
: send_time_history_(clock, kSendTimeHistoryWindowMs),
clock_(clock),
current_offset_ms_(kNoTimestamp),
last_timestamp_us_(kNoTimestamp),
bitrate_controller_(bitrate_controller) {}
TransportFeedbackAdapter::~TransportFeedbackAdapter() {}
void TransportFeedbackAdapter::InitBwe() {
rtc::CritScope cs(&bwe_lock_);
delay_based_bwe_.reset(new DelayBasedBwe(clock_));
}
void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
size_t length,
int probe_cluster_id) {
rtc::CritScope cs(&lock_);
send_time_history_.AddAndRemoveOld(sequence_number, length, probe_cluster_id);
}
void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
int64_t send_time_ms) {
rtc::CritScope cs(&lock_);
send_time_history_.OnSentPacket(sequence_number, send_time_ms);
}
void TransportFeedbackAdapter::SetMinBitrate(int min_bitrate_bps) {
rtc::CritScope cs(&bwe_lock_);
delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
}
std::vector<PacketInfo> TransportFeedbackAdapter::GetPacketFeedbackVector(
const rtcp::TransportFeedback& feedback) {
int64_t timestamp_us = feedback.GetBaseTimeUs();
// Add timestamp deltas to a local time base selected on first packet arrival.
// This won't be the true time base, but makes it easier to manually inspect
// time stamps.
if (last_timestamp_us_ == kNoTimestamp) {
current_offset_ms_ = clock_->TimeInMilliseconds();
} else {
int64_t delta = timestamp_us - last_timestamp_us_;
// Detect and compensate for wrap-arounds in base time.
if (std::abs(delta - kBaseTimestampRangeSizeUs) < std::abs(delta)) {
delta -= kBaseTimestampRangeSizeUs; // Wrap backwards.
} else if (std::abs(delta + kBaseTimestampRangeSizeUs) < std::abs(delta)) {
delta += kBaseTimestampRangeSizeUs; // Wrap forwards.
}
current_offset_ms_ += delta / 1000;
}
last_timestamp_us_ = timestamp_us;
uint16_t sequence_number = feedback.GetBaseSequence();
std::vector<int64_t> delta_vec = feedback.GetReceiveDeltasUs();
auto delta_it = delta_vec.begin();
std::vector<PacketInfo> packet_feedback_vector;
packet_feedback_vector.reserve(delta_vec.size());
{
rtc::CritScope cs(&lock_);
size_t failed_lookups = 0;
int64_t offset_us = 0;
for (auto symbol : feedback.GetStatusVector()) {
if (symbol != rtcp::TransportFeedback::StatusSymbol::kNotReceived) {
RTC_DCHECK(delta_it != delta_vec.end());
offset_us += *(delta_it++);
int64_t timestamp_ms = current_offset_ms_ + (offset_us / 1000);
PacketInfo info(timestamp_ms, sequence_number);
if (send_time_history_.GetInfo(&info, true) && info.send_time_ms >= 0) {
packet_feedback_vector.push_back(info);
} else {
++failed_lookups;
}
}
++sequence_number;
}
std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
PacketInfoComparator());
RTC_DCHECK(delta_it == delta_vec.end());
if (failed_lookups > 0) {
LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
<< " packet" << (failed_lookups > 1 ? "s" : "")
<< ". Send time history too small?";
}
}
return packet_feedback_vector;
}
void TransportFeedbackAdapter::OnTransportFeedback(
const rtcp::TransportFeedback& feedback) {
last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
DelayBasedBwe::Result result;
{
rtc::CritScope cs(&bwe_lock_);
result = delay_based_bwe_->IncomingPacketFeedbackVector(
last_packet_feedback_vector_);
}
if (result.updated)
bitrate_controller_->OnDelayBasedBweResult(result);
}
std::vector<PacketInfo> TransportFeedbackAdapter::GetTransportFeedbackVector()
const {
return last_packet_feedback_vector_;
}
void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
int64_t max_rtt_ms) {
rtc::CritScope cs(&bwe_lock_);
delay_based_bwe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
}
} // namespace webrtc