webrtc_m130/webrtc/modules/congestion_controller/transport_feedback_adapter.cc

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/*
* 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_);
Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) Reason for revert: Fix already landed in google3, this revert actually breaks the import. Original issue's description: > Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) > > Reason for revert: > Revert this because it broke the google3 import build. > http://webrtc-buildbot-master.mtv.corp.google.com:21000/builders/WebRTC%20google3%20Importer%20%28Shem%20TOT%29/builds/67/steps/blaze_regular_tests/logs/stdio > > Original issue's description: > > Remove audio/video distinction for probe packets. > > > > Allows detecting large-enough audio packets as part of a probe, > > speculative fix for a rampup-time regression in M50. These packets are > > accounted on the send side when probing. > > > > BUG=webrtc:5985 > > R=mflodman@webrtc.org, philipel@webrtc.org > > > > Committed: https://crrev.com/a7d88d38448f6a5677a017562765ab505b89d468 > > Cr-Commit-Position: refs/heads/master@{#13210} > > TBR=mflodman@webrtc.org,philipel@webrtc.org,pbos@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5985 > > Committed: https://crrev.com/17bde8c96ee8b5a7e496a7dc98828b84f9756925 > Cr-Commit-Position: refs/heads/master@{#13221} TBR=mflodman@webrtc.org,philipel@webrtc.org,honghaiz@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5985 Review-Url: https://codereview.webrtc.org/2085653002 Cr-Commit-Position: refs/heads/master@{#13223}
2016-06-20 11:53:02 -07:00
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