blob: 4f8940999526b8c73eb717bacc438f15e4eee062 [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 <deque>
#include <functional>
#include <memory>
#include <vector>
#include "api/transport/network_control.h"
#include "api/transport/webrtc_key_value_config.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/packet_arrival_map.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
class Clock;
namespace rtcp {
class TransportFeedback;
// Class used when send-side BWE is enabled: This proxy is instantiated on the
// receive side. It buffers a number of receive timestamps and then sends
// transport feedback messages back too the send side.
class RemoteEstimatorProxy : public RemoteBitrateEstimator {
// Used for sending transport feedback messages when send side
// BWE is used.
using TransportFeedbackSender = std::function<void(
std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets)>;
RemoteEstimatorProxy(Clock* clock,
TransportFeedbackSender feedback_sender,
const WebRtcKeyValueConfig* key_value_config,
NetworkStateEstimator* network_state_estimator);
~RemoteEstimatorProxy() override;
void IncomingPacket(int64_t arrival_time_ms,
size_t payload_size,
const RTPHeader& header) override;
void RemoveStream(uint32_t ssrc) override {}
bool LatestEstimate(std::vector<unsigned int>* ssrcs,
unsigned int* bitrate_bps) const override;
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override {}
void SetMinBitrate(int min_bitrate_bps) override {}
int64_t TimeUntilNextProcess() override;
void Process() override;
void OnBitrateChanged(int bitrate);
void SetSendPeriodicFeedback(bool send_periodic_feedback);
struct TransportWideFeedbackConfig {
FieldTrialParameter<TimeDelta> back_window{"wind", TimeDelta::Millis(500)};
FieldTrialParameter<TimeDelta> min_interval{"min", TimeDelta::Millis(50)};
FieldTrialParameter<TimeDelta> max_interval{"max", TimeDelta::Millis(250)};
FieldTrialParameter<TimeDelta> default_interval{"def",
FieldTrialParameter<double> bandwidth_fraction{"frac", 0.05};
explicit TransportWideFeedbackConfig(
const WebRtcKeyValueConfig* key_value_config) {
ParseFieldTrial({&back_window, &min_interval, &max_interval,
&default_interval, &bandwidth_fraction},
void MaybeCullOldPackets(int64_t sequence_number, int64_t arrival_time_ms)
void SendPeriodicFeedbacks() RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
void SendFeedbackOnRequest(int64_t sequence_number,
const FeedbackRequest& feedback_request)
// Returns a Transport Feedback packet with information about as many packets
// that has been received between [`begin_sequence_number_incl`,
// `end_sequence_number_excl`) that can fit in it. If `is_periodic_update`,
// this represents sending a periodic feedback message, which will make it
// update the `periodic_window_start_seq_` variable with the first packet that
// was not included in the feedback packet, so that the next update can
// continue from that sequence number.
// If no incoming packets were added, nullptr is returned.
// `include_timestamps` decide if the returned TransportFeedback should
// include timestamps.
std::unique_ptr<rtcp::TransportFeedback> MaybeBuildFeedbackPacket(
bool include_timestamps,
int64_t begin_sequence_number_inclusive,
int64_t end_sequence_number_exclusive,
bool is_periodic_update) RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
Clock* const clock_;
const TransportFeedbackSender feedback_sender_;
const TransportWideFeedbackConfig send_config_;
int64_t last_process_time_ms_;
Mutex lock_;
// |network_state_estimator_| may be null.
NetworkStateEstimator* const network_state_estimator_
uint32_t media_ssrc_ RTC_GUARDED_BY(&lock_);
uint8_t feedback_packet_count_ RTC_GUARDED_BY(&lock_);
SeqNumUnwrapper<uint16_t> unwrapper_ RTC_GUARDED_BY(&lock_);
// The next sequence number that should be the start sequence number during
// periodic reporting. Will be absl::nullopt before the first seen packet.
absl::optional<int64_t> periodic_window_start_seq_ RTC_GUARDED_BY(&lock_);
// Packet arrival times, by sequence number.
PacketArrivalTimeMap packet_arrival_times_ RTC_GUARDED_BY(&lock_);
int64_t send_interval_ms_ RTC_GUARDED_BY(&lock_);
bool send_periodic_feedback_ RTC_GUARDED_BY(&lock_);
// Unwraps absolute send times.
uint32_t previous_abs_send_time_ RTC_GUARDED_BY(&lock_);
Timestamp abs_send_timestamp_ RTC_GUARDED_BY(&lock_);
} // namespace webrtc