| /* |
| * 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. |
| */ |
| |
| #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_ |
| #define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_ |
| |
| #include <deque> |
| #include <functional> |
| #include <memory> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/field_trials_view.h" |
| #include "api/rtp_headers.h" |
| #include "api/transport/network_control.h" |
| #include "api/units/data_size.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "modules/remote_bitrate_estimator/packet_arrival_map.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "rtc_base/numerics/sequence_number_unwrapper.h" |
| #include "rtc_base/synchronization/mutex.h" |
| |
| namespace webrtc { |
| |
| // 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: |
| // 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(TransportFeedbackSender feedback_sender, |
| NetworkStateEstimator* network_state_estimator); |
| ~RemoteEstimatorProxy(); |
| |
| void IncomingPacket(const RtpPacketReceived& packet); |
| |
| // TODO(perkj, bugs.webrtc.org/14859): Remove all usage. This method is |
| // currently not used by PeerConnections. |
| void IncomingPacket(int64_t arrival_time_ms, |
| size_t payload_size, |
| const RTPHeader& header); |
| |
| // Sends periodic feedback if it is time to send it. |
| // Returns time until next call to Process should be made. |
| TimeDelta Process(Timestamp now); |
| |
| void OnBitrateChanged(int bitrate); |
| void SetTransportOverhead(DataSize overhead_per_packet); |
| |
| private: |
| struct Packet { |
| Timestamp arrival_time; |
| DataSize size; |
| uint32_t ssrc; |
| absl::optional<uint32_t> absolute_send_time_24bits; |
| absl::optional<uint16_t> transport_sequence_number; |
| absl::optional<FeedbackRequest> feedback_request; |
| }; |
| void IncomingPacket(Packet packet) RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_); |
| |
| void MaybeCullOldPackets(int64_t sequence_number, Timestamp arrival_time) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_); |
| void SendPeriodicFeedbacks() RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_); |
| void SendFeedbackOnRequest(int64_t sequence_number, |
| const FeedbackRequest& feedback_request) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_); |
| |
| // 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_); |
| |
| const TransportFeedbackSender feedback_sender_; |
| Timestamp last_process_time_; |
| |
| Mutex lock_; |
| // `network_state_estimator_` may be null. |
| NetworkStateEstimator* const network_state_estimator_ |
| RTC_PT_GUARDED_BY(&lock_); |
| 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_); |
| DataSize packet_overhead_ 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_); |
| |
| TimeDelta send_interval_ 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 |
| |
| #endif // MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_ |