Repo created

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/OWNERS

@@ -0,0 +1,6 @@
+stefan@webrtc.org
+henrik.lundin@webrtc.org
+mflodman@webrtc.org
+asapersson@webrtc.org
+danilchap@webrtc.org
+sprang@webrtc.org

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
+#define MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/ulpfec_receiver.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+
+class FlexfecReceiver {
+ public:
+  FlexfecReceiver(uint32_t ssrc,
+                  uint32_t protected_media_ssrc,
+                  RecoveredPacketReceiver* recovered_packet_receiver);
+  FlexfecReceiver(Clock* clock,
+                  uint32_t ssrc,
+                  uint32_t protected_media_ssrc,
+                  RecoveredPacketReceiver* recovered_packet_receiver);
+  ~FlexfecReceiver();
+
+  // Inserts a received packet (can be either media or FlexFEC) into the
+  // internal buffer, and sends the received packets to the erasure code.
+  // All newly recovered packets are sent back through the callback.
+  void OnRtpPacket(const RtpPacketReceived& packet);
+
+  // Returns a counter describing the added and recovered packets.
+  FecPacketCounter GetPacketCounter() const;
+
+  // Protected to aid testing.
+ protected:
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> AddReceivedPacket(
+      const RtpPacketReceived& packet);
+  void ProcessReceivedPacket(
+      const ForwardErrorCorrection::ReceivedPacket& received_packet);
+
+ private:
+  // Config.
+  const uint32_t ssrc_;
+  const uint32_t protected_media_ssrc_;
+
+  // Erasure code interfacing and callback.
+  std::unique_ptr<ForwardErrorCorrection> erasure_code_
+      RTC_GUARDED_BY(sequence_checker_);
+  ForwardErrorCorrection::RecoveredPacketList recovered_packets_
+      RTC_GUARDED_BY(sequence_checker_);
+  RecoveredPacketReceiver* const recovered_packet_receiver_;
+
+  // Logging and stats.
+  Clock* const clock_;
+  Timestamp last_recovered_packet_ RTC_GUARDED_BY(sequence_checker_) =
+      Timestamp::MinusInfinity();
+  FecPacketCounter packet_counter_ RTC_GUARDED_BY(sequence_checker_);
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/flexfec_sender.h

@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_
+#define MODULES_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
+#include "modules/rtp_rtcp/source/ulpfec_generator.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
+#include "rtc_base/bitrate_tracker.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class Clock;
+class RtpPacketToSend;
+
+// Note that this class is not thread safe, and thus requires external
+// synchronization. Currently, this is done using the lock in PayloadRouter.
+
+class FlexfecSender : public VideoFecGenerator {
+ public:
+  FlexfecSender(int payload_type,
+                uint32_t ssrc,
+                uint32_t protected_media_ssrc,
+                absl::string_view mid,
+                const std::vector<RtpExtension>& rtp_header_extensions,
+                rtc::ArrayView<const RtpExtensionSize> extension_sizes,
+                const RtpState* rtp_state,
+                Clock* clock);
+  ~FlexfecSender();
+
+  FecType GetFecType() const override {
+    return VideoFecGenerator::FecType::kFlexFec;
+  }
+  absl::optional<uint32_t> FecSsrc() override { return ssrc_; }
+
+  // Sets the FEC rate, max frames sent before FEC packets are sent,
+  // and what type of generator matrices are used.
+  void SetProtectionParameters(const FecProtectionParams& delta_params,
+                               const FecProtectionParams& key_params) override;
+
+  // Adds a media packet to the internal buffer. When enough media packets
+  // have been added, the FEC packets are generated and stored internally.
+  // These FEC packets are then obtained by calling GetFecPackets().
+  void AddPacketAndGenerateFec(const RtpPacketToSend& packet) override;
+
+  // Returns generated FlexFEC packets.
+  std::vector<std::unique_ptr<RtpPacketToSend>> GetFecPackets() override;
+
+  // Returns the overhead, per packet, for FlexFEC.
+  size_t MaxPacketOverhead() const override;
+
+  DataRate CurrentFecRate() const override;
+
+  // Only called on the VideoSendStream queue, after operation has shut down.
+  absl::optional<RtpState> GetRtpState() override;
+
+ private:
+  // Utility.
+  Clock* const clock_;
+  Random random_;
+  Timestamp last_generated_packet_ = Timestamp::MinusInfinity();
+
+  // Config.
+  const int payload_type_;
+  const uint32_t timestamp_offset_;
+  const uint32_t ssrc_;
+  const uint32_t protected_media_ssrc_;
+  // MID value to send in the MID header extension.
+  const std::string mid_;
+  // Sequence number of next packet to generate.
+  uint16_t seq_num_;
+
+  // Implementation.
+  UlpfecGenerator ulpfec_generator_;
+  const RtpHeaderExtensionMap rtp_header_extension_map_;
+  const size_t header_extensions_size_;
+
+  mutable Mutex mutex_;
+  BitrateTracker fec_bitrate_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/receive_statistics.h

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+/*
+ *  Copyright (c) 2013 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_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+namespace webrtc {
+
+class Clock;
+
+class ReceiveStatisticsProvider {
+ public:
+  virtual ~ReceiveStatisticsProvider() = default;
+  // Collects receive statistic in a form of rtcp report blocks.
+  // Returns at most `max_blocks` report blocks.
+  virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
+      size_t max_blocks) = 0;
+};
+
+class StreamStatistician {
+ public:
+  virtual ~StreamStatistician();
+
+  virtual RtpReceiveStats GetStats() const = 0;
+
+  // Returns average over the stream life time.
+  virtual absl::optional<int> GetFractionLostInPercent() const = 0;
+
+  // TODO(bugs.webrtc.org/10679): Delete, migrate users to the above GetStats
+  // method (and extend RtpReceiveStats if needed).
+  // Gets receive stream data counters.
+  virtual StreamDataCounters GetReceiveStreamDataCounters() const = 0;
+
+  virtual uint32_t BitrateReceived() const = 0;
+};
+
+class ReceiveStatistics : public ReceiveStatisticsProvider,
+                          public RtpPacketSinkInterface {
+ public:
+  ~ReceiveStatistics() override = default;
+
+  // Returns a thread-safe instance of ReceiveStatistics.
+  // https://chromium.googlesource.com/chromium/src/+/lkgr/docs/threading_and_tasks.md#threading-lexicon
+  static std::unique_ptr<ReceiveStatistics> Create(Clock* clock);
+  // Returns a thread-compatible instance of ReceiveStatistics.
+  static std::unique_ptr<ReceiveStatistics> CreateThreadCompatible(
+      Clock* clock);
+
+  // Returns a pointer to the statistician of an ssrc.
+  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
+
+  // TODO(bugs.webrtc.org/10669): Deprecated, delete as soon as downstream
+  // projects are updated. This method sets the max reordering threshold of all
+  // current and future streams.
+  virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
+
+  // Sets the max reordering threshold in number of packets.
+  virtual void SetMaxReorderingThreshold(uint32_t ssrc,
+                                         int max_reordering_threshold) = 0;
+  // Detect retransmissions, enabling updates of the retransmitted counters. The
+  // default is false.
+  virtual void EnableRetransmitDetection(uint32_t ssrc, bool enable) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/recovered_packet_receiver.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2022 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_RTP_RTCP_INCLUDE_RECOVERED_PACKET_RECEIVER_H_
+#define MODULES_RTP_RTCP_INCLUDE_RECOVERED_PACKET_RECEIVER_H_
+
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Callback interface for packets recovered by FlexFEC or ULPFEC. In
+// the FlexFEC case, the implementation should be able to demultiplex
+// the recovered RTP packets based on SSRC.
+class RecoveredPacketReceiver {
+ public:
+  virtual void OnRecoveredPacket(const RtpPacketReceived& packet) = 0;
+
+ protected:
+  virtual ~RecoveredPacketReceiver() = default;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RECOVERED_PACKET_RECEIVER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2014 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_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/numerics/moving_percentile_filter.h"
+#include "system_wrappers/include/rtp_to_ntp_estimator.h"
+
+namespace webrtc {
+
+class Clock;
+
+// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP
+// time in local timebase.
+// Note that it needs to be trained with at least 2 RTCP SR (by calling
+// `UpdateRtcpTimestamp`) before it can be used.
+class RemoteNtpTimeEstimator {
+ public:
+  explicit RemoteNtpTimeEstimator(Clock* clock);
+  RemoteNtpTimeEstimator(const RemoteNtpTimeEstimator&) = delete;
+  RemoteNtpTimeEstimator& operator=(const RemoteNtpTimeEstimator&) = delete;
+  ~RemoteNtpTimeEstimator() = default;
+
+  // Updates the estimator with round trip time `rtt` and
+  // new NTP time <-> RTP timestamp mapping from an RTCP sender report.
+  bool UpdateRtcpTimestamp(TimeDelta rtt,
+                           NtpTime sender_send_time,
+                           uint32_t rtp_timestamp);
+
+  // Estimates the NTP timestamp in local timebase from `rtp_timestamp`.
+  // Returns the NTP timestamp in ms when success. -1 if failed.
+  int64_t Estimate(uint32_t rtp_timestamp) {
+    NtpTime ntp_time = EstimateNtp(rtp_timestamp);
+    if (!ntp_time.Valid()) {
+      return -1;
+    }
+    return ntp_time.ToMs();
+  }
+
+  // Estimates the NTP timestamp in local timebase from `rtp_timestamp`.
+  // Returns invalid NtpTime (i.e. NtpTime(0)) on failure.
+  NtpTime EstimateNtp(uint32_t rtp_timestamp);
+
+  // Estimates the offset between the remote clock and the
+  // local one. This is equal to local NTP clock - remote NTP clock.
+  // The offset is returned in ntp time resolution, i.e. 1/2^32 sec ~= 0.2 ns.
+  // Returns nullopt on failure.
+  absl::optional<int64_t> EstimateRemoteToLocalClockOffset();
+
+ private:
+  Clock* clock_;
+  // Offset is measured with the same precision as NtpTime: in 1/2^32 seconds ~=
+  // 0.2 ns.
+  MovingMedianFilter<int64_t> ntp_clocks_offset_estimator_;
+  RtpToNtpEstimator rtp_to_ntp_;
+  Timestamp last_timing_log_ = Timestamp::MinusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/report_block_data.cc

@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2019 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 "modules/rtp_rtcp/include/report_block_data.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+TimeDelta ReportBlockData::jitter(int rtp_clock_rate_hz) const {
+  RTC_DCHECK_GT(rtp_clock_rate_hz, 0);
+  // Conversion to TimeDelta and division are swapped to avoid conversion
+  // to/from floating point types.
+  return TimeDelta::Seconds(jitter()) / rtp_clock_rate_hz;
+}
+
+void ReportBlockData::SetReportBlock(uint32_t sender_ssrc,
+                                     const rtcp::ReportBlock& report_block,
+                                     Timestamp report_block_timestamp_utc) {
+  sender_ssrc_ = sender_ssrc;
+  source_ssrc_ = report_block.source_ssrc();
+  fraction_lost_raw_ = report_block.fraction_lost();
+  cumulative_lost_ = report_block.cumulative_lost();
+  extended_highest_sequence_number_ = report_block.extended_high_seq_num();
+  jitter_ = report_block.jitter();
+  report_block_timestamp_utc_ = report_block_timestamp_utc;
+}
+
+void ReportBlockData::AddRoundTripTimeSample(TimeDelta rtt) {
+  last_rtt_ = rtt;
+  sum_rtt_ += rtt;
+  ++num_rtts_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/report_block_data.h

@@ -0,0 +1,123 @@
+/*
+ *  Copyright 2019 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_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
+#define MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+namespace webrtc {
+
+// Represents fields and derived information received in RTCP report block
+// attached to RTCP sender report or RTCP receiver report, as described in
+// https://www.rfc-editor.org/rfc/rfc3550#section-6.4.1
+class ReportBlockData {
+ public:
+  ReportBlockData() = default;
+
+  ReportBlockData(const ReportBlockData&) = default;
+  ReportBlockData& operator=(const ReportBlockData&) = default;
+
+  // The SSRC identifier for the originator of this report block,
+  // i.e. remote receiver of the RTP stream.
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+
+  // The SSRC identifier of the source to which the information in this
+  // reception report block pertains, i.e. local sender of the RTP stream.
+  uint32_t source_ssrc() const { return source_ssrc_; }
+
+  // The fraction of RTP data packets from 'source_ssrc()' lost since the
+  // previous report block was sent.
+  // Fraction loss in range [0.0, 1.0].
+  float fraction_lost() const { return fraction_lost_raw() / 256.0; }
+
+  // Fraction loss as was written in the raw packet: range is [0, 255] where 0
+  // represents no loss, and 255 represents 99.6% loss (255/256 * 100%).
+  uint8_t fraction_lost_raw() const { return fraction_lost_raw_; }
+
+  // The total number of RTP data packets from 'source_ssrc()' that have been
+  // lost since the beginning of reception.  This number is defined to be the
+  // number of packets expected less the number of packets actually received,
+  // where the number of packets received includes any which are late or
+  // duplicates. Thus, packets that arrive late are not counted as lost, and the
+  // loss may be negative if there are duplicates.
+  int cumulative_lost() const { return cumulative_lost_; }
+
+  // The low 16 bits contain the highest sequence number received in an RTP data
+  // packet from 'source_ssrc()', and the most significant 16 bits extend that
+  // sequence number with the corresponding count of sequence number cycles.
+  uint32_t extended_highest_sequence_number() const {
+    return extended_highest_sequence_number_;
+  }
+
+  // An estimate of the statistical variance of the RTP data packet interarrival
+  // time, measured in RTP timestamp units. The interarrival jitter J is defined
+  // to be the mean deviation (smoothed absolute value) of the difference D in
+  // packet spacing at the receiver compared to the sender for a pair of
+  // packets.
+  uint32_t jitter() const { return jitter_; }
+
+  // Jitter converted to common time units.
+  TimeDelta jitter(int rtp_clock_rate_hz) const;
+
+  // Time in utc epoch (Jan 1st, 1970) the report block was received.
+  Timestamp report_block_timestamp_utc() const {
+    return report_block_timestamp_utc_;
+  }
+
+  // Round Trip Time measurments for given (sender_ssrc, source_ssrc) pair.
+  // Min, max, sum, number of measurements are since beginning of the call.
+  TimeDelta last_rtt() const { return last_rtt_; }
+  TimeDelta sum_rtts() const { return sum_rtt_; }
+  size_t num_rtts() const { return num_rtts_; }
+  bool has_rtt() const { return num_rtts_ != 0; }
+
+  void set_sender_ssrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  void set_source_ssrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
+  void set_fraction_lost_raw(uint8_t lost) { fraction_lost_raw_ = lost; }
+  void set_cumulative_lost(int lost) { cumulative_lost_ = lost; }
+  void set_extended_highest_sequence_number(uint32_t sn) {
+    extended_highest_sequence_number_ = sn;
+  }
+  void set_jitter(uint32_t jitter) { jitter_ = jitter; }
+  void set_report_block_timestamp_utc(Timestamp arrival_time) {
+    report_block_timestamp_utc_ = arrival_time;
+  }
+
+  void SetReportBlock(uint32_t sender_ssrc,
+                      const rtcp::ReportBlock& report_block,
+                      Timestamp report_block_timestamp_utc);
+  void AddRoundTripTimeSample(TimeDelta rtt);
+
+ private:
+  uint32_t sender_ssrc_ = 0;
+  uint32_t source_ssrc_ = 0;
+  uint8_t fraction_lost_raw_ = 0;
+  int32_t cumulative_lost_ = 0;
+  uint32_t extended_highest_sequence_number_ = 0;
+  uint32_t jitter_ = 0;
+  Timestamp report_block_timestamp_utc_ = Timestamp::Zero();
+  TimeDelta last_rtt_ = TimeDelta::Zero();
+  TimeDelta sum_rtt_ = TimeDelta::Zero();
+  size_t num_rtts_ = 0;
+};
+
+class ReportBlockDataObserver {
+ public:
+  virtual ~ReportBlockDataObserver() = default;
+
+  virtual void OnReportBlockDataUpdated(ReportBlockData report_block_data) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtcp_statistics.h

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2018 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_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
+
+#include <stdint.h>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+// Statistics for RTCP packet types.
+struct RtcpPacketTypeCounter {
+  RtcpPacketTypeCounter()
+      : nack_packets(0),
+        fir_packets(0),
+        pli_packets(0),
+        nack_requests(0),
+        unique_nack_requests(0) {}
+
+  void Add(const RtcpPacketTypeCounter& other) {
+    nack_packets += other.nack_packets;
+    fir_packets += other.fir_packets;
+    pli_packets += other.pli_packets;
+    nack_requests += other.nack_requests;
+    unique_nack_requests += other.unique_nack_requests;
+  }
+
+  void Subtract(const RtcpPacketTypeCounter& other) {
+    nack_packets -= other.nack_packets;
+    fir_packets -= other.fir_packets;
+    pli_packets -= other.pli_packets;
+    nack_requests -= other.nack_requests;
+    unique_nack_requests -= other.unique_nack_requests;
+  }
+
+  int UniqueNackRequestsInPercent() const {
+    if (nack_requests == 0) {
+      return 0;
+    }
+    return static_cast<int>((unique_nack_requests * 100.0f / nack_requests) +
+                            0.5f);
+  }
+
+  uint32_t nack_packets;          // Number of RTCP NACK packets.
+  uint32_t fir_packets;           // Number of RTCP FIR packets.
+  uint32_t pli_packets;           // Number of RTCP PLI packets.
+  uint32_t nack_requests;         // Number of NACKed RTP packets.
+  uint32_t unique_nack_requests;  // Number of unique NACKed RTP packets.
+};
+
+class RtcpPacketTypeCounterObserver {
+ public:
+  virtual ~RtcpPacketTypeCounterObserver() {}
+  virtual void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) = 0;
+};
+
+// Invoked for each cname passed in RTCP SDES blocks.
+class RtcpCnameCallback {
+ public:
+  virtual ~RtcpCnameCallback() = default;
+
+  virtual void OnCname(uint32_t ssrc, absl::string_view cname) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtp_cvo.h

@@ -0,0 +1,56 @@
+/*
+ *  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_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+
+#include "api/video/video_rotation.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/
+// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is
+// the clockwise angle the frames must be rotated in order to display the frames
+// correctly if the display is rotated in its natural orientation.
+inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) {
+  switch (rotation) {
+    case kVideoRotation_0:
+      return 0;
+    case kVideoRotation_90:
+      return 1;
+    case kVideoRotation_180:
+      return 2;
+    case kVideoRotation_270:
+      return 3;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t cvo_byte) {
+  // CVO byte: |0 0 0 0 C F R R|.
+  const uint8_t rotation_bits = cvo_byte & 0x3;
+  switch (rotation_bits) {
+    case 0:
+      return kVideoRotation_0;
+    case 1:
+      return kVideoRotation_90;
+    case 2:
+      return kVideoRotation_180;
+    case 3:
+      return kVideoRotation_270;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kVideoRotation_0;
+  }
+}
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class RtpHeaderExtensionMap {
+ public:
+  static constexpr RTPExtensionType kInvalidType = kRtpExtensionNone;
+  static constexpr int kInvalidId = 0;
+
+  RtpHeaderExtensionMap();
+  explicit RtpHeaderExtensionMap(bool extmap_allow_mixed);
+  explicit RtpHeaderExtensionMap(rtc::ArrayView<const RtpExtension> extensions);
+
+  void Reset(rtc::ArrayView<const RtpExtension> extensions);
+
+  template <typename Extension>
+  bool Register(int id) {
+    return Register(id, Extension::kId, Extension::Uri());
+  }
+  bool RegisterByType(int id, RTPExtensionType type);
+  bool RegisterByUri(int id, absl::string_view uri);
+
+  bool IsRegistered(RTPExtensionType type) const {
+    return GetId(type) != kInvalidId;
+  }
+  // Return kInvalidType if not found.
+  RTPExtensionType GetType(int id) const;
+  // Return kInvalidId if not found.
+  uint8_t GetId(RTPExtensionType type) const {
+    RTC_DCHECK_GT(type, kRtpExtensionNone);
+    RTC_DCHECK_LT(type, kRtpExtensionNumberOfExtensions);
+    return ids_[type];
+  }
+
+  void Deregister(absl::string_view uri);
+
+  // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
+  // Set to true if it's allowed to mix one- and two-byte RTP header extensions
+  // in the same stream.
+  bool ExtmapAllowMixed() const { return extmap_allow_mixed_; }
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) {
+    extmap_allow_mixed_ = extmap_allow_mixed;
+  }
+
+ private:
+  bool Register(int id, RTPExtensionType type, absl::string_view uri);
+
+  uint8_t ids_[kRtpExtensionNumberOfExtensions];
+  bool extmap_allow_mixed_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtp_packet_sender.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2019 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_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+class RtpPacketSender {
+ public:
+  virtual ~RtpPacketSender() = default;
+
+  // Insert a set of packets into queue, for eventual transmission. Based on the
+  // type of packets, they will be prioritized and scheduled relative to other
+  // packets and the current target send rate.
+  virtual void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) = 0;
+
+  // Clear any pending packets with the given SSRC from the queue.
+  // TODO(crbug.com/1395081): Make pure virtual when downstream code has been
+  // updated.
+  virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+
+#include <memory>
+
+#include "absl/base/attributes.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+
+namespace webrtc {
+
+class ABSL_DEPRECATED("") RtpRtcp : public RtpRtcpInterface {
+ public:
+  // Instantiates a deprecated version of the RtpRtcp module.
+  static std::unique_ptr<RtpRtcp> ABSL_DEPRECATED("")
+      Create(const Configuration& configuration) {
+    return DEPRECATED_Create(configuration);
+  }
+
+  static std::unique_ptr<RtpRtcp> DEPRECATED_Create(
+      const Configuration& configuration);
+
+  // Process any pending tasks such as timeouts.
+  virtual void Process() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2017 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 "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+#include <string.h>
+
+#include <type_traits>
+
+#include "absl/algorithm/container.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+namespace {
+constexpr size_t kMidRsidMaxSize = 16;
+
+// Check if passed character is a "token-char" from RFC 4566.
+// https://datatracker.ietf.org/doc/html/rfc4566#section-9
+//    token-char =          %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
+//                         / %x41-5A / %x5E-7E
+bool IsTokenChar(char ch) {
+  return ch == 0x21 || (ch >= 0x23 && ch <= 0x27) || ch == 0x2a || ch == 0x2b ||
+         ch == 0x2d || ch == 0x2e || (ch >= 0x30 && ch <= 0x39) ||
+         (ch >= 0x41 && ch <= 0x5a) || (ch >= 0x5e && ch <= 0x7e);
+}
+}  // namespace
+
+bool IsLegalMidName(absl::string_view name) {
+  return (name.size() <= kMidRsidMaxSize && !name.empty() &&
+          absl::c_all_of(name, IsTokenChar));
+}
+
+bool IsLegalRsidName(absl::string_view name) {
+  return (name.size() <= kMidRsidMaxSize && !name.empty() &&
+          absl::c_all_of(name, isalnum));
+}
+
+StreamDataCounters::StreamDataCounters() = default;
+
+RtpPacketCounter::RtpPacketCounter(const RtpPacket& packet)
+    : header_bytes(packet.headers_size()),
+      payload_bytes(packet.payload_size()),
+      padding_bytes(packet.padding_size()),
+      packets(1) {}
+
+RtpPacketCounter::RtpPacketCounter(const RtpPacketToSend& packet_to_send)
+    : RtpPacketCounter(static_cast<const RtpPacket&>(packet_to_send)) {
+  total_packet_delay =
+      packet_to_send.time_in_send_queue().value_or(TimeDelta::Zero());
+}
+
+void RtpPacketCounter::AddPacket(const RtpPacket& packet) {
+  ++packets;
+  header_bytes += packet.headers_size();
+  padding_bytes += packet.padding_size();
+  payload_bytes += packet.payload_size();
+}
+
+void RtpPacketCounter::AddPacket(const RtpPacketToSend& packet_to_send) {
+  AddPacket(static_cast<const RtpPacket&>(packet_to_send));
+  total_packet_delay +=
+      packet_to_send.time_in_send_queue().value_or(TimeDelta::Zero());
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -0,0 +1,420 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "absl/types/variant.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/rtp_headers.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+#include "system_wrappers/include/clock.h"
+
+#define RTCP_CNAME_SIZE 256  // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500  // we assume ethernet
+
+namespace webrtc {
+class RtpPacket;
+class RtpPacketToSend;
+namespace rtcp {
+class TransportFeedback;
+}
+
+const int kVideoPayloadTypeFrequency = 90000;
+
+// TODO(bugs.webrtc.org/6458): Remove this when all the depending projects are
+// updated to correctly set rtp rate for RtcpSender.
+const int kBogusRtpRateForAudioRtcp = 8000;
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+bool IsLegalMidName(absl::string_view name);
+bool IsLegalRsidName(absl::string_view name);
+
+// This enum must not have any gaps, i.e., all integers between
+// kRtpExtensionNone and kRtpExtensionNumberOfExtensions must be valid enum
+// entries.
+enum RTPExtensionType : int {
+  kRtpExtensionNone,
+  kRtpExtensionTransmissionTimeOffset,
+  kRtpExtensionAudioLevel,
+  kRtpExtensionCsrcAudioLevel,
+  kRtpExtensionInbandComfortNoise,
+  kRtpExtensionAbsoluteSendTime,
+  kRtpExtensionAbsoluteCaptureTime,
+  kRtpExtensionVideoRotation,
+  kRtpExtensionTransportSequenceNumber,
+  kRtpExtensionTransportSequenceNumber02,
+  kRtpExtensionPlayoutDelay,
+  kRtpExtensionVideoContentType,
+  kRtpExtensionVideoLayersAllocation,
+  kRtpExtensionVideoTiming,
+  kRtpExtensionRtpStreamId,
+  kRtpExtensionRepairedRtpStreamId,
+  kRtpExtensionMid,
+  kRtpExtensionGenericFrameDescriptor,
+  kRtpExtensionGenericFrameDescriptor00 [[deprecated]] =
+      kRtpExtensionGenericFrameDescriptor,
+  kRtpExtensionDependencyDescriptor,
+  kRtpExtensionGenericFrameDescriptor02 [[deprecated]] =
+      kRtpExtensionDependencyDescriptor,
+  kRtpExtensionColorSpace,
+  kRtpExtensionVideoFrameTrackingId,
+  kRtpExtensionNumberOfExtensions  // Must be the last entity in the enum.
+};
+
+enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
+
+// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
+enum RTCPPacketType : uint32_t {
+  kRtcpReport = 0x0001,
+  kRtcpSr = 0x0002,
+  kRtcpRr = 0x0004,
+  kRtcpSdes = 0x0008,
+  kRtcpBye = 0x0010,
+  kRtcpPli = 0x0020,
+  kRtcpNack = 0x0040,
+  kRtcpFir = 0x0080,
+  kRtcpTmmbr = 0x0100,
+  kRtcpTmmbn = 0x0200,
+  kRtcpSrReq = 0x0400,
+  kRtcpLossNotification = 0x2000,
+  kRtcpRemb = 0x10000,
+  kRtcpTransmissionTimeOffset = 0x20000,
+  kRtcpXrReceiverReferenceTime = 0x40000,
+  kRtcpXrDlrrReportBlock = 0x80000,
+  kRtcpTransportFeedback = 0x100000,
+  kRtcpXrTargetBitrate = 0x200000
+};
+
+enum class KeyFrameReqMethod : uint8_t {
+  kNone,     // Don't request keyframes.
+  kPliRtcp,  // Request keyframes through Picture Loss Indication.
+  kFirRtcp   // Request keyframes through Full Intra-frame Request.
+};
+
+enum RtxMode {
+  kRtxOff = 0x0,
+  kRtxRetransmitted = 0x1,     // Only send retransmissions over RTX.
+  kRtxRedundantPayloads = 0x2  // Preventively send redundant payloads
+                               // instead of padding.
+};
+
+const size_t kRtxHeaderSize = 2;
+
+struct RtpState {
+  uint16_t sequence_number = 0;
+  uint32_t start_timestamp = 0;
+  uint32_t timestamp = 0;
+  Timestamp capture_time = Timestamp::MinusInfinity();
+  Timestamp last_timestamp_time = Timestamp::MinusInfinity();
+  bool ssrc_has_acked = false;
+};
+
+class RtcpIntraFrameObserver {
+ public:
+  virtual ~RtcpIntraFrameObserver() {}
+
+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
+};
+
+// Observer for incoming LossNotification RTCP messages.
+// See the documentation of LossNotification for details.
+class RtcpLossNotificationObserver {
+ public:
+  virtual ~RtcpLossNotificationObserver() = default;
+
+  virtual void OnReceivedLossNotification(uint32_t ssrc,
+                                          uint16_t seq_num_of_last_decodable,
+                                          uint16_t seq_num_of_last_received,
+                                          bool decodability_flag) = 0;
+};
+
+// Interface to watch incoming rtcp packets related to the link in general.
+// All message handlers have default empty implementation. This way users only
+// need to implement the ones they are interested in.
+// All message handles pass `receive_time` parameter, which is receive time
+// of the rtcp packet that triggered the update.
+class NetworkLinkRtcpObserver {
+ public:
+  virtual ~NetworkLinkRtcpObserver() = default;
+
+  virtual void OnTransportFeedback(Timestamp receive_time,
+                                   const rtcp::TransportFeedback& feedback) {}
+  virtual void OnReceiverEstimatedMaxBitrate(Timestamp receive_time,
+                                             DataRate bitrate) {}
+
+  // Called on an RTCP packet with sender or receiver reports with non zero
+  // report blocks. Report blocks are combined from all reports into one array.
+  virtual void OnReport(Timestamp receive_time,
+                        rtc::ArrayView<const ReportBlockData> report_blocks) {}
+  virtual void OnRttUpdate(Timestamp receive_time, TimeDelta rtt) {}
+};
+
+// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
+static constexpr size_t kNumMediaTypes = 5;
+enum class RtpPacketMediaType : size_t {
+  kAudio,                         // Audio media packets.
+  kVideo,                         // Video media packets.
+  kRetransmission,                // Retransmisions, sent as response to NACK.
+  kForwardErrorCorrection,        // FEC packets.
+  kPadding = kNumMediaTypes - 1,  // RTX or plain padding sent to maintain BWE.
+  // Again, don't forget to update `kNumMediaTypes` if you add another value!
+};
+
+struct RtpPacketSendInfo {
+  uint16_t transport_sequence_number = 0;
+  absl::optional<uint32_t> media_ssrc;
+  uint16_t rtp_sequence_number = 0;  // Only valid if `media_ssrc` is set.
+  uint32_t rtp_timestamp = 0;
+  size_t length = 0;
+  absl::optional<RtpPacketMediaType> packet_type;
+  PacedPacketInfo pacing_info;
+};
+
+class NetworkStateEstimateObserver {
+ public:
+  virtual void OnRemoteNetworkEstimate(NetworkStateEstimate estimate) = 0;
+  virtual ~NetworkStateEstimateObserver() = default;
+};
+
+class TransportFeedbackObserver {
+ public:
+  virtual ~TransportFeedbackObserver() = default;
+
+  virtual void OnAddPacket(const RtpPacketSendInfo& packet_info) = 0;
+};
+
+// Interface for PacketRouter to send rtcp feedback on behalf of
+// congestion controller.
+// TODO(bugs.webrtc.org/8239): Remove and use RtcpTransceiver directly
+// when RtcpTransceiver always present in rtp transport.
+class RtcpFeedbackSenderInterface {
+ public:
+  virtual ~RtcpFeedbackSenderInterface() = default;
+  virtual void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) = 0;
+  virtual void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) = 0;
+  virtual void UnsetRemb() = 0;
+};
+
+class StreamFeedbackObserver {
+ public:
+  struct StreamPacketInfo {
+    bool received;
+
+    // `rtp_sequence_number` and `is_retransmission` are only valid if `ssrc`
+    // is populated.
+    absl::optional<uint32_t> ssrc;
+    uint16_t rtp_sequence_number;
+    bool is_retransmission;
+  };
+  virtual ~StreamFeedbackObserver() = default;
+
+  virtual void OnPacketFeedbackVector(
+      std::vector<StreamPacketInfo> packet_feedback_vector) = 0;
+};
+
+class StreamFeedbackProvider {
+ public:
+  virtual void RegisterStreamFeedbackObserver(
+      std::vector<uint32_t> ssrcs,
+      StreamFeedbackObserver* observer) = 0;
+  virtual void DeRegisterStreamFeedbackObserver(
+      StreamFeedbackObserver* observer) = 0;
+  virtual ~StreamFeedbackProvider() = default;
+};
+
+class RtcpRttStats {
+ public:
+  virtual void OnRttUpdate(int64_t rtt) = 0;
+
+  virtual int64_t LastProcessedRtt() const = 0;
+
+  virtual ~RtcpRttStats() {}
+};
+
+struct RtpPacketCounter {
+  RtpPacketCounter()
+      : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
+
+  explicit RtpPacketCounter(const RtpPacket& packet);
+  explicit RtpPacketCounter(const RtpPacketToSend& packet_to_send);
+
+  void Add(const RtpPacketCounter& other) {
+    header_bytes += other.header_bytes;
+    payload_bytes += other.payload_bytes;
+    padding_bytes += other.padding_bytes;
+    packets += other.packets;
+    total_packet_delay += other.total_packet_delay;
+  }
+
+  bool operator==(const RtpPacketCounter& other) const {
+    return header_bytes == other.header_bytes &&
+           payload_bytes == other.payload_bytes &&
+           padding_bytes == other.padding_bytes && packets == other.packets &&
+           total_packet_delay == other.total_packet_delay;
+  }
+
+  // Not inlined, since use of RtpPacket would result in circular includes.
+  void AddPacket(const RtpPacket& packet);
+  void AddPacket(const RtpPacketToSend& packet_to_send);
+
+  size_t TotalBytes() const {
+    return header_bytes + payload_bytes + padding_bytes;
+  }
+
+  size_t header_bytes;   // Number of bytes used by RTP headers.
+  size_t payload_bytes;  // Payload bytes, excluding RTP headers and padding.
+  size_t padding_bytes;  // Number of padding bytes.
+  size_t packets;        // Number of packets.
+  // The total delay of all `packets`. For RtpPacketToSend packets, this is
+  // `time_in_send_queue()`. For receive packets, this is zero.
+  webrtc::TimeDelta total_packet_delay = webrtc::TimeDelta::Zero();
+};
+
+// Data usage statistics for a (rtp) stream.
+struct StreamDataCounters {
+  StreamDataCounters();
+
+  void Add(const StreamDataCounters& other) {
+    transmitted.Add(other.transmitted);
+    retransmitted.Add(other.retransmitted);
+    fec.Add(other.fec);
+    if (other.first_packet_time < first_packet_time) {
+      // Use oldest time (excluding unsed value represented as plus infinity.
+      first_packet_time = other.first_packet_time;
+    }
+  }
+
+  void MaybeSetFirstPacketTime(Timestamp now) {
+    if (first_packet_time == Timestamp::PlusInfinity()) {
+      first_packet_time = now;
+    }
+  }
+
+  // Return time since first packet is send/received, or zero if such event
+  // haven't happen.
+  TimeDelta TimeSinceFirstPacket(Timestamp now) const {
+    return first_packet_time == Timestamp::PlusInfinity()
+               ? TimeDelta::Zero()
+               : now - first_packet_time;
+  }
+
+  // Returns the number of bytes corresponding to the actual media payload (i.e.
+  // RTP headers, padding, retransmissions and fec packets are excluded).
+  // Note this function does not have meaning for an RTX stream.
+  size_t MediaPayloadBytes() const {
+    return transmitted.payload_bytes - retransmitted.payload_bytes -
+           fec.payload_bytes;
+  }
+
+  // Time when first packet is sent/received.
+  Timestamp first_packet_time = Timestamp::PlusInfinity();
+
+  RtpPacketCounter transmitted;    // Number of transmitted packets/bytes.
+  RtpPacketCounter retransmitted;  // Number of retransmitted packets/bytes.
+  RtpPacketCounter fec;            // Number of redundancy packets/bytes.
+};
+
+class RtpSendRates {
+  template <std::size_t... Is>
+  constexpr std::array<DataRate, sizeof...(Is)> make_zero_array(
+      std::index_sequence<Is...>) {
+    return {{(static_cast<void>(Is), DataRate::Zero())...}};
+  }
+
+ public:
+  RtpSendRates()
+      : send_rates_(
+            make_zero_array(std::make_index_sequence<kNumMediaTypes>())) {}
+  RtpSendRates(const RtpSendRates& rhs) = default;
+  RtpSendRates& operator=(const RtpSendRates&) = default;
+
+  DataRate& operator[](RtpPacketMediaType type) {
+    return send_rates_[static_cast<size_t>(type)];
+  }
+  const DataRate& operator[](RtpPacketMediaType type) const {
+    return send_rates_[static_cast<size_t>(type)];
+  }
+  DataRate Sum() const {
+    return absl::c_accumulate(send_rates_, DataRate::Zero());
+  }
+
+ private:
+  std::array<DataRate, kNumMediaTypes> send_rates_;
+};
+
+// Callback, called whenever byte/packet counts have been updated.
+class StreamDataCountersCallback {
+ public:
+  virtual ~StreamDataCountersCallback() {}
+
+  virtual void DataCountersUpdated(const StreamDataCounters& counters,
+                                   uint32_t ssrc) = 0;
+};
+
+// Information exposed through the GetStats api.
+struct RtpReceiveStats {
+  // `packets_lost` and `jitter` are defined by RFC 3550, and exposed in the
+  // RTCReceivedRtpStreamStats dictionary, see
+  // https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*
+  int32_t packets_lost = 0;
+  // Interarrival jitter in samples.
+  uint32_t jitter = 0;
+  // Interarrival jitter in time.
+  webrtc::TimeDelta interarrival_jitter = webrtc::TimeDelta::Zero();
+
+  // Time of the last packet received in unix epoch,
+  // i.e. Timestamp::Zero() represents 1st Jan 1970 00:00
+  absl::optional<Timestamp> last_packet_received;
+
+  // Counters exposed in RTCInboundRtpStreamStats, see
+  // https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
+  RtpPacketCounter packet_counter;
+};
+
+// Callback, used to notify an observer whenever new rates have been estimated.
+class BitrateStatisticsObserver {
+ public:
+  virtual ~BitrateStatisticsObserver() {}
+
+  virtual void Notify(uint32_t total_bitrate_bps,
+                      uint32_t retransmit_bitrate_bps,
+                      uint32_t ssrc) = 0;
+};
+
+// Callback, used to notify an observer whenever a packet is sent to the
+// transport.
+class SendPacketObserver {
+ public:
+  virtual ~SendPacketObserver() = default;
+  virtual void OnSendPacket(absl::optional<uint16_t> packet_id,
+                            Timestamp capture_time,
+                            uint32_t ssrc) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2023 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_RTP_RTCP_MOCKS_MOCK_NETWORK_LINK_RTCP_OBSERVER_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_NETWORK_LINK_RTCP_OBSERVER_H_
+
+#include "api/array_view.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockNetworkLinkRtcpObserver : public NetworkLinkRtcpObserver {
+ public:
+  MOCK_METHOD(void,
+              OnRttUpdate,
+              (Timestamp receive_time, TimeDelta rtt),
+              (override));
+  MOCK_METHOD(void,
+              OnTransportFeedback,
+              (Timestamp receive_time, const rtcp::TransportFeedback& feedback),
+              (override));
+  MOCK_METHOD(void,
+              OnReceiverEstimatedMaxBitrate,
+              (Timestamp receive_time, DataRate bitrate),
+              (override));
+  MOCK_METHOD(void,
+              OnReport,
+              (Timestamp receive_time,
+               rtc::ArrayView<const ReportBlockData> report_blocks),
+              (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_NETWORK_LINK_RTCP_OBSERVER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_
+
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRecoveredPacketReceiver : public RecoveredPacketReceiver {
+ public:
+  MOCK_METHOD(void,
+              OnRecoveredPacket,
+              (const RtpPacketReceived& packet),
+              (override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h

@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRtcpRttStats : public RtcpRttStats {
+ public:
+  MOCK_METHOD(void, OnRttUpdate, (int64_t rtt), (override));
+  MOCK_METHOD(int64_t, LastProcessedRtt, (), (const, override));
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
+
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRtpRtcpInterface : public RtpRtcpInterface {
+ public:
+  MOCK_METHOD(void,
+              IncomingRtcpPacket,
+              (rtc::ArrayView<const uint8_t> packet),
+              (override));
+  MOCK_METHOD(void, SetRemoteSSRC, (uint32_t ssrc), (override));
+  MOCK_METHOD(void, SetLocalSsrc, (uint32_t ssrc), (override));
+  MOCK_METHOD(void, SetMaxRtpPacketSize, (size_t size), (override));
+  MOCK_METHOD(size_t, MaxRtpPacketSize, (), (const, override));
+  MOCK_METHOD(void,
+              RegisterSendPayloadFrequency,
+              (int payload_type, int frequency),
+              (override));
+  MOCK_METHOD(int32_t,
+              DeRegisterSendPayload,
+              (int8_t payload_type),
+              (override));
+  MOCK_METHOD(void, SetExtmapAllowMixed, (bool extmap_allow_mixed), (override));
+  MOCK_METHOD(void,
+              RegisterRtpHeaderExtension,
+              (absl::string_view uri, int id),
+              (override));
+  MOCK_METHOD(void,
+              DeregisterSendRtpHeaderExtension,
+              (absl::string_view uri),
+              (override));
+  MOCK_METHOD(bool, SupportsPadding, (), (const, override));
+  MOCK_METHOD(bool, SupportsRtxPayloadPadding, (), (const, override));
+  MOCK_METHOD(uint32_t, StartTimestamp, (), (const, override));
+  MOCK_METHOD(void, SetStartTimestamp, (uint32_t timestamp), (override));
+  MOCK_METHOD(uint16_t, SequenceNumber, (), (const, override));
+  MOCK_METHOD(void, SetSequenceNumber, (uint16_t seq), (override));
+  MOCK_METHOD(void, SetRtpState, (const RtpState& rtp_state), (override));
+  MOCK_METHOD(void, SetRtxState, (const RtpState& rtp_state), (override));
+  MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
+  MOCK_METHOD(RtpState, GetRtpState, (), (const, override));
+  MOCK_METHOD(RtpState, GetRtxState, (), (const, override));
+  MOCK_METHOD(uint32_t, SSRC, (), (const, override));
+  MOCK_METHOD(void, SetMid, (absl::string_view mid), (override));
+  MOCK_METHOD(void, SetRtxSendStatus, (int modes), (override));
+  MOCK_METHOD(int, RtxSendStatus, (), (const, override));
+  MOCK_METHOD(absl::optional<uint32_t>, RtxSsrc, (), (const, override));
+  MOCK_METHOD(void, SetRtxSendPayloadType, (int, int), (override));
+  MOCK_METHOD(absl::optional<uint32_t>, FlexfecSsrc, (), (const, override));
+  MOCK_METHOD(int32_t, SetSendingStatus, (bool sending), (override));
+  MOCK_METHOD(bool, Sending, (), (const, override));
+  MOCK_METHOD(void, SetSendingMediaStatus, (bool sending), (override));
+  MOCK_METHOD(bool, SendingMedia, (), (const, override));
+  MOCK_METHOD(bool, IsAudioConfigured, (), (const, override));
+  MOCK_METHOD(void, SetAsPartOfAllocation, (bool), (override));
+  MOCK_METHOD(RtpSendRates, GetSendRates, (), (const, override));
+  MOCK_METHOD(bool,
+              OnSendingRtpFrame,
+              (uint32_t, int64_t, int, bool),
+              (override));
+  MOCK_METHOD(bool,
+              TrySendPacket,
+              (std::unique_ptr<RtpPacketToSend> packet,
+               const PacedPacketInfo& pacing_info),
+              (override));
+  MOCK_METHOD(void, OnBatchComplete, (), (override));
+  MOCK_METHOD(void,
+              SetFecProtectionParams,
+              (const FecProtectionParams& delta_params,
+               const FecProtectionParams& key_params),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFecPackets,
+              (),
+              (override));
+  MOCK_METHOD(void,
+              OnAbortedRetransmissions,
+              (rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(void,
+              OnPacketsAcknowledged,
+              (rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              GeneratePadding,
+              (size_t target_size_bytes),
+              (override));
+  MOCK_METHOD(std::vector<RtpSequenceNumberMap::Info>,
+              GetSentRtpPacketInfos,
+              (rtc::ArrayView<const uint16_t> sequence_numbers),
+              (const, override));
+  MOCK_METHOD(size_t, ExpectedPerPacketOverhead, (), (const, override));
+  MOCK_METHOD(void, OnPacketSendingThreadSwitched, (), (override));
+  MOCK_METHOD(RtcpMode, RTCP, (), (const, override));
+  MOCK_METHOD(void, SetRTCPStatus, (RtcpMode method), (override));
+  MOCK_METHOD(int32_t, SetCNAME, (absl::string_view cname), (override));
+  MOCK_METHOD(absl::optional<TimeDelta>, LastRtt, (), (const, override));
+  MOCK_METHOD(TimeDelta, ExpectedRetransmissionTime, (), (const, override));
+  MOCK_METHOD(int32_t, SendRTCP, (RTCPPacketType packet_type), (override));
+  MOCK_METHOD(void,
+              GetSendStreamDataCounters,
+              (StreamDataCounters*, StreamDataCounters*),
+              (const, override));
+  MOCK_METHOD(std::vector<ReportBlockData>,
+              GetLatestReportBlockData,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<SenderReportStats>,
+              GetSenderReportStats,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<NonSenderRttStats>,
+              GetNonSenderRttStats,
+              (),
+              (const, override));
+  MOCK_METHOD(void,
+              SetRemb,
+              (int64_t bitrate, std::vector<uint32_t> ssrcs),
+              (override));
+  MOCK_METHOD(void, UnsetRemb, (), (override));
+  MOCK_METHOD(int32_t,
+              SendNACK,
+              (const uint16_t* nack_list, uint16_t size),
+              (override));
+  MOCK_METHOD(void,
+              SendNack,
+              (const std::vector<uint16_t>& sequence_numbers),
+              (override));
+  MOCK_METHOD(void,
+              SetStorePacketsStatus,
+              (bool enable, uint16_t number_to_store),
+              (override));
+  MOCK_METHOD(void,
+              SendCombinedRtcpPacket,
+              (std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets),
+              (override));
+  MOCK_METHOD(int32_t,
+              SendLossNotification,
+              (uint16_t last_decoded_seq_num,
+               uint16_t last_received_seq_num,
+               bool decodability_flag,
+               bool buffering_allowed),
+              (override));
+  MOCK_METHOD(void,
+              SetVideoBitrateAllocation,
+              (const VideoBitrateAllocation&),
+              (override));
+  MOCK_METHOD(RTPSender*, RtpSender, (), (override));
+  MOCK_METHOD(const RTPSender*, RtpSender, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc

@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+
+#include <limits>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
+    : clock_(clock) {}
+
+uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
+    uint32_t ssrc,
+    rtc::ArrayView<const uint32_t> csrcs) {
+  if (csrcs.empty()) {
+    return ssrc;
+  }
+
+  return csrcs[0];
+}
+
+absl::optional<AbsoluteCaptureTime>
+AbsoluteCaptureTimeInterpolator::OnReceivePacket(
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz,
+    const absl::optional<AbsoluteCaptureTime>& received_extension) {
+  const Timestamp receive_time = clock_->CurrentTime();
+
+  MutexLock lock(&mutex_);
+
+  if (received_extension == absl::nullopt) {
+    if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
+                                    rtp_clock_frequency_hz)) {
+      last_receive_time_ = Timestamp::MinusInfinity();
+      return absl::nullopt;
+    }
+
+    return AbsoluteCaptureTime{
+        .absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp(
+            rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
+            last_received_extension_.absolute_capture_timestamp),
+        .estimated_capture_clock_offset =
+            last_received_extension_.estimated_capture_clock_offset,
+    };
+  } else {
+    last_source_ = source;
+    last_rtp_timestamp_ = rtp_timestamp;
+    last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz;
+    last_received_extension_ = *received_extension;
+
+    last_receive_time_ = receive_time;
+
+    return received_extension;
+  }
+}
+
+uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz,
+    uint32_t last_rtp_timestamp,
+    uint64_t last_absolute_capture_timestamp) {
+  RTC_DCHECK_GT(rtp_clock_frequency_hz, 0);
+
+  return last_absolute_capture_timestamp +
+         static_cast<int64_t>(uint64_t{rtp_timestamp - last_rtp_timestamp}
+                              << 32) /
+             rtp_clock_frequency_hz;
+}
+
+bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension(
+    Timestamp receive_time,
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz) const {
+  // Shouldn't if the last received extension is not eligible for interpolation,
+  // in particular if we don't have a previously received extension stored.
+  if (receive_time - last_receive_time_ > kInterpolationMaxInterval) {
+    return false;
+  }
+
+  // Shouldn't if the source has changed.
+  if (last_source_ != source) {
+    return false;
+  }
+
+  // Shouldn't if the RTP clock frequency has changed.
+  if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) {
+    return false;
+  }
+
+  // Shouldn't if the RTP clock frequency is invalid.
+  if (rtp_clock_frequency_hz <= 0) {
+    return false;
+  }
+
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h

@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2021 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_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
+#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+//
+// Helper class for interpolating the `AbsoluteCaptureTime` header extension.
+//
+// Supports the "timestamp interpolation" optimization:
+//   A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
+//   timestamp, and RTP timestamp of the most recently received abs-capture-time
+//   packet on each received stream. It can then use that information, in
+//   combination with RTP timestamps of packets without abs-capture-time, to
+//   extrapolate missing capture timestamps.
+//
+// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+//
+class AbsoluteCaptureTimeInterpolator {
+ public:
+  static constexpr TimeDelta kInterpolationMaxInterval = TimeDelta::Seconds(5);
+
+  explicit AbsoluteCaptureTimeInterpolator(Clock* clock);
+
+  // Returns the source (i.e. SSRC or CSRC) of the capture system.
+  static uint32_t GetSource(uint32_t ssrc,
+                            rtc::ArrayView<const uint32_t> csrcs);
+
+  // Returns a received header extension, an interpolated header extension, or
+  // `absl::nullopt` if it's not possible to interpolate a header extension.
+  absl::optional<AbsoluteCaptureTime> OnReceivePacket(
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      int rtp_clock_frequency_hz,
+      const absl::optional<AbsoluteCaptureTime>& received_extension);
+
+ private:
+  friend class AbsoluteCaptureTimeSender;
+
+  static uint64_t InterpolateAbsoluteCaptureTimestamp(
+      uint32_t rtp_timestamp,
+      int rtp_clock_frequency_hz,
+      uint32_t last_rtp_timestamp,
+      uint64_t last_absolute_capture_timestamp);
+
+  bool ShouldInterpolateExtension(Timestamp receive_time,
+                                  uint32_t source,
+                                  uint32_t rtp_timestamp,
+                                  int rtp_clock_frequency_hz) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+
+  Mutex mutex_;
+
+  // Time of the last received header extension eligible for interpolation,
+  // MinusInfinity() if no extension was received, or last received one is
+  // not eligible for interpolation.
+  Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_) =
+      Timestamp::MinusInfinity();
+
+  uint32_t last_source_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
+  int last_rtp_clock_frequency_hz_ RTC_GUARDED_BY(mutex_);
+  AbsoluteCaptureTime last_received_extension_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.cc

@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+
+#include <limits>
+
+#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+static_assert(
+    AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval >=
+        AbsoluteCaptureTimeSender::kInterpolationMaxInterval,
+    "Receivers should be as willing to interpolate timestamps as senders.");
+
+AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
+    : clock_(clock) {}
+
+uint32_t AbsoluteCaptureTimeSender::GetSource(
+    uint32_t ssrc,
+    rtc::ArrayView<const uint32_t> csrcs) {
+  return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
+}
+
+absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    uint32_t rtp_clock_frequency,
+    uint64_t absolute_capture_timestamp,
+    absl::optional<int64_t> estimated_capture_clock_offset) {
+  return OnSendPacket(source, rtp_timestamp, rtp_clock_frequency,
+                      NtpTime(absolute_capture_timestamp),
+                      estimated_capture_clock_offset, /*force=*/false);
+}
+
+absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz,
+    NtpTime absolute_capture_time,
+    absl::optional<int64_t> estimated_capture_clock_offset,
+    bool force) {
+  Timestamp send_time = clock_->CurrentTime();
+  if (!(force || ShouldSendExtension(
+                     send_time, source, rtp_timestamp, rtp_clock_frequency_hz,
+                     absolute_capture_time, estimated_capture_clock_offset))) {
+    return absl::nullopt;
+  }
+
+  last_source_ = source;
+  last_rtp_timestamp_ = rtp_timestamp;
+  last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz;
+  last_absolute_capture_time_ = absolute_capture_time;
+  last_estimated_capture_clock_offset_ = estimated_capture_clock_offset;
+  last_send_time_ = send_time;
+
+  return AbsoluteCaptureTime{
+      .absolute_capture_timestamp = uint64_t{absolute_capture_time},
+      .estimated_capture_clock_offset = estimated_capture_clock_offset,
+  };
+}
+
+bool AbsoluteCaptureTimeSender::ShouldSendExtension(
+    Timestamp send_time,
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz,
+    NtpTime absolute_capture_time,
+    absl::optional<int64_t> estimated_capture_clock_offset) const {
+  // Should if the last sent extension is too old, in particular if we've never
+  // sent anything before.
+  if (send_time - last_send_time_ > kInterpolationMaxInterval) {
+    return true;
+  }
+
+  // Should if the source has changed.
+  if (last_source_ != source) {
+    return true;
+  }
+
+  // Should if the RTP clock frequency has changed.
+  if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) {
+    return true;
+  }
+
+  // Should if the RTP clock frequency is invalid.
+  if (rtp_clock_frequency_hz <= 0) {
+    return true;
+  }
+
+  // Should if the estimated capture clock offset has changed.
+  if (last_estimated_capture_clock_offset_ != estimated_capture_clock_offset) {
+    return true;
+  }
+
+  // Should if interpolation would introduce too much error.
+  const uint64_t interpolated_absolute_capture_timestamp =
+      AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
+          rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
+          uint64_t{last_absolute_capture_time_});
+  const uint64_t absolute_capture_timestamp = uint64_t{absolute_capture_time};
+  const int64_t interpolation_error_ms = UQ32x32ToInt64Ms(std::min(
+      interpolated_absolute_capture_timestamp - absolute_capture_timestamp,
+      absolute_capture_timestamp - interpolated_absolute_capture_timestamp));
+  if (interpolation_error_ms > kInterpolationMaxError.ms()) {
+    return true;
+  }
+
+  return false;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.h

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2019 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_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
+#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+//
+// Helper class for sending the `AbsoluteCaptureTime` header extension.
+//
+// Supports the "timestamp interpolation" optimization:
+//   A sender SHOULD save bandwidth by not sending abs-capture-time with every
+//   RTP packet. It SHOULD still send them at regular intervals (e.g. every
+//   second) to help mitigate the impact of clock drift and packet loss. Mixers
+//   SHOULD always send abs-capture-time with the first RTP packet after
+//   changing capture system.
+//
+//   Timestamp interpolation works fine as long as there’s reasonably low
+//   NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
+//   between its NTP and RTP clock mappings SHOULD send abs-capture-time with
+//   the first RTP packet after such a thing happening.
+//
+// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+//
+class AbsoluteCaptureTimeSender {
+ public:
+  static constexpr TimeDelta kInterpolationMaxInterval = TimeDelta::Seconds(1);
+  static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1);
+
+  explicit AbsoluteCaptureTimeSender(Clock* clock);
+
+  // Returns the source (i.e. SSRC or CSRC) of the capture system.
+  static uint32_t GetSource(uint32_t ssrc,
+                            rtc::ArrayView<const uint32_t> csrcs);
+
+  // Returns value to write into AbsoluteCaptureTime RTP header extension to be
+  // sent, or `absl::nullopt` if the header extension shouldn't be attached to
+  // the outgoing packet.
+  //
+  // - `source` - id of the capture system.
+  // - `rtp_timestamp` - capture time represented as rtp timestamp in the
+  // outgoing packet
+  // - `rtp_clock_frequency_hz` - description of the `rtp_timestamp` units -
+  // `rtp_timetamp` delta of `rtp_clock_freqnecy_hz` represents 1 second.
+  // - `absolute_capture_time` - time when a frame was captured by the capture
+  // system.
+  // - `estimated_capture_clock_offset` - estimated offset between capture
+  // system clock and local `clock` passed as the AbsoluteCaptureTimeSender
+  // construction paramter. Uses the same units as `absolute_capture_time`,
+  // i.e. delta of 2^32 represents 1 second. See AbsoluteCaptureTime type
+  // comments for more details.
+  // - `force` - when set to true, OnSendPacket is forced to return non-nullopt.
+  absl::optional<AbsoluteCaptureTime> OnSendPacket(
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      int rtp_clock_frequency_hz,
+      NtpTime absolute_capture_time,
+      absl::optional<int64_t> estimated_capture_clock_offset,
+      bool force = false);
+
+  // Returns a header extension to be sent, or `absl::nullopt` if the header
+  // extension shouldn't be sent.
+  [[deprecated]] absl::optional<AbsoluteCaptureTime> OnSendPacket(
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      uint64_t absolute_capture_timestamp,
+      absl::optional<int64_t> estimated_capture_clock_offset);
+
+ private:
+  bool ShouldSendExtension(
+      Timestamp send_time,
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      int rtp_clock_frequency_hz,
+      NtpTime absolute_capture_time,
+      absl::optional<int64_t> estimated_capture_clock_offset) const;
+
+  Clock* const clock_;
+
+  Timestamp last_send_time_ = Timestamp::MinusInfinity();
+
+  uint32_t last_source_;
+  uint32_t last_rtp_timestamp_;
+  int last_rtp_clock_frequency_hz_;
+  NtpTime last_absolute_capture_time_;
+  absl::optional<int64_t> last_estimated_capture_clock_offset_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/active_decode_targets_helper.cc

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2020 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 "modules/rtp_rtcp/source/active_decode_targets_helper.h"
+
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+// Returns mask of ids of chains previous frame is part of.
+// Assumes for each chain frames are seen in order and no frame on any chain is
+// missing. That assumptions allows a simple detection when previous frame is
+// part of a chain.
+std::bitset<32> LastSendOnChain(int frame_diff,
+                                rtc::ArrayView<const int> chain_diffs) {
+  std::bitset<32> bitmask = 0;
+  for (size_t i = 0; i < chain_diffs.size(); ++i) {
+    if (frame_diff == chain_diffs[i]) {
+      bitmask.set(i);
+    }
+  }
+  return bitmask;
+}
+
+// Returns bitmask with first `num` bits set to 1.
+std::bitset<32> AllActive(size_t num) {
+  RTC_DCHECK_LE(num, 32);
+  return (~uint32_t{0}) >> (32 - num);
+}
+
+// Returns bitmask of chains that protect at least one active decode target.
+std::bitset<32> ActiveChains(
+    rtc::ArrayView<const int> decode_target_protected_by_chain,
+    int num_chains,
+    std::bitset<32> active_decode_targets) {
+  std::bitset<32> active_chains = 0;
+  for (size_t dt = 0; dt < decode_target_protected_by_chain.size(); ++dt) {
+    if (dt < active_decode_targets.size() && !active_decode_targets[dt]) {
+      continue;
+    }
+    int chain_idx = decode_target_protected_by_chain[dt];
+    RTC_DCHECK_LT(chain_idx, num_chains);
+    active_chains.set(chain_idx);
+  }
+  return active_chains;
+}
+
+}  // namespace
+
+void ActiveDecodeTargetsHelper::OnFrame(
+    rtc::ArrayView<const int> decode_target_protected_by_chain,
+    std::bitset<32> active_decode_targets,
+    bool is_keyframe,
+    int64_t frame_id,
+    rtc::ArrayView<const int> chain_diffs) {
+  const int num_chains = chain_diffs.size();
+  if (num_chains == 0) {
+    // Avoid printing the warning
+    // when already printed the warning for the same active decode targets, or
+    // when active_decode_targets are not changed from it's default value of
+    // all are active, including non-existent decode targets.
+    if (last_active_decode_targets_ != active_decode_targets &&
+        !active_decode_targets.all()) {
+      RTC_LOG(LS_WARNING) << "No chains are configured, but some decode "
+                             "targets might be inactive. Unsupported.";
+    }
+    last_active_decode_targets_ = active_decode_targets;
+    return;
+  }
+  const size_t num_decode_targets = decode_target_protected_by_chain.size();
+  RTC_DCHECK_GT(num_decode_targets, 0);
+  std::bitset<32> all_decode_targets = AllActive(num_decode_targets);
+  // Default value for active_decode_targets is 'all are active', i.e. all bits
+  // are set. Default value is set before number of decode targets is known.
+  // It is up to this helper to make the value cleaner and unset unused bits.
+  active_decode_targets &= all_decode_targets;
+
+  if (is_keyframe) {
+    // Key frame resets the state.
+    last_active_decode_targets_ = all_decode_targets;
+    last_active_chains_ = AllActive(num_chains);
+    unsent_on_chain_.reset();
+  } else {
+    // Update state assuming previous frame was sent.
+    unsent_on_chain_ &=
+        ~LastSendOnChain(frame_id - last_frame_id_, chain_diffs);
+  }
+  // Save for the next call to OnFrame.
+  // Though usually `frame_id == last_frame_id_ + 1`, it might not be so when
+  // frame id space is shared by several simulcast rtp streams.
+  last_frame_id_ = frame_id;
+
+  if (active_decode_targets == last_active_decode_targets_) {
+    return;
+  }
+  last_active_decode_targets_ = active_decode_targets;
+
+  if (active_decode_targets.none()) {
+    RTC_LOG(LS_ERROR) << "It is invalid to produce a frame (" << frame_id
+                      << ") while there are no active decode targets";
+    return;
+  }
+  last_active_chains_ = ActiveChains(decode_target_protected_by_chain,
+                                     num_chains, active_decode_targets);
+  // Frames that are part of inactive chains might not be produced by the
+  // encoder. Thus stop sending `active_decode_target` bitmask when it is sent
+  // on all active chains rather than on all chains.
+  unsent_on_chain_ = last_active_chains_;
+  RTC_DCHECK(!unsent_on_chain_.none());
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/active_decode_targets_helper.h

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2020 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_RTP_RTCP_SOURCE_ACTIVE_DECODE_TARGETS_HELPER_H_
+#define MODULES_RTP_RTCP_SOURCE_ACTIVE_DECODE_TARGETS_HELPER_H_
+
+#include <stdint.h>
+
+#include <bitset>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Helper class that decides when active_decode_target_bitmask should be written
+// into the dependency descriptor rtp header extension.
+// See: https://aomediacodec.github.io/av1-rtp-spec/#a44-switching
+// This class is thread-compatible
+class ActiveDecodeTargetsHelper {
+ public:
+  ActiveDecodeTargetsHelper() = default;
+  ActiveDecodeTargetsHelper(const ActiveDecodeTargetsHelper&) = delete;
+  ActiveDecodeTargetsHelper& operator=(const ActiveDecodeTargetsHelper&) =
+      delete;
+  ~ActiveDecodeTargetsHelper() = default;
+
+  // Decides if active decode target bitmask should be attached to the frame
+  // that is about to be sent.
+  void OnFrame(rtc::ArrayView<const int> decode_target_protected_by_chain,
+               std::bitset<32> active_decode_targets,
+               bool is_keyframe,
+               int64_t frame_id,
+               rtc::ArrayView<const int> chain_diffs);
+
+  // Returns active decode target to attach to the dependency descriptor.
+  absl::optional<uint32_t> ActiveDecodeTargetsBitmask() const {
+    if (unsent_on_chain_.none())
+      return absl::nullopt;
+    return last_active_decode_targets_.to_ulong();
+  }
+
+  std::bitset<32> ActiveChainsBitmask() const { return last_active_chains_; }
+
+ private:
+  // `unsent_on_chain_[i]` indicates last active decode
+  // target bitmask wasn't attached to a packet on the chain with id `i`.
+  std::bitset<32> unsent_on_chain_ = 0;
+  std::bitset<32> last_active_decode_targets_ = 0;
+  std::bitset<32> last_active_chains_ = 0;
+  int64_t last_frame_id_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ACTIVE_DECODE_TARGETS_HELPER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/byte_io.h

@@ -0,0 +1,402 @@
+/*
+ *  Copyright (c) 2013 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_RTP_RTCP_SOURCE_BYTE_IO_H_
+#define MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_
+
+// This file contains classes for reading and writing integer types from/to
+// byte array representations. Signed/unsigned, partial (whole byte) sizes,
+// and big/little endian byte order is all supported.
+//
+// Usage examples:
+//
+// uint8_t* buffer = ...;
+//
+// // Read an unsigned 4 byte integer in big endian format
+// uint32_t val = ByteReader<uint32_t>::ReadBigEndian(buffer);
+//
+// // Read a signed 24-bit (3 byte) integer in little endian format
+// int32_t val = ByteReader<int32_t, 3>::ReadLittle(buffer);
+//
+// // Write an unsigned 8 byte integer in little endian format
+// ByteWriter<uint64_t>::WriteLittleEndian(buffer, val);
+//
+// Write an unsigned 40-bit (5 byte) integer in big endian format
+// ByteWriter<uint64_t, 5>::WriteBigEndian(buffer, val);
+//
+// These classes are implemented as recursive templetizations, intended to make
+// it easy for the compiler to completely inline the reading/writing.
+
+#include <stdint.h>
+
+#include <limits>
+
+namespace webrtc {
+
+// According to ISO C standard ISO/IEC 9899, section 6.2.6.2 (2), the three
+// representations of signed integers allowed are two's complement, one's
+// complement and sign/magnitude. We can detect which is used by looking at
+// the two last bits of -1, which will be 11 in two's complement, 10 in one's
+// complement and 01 in sign/magnitude.
+// TODO(sprang): In the unlikely event that we actually need to support a
+// platform that doesn't use two's complement, implement conversion to/from
+// wire format.
+
+// Assume the if any one signed integer type is two's complement, then all
+// other will be too.
+static_assert(
+    (-1 & 0x03) == 0x03,
+    "Only two's complement representation of signed integers supported.");
+
+// Plain const char* won't work for static_assert, use #define instead.
+#define kSizeErrorMsg "Byte size must be less than or equal to data type size."
+
+// Utility class for getting the unsigned equivalent of a signed type.
+template <typename T>
+struct UnsignedOf;
+
+// Class for reading integers from a sequence of bytes.
+// T = type of integer, B = bytes to read, is_signed = true if signed integer.
+// If is_signed is true and B < sizeof(T), sign extension might be needed.
+template <typename T,
+          unsigned int B = sizeof(T),
+          bool is_signed = std::numeric_limits<T>::is_signed>
+class ByteReader;
+
+// Specialization of ByteReader for unsigned types.
+template <typename T, unsigned int B>
+class ByteReader<T, B, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    return InternalReadBigEndian(data);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    return InternalReadLittleEndian(data);
+  }
+
+ private:
+  static T InternalReadBigEndian(const uint8_t* data) {
+    T val(0);
+    for (unsigned int i = 0; i < B; ++i)
+      val |= static_cast<T>(data[i]) << ((B - 1 - i) * 8);
+    return val;
+  }
+
+  static T InternalReadLittleEndian(const uint8_t* data) {
+    T val(0);
+    for (unsigned int i = 0; i < B; ++i)
+      val |= static_cast<T>(data[i]) << (i * 8);
+    return val;
+  }
+};
+
+// Specialization of ByteReader for signed types.
+template <typename T, unsigned int B>
+class ByteReader<T, B, true> {
+ public:
+  typedef typename UnsignedOf<T>::Type U;
+
+  static T ReadBigEndian(const uint8_t* data) {
+    U unsigned_val = ByteReader<T, B, false>::ReadBigEndian(data);
+    if (B < sizeof(T))
+      unsigned_val = SignExtend(unsigned_val);
+    return ReinterpretAsSigned(unsigned_val);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    U unsigned_val = ByteReader<T, B, false>::ReadLittleEndian(data);
+    if (B < sizeof(T))
+      unsigned_val = SignExtend(unsigned_val);
+    return ReinterpretAsSigned(unsigned_val);
+  }
+
+ private:
+  // As a hack to avoid implementation-specific or undefined behavior when
+  // bit-shifting or casting signed integers, read as a signed equivalent
+  // instead and convert to signed. This is safe since we have asserted that
+  // two's complement for is used.
+  static T ReinterpretAsSigned(U unsigned_val) {
+    // An unsigned value with only the highest order bit set (ex 0x80).
+    const U kUnsignedHighestBitMask = static_cast<U>(1)
+                                      << ((sizeof(U) * 8) - 1);
+    // A signed value with only the highest bit set. Since this is two's
+    // complement form, we can use the min value from std::numeric_limits.
+    const T kSignedHighestBitMask = std::numeric_limits<T>::min();
+
+    T val;
+    if ((unsigned_val & kUnsignedHighestBitMask) != 0) {
+      // Casting is only safe when unsigned value can be represented in the
+      // signed target type, so mask out highest bit and mask it back manually.
+      val = static_cast<T>(unsigned_val & ~kUnsignedHighestBitMask);
+      val |= kSignedHighestBitMask;
+    } else {
+      val = static_cast<T>(unsigned_val);
+    }
+    return val;
+  }
+
+  // If number of bytes is less than native data type (eg 24 bit, in int32_t),
+  // and the most significant bit of the actual data is set, we must sign
+  // extend the remaining byte(s) with ones so that the correct negative
+  // number is retained.
+  // Ex: 0x810A0B -> 0xFF810A0B, but 0x710A0B -> 0x00710A0B
+  static U SignExtend(const U val) {
+    const uint8_t kMsb = static_cast<uint8_t>(val >> ((B - 1) * 8));
+    if ((kMsb & 0x80) != 0) {
+      // Create a mask where all bits used by the B bytes are set to one,
+      // for instance 0x00FFFFFF for B = 3. Bit-wise invert that mask (to
+      // (0xFF000000 in the example above) and add it to the input value.
+      // The "B % sizeof(T)" is a workaround to undefined values warnings for
+      // B == sizeof(T), in which case this code won't be called anyway.
+      const U kUsedBitsMask = (1 << ((B % sizeof(T)) * 8)) - 1;
+      return ~kUsedBitsMask | val;
+    }
+    return val;
+  }
+};
+
+// Class for writing integers to a sequence of bytes
+// T = type of integer, B = bytes to write
+template <typename T,
+          unsigned int B = sizeof(T),
+          bool is_signed = std::numeric_limits<T>::is_signed>
+class ByteWriter;
+
+// Specialization of ByteWriter for unsigned types.
+template <typename T, unsigned int B>
+class ByteWriter<T, B, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    for (unsigned int i = 0; i < B; ++i) {
+      data[i] = val >> ((B - 1 - i) * 8);
+    }
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    for (unsigned int i = 0; i < B; ++i) {
+      data[i] = val >> (i * 8);
+    }
+  }
+};
+
+// Specialization of ByteWriter for signed types.
+template <typename T, unsigned int B>
+class ByteWriter<T, B, true> {
+ public:
+  typedef typename UnsignedOf<T>::Type U;
+
+  static void WriteBigEndian(uint8_t* data, T val) {
+    ByteWriter<U, B, false>::WriteBigEndian(data, ReinterpretAsUnsigned(val));
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    ByteWriter<U, B, false>::WriteLittleEndian(data,
+                                               ReinterpretAsUnsigned(val));
+  }
+
+ private:
+  static U ReinterpretAsUnsigned(T val) {
+    // According to ISO C standard ISO/IEC 9899, section 6.3.1.3 (1, 2) a
+    // conversion from signed to unsigned keeps the value if the new type can
+    // represent it, and otherwise adds one more than the max value of T until
+    // the value is in range. For two's complement, this fortunately means
+    // that the bit-wise value will be intact. Thus, since we have asserted that
+    // two's complement form is actually used, a simple cast is sufficient.
+    return static_cast<U>(val);
+  }
+};
+
+// ----- Below follows specializations of UnsignedOf utility class -----
+
+template <>
+struct UnsignedOf<int8_t> {
+  typedef uint8_t Type;
+};
+template <>
+struct UnsignedOf<int16_t> {
+  typedef uint16_t Type;
+};
+template <>
+struct UnsignedOf<int32_t> {
+  typedef uint32_t Type;
+};
+template <>
+struct UnsignedOf<int64_t> {
+  typedef uint64_t Type;
+};
+
+// ----- Below follows specializations for unsigned, B in { 1, 2, 4, 8 } -----
+
+// TODO(sprang): Check if these actually help or if generic cases will be
+// unrolled to and optimized to similar performance.
+
+// Specializations for single bytes
+template <typename T>
+class ByteReader<T, 1, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    return data[0];
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    return data[0];
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 1, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    data[0] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    data[0] = val;
+  }
+};
+
+// Specializations for two byte words
+template <typename T>
+class ByteReader<T, 2, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    return (data[0] << 8) | data[1];
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    return data[0] | (data[1] << 8);
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 2, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    data[0] = val >> 8;
+    data[1] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    data[0] = val;
+    data[1] = val >> 8;
+  }
+};
+
+// Specializations for four byte words.
+template <typename T>
+class ByteReader<T, 4, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    return (Get(data, 0) << 24) | (Get(data, 1) << 16) | (Get(data, 2) << 8) |
+           Get(data, 3);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    return Get(data, 0) | (Get(data, 1) << 8) | (Get(data, 2) << 16) |
+           (Get(data, 3) << 24);
+  }
+
+ private:
+  inline static T Get(const uint8_t* data, unsigned int index) {
+    return static_cast<T>(data[index]);
+  }
+};
+
+// Specializations for four byte words.
+template <typename T>
+class ByteWriter<T, 4, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    data[0] = val >> 24;
+    data[1] = val >> 16;
+    data[2] = val >> 8;
+    data[3] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    data[0] = val;
+    data[1] = val >> 8;
+    data[2] = val >> 16;
+    data[3] = val >> 24;
+  }
+};
+
+// Specializations for eight byte words.
+template <typename T>
+class ByteReader<T, 8, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    return (Get(data, 0) << 56) | (Get(data, 1) << 48) | (Get(data, 2) << 40) |
+           (Get(data, 3) << 32) | (Get(data, 4) << 24) | (Get(data, 5) << 16) |
+           (Get(data, 6) << 8) | Get(data, 7);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    return Get(data, 0) | (Get(data, 1) << 8) | (Get(data, 2) << 16) |
+           (Get(data, 3) << 24) | (Get(data, 4) << 32) | (Get(data, 5) << 40) |
+           (Get(data, 6) << 48) | (Get(data, 7) << 56);
+  }
+
+ private:
+  inline static T Get(const uint8_t* data, unsigned int index) {
+    return static_cast<T>(data[index]);
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 8, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    data[0] = val >> 56;
+    data[1] = val >> 48;
+    data[2] = val >> 40;
+    data[3] = val >> 32;
+    data[4] = val >> 24;
+    data[5] = val >> 16;
+    data[6] = val >> 8;
+    data[7] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    data[0] = val;
+    data[1] = val >> 8;
+    data[2] = val >> 16;
+    data[3] = val >> 24;
+    data[4] = val >> 32;
+    data[5] = val >> 40;
+    data[6] = val >> 48;
+    data[7] = val >> 56;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.cc

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2021 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 "modules/rtp_rtcp/source/capture_clock_offset_updater.h"
+
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+absl::optional<int64_t>
+CaptureClockOffsetUpdater::AdjustEstimatedCaptureClockOffset(
+    absl::optional<int64_t> remote_capture_clock_offset) const {
+  if (remote_capture_clock_offset == absl::nullopt ||
+      remote_to_local_clock_offset_ == absl::nullopt) {
+    return absl::nullopt;
+  }
+
+  // Do calculations as "unsigned" to make overflows deterministic.
+  return static_cast<uint64_t>(*remote_capture_clock_offset) +
+         static_cast<uint64_t>(*remote_to_local_clock_offset_);
+}
+
+absl::optional<TimeDelta> CaptureClockOffsetUpdater::ConvertsToTimeDela(
+    absl::optional<int64_t> q32x32) {
+  if (q32x32 == absl::nullopt) {
+    return absl::nullopt;
+  }
+  return TimeDelta::Millis(Q32x32ToInt64Ms(*q32x32));
+}
+
+void CaptureClockOffsetUpdater::SetRemoteToLocalClockOffset(
+    absl::optional<int64_t> offset_q32x32) {
+  remote_to_local_clock_offset_ = offset_q32x32;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.h

@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 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_RTP_RTCP_SOURCE_CAPTURE_CLOCK_OFFSET_UPDATER_H_
+#define MODULES_RTP_RTCP_SOURCE_CAPTURE_CLOCK_OFFSET_UPDATER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+//
+// Helper class for calculating the clock offset against the capturer's clock.
+//
+// This is achieved by adjusting the estimated capture clock offset in received
+// Absolute Capture Time RTP header extension (see
+// https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/), which
+// represents the clock offset between a remote sender and the capturer, by
+// adding local-to-remote clock offset.
+
+class CaptureClockOffsetUpdater {
+ public:
+  // Adjusts remote_capture_clock_offset, which originates from Absolute Capture
+  // Time RTP header extension, to get the local clock offset against the
+  // capturer's clock.
+  absl::optional<int64_t> AdjustEstimatedCaptureClockOffset(
+      absl::optional<int64_t> remote_capture_clock_offset) const;
+
+  // Sets the NTP clock offset between the sender system (which may be different
+  // from the capture system) and the local system. This information is normally
+  // provided by passing half the value of the Round-Trip Time estimation given
+  // by RTCP sender reports (see DLSR/DLRR).
+  //
+  // Note that the value must be in Q32.32-formatted fixed-point seconds.
+  void SetRemoteToLocalClockOffset(absl::optional<int64_t> offset_q32x32);
+
+  // Converts a signed Q32.32-formatted fixed-point to a TimeDelta.
+  static absl::optional<TimeDelta> ConvertsToTimeDela(
+      absl::optional<int64_t> q32x32);
+
+ private:
+  absl::optional<int64_t> remote_to_local_clock_offset_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_CAPTURE_CLOCK_OFFSET_UPDATER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc

@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
+
+#include <memory>
+
+#include "api/video/video_codec_type.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+#ifdef RTC_ENABLE_H265
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h"
+#endif
+
+namespace webrtc {
+
+std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
+    VideoCodecType codec) {
+  switch (codec) {
+    case kVideoCodecH264:
+      return std::make_unique<VideoRtpDepacketizerH264>();
+    case kVideoCodecVP8:
+      return std::make_unique<VideoRtpDepacketizerVp8>();
+    case kVideoCodecVP9:
+      return std::make_unique<VideoRtpDepacketizerVp9>();
+    case kVideoCodecAV1:
+      return std::make_unique<VideoRtpDepacketizerAv1>();
+    case kVideoCodecH265:
+#ifdef RTC_ENABLE_H265
+      return std::make_unique<VideoRtpDepacketizerH265>();
+#else
+      return nullptr;
+#endif
+    case kVideoCodecGeneric:
+    case kVideoCodecMultiplex:
+      return std::make_unique<VideoRtpDepacketizerGeneric>();
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2019 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_RTP_RTCP_SOURCE_CREATE_VIDEO_RTP_DEPACKETIZER_H_
+#define MODULES_RTP_RTCP_SOURCE_CREATE_VIDEO_RTP_DEPACKETIZER_H_
+
+#include <memory>
+
+#include "api/video/video_codec_type.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
+    VideoCodecType codec);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_CREATE_VIDEO_RTP_DEPACKETIZER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc

@@ -0,0 +1,374 @@
+/*
+ *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+constexpr uint32_t kTimestampTicksPerMs = 90;
+constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1);
+constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
+
+}  // namespace
+
+DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
+    DEPRECATED_RtpSenderEgress* sender,
+    PacketSequencer* sequence_number_assigner)
+    : transport_sequence_number_(0),
+      sender_(sender),
+      sequence_number_assigner_(sequence_number_assigner) {
+  RTC_DCHECK(sequence_number_assigner_);
+}
+DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() =
+    default;
+
+void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+  for (auto& packet : packets) {
+    // Assign sequence numbers, but not for flexfec which is already running on
+    // an internally maintained sequence number series.
+    if (packet->Ssrc() != sender_->FlexFecSsrc()) {
+      sequence_number_assigner_->Sequence(*packet);
+    }
+    if (!packet->SetExtension<TransportSequenceNumber>(
+            ++transport_sequence_number_)) {
+      --transport_sequence_number_;
+    }
+    packet->ReserveExtension<TransmissionOffset>();
+    packet->ReserveExtension<AbsoluteSendTime>();
+    sender_->SendPacket(packet.get(), PacedPacketInfo());
+  }
+}
+
+DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
+    const RtpRtcpInterface::Configuration& config,
+    RtpPacketHistory* packet_history)
+    : ssrc_(config.local_media_ssrc),
+      rtx_ssrc_(config.rtx_send_ssrc),
+      flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
+                                         : absl::nullopt),
+      populate_network2_timestamp_(config.populate_network2_timestamp),
+      clock_(config.clock),
+      packet_history_(packet_history),
+      transport_(config.outgoing_transport),
+      event_log_(config.event_log),
+      is_audio_(config.audio),
+      need_rtp_packet_infos_(config.need_rtp_packet_infos),
+      transport_feedback_observer_(config.transport_feedback_callback),
+      send_packet_observer_(config.send_packet_observer),
+      rtp_stats_callback_(config.rtp_stats_callback),
+      bitrate_callback_(config.send_bitrate_observer),
+      media_has_been_sent_(false),
+      force_part_of_allocation_(false),
+      timestamp_offset_(0),
+      send_rates_(kNumMediaTypes, BitrateTracker(kBitrateStatisticsWindow)),
+      rtp_sequence_number_map_(need_rtp_packet_infos_
+                                   ? std::make_unique<RtpSequenceNumberMap>(
+                                         kRtpSequenceNumberMapMaxEntries)
+                                   : nullptr) {}
+
+void DEPRECATED_RtpSenderEgress::SendPacket(
+    RtpPacketToSend* packet,
+    const PacedPacketInfo& pacing_info) {
+  RTC_DCHECK(packet);
+
+  const uint32_t packet_ssrc = packet->Ssrc();
+  RTC_DCHECK(packet->packet_type().has_value());
+  RTC_DCHECK(HasCorrectSsrc(*packet));
+  Timestamp now = clock_->CurrentTime();
+  int64_t now_ms = now.ms();
+
+  if (is_audio_) {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
+                                    GetSendRates().Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "AudioNackBitrate_kbps", now_ms,
+        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
+        packet_ssrc);
+#endif
+  } else {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
+                                    GetSendRates().Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "VideoNackBitrate_kbps", now_ms,
+        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
+        packet_ssrc);
+#endif
+  }
+
+  PacketOptions options;
+  {
+    MutexLock lock(&lock_);
+    options.included_in_allocation = force_part_of_allocation_;
+
+    if (need_rtp_packet_infos_ &&
+        packet->packet_type() == RtpPacketToSend::Type::kVideo) {
+      RTC_DCHECK(rtp_sequence_number_map_);
+      // Last packet of a frame, add it to sequence number info map.
+      const uint32_t timestamp = packet->Timestamp() - timestamp_offset_;
+      bool is_first_packet_of_frame = packet->is_first_packet_of_frame();
+      bool is_last_packet_of_frame = packet->Marker();
+
+      rtp_sequence_number_map_->InsertPacket(
+          packet->SequenceNumber(),
+          RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
+                                     is_last_packet_of_frame));
+    }
+  }
+
+  // Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
+  // the pacer, these modifications of the header below are happening after the
+  // FEC protection packets are calculated. This will corrupt recovered packets
+  // at the same place. It's not an issue for extensions, which are present in
+  // all the packets (their content just may be incorrect on recovered packets).
+  // In case of VideoTimingExtension, since it's present not in every packet,
+  // data after rtp header may be corrupted if these packets are protected by
+  // the FEC.
+  int64_t diff_ms = now_ms - packet->capture_time().ms();
+  if (packet->HasExtension<TransmissionOffset>()) {
+    packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms);
+  }
+  if (packet->HasExtension<AbsoluteSendTime>()) {
+    packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
+  }
+
+  if (packet->HasExtension<VideoTimingExtension>()) {
+    if (populate_network2_timestamp_) {
+      packet->set_network2_time(now);
+    } else {
+      packet->set_pacer_exit_time(now);
+    }
+  }
+
+  const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio ||
+                        packet->packet_type() == RtpPacketMediaType::kVideo;
+
+  // Downstream code actually uses this flag to distinguish between media and
+  // everything else.
+  options.is_retransmit = !is_media;
+  if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
+    options.packet_id = *packet_id;
+    options.included_in_feedback = true;
+    options.included_in_allocation = true;
+    AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
+  }
+
+  options.additional_data = packet->additional_data();
+
+  if (packet->packet_type() != RtpPacketMediaType::kPadding &&
+      packet->packet_type() != RtpPacketMediaType::kRetransmission) {
+    UpdateOnSendPacket(options.packet_id, packet->capture_time().ms(),
+                       packet_ssrc);
+  }
+
+  const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
+
+  // Put packet in retransmission history or update pending status even if
+  // actual sending fails.
+  if (is_media && packet->allow_retransmission()) {
+    packet_history_->PutRtpPacket(std::make_unique<RtpPacketToSend>(*packet),
+                                  now);
+  } else if (packet->retransmitted_sequence_number()) {
+    packet_history_->MarkPacketAsSent(*packet->retransmitted_sequence_number());
+  }
+
+  if (send_success) {
+    MutexLock lock(&lock_);
+    UpdateRtpStats(*packet);
+    media_has_been_sent_ = true;
+  }
+}
+
+void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() {
+  if (!bitrate_callback_)
+    return;
+
+  MutexLock lock(&lock_);
+  RtpSendRates send_rates = GetSendRatesLocked();
+  bitrate_callback_->Notify(
+      send_rates.Sum().bps(),
+      send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
+}
+
+RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const {
+  MutexLock lock(&lock_);
+  return GetSendRatesLocked();
+}
+
+RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const {
+  const Timestamp now = clock_->CurrentTime();
+  RtpSendRates current_rates;
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
+    current_rates[type] = send_rates_[i].Rate(now).value_or(DataRate::Zero());
+  }
+  return current_rates;
+}
+
+void DEPRECATED_RtpSenderEgress::GetDataCounters(
+    StreamDataCounters* rtp_stats,
+    StreamDataCounters* rtx_stats) const {
+  MutexLock lock(&lock_);
+  *rtp_stats = rtp_stats_;
+  *rtx_stats = rtx_rtp_stats_;
+}
+
+void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
+    bool part_of_allocation) {
+  MutexLock lock(&lock_);
+  force_part_of_allocation_ = part_of_allocation;
+}
+
+bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const {
+  MutexLock lock(&lock_);
+  return media_has_been_sent_;
+}
+
+void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
+  MutexLock lock(&lock_);
+  media_has_been_sent_ = media_sent;
+}
+
+void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
+  MutexLock lock(&lock_);
+  timestamp_offset_ = timestamp;
+}
+
+std::vector<RtpSequenceNumberMap::Info>
+DEPRECATED_RtpSenderEgress::GetSentRtpPacketInfos(
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
+  RTC_DCHECK(!sequence_numbers.empty());
+  if (!need_rtp_packet_infos_) {
+    return std::vector<RtpSequenceNumberMap::Info>();
+  }
+
+  std::vector<RtpSequenceNumberMap::Info> results;
+  results.reserve(sequence_numbers.size());
+
+  MutexLock lock(&lock_);
+  for (uint16_t sequence_number : sequence_numbers) {
+    const auto& info = rtp_sequence_number_map_->Get(sequence_number);
+    if (!info) {
+      // The empty vector will be returned. We can delay the clearing
+      // of the vector until after we exit the critical section.
+      return std::vector<RtpSequenceNumberMap::Info>();
+    }
+    results.push_back(*info);
+  }
+
+  return results;
+}
+
+bool DEPRECATED_RtpSenderEgress::HasCorrectSsrc(
+    const RtpPacketToSend& packet) const {
+  switch (*packet.packet_type()) {
+    case RtpPacketMediaType::kAudio:
+    case RtpPacketMediaType::kVideo:
+      return packet.Ssrc() == ssrc_;
+    case RtpPacketMediaType::kRetransmission:
+    case RtpPacketMediaType::kPadding:
+      // Both padding and retransmission must be on either the media or the
+      // RTX stream.
+      return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_;
+    case RtpPacketMediaType::kForwardErrorCorrection:
+      // FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
+      return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_;
+  }
+  return false;
+}
+
+void DEPRECATED_RtpSenderEgress::AddPacketToTransportFeedback(
+    uint16_t packet_id,
+    const RtpPacketToSend& packet,
+    const PacedPacketInfo& pacing_info) {
+  if (transport_feedback_observer_) {
+    RtpPacketSendInfo packet_info;
+    packet_info.media_ssrc = ssrc_;
+    packet_info.transport_sequence_number = packet_id;
+    packet_info.rtp_sequence_number = packet.SequenceNumber();
+    packet_info.length = packet.size();
+    packet_info.pacing_info = pacing_info;
+    packet_info.packet_type = packet.packet_type();
+    transport_feedback_observer_->OnAddPacket(packet_info);
+  }
+}
+
+void DEPRECATED_RtpSenderEgress::UpdateOnSendPacket(int packet_id,
+                                                    int64_t capture_time_ms,
+                                                    uint32_t ssrc) {
+  if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1) {
+    return;
+  }
+
+  send_packet_observer_->OnSendPacket(packet_id,
+                                      Timestamp::Millis(capture_time_ms), ssrc);
+}
+
+bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork(
+    const RtpPacketToSend& packet,
+    const PacketOptions& options,
+    const PacedPacketInfo& pacing_info) {
+  int bytes_sent = -1;
+  if (transport_) {
+    bytes_sent = transport_->SendRtp(packet, options)
+                     ? static_cast<int>(packet.size())
+                     : -1;
+    if (event_log_ && bytes_sent > 0) {
+      event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
+          packet, pacing_info.probe_cluster_id));
+    }
+  }
+
+  if (bytes_sent <= 0) {
+    RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
+    return false;
+  }
+  return true;
+}
+
+void DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) {
+  Timestamp now = clock_->CurrentTime();
+
+  StreamDataCounters* counters =
+      packet.Ssrc() == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;
+
+  counters->MaybeSetFirstPacketTime(now);
+
+  if (packet.packet_type() == RtpPacketMediaType::kForwardErrorCorrection) {
+    counters->fec.AddPacket(packet);
+  }
+
+  if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
+    counters->retransmitted.AddPacket(packet);
+  }
+  counters->transmitted.AddPacket(packet);
+
+  RTC_DCHECK(packet.packet_type().has_value());
+  send_rates_[static_cast<size_t>(*packet.packet_type())].Update(packet.size(),
+                                                                 now);
+
+  if (rtp_stats_callback_) {
+    rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h

@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2019 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_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_
+#define MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/call/transport.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/units/data_rate.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+#include "rtc_base/bitrate_tracker.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class DEPRECATED_RtpSenderEgress {
+ public:
+  // Helper class that redirects packets directly to the send part of this class
+  // without passing through an actual paced sender.
+  class NonPacedPacketSender : public RtpPacketSender {
+   public:
+    NonPacedPacketSender(DEPRECATED_RtpSenderEgress* sender,
+                         PacketSequencer* sequence_number_assigner);
+    virtual ~NonPacedPacketSender();
+
+    void EnqueuePackets(
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
+    void RemovePacketsForSsrc(uint32_t ssrc) override {}
+
+   private:
+    uint16_t transport_sequence_number_;
+    DEPRECATED_RtpSenderEgress* const sender_;
+    PacketSequencer* sequence_number_assigner_;
+  };
+
+  DEPRECATED_RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
+                             RtpPacketHistory* packet_history);
+  ~DEPRECATED_RtpSenderEgress() = default;
+
+  void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info)
+      RTC_LOCKS_EXCLUDED(lock_);
+  uint32_t Ssrc() const { return ssrc_; }
+  absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
+  absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
+
+  void ProcessBitrateAndNotifyObservers() RTC_LOCKS_EXCLUDED(lock_);
+  RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_);
+  void GetDataCounters(StreamDataCounters* rtp_stats,
+                       StreamDataCounters* rtx_stats) const
+      RTC_LOCKS_EXCLUDED(lock_);
+
+  void ForceIncludeSendPacketsInAllocation(bool part_of_allocation)
+      RTC_LOCKS_EXCLUDED(lock_);
+  bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_);
+  void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
+  void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
+
+  // For each sequence number in `sequence_number`, recall the last RTP packet
+  // which bore it - its timestamp and whether it was the first and/or last
+  // packet in that frame. If all of the given sequence numbers could be
+  // recalled, return a vector with all of them (in corresponding order).
+  // If any could not be recalled, return an empty vector.
+  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      rtc::ArrayView<const uint16_t> sequence_numbers) const
+      RTC_LOCKS_EXCLUDED(lock_);
+
+ private:
+  RtpSendRates GetSendRatesLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
+  void AddPacketToTransportFeedback(uint16_t packet_id,
+                                    const RtpPacketToSend& packet,
+                                    const PacedPacketInfo& pacing_info);
+  void UpdateOnSendPacket(int packet_id,
+                          int64_t capture_time_ms,
+                          uint32_t ssrc);
+  // Sends packet on to `transport_`, leaving the RTP module.
+  bool SendPacketToNetwork(const RtpPacketToSend& packet,
+                           const PacketOptions& options,
+                           const PacedPacketInfo& pacing_info);
+  void UpdateRtpStats(const RtpPacketToSend& packet)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+
+  const uint32_t ssrc_;
+  const absl::optional<uint32_t> rtx_ssrc_;
+  const absl::optional<uint32_t> flexfec_ssrc_;
+  const bool populate_network2_timestamp_;
+  Clock* const clock_;
+  RtpPacketHistory* const packet_history_;
+  Transport* const transport_;
+  RtcEventLog* const event_log_;
+  const bool is_audio_;
+  const bool need_rtp_packet_infos_;
+
+  TransportFeedbackObserver* const transport_feedback_observer_;
+  SendPacketObserver* const send_packet_observer_;
+  StreamDataCountersCallback* const rtp_stats_callback_;
+  BitrateStatisticsObserver* const bitrate_callback_;
+
+  mutable Mutex lock_;
+  bool media_has_been_sent_ RTC_GUARDED_BY(lock_);
+  bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
+  uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_);
+
+  StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
+  StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
+  // One element per value in RtpPacketMediaType, with index matching value.
+  std::vector<BitrateTracker> send_rates_ RTC_GUARDED_BY(lock_);
+
+  // Maps sent packets' sequence numbers to a tuple consisting of:
+  // 1. The timestamp, without the randomizing offset mandated by the RFC.
+  // 2. Whether the packet was the first in its frame.
+  // 3. Whether the packet was the last in its frame.
+  const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
+      RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 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 "modules/rtp_rtcp/source/dtmf_queue.h"
+
+#include <stddef.h>
+
+#include "rtc_base/checks.h"
+
+namespace {
+constexpr size_t kDtmfOutbandMax = 20;
+}
+
+namespace webrtc {
+DtmfQueue::DtmfQueue() {}
+
+DtmfQueue::~DtmfQueue() {}
+
+bool DtmfQueue::AddDtmf(const Event& event) {
+  MutexLock lock(&dtmf_mutex_);
+  if (queue_.size() >= kDtmfOutbandMax) {
+    return false;
+  }
+  queue_.push_back(event);
+  return true;
+}
+
+bool DtmfQueue::NextDtmf(Event* event) {
+  RTC_DCHECK(event);
+  MutexLock lock(&dtmf_mutex_);
+  if (queue_.empty()) {
+    return false;
+  }
+
+  *event = queue_.front();
+  queue_.pop_front();
+  return true;
+}
+
+bool DtmfQueue::PendingDtmf() const {
+  MutexLock lock(&dtmf_mutex_);
+  return !queue_.empty();
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/dtmf_queue.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2011 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_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+#define MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+
+#include <stdint.h>
+
+#include <list>
+
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+class DtmfQueue {
+ public:
+  struct Event {
+    uint16_t duration_ms = 0;
+    uint8_t payload_type = 0;
+    uint8_t key = 0;
+    uint8_t level = 0;
+  };
+
+  DtmfQueue();
+  ~DtmfQueue();
+
+  bool AddDtmf(const Event& event);
+  bool NextDtmf(Event* event);
+  bool PendingDtmf() const;
+
+ private:
+  mutable Mutex dtmf_mutex_;
+  std::list<Event> queue_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc

@@ -0,0 +1,660 @@
+/*
+ *  Copyright (c) 2018 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 "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+
+namespace {
+// clang-format off
+#define kMaskBursty1_1 \
+  0x80, 0x00
+
+#define kMaskBursty2_1 \
+  0xc0, 0x00
+
+#define kMaskBursty2_2 \
+  0x80, 0x00, \
+  0xc0, 0x00
+
+#define kMaskBursty3_1 \
+  0xe0, 0x00
+
+#define kMaskBursty3_2 \
+  0xc0, 0x00, \
+  0xa0, 0x00
+
+#define kMaskBursty3_3 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00
+
+#define kMaskBursty4_1 \
+  0xf0, 0x00
+
+#define kMaskBursty4_2 \
+  0xa0, 0x00, \
+  0xd0, 0x00
+
+#define kMaskBursty4_3 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x90, 0x00
+
+#define kMaskBursty4_4 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00
+
+#define kMaskBursty5_1 \
+  0xf8, 0x00
+
+#define kMaskBursty5_2 \
+  0xd0, 0x00, \
+  0xa8, 0x00
+
+#define kMaskBursty5_3 \
+  0x70, 0x00, \
+  0x90, 0x00, \
+  0xc8, 0x00
+
+#define kMaskBursty5_4 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x88, 0x00
+
+#define kMaskBursty5_5 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00
+
+#define kMaskBursty6_1 \
+  0xfc, 0x00
+
+#define kMaskBursty6_2 \
+  0xa8, 0x00, \
+  0xd4, 0x00
+
+#define kMaskBursty6_3 \
+  0x94, 0x00, \
+  0xc8, 0x00, \
+  0x64, 0x00
+
+#define kMaskBursty6_4 \
+  0x60, 0x00, \
+  0x38, 0x00, \
+  0x88, 0x00, \
+  0xc4, 0x00
+
+#define kMaskBursty6_5 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x84, 0x00
+
+#define kMaskBursty6_6 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00
+
+#define kMaskBursty7_1 \
+  0xfe, 0x00
+
+#define kMaskBursty7_2 \
+  0xd4, 0x00, \
+  0xaa, 0x00
+
+#define kMaskBursty7_3 \
+  0xc8, 0x00, \
+  0x74, 0x00, \
+  0x92, 0x00
+
+#define kMaskBursty7_4 \
+  0x38, 0x00, \
+  0x8a, 0x00, \
+  0xc4, 0x00, \
+  0x62, 0x00
+
+#define kMaskBursty7_5 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x1c, 0x00, \
+  0x84, 0x00, \
+  0xc2, 0x00
+
+#define kMaskBursty7_6 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x82, 0x00
+
+#define kMaskBursty7_7 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00
+
+#define kMaskBursty8_1 \
+  0xff, 0x00
+
+#define kMaskBursty8_2 \
+  0xaa, 0x00, \
+  0xd5, 0x00
+
+#define kMaskBursty8_3 \
+  0x74, 0x00, \
+  0x92, 0x00, \
+  0xc9, 0x00
+
+#define kMaskBursty8_4 \
+  0x8a, 0x00, \
+  0xc5, 0x00, \
+  0x62, 0x00, \
+  0x31, 0x00
+
+#define kMaskBursty8_5 \
+  0x30, 0x00, \
+  0x1c, 0x00, \
+  0x85, 0x00, \
+  0xc2, 0x00, \
+  0x61, 0x00
+
+#define kMaskBursty8_6 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0e, 0x00, \
+  0x82, 0x00, \
+  0xc1, 0x00
+
+#define kMaskBursty8_7 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x81, 0x00
+
+#define kMaskBursty8_8 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00
+
+#define kMaskBursty9_1 \
+  0xff, 0x80
+
+#define kMaskBursty9_2 \
+  0xd5, 0x00, \
+  0xaa, 0x80
+
+#define kMaskBursty9_3 \
+  0x92, 0x00, \
+  0xc9, 0x00, \
+  0x74, 0x80
+
+#define kMaskBursty9_4 \
+  0xc5, 0x00, \
+  0x62, 0x00, \
+  0x39, 0x00, \
+  0x8a, 0x80
+
+#define kMaskBursty9_5 \
+  0x1c, 0x00, \
+  0x85, 0x00, \
+  0xc2, 0x80, \
+  0x61, 0x00, \
+  0x30, 0x80
+
+#define kMaskBursty9_6 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0e, 0x00, \
+  0x82, 0x80, \
+  0xc1, 0x00, \
+  0x60, 0x80
+
+#define kMaskBursty9_7 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x00, \
+  0xc0, 0x80
+
+#define kMaskBursty9_8 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x80, 0x80
+
+#define kMaskBursty9_9 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80
+
+#define kMaskBursty10_1 \
+  0xff, 0xc0
+
+#define kMaskBursty10_2 \
+  0xaa, 0x80, \
+  0xd5, 0x40
+
+#define kMaskBursty10_3 \
+  0xc9, 0x00, \
+  0x74, 0x80, \
+  0x92, 0x40
+
+#define kMaskBursty10_4 \
+  0x62, 0x00, \
+  0x39, 0x00, \
+  0x8a, 0x80, \
+  0xc5, 0x40
+
+#define kMaskBursty10_5 \
+  0x85, 0x00, \
+  0xc2, 0x80, \
+  0x61, 0x40, \
+  0x30, 0x80, \
+  0x18, 0x40
+
+#define kMaskBursty10_6 \
+  0x18, 0x00, \
+  0x0e, 0x00, \
+  0x82, 0x80, \
+  0xc1, 0x40, \
+  0x60, 0x80, \
+  0x30, 0x40
+
+#define kMaskBursty10_7 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x40, \
+  0xc0, 0x80, \
+  0x60, 0x40
+
+#define kMaskBursty10_8 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x80, 0x80, \
+  0xc0, 0x40
+
+#define kMaskBursty10_9 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x80, 0x40
+
+#define kMaskBursty10_10 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0
+
+#define kMaskBursty11_1 \
+  0xff, 0xe0
+
+#define kMaskBursty11_2 \
+  0xd5, 0x40, \
+  0xaa, 0xa0
+
+#define kMaskBursty11_3 \
+  0x74, 0x80, \
+  0x92, 0x40, \
+  0xc9, 0x20
+
+#define kMaskBursty11_4 \
+  0x39, 0x00, \
+  0x8a, 0x80, \
+  0xc5, 0x40, \
+  0x62, 0x20
+
+#define kMaskBursty11_5 \
+  0xc2, 0xc0, \
+  0x61, 0x00, \
+  0x30, 0xa0, \
+  0x1c, 0x40, \
+  0x85, 0x20
+
+#define kMaskBursty11_6 \
+  0x0e, 0x00, \
+  0x82, 0x80, \
+  0xc1, 0x40, \
+  0x60, 0xa0, \
+  0x30, 0x40, \
+  0x18, 0x20
+
+#define kMaskBursty11_7 \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x40, \
+  0xc0, 0xa0, \
+  0x60, 0x40, \
+  0x30, 0x20
+
+#define kMaskBursty11_8 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x40, \
+  0x80, 0xa0, \
+  0xc0, 0x40, \
+  0x60, 0x20
+
+#define kMaskBursty11_9 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x80, 0x40, \
+  0xc0, 0x20
+
+#define kMaskBursty11_10 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x80, 0x20
+
+#define kMaskBursty11_11 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x00, 0x60
+
+#define kMaskBursty12_1 \
+  0xff, 0xf0
+
+#define kMaskBursty12_2 \
+  0xaa, 0xa0, \
+  0xd5, 0x50
+
+#define kMaskBursty12_3 \
+  0x92, 0x40, \
+  0xc9, 0x20, \
+  0x74, 0x90
+
+#define kMaskBursty12_4 \
+  0x8a, 0x80, \
+  0xc5, 0x40, \
+  0x62, 0x20, \
+  0x39, 0x10
+
+#define kMaskBursty12_5 \
+  0x61, 0x00, \
+  0x30, 0xa0, \
+  0x1c, 0x50, \
+  0x85, 0x20, \
+  0xc2, 0x90
+
+#define kMaskBursty12_6 \
+  0x82, 0x90, \
+  0xc1, 0x40, \
+  0x60, 0xa0, \
+  0x30, 0x50, \
+  0x18, 0x20, \
+  0x0c, 0x10
+
+#define kMaskBursty12_7 \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x40, \
+  0xc0, 0xa0, \
+  0x60, 0x50, \
+  0x30, 0x20, \
+  0x18, 0x10
+
+#define kMaskBursty12_8 \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x80, 0xa0, \
+  0xc0, 0x50, \
+  0x60, 0x20, \
+  0x30, 0x10
+
+#define kMaskBursty12_9 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x80, 0x50, \
+  0xc0, 0x20, \
+  0x60, 0x10
+
+#define kMaskBursty12_10 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x80, 0x20, \
+  0xc0, 0x10
+
+#define kMaskBursty12_11 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x00, 0x60, \
+  0x80, 0x10
+
+#define kMaskBursty12_12 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x00, 0x60, \
+  0x00, 0x30
+
+#define kPacketMaskBursty1 1, \
+  kMaskBursty1_1
+
+#define kPacketMaskBursty2 2, \
+  kMaskBursty2_1, \
+  kMaskBursty2_2
+
+#define kPacketMaskBursty3 3, \
+  kMaskBursty3_1, \
+  kMaskBursty3_2, \
+  kMaskBursty3_3
+
+#define kPacketMaskBursty4 4, \
+  kMaskBursty4_1, \
+  kMaskBursty4_2, \
+  kMaskBursty4_3, \
+  kMaskBursty4_4
+
+#define kPacketMaskBursty5 5, \
+  kMaskBursty5_1, \
+  kMaskBursty5_2, \
+  kMaskBursty5_3, \
+  kMaskBursty5_4, \
+  kMaskBursty5_5
+
+#define kPacketMaskBursty6 6, \
+  kMaskBursty6_1, \
+  kMaskBursty6_2, \
+  kMaskBursty6_3, \
+  kMaskBursty6_4, \
+  kMaskBursty6_5, \
+  kMaskBursty6_6
+
+#define kPacketMaskBursty7 7, \
+  kMaskBursty7_1, \
+  kMaskBursty7_2, \
+  kMaskBursty7_3, \
+  kMaskBursty7_4, \
+  kMaskBursty7_5, \
+  kMaskBursty7_6, \
+  kMaskBursty7_7
+
+#define kPacketMaskBursty8 8, \
+  kMaskBursty8_1, \
+  kMaskBursty8_2, \
+  kMaskBursty8_3, \
+  kMaskBursty8_4, \
+  kMaskBursty8_5, \
+  kMaskBursty8_6, \
+  kMaskBursty8_7, \
+  kMaskBursty8_8
+
+#define kPacketMaskBursty9 9, \
+  kMaskBursty9_1, \
+  kMaskBursty9_2, \
+  kMaskBursty9_3, \
+  kMaskBursty9_4, \
+  kMaskBursty9_5, \
+  kMaskBursty9_6, \
+  kMaskBursty9_7, \
+  kMaskBursty9_8, \
+  kMaskBursty9_9
+
+#define kPacketMaskBursty10 10, \
+  kMaskBursty10_1, \
+  kMaskBursty10_2, \
+  kMaskBursty10_3, \
+  kMaskBursty10_4, \
+  kMaskBursty10_5, \
+  kMaskBursty10_6, \
+  kMaskBursty10_7, \
+  kMaskBursty10_8, \
+  kMaskBursty10_9, \
+  kMaskBursty10_10
+
+#define kPacketMaskBursty11 11, \
+  kMaskBursty11_1, \
+  kMaskBursty11_2, \
+  kMaskBursty11_3, \
+  kMaskBursty11_4, \
+  kMaskBursty11_5, \
+  kMaskBursty11_6, \
+  kMaskBursty11_7, \
+  kMaskBursty11_8, \
+  kMaskBursty11_9, \
+  kMaskBursty11_10, \
+  kMaskBursty11_11
+
+#define kPacketMaskBursty12 12, \
+  kMaskBursty12_1, \
+  kMaskBursty12_2, \
+  kMaskBursty12_3, \
+  kMaskBursty12_4, \
+  kMaskBursty12_5, \
+  kMaskBursty12_6, \
+  kMaskBursty12_7, \
+  kMaskBursty12_8, \
+  kMaskBursty12_9, \
+  kMaskBursty12_10, \
+  kMaskBursty12_11, \
+  kMaskBursty12_12
+
+// clang-format on
+}  // namespace
+
+namespace webrtc {
+namespace fec_private_tables {
+
+const uint8_t kPacketMaskBurstyTbl[] = {
+    12,
+    kPacketMaskBursty1,
+    kPacketMaskBursty2,
+    kPacketMaskBursty3,
+    kPacketMaskBursty4,
+    kPacketMaskBursty5,
+    kPacketMaskBursty6,
+    kPacketMaskBursty7,
+    kPacketMaskBursty8,
+    kPacketMaskBursty9,
+    kPacketMaskBursty10,
+    kPacketMaskBursty11,
+    kPacketMaskBursty12,
+};
+
+}  // namespace fec_private_tables
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h

@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
+#define MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
+
+// This file contains a set of packets masks for the FEC code. The masks in
+// this table are specifically designed to favor recovery of bursty/consecutive
+// loss network conditions. The tradeoff is worse recovery for random losses.
+// These packet masks are currently defined to protect up to 12 media packets.
+// They have the following property: for any packet mask defined by the
+// parameters (k,m), where k = number of media packets, m = number of FEC
+// packets, all "consecutive" losses of size <= m are completely recoverable.
+// By consecutive losses we mean consecutive with respect to the sequence
+// number ordering of the list (media and FEC) of packets. The difference
+// between these masks (`kFecMaskBursty`) and `kFecMaskRandom` type, defined
+// in fec_private_tables.h, is more significant for longer codes
+// (i.e., more packets/symbols in the code, so larger (k,m), i.e.,  k > 4,
+// m > 3).
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace fec_private_tables {
+
+extern const uint8_t kPacketMaskBurstyTbl[];
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc

@@ -0,0 +1,660 @@
+/*
+ *  Copyright (c) 2018 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 "modules/rtp_rtcp/source/fec_private_tables_random.h"
+
+namespace {
+// clang-format off
+#define kMaskRandom1_1 \
+  0x80, 0x00
+
+#define kMaskRandom2_1 \
+  0xc0, 0x00
+
+#define kMaskRandom2_2 \
+  0xc0, 0x00, \
+  0x80, 0x00
+
+#define kMaskRandom3_1 \
+  0xe0, 0x00
+
+#define kMaskRandom3_2 \
+  0xc0, 0x00, \
+  0xa0, 0x00
+
+#define kMaskRandom3_3 \
+  0xc0, 0x00, \
+  0xa0, 0x00, \
+  0x60, 0x00
+
+#define kMaskRandom4_1 \
+  0xf0, 0x00
+
+#define kMaskRandom4_2 \
+  0xc0, 0x00, \
+  0xb0, 0x00
+
+#define kMaskRandom4_3 \
+  0xc0, 0x00, \
+  0xb0, 0x00, \
+  0x60, 0x00
+
+#define kMaskRandom4_4 \
+  0xc0, 0x00, \
+  0xa0, 0x00, \
+  0x30, 0x00, \
+  0x50, 0x00
+
+#define kMaskRandom5_1 \
+  0xf8, 0x00
+
+#define kMaskRandom5_2 \
+  0xa8, 0x00, \
+  0xd0, 0x00
+
+#define kMaskRandom5_3 \
+  0xb0, 0x00, \
+  0xc8, 0x00, \
+  0x50, 0x00
+
+#define kMaskRandom5_4 \
+  0xc8, 0x00, \
+  0xb0, 0x00, \
+  0x50, 0x00, \
+  0x28, 0x00
+
+#define kMaskRandom5_5 \
+  0xc0, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0xa0, 0x00, \
+  0x48, 0x00
+
+#define kMaskRandom6_1 \
+  0xfc, 0x00
+
+#define kMaskRandom6_2 \
+  0xa8, 0x00, \
+  0xd4, 0x00
+
+#define kMaskRandom6_3 \
+  0xd0, 0x00, \
+  0x68, 0x00, \
+  0xa4, 0x00
+
+#define kMaskRandom6_4 \
+  0xa8, 0x00, \
+  0x58, 0x00, \
+  0x64, 0x00, \
+  0x94, 0x00
+
+#define kMaskRandom6_5 \
+  0xa8, 0x00, \
+  0x84, 0x00, \
+  0x64, 0x00, \
+  0x90, 0x00, \
+  0x58, 0x00
+
+#define kMaskRandom6_6 \
+  0x98, 0x00, \
+  0x64, 0x00, \
+  0x50, 0x00, \
+  0x14, 0x00, \
+  0xa8, 0x00, \
+  0xe0, 0x00
+
+#define kMaskRandom7_1 \
+  0xfe, 0x00
+
+#define kMaskRandom7_2 \
+  0xd4, 0x00, \
+  0xaa, 0x00
+
+#define kMaskRandom7_3 \
+  0xd0, 0x00, \
+  0xaa, 0x00, \
+  0x64, 0x00
+
+#define kMaskRandom7_4 \
+  0xd0, 0x00, \
+  0xaa, 0x00, \
+  0x64, 0x00, \
+  0x1c, 0x00
+
+#define kMaskRandom7_5 \
+  0x0c, 0x00, \
+  0xb0, 0x00, \
+  0x1a, 0x00, \
+  0xc4, 0x00, \
+  0x62, 0x00
+
+#define kMaskRandom7_6 \
+  0x8c, 0x00, \
+  0x4a, 0x00, \
+  0x64, 0x00, \
+  0xd0, 0x00, \
+  0xa0, 0x00, \
+  0x32, 0x00
+
+#define kMaskRandom7_7 \
+  0x4a, 0x00, \
+  0x94, 0x00, \
+  0x1a, 0x00, \
+  0xc4, 0x00, \
+  0x28, 0x00, \
+  0xc2, 0x00, \
+  0x34, 0x00
+
+#define kMaskRandom8_1 \
+  0xff, 0x00
+
+#define kMaskRandom8_2 \
+  0xaa, 0x00, \
+  0xd5, 0x00
+
+#define kMaskRandom8_3 \
+  0xc5, 0x00, \
+  0x92, 0x00, \
+  0x6a, 0x00
+
+#define kMaskRandom8_4 \
+  0x45, 0x00, \
+  0xb4, 0x00, \
+  0x6a, 0x00, \
+  0x89, 0x00
+
+#define kMaskRandom8_5 \
+  0x8c, 0x00, \
+  0x92, 0x00, \
+  0x2b, 0x00, \
+  0x51, 0x00, \
+  0x64, 0x00
+
+#define kMaskRandom8_6 \
+  0xa1, 0x00, \
+  0x52, 0x00, \
+  0x91, 0x00, \
+  0x2a, 0x00, \
+  0xc4, 0x00, \
+  0x4c, 0x00
+
+#define kMaskRandom8_7 \
+  0x15, 0x00, \
+  0xc2, 0x00, \
+  0x25, 0x00, \
+  0x62, 0x00, \
+  0x58, 0x00, \
+  0x8c, 0x00, \
+  0xa3, 0x00
+
+#define kMaskRandom8_8 \
+  0x25, 0x00, \
+  0x8a, 0x00, \
+  0x91, 0x00, \
+  0x68, 0x00, \
+  0x32, 0x00, \
+  0x43, 0x00, \
+  0xc4, 0x00, \
+  0x1c, 0x00
+
+#define kMaskRandom9_1 \
+  0xff, 0x80
+
+#define kMaskRandom9_2 \
+  0xaa, 0x80, \
+  0xd5, 0x00
+
+#define kMaskRandom9_3 \
+  0xa5, 0x00, \
+  0xc8, 0x00, \
+  0x52, 0x80
+
+#define kMaskRandom9_4 \
+  0xa2, 0x00, \
+  0xc9, 0x00, \
+  0x52, 0x80, \
+  0x24, 0x80
+
+#define kMaskRandom9_5 \
+  0x8c, 0x00, \
+  0x25, 0x00, \
+  0x92, 0x80, \
+  0x41, 0x80, \
+  0x58, 0x00
+
+#define kMaskRandom9_6 \
+  0x84, 0x80, \
+  0x27, 0x00, \
+  0x51, 0x80, \
+  0x1a, 0x00, \
+  0x68, 0x00, \
+  0x89, 0x00
+
+#define kMaskRandom9_7 \
+  0x8c, 0x00, \
+  0x47, 0x00, \
+  0x81, 0x80, \
+  0x12, 0x80, \
+  0x58, 0x00, \
+  0x28, 0x80, \
+  0xb4, 0x00
+
+#define kMaskRandom9_8 \
+  0x2c, 0x00, \
+  0x91, 0x00, \
+  0x40, 0x80, \
+  0x06, 0x80, \
+  0xc8, 0x00, \
+  0x45, 0x00, \
+  0x30, 0x80, \
+  0xa2, 0x00
+
+#define kMaskRandom9_9 \
+  0x4c, 0x00, \
+  0x62, 0x00, \
+  0x91, 0x00, \
+  0x42, 0x80, \
+  0xa4, 0x00, \
+  0x13, 0x00, \
+  0x30, 0x80, \
+  0x88, 0x80, \
+  0x09, 0x00
+
+#define kMaskRandom10_1 \
+  0xff, 0xc0
+
+#define kMaskRandom10_10 \
+  0x4c, 0x00, \
+  0x51, 0x00, \
+  0xa0, 0x40, \
+  0x04, 0xc0, \
+  0x03, 0x80, \
+  0x86, 0x00, \
+  0x29, 0x00, \
+  0x42, 0x40, \
+  0x98, 0x00, \
+  0x30, 0x80
+
+#define kMaskRandom10_2 \
+  0xaa, 0x80, \
+  0xd5, 0x40
+
+#define kMaskRandom10_3 \
+  0xa4, 0x40, \
+  0xc9, 0x00, \
+  0x52, 0x80
+
+#define kMaskRandom10_4 \
+  0xca, 0x00, \
+  0x32, 0x80, \
+  0xa1, 0x40, \
+  0x55, 0x00
+
+#define kMaskRandom10_5 \
+  0xca, 0x00, \
+  0x32, 0x80, \
+  0xa1, 0x40, \
+  0x55, 0x00, \
+  0x08, 0xc0
+
+#define kMaskRandom10_6 \
+  0x0e, 0x00, \
+  0x33, 0x00, \
+  0x10, 0xc0, \
+  0x45, 0x40, \
+  0x88, 0x80, \
+  0xe0, 0x00
+
+#define kMaskRandom10_7 \
+  0x46, 0x00, \
+  0x33, 0x00, \
+  0x80, 0xc0, \
+  0x0c, 0x40, \
+  0x28, 0x80, \
+  0x94, 0x00, \
+  0xc1, 0x00
+
+#define kMaskRandom10_8 \
+  0x2c, 0x00, \
+  0x81, 0x80, \
+  0xa0, 0x40, \
+  0x05, 0x40, \
+  0x18, 0x80, \
+  0xc2, 0x00, \
+  0x22, 0x80, \
+  0x50, 0x40
+
+#define kMaskRandom10_9 \
+  0x4c, 0x00, \
+  0x23, 0x00, \
+  0x88, 0xc0, \
+  0x21, 0x40, \
+  0x52, 0x80, \
+  0x94, 0x00, \
+  0x26, 0x00, \
+  0x48, 0x40, \
+  0x91, 0x80
+
+#define kMaskRandom11_1 \
+  0xff, 0xe0
+
+#define kMaskRandom11_10 \
+  0x64, 0x40, \
+  0x51, 0x40, \
+  0xa9, 0x00, \
+  0x04, 0xc0, \
+  0xd0, 0x00, \
+  0x82, 0x40, \
+  0x21, 0x20, \
+  0x0c, 0x20, \
+  0x4a, 0x00, \
+  0x12, 0xa0
+
+#define kMaskRandom11_11 \
+  0x46, 0x40, \
+  0x33, 0x20, \
+  0x99, 0x00, \
+  0x05, 0x80, \
+  0x80, 0xa0, \
+  0x84, 0x40, \
+  0x40, 0x60, \
+  0x0a, 0x80, \
+  0x68, 0x00, \
+  0x10, 0x20, \
+  0x30, 0x40
+
+#define kMaskRandom11_2 \
+  0xec, 0xc0, \
+  0x9b, 0xa0
+
+#define kMaskRandom11_3 \
+  0xca, 0xc0, \
+  0xf1, 0x40, \
+  0xb6, 0x20
+
+#define kMaskRandom11_4 \
+  0xc4, 0xc0, \
+  0x31, 0x60, \
+  0x4b, 0x20, \
+  0x2c, 0xa0
+
+#define kMaskRandom11_5 \
+  0x86, 0x80, \
+  0x23, 0x20, \
+  0x16, 0x20, \
+  0x4c, 0x20, \
+  0x41, 0xc0
+
+#define kMaskRandom11_6 \
+  0x64, 0x40, \
+  0x51, 0x40, \
+  0x0c, 0xa0, \
+  0xa1, 0x20, \
+  0x12, 0xa0, \
+  0x8a, 0x40
+
+#define kMaskRandom11_7 \
+  0x46, 0x40, \
+  0x33, 0x20, \
+  0x91, 0x80, \
+  0xa4, 0x20, \
+  0x50, 0xa0, \
+  0x84, 0xc0, \
+  0x09, 0x60
+
+#define kMaskRandom11_8 \
+  0x0c, 0x80, \
+  0x80, 0x60, \
+  0xa0, 0x80, \
+  0x05, 0x40, \
+  0x43, 0x00, \
+  0x1a, 0x00, \
+  0x60, 0x20, \
+  0x14, 0x20
+
+#define kMaskRandom11_9 \
+  0x46, 0x40, \
+  0x62, 0x60, \
+  0x8c, 0x00, \
+  0x01, 0x60, \
+  0x07, 0x80, \
+  0xa0, 0x80, \
+  0x18, 0xa0, \
+  0x91, 0x00, \
+  0x78, 0x00
+
+#define kMaskRandom12_1 \
+  0xff, 0xf0
+
+#define kMaskRandom12_10 \
+  0x51, 0x40, \
+  0x45, 0x10, \
+  0x80, 0xd0, \
+  0x24, 0x20, \
+  0x0a, 0x20, \
+  0x00, 0xe0, \
+  0xb8, 0x00, \
+  0x09, 0x10, \
+  0x56, 0x00, \
+  0xa2, 0x80
+
+#define kMaskRandom12_11 \
+  0x53, 0x60, \
+  0x21, 0x30, \
+  0x10, 0x90, \
+  0x00, 0x70, \
+  0x0c, 0x10, \
+  0x40, 0xc0, \
+  0x6a, 0x00, \
+  0x86, 0x00, \
+  0x24, 0x80, \
+  0x89, 0x00, \
+  0xc0, 0x20
+
+#define kMaskRandom12_12 \
+  0x10, 0x60, \
+  0x02, 0x30, \
+  0x40, 0x50, \
+  0x21, 0x80, \
+  0x81, 0x10, \
+  0x14, 0x80, \
+  0x98, 0x00, \
+  0x08, 0x90, \
+  0x62, 0x00, \
+  0x24, 0x20, \
+  0x8a, 0x00, \
+  0x84, 0x40
+
+#define kMaskRandom12_2 \
+  0xec, 0xc0, \
+  0x93, 0xb0
+
+#define kMaskRandom12_3 \
+  0x9b, 0x80, \
+  0x4f, 0x10, \
+  0x3c, 0x60
+
+#define kMaskRandom12_4 \
+  0x8b, 0x20, \
+  0x14, 0xb0, \
+  0x22, 0xd0, \
+  0x45, 0x50
+
+#define kMaskRandom12_5 \
+  0x53, 0x60, \
+  0x64, 0x20, \
+  0x0c, 0xc0, \
+  0x82, 0xa0, \
+  0x09, 0x30
+
+#define kMaskRandom12_6 \
+  0x51, 0x40, \
+  0xc5, 0x10, \
+  0x21, 0x80, \
+  0x12, 0x30, \
+  0x08, 0xe0, \
+  0x2e, 0x00
+
+#define kMaskRandom12_7 \
+  0x53, 0x60, \
+  0x21, 0x30, \
+  0x90, 0x90, \
+  0x02, 0x50, \
+  0x06, 0xa0, \
+  0x2c, 0x00, \
+  0x88, 0x60
+
+#define kMaskRandom12_8 \
+  0x20, 0x60, \
+  0x80, 0x30, \
+  0x42, 0x40, \
+  0x01, 0x90, \
+  0x14, 0x10, \
+  0x0a, 0x80, \
+  0x38, 0x00, \
+  0xc5, 0x00
+
+#define kMaskRandom12_9 \
+  0x53, 0x60, \
+  0xe4, 0x20, \
+  0x24, 0x40, \
+  0xa1, 0x10, \
+  0x18, 0x30, \
+  0x03, 0x90, \
+  0x8a, 0x10, \
+  0x04, 0x90, \
+  0x00, 0xe0
+
+#define kPacketMaskRandom1 1, \
+  kMaskRandom1_1
+
+#define kPacketMaskRandom2 2, \
+  kMaskRandom2_1, \
+  kMaskRandom2_2
+
+#define kPacketMaskRandom3 3, \
+  kMaskRandom3_1, \
+  kMaskRandom3_2, \
+  kMaskRandom3_3
+
+#define kPacketMaskRandom4 4, \
+  kMaskRandom4_1, \
+  kMaskRandom4_2, \
+  kMaskRandom4_3, \
+  kMaskRandom4_4
+
+#define kPacketMaskRandom5 5, \
+  kMaskRandom5_1, \
+  kMaskRandom5_2, \
+  kMaskRandom5_3, \
+  kMaskRandom5_4, \
+  kMaskRandom5_5
+
+#define kPacketMaskRandom6 6, \
+  kMaskRandom6_1, \
+  kMaskRandom6_2, \
+  kMaskRandom6_3, \
+  kMaskRandom6_4, \
+  kMaskRandom6_5, \
+  kMaskRandom6_6
+
+#define kPacketMaskRandom7 7, \
+  kMaskRandom7_1, \
+  kMaskRandom7_2, \
+  kMaskRandom7_3, \
+  kMaskRandom7_4, \
+  kMaskRandom7_5, \
+  kMaskRandom7_6, \
+  kMaskRandom7_7
+
+#define kPacketMaskRandom8 8, \
+  kMaskRandom8_1, \
+  kMaskRandom8_2, \
+  kMaskRandom8_3, \
+  kMaskRandom8_4, \
+  kMaskRandom8_5, \
+  kMaskRandom8_6, \
+  kMaskRandom8_7, \
+  kMaskRandom8_8
+
+#define kPacketMaskRandom9 9, \
+  kMaskRandom9_1, \
+  kMaskRandom9_2, \
+  kMaskRandom9_3, \
+  kMaskRandom9_4, \
+  kMaskRandom9_5, \
+  kMaskRandom9_6, \
+  kMaskRandom9_7, \
+  kMaskRandom9_8, \
+  kMaskRandom9_9
+
+#define kPacketMaskRandom10 10, \
+  kMaskRandom10_1, \
+  kMaskRandom10_2, \
+  kMaskRandom10_3, \
+  kMaskRandom10_4, \
+  kMaskRandom10_5, \
+  kMaskRandom10_6, \
+  kMaskRandom10_7, \
+  kMaskRandom10_8, \
+  kMaskRandom10_9, \
+  kMaskRandom10_10
+
+#define kPacketMaskRandom11 11, \
+  kMaskRandom11_1, \
+  kMaskRandom11_2, \
+  kMaskRandom11_3, \
+  kMaskRandom11_4, \
+  kMaskRandom11_5, \
+  kMaskRandom11_6, \
+  kMaskRandom11_7, \
+  kMaskRandom11_8, \
+  kMaskRandom11_9, \
+  kMaskRandom11_10, \
+  kMaskRandom11_11
+
+#define kPacketMaskRandom12 12, \
+  kMaskRandom12_1, \
+  kMaskRandom12_2, \
+  kMaskRandom12_3, \
+  kMaskRandom12_4, \
+  kMaskRandom12_5, \
+  kMaskRandom12_6, \
+  kMaskRandom12_7, \
+  kMaskRandom12_8, \
+  kMaskRandom12_9, \
+  kMaskRandom12_10, \
+  kMaskRandom12_11, \
+  kMaskRandom12_12
+
+// clang-format on
+}  // namespace
+
+namespace webrtc {
+namespace fec_private_tables {
+
+const uint8_t kPacketMaskRandomTbl[] = {
+    12,
+    kPacketMaskRandom1,  // 2 byte entries.
+    kPacketMaskRandom2,
+    kPacketMaskRandom3,
+    kPacketMaskRandom4,
+    kPacketMaskRandom5,
+    kPacketMaskRandom6,
+    kPacketMaskRandom7,
+    kPacketMaskRandom8,
+    kPacketMaskRandom9,
+    kPacketMaskRandom10,
+    kPacketMaskRandom11,
+    kPacketMaskRandom12,
+};
+
+}  // namespace fec_private_tables
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h

@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
+#define MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
+
+// This file contains a set of packets masks for the FEC code. The masks in
+// this table are specifically designed to favor recovery to random loss.
+// These packet masks are defined to protect up to maximum of 48 media packets.
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace fec_private_tables {
+
+extern const uint8_t kPacketMaskRandomTbl[];
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc

@@ -0,0 +1,230 @@
+/*
+ *  Copyright (c) 2012 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 "modules/rtp_rtcp/source/fec_test_helper.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+namespace fec {
+
+namespace {
+
+constexpr uint8_t kRtpMarkerBitMask = 0x80;
+
+constexpr uint8_t kFecPayloadType = 96;
+constexpr uint8_t kRedPayloadType = 97;
+constexpr uint8_t kVp8PayloadType = 120;
+
+constexpr int kPacketTimestampIncrement = 3000;
+}  // namespace
+
+MediaPacketGenerator::MediaPacketGenerator(uint32_t min_packet_size,
+                                           uint32_t max_packet_size,
+                                           uint32_t ssrc,
+                                           Random* random)
+    : min_packet_size_(min_packet_size),
+      max_packet_size_(max_packet_size),
+      ssrc_(ssrc),
+      random_(random) {}
+
+MediaPacketGenerator::~MediaPacketGenerator() = default;
+
+ForwardErrorCorrection::PacketList MediaPacketGenerator::ConstructMediaPackets(
+    int num_media_packets,
+    uint16_t start_seq_num) {
+  RTC_DCHECK_GT(num_media_packets, 0);
+  uint16_t seq_num = start_seq_num;
+  int time_stamp = random_->Rand<int>();
+
+  ForwardErrorCorrection::PacketList media_packets;
+
+  for (int i = 0; i < num_media_packets; ++i) {
+    std::unique_ptr<ForwardErrorCorrection::Packet> media_packet(
+        new ForwardErrorCorrection::Packet());
+    media_packet->data.SetSize(
+        random_->Rand(min_packet_size_, max_packet_size_));
+
+    uint8_t* data = media_packet->data.MutableData();
+    // Generate random values for the first 2 bytes
+    data[0] = random_->Rand<uint8_t>();
+    data[1] = random_->Rand<uint8_t>();
+
+    // The first two bits are assumed to be 10 by the FEC encoder.
+    // In fact the FEC decoder will set the two first bits to 10 regardless of
+    // what they actually were. Set the first two bits to 10 so that a memcmp
+    // can be performed for the whole restored packet.
+    data[0] |= 0x80;
+    data[0] &= 0xbf;
+
+    // FEC is applied to a whole frame.
+    // A frame is signaled by multiple packets without the marker bit set
+    // followed by the last packet of the frame for which the marker bit is set.
+    // Only push one (fake) frame to the FEC.
+    data[1] &= 0x7f;
+
+    webrtc::ByteWriter<uint16_t>::WriteBigEndian(&data[2], seq_num);
+    webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[4], time_stamp);
+    webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[8], ssrc_);
+
+    // Generate random values for payload.
+    for (size_t j = 12; j < media_packet->data.size(); ++j)
+      data[j] = random_->Rand<uint8_t>();
+    seq_num++;
+    media_packets.push_back(std::move(media_packet));
+  }
+  // Last packet, set marker bit.
+  ForwardErrorCorrection::Packet* media_packet = media_packets.back().get();
+  RTC_DCHECK(media_packet);
+  media_packet->data.MutableData()[1] |= 0x80;
+
+  next_seq_num_ = seq_num;
+
+  return media_packets;
+}
+
+ForwardErrorCorrection::PacketList MediaPacketGenerator::ConstructMediaPackets(
+    int num_media_packets) {
+  return ConstructMediaPackets(num_media_packets, random_->Rand<uint16_t>());
+}
+
+uint16_t MediaPacketGenerator::GetNextSeqNum() {
+  return next_seq_num_;
+}
+
+AugmentedPacketGenerator::AugmentedPacketGenerator(uint32_t ssrc)
+    : num_packets_(0), ssrc_(ssrc), seq_num_(0), timestamp_(0) {}
+
+void AugmentedPacketGenerator::NewFrame(size_t num_packets) {
+  num_packets_ = num_packets;
+  timestamp_ += kPacketTimestampIncrement;
+}
+
+uint16_t AugmentedPacketGenerator::NextPacketSeqNum() {
+  return ++seq_num_;
+}
+
+std::unique_ptr<AugmentedPacket> AugmentedPacketGenerator::NextPacket(
+    size_t offset,
+    size_t length) {
+  std::unique_ptr<AugmentedPacket> packet(new AugmentedPacket());
+
+  packet->data.SetSize(length + kRtpHeaderSize);
+  uint8_t* data = packet->data.MutableData();
+  for (size_t i = 0; i < length; ++i)
+    data[i + kRtpHeaderSize] = offset + i;
+  packet->data.SetSize(length + kRtpHeaderSize);
+  packet->header.headerLength = kRtpHeaderSize;
+  packet->header.markerBit = (num_packets_ == 1);
+  packet->header.payloadType = kVp8PayloadType;
+  packet->header.sequenceNumber = seq_num_;
+  packet->header.timestamp = timestamp_;
+  packet->header.ssrc = ssrc_;
+  WriteRtpHeader(packet->header, data);
+  ++seq_num_;
+  --num_packets_;
+
+  return packet;
+}
+
+void AugmentedPacketGenerator::WriteRtpHeader(const RTPHeader& header,
+                                              uint8_t* data) {
+  data[0] = 0x80;  // Version 2.
+  data[1] = header.payloadType;
+  data[1] |= (header.markerBit ? kRtpMarkerBitMask : 0);
+  ByteWriter<uint16_t>::WriteBigEndian(data + 2, header.sequenceNumber);
+  ByteWriter<uint32_t>::WriteBigEndian(data + 4, header.timestamp);
+  ByteWriter<uint32_t>::WriteBigEndian(data + 8, header.ssrc);
+}
+
+FlexfecPacketGenerator::FlexfecPacketGenerator(uint32_t media_ssrc,
+                                               uint32_t flexfec_ssrc)
+    : AugmentedPacketGenerator(media_ssrc),
+      flexfec_ssrc_(flexfec_ssrc),
+      flexfec_seq_num_(0),
+      flexfec_timestamp_(0) {}
+
+std::unique_ptr<AugmentedPacket> FlexfecPacketGenerator::BuildFlexfecPacket(
+    const ForwardErrorCorrection::Packet& packet) {
+  RTC_DCHECK_LE(packet.data.size(),
+                static_cast<size_t>(IP_PACKET_SIZE - kRtpHeaderSize));
+
+  RTPHeader header;
+  header.sequenceNumber = flexfec_seq_num_;
+  ++flexfec_seq_num_;
+  header.timestamp = flexfec_timestamp_;
+  flexfec_timestamp_ += kPacketTimestampIncrement;
+  header.ssrc = flexfec_ssrc_;
+
+  std::unique_ptr<AugmentedPacket> packet_with_rtp_header(
+      new AugmentedPacket());
+  packet_with_rtp_header->data.SetSize(kRtpHeaderSize + packet.data.size());
+  WriteRtpHeader(header, packet_with_rtp_header->data.MutableData());
+  memcpy(packet_with_rtp_header->data.MutableData() + kRtpHeaderSize,
+         packet.data.cdata(), packet.data.size());
+
+  return packet_with_rtp_header;
+}
+
+UlpfecPacketGenerator::UlpfecPacketGenerator(uint32_t ssrc)
+    : AugmentedPacketGenerator(ssrc) {}
+
+RtpPacketReceived UlpfecPacketGenerator::BuildMediaRedPacket(
+    const AugmentedPacket& packet,
+    bool is_recovered) {
+  // Create a temporary buffer used to wrap the media packet in RED.
+  rtc::CopyOnWriteBuffer red_buffer;
+  const size_t kHeaderLength = packet.header.headerLength;
+  // Append header.
+  red_buffer.SetData(packet.data.data(), kHeaderLength);
+  // Find payload type and add it as RED header.
+  uint8_t media_payload_type = red_buffer[1] & 0x7F;
+  red_buffer.AppendData({media_payload_type});
+  // Append rest of payload/padding.
+  red_buffer.AppendData(
+      packet.data.Slice(kHeaderLength, packet.data.size() - kHeaderLength));
+
+  RtpPacketReceived red_packet;
+  RTC_CHECK(red_packet.Parse(std::move(red_buffer)));
+  red_packet.SetPayloadType(kRedPayloadType);
+  red_packet.set_recovered(is_recovered);
+
+  return red_packet;
+}
+
+RtpPacketReceived UlpfecPacketGenerator::BuildUlpfecRedPacket(
+    const ForwardErrorCorrection::Packet& packet) {
+  // Create a fake media packet to get a correct header. 1 byte RED header.
+  ++num_packets_;
+  std::unique_ptr<AugmentedPacket> fake_packet =
+      NextPacket(0, packet.data.size() + 1);
+
+  RtpPacketReceived red_packet;
+  red_packet.Parse(fake_packet->data);
+  red_packet.SetMarker(false);
+  uint8_t* rtp_payload = red_packet.AllocatePayload(packet.data.size() + 1);
+  rtp_payload[0] = kFecPayloadType;
+  red_packet.SetPayloadType(kRedPayloadType);
+  memcpy(rtp_payload + 1, packet.data.cdata(), packet.data.size());
+  red_packet.set_recovered(false);
+
+  return red_packet;
+}
+
+}  // namespace fec
+}  // namespace test
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/fec_test_helper.h

@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
+#define MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
+
+#include <memory>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace test {
+namespace fec {
+
+struct AugmentedPacket : public ForwardErrorCorrection::Packet {
+  RTPHeader header;
+};
+
+// TODO(brandtr): Consider merging MediaPacketGenerator and
+// AugmentedPacketGenerator into a single class, since their functionality is
+// similar.
+
+// This class generates media packets corresponding to a single frame.
+class MediaPacketGenerator {
+ public:
+  MediaPacketGenerator(uint32_t min_packet_size,
+                       uint32_t max_packet_size,
+                       uint32_t ssrc,
+                       Random* random);
+  ~MediaPacketGenerator();
+
+  // Construct the media packets, up to `num_media_packets` packets.
+  ForwardErrorCorrection::PacketList ConstructMediaPackets(
+      int num_media_packets,
+      uint16_t start_seq_num);
+  ForwardErrorCorrection::PacketList ConstructMediaPackets(
+      int num_media_packets);
+
+  uint16_t GetNextSeqNum();
+
+ private:
+  uint32_t min_packet_size_;
+  uint32_t max_packet_size_;
+  uint32_t ssrc_;
+  Random* random_;
+
+  ForwardErrorCorrection::PacketList media_packets_;
+  uint16_t next_seq_num_;
+};
+
+// This class generates media packets with a certain structure of the payload.
+class AugmentedPacketGenerator {
+ public:
+  explicit AugmentedPacketGenerator(uint32_t ssrc);
+
+  // Prepare for generating a new set of packets, corresponding to a frame.
+  void NewFrame(size_t num_packets);
+
+  // Increment and return the newly incremented sequence number.
+  uint16_t NextPacketSeqNum();
+
+  // Return the next packet in the current frame.
+  std::unique_ptr<AugmentedPacket> NextPacket(size_t offset, size_t length);
+
+ protected:
+  // Given `header`, writes the appropriate RTP header fields in `data`.
+  static void WriteRtpHeader(const RTPHeader& header, uint8_t* data);
+
+  // Number of packets left to generate, in the current frame.
+  size_t num_packets_;
+
+ private:
+  uint32_t ssrc_;
+  uint16_t seq_num_;
+  uint32_t timestamp_;
+};
+
+// This class generates media and FlexFEC packets for a single frame.
+class FlexfecPacketGenerator : public AugmentedPacketGenerator {
+ public:
+  FlexfecPacketGenerator(uint32_t media_ssrc, uint32_t flexfec_ssrc);
+
+  // Creates a new AugmentedPacket (with RTP headers) from a
+  // FlexFEC packet (without RTP headers).
+  std::unique_ptr<AugmentedPacket> BuildFlexfecPacket(
+      const ForwardErrorCorrection::Packet& packet);
+
+ private:
+  uint32_t flexfec_ssrc_;
+  uint16_t flexfec_seq_num_;
+  uint32_t flexfec_timestamp_;
+};
+
+// This class generates media and ULPFEC packets (both encapsulated in RED)
+// for a single frame.
+class UlpfecPacketGenerator : public AugmentedPacketGenerator {
+ public:
+  explicit UlpfecPacketGenerator(uint32_t ssrc);
+
+  // Creates a new RtpPacket with the RED header added to the packet.
+  static RtpPacketReceived BuildMediaRedPacket(const AugmentedPacket& packet,
+                                               bool is_recovered);
+
+  // Creates a new RtpPacket with FEC payload and RED header. Does this by
+  // creating a new fake media AugmentedPacket, clears the marker bit and adds a
+  // RED header. Finally replaces the payload with the content of
+  // `packet->data`.
+  RtpPacketReceived BuildUlpfecRedPacket(
+      const ForwardErrorCorrection::Packet& packet);
+};
+
+}  // namespace fec
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc

@@ -0,0 +1,319 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h"
+
+#include <string.h>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Maximum number of media packets that can be protected in one batch.
+constexpr size_t kMaxMediaPackets = 48;  // Since we are reusing ULPFEC masks.
+
+// Maximum number of media packets tracked by FEC decoder.
+// Maintain a sufficiently larger tracking window than `kMaxMediaPackets`
+// to account for packet reordering in pacer/ network.
+constexpr size_t kMaxTrackedMediaPackets = 4 * kMaxMediaPackets;
+
+// Maximum number of FEC packets stored inside ForwardErrorCorrection.
+constexpr size_t kMaxFecPackets = kMaxMediaPackets;
+
+// Size (in bytes) of packet masks, given number of K bits set.
+constexpr size_t kFlexfecPacketMaskSizes[] = {2, 6, 14};
+
+// Size (in bytes) of part of header which is not packet mask specific.
+constexpr size_t kBaseHeaderSize = 12;
+
+// Size (in bytes) of part of header which is stream specific.
+constexpr size_t kStreamSpecificHeaderSize = 6;
+
+// Size (in bytes) of header, given the single stream packet mask size, i.e.
+// the number of K-bits set.
+constexpr size_t kHeaderSizes[] = {
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[0],
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[1],
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[2]};
+
+// Flexfec03 implementation only support single-stream protection.
+// For multi-stream protection use implementation of the FlexFEC final standard.
+constexpr uint8_t kSsrcCount = 1;
+
+// There are three reserved bytes that MUST be set to zero in the header.
+constexpr uint32_t kReservedBits = 0;
+
+constexpr size_t kPacketMaskOffset =
+    kBaseHeaderSize + kStreamSpecificHeaderSize;
+
+// Here we count the K-bits as belonging to the packet mask.
+// This can be used in conjunction with
+// Flexfec03HeaderWriter::MinPacketMaskSize, which calculates a bound on the
+// needed packet mask size including K-bits, given a packet mask without K-bits.
+size_t FlexfecHeaderSize(size_t packet_mask_size) {
+  RTC_DCHECK_LE(packet_mask_size, kFlexfecPacketMaskSizes[2]);
+  if (packet_mask_size <= kFlexfecPacketMaskSizes[0]) {
+    return kHeaderSizes[0];
+  } else if (packet_mask_size <= kFlexfecPacketMaskSizes[1]) {
+    return kHeaderSizes[1];
+  }
+  return kHeaderSizes[2];
+}
+
+}  // namespace
+
+Flexfec03HeaderReader::Flexfec03HeaderReader()
+    : FecHeaderReader(kMaxTrackedMediaPackets, kMaxFecPackets) {}
+
+Flexfec03HeaderReader::~Flexfec03HeaderReader() = default;
+
+// Flexfec03 implementation doesn't support retransmission and flexible masks.
+// When that features are desired, they should be implemented in the class that
+// follows final version of the FlexFEC standard.
+bool Flexfec03HeaderReader::ReadFecHeader(
+    ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const {
+  if (fec_packet->pkt->data.size() <=
+      kBaseHeaderSize + kStreamSpecificHeaderSize) {
+    RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+    return false;
+  }
+  uint8_t* const data = fec_packet->pkt->data.MutableData();
+  bool r_bit = (data[0] & 0x80) != 0;
+  if (r_bit) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC03 packet with retransmission bit set. We do not "
+           "support this, thus discarding the packet.";
+    return false;
+  }
+  bool f_bit = (data[0] & 0x40) != 0;
+  if (f_bit) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC03 packet with inflexible generator matrix. We do "
+           "not support this, thus discarding packet.";
+    return false;
+  }
+  uint8_t ssrc_count = ByteReader<uint8_t>::ReadBigEndian(&data[8]);
+  if (ssrc_count != 1) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC03 packet protecting multiple media SSRCs. We do not "
+           "support this, thus discarding packet.";
+    return false;
+  }
+  uint32_t protected_ssrc = ByteReader<uint32_t>::ReadBigEndian(&data[12]);
+  uint16_t seq_num_base = ByteReader<uint16_t>::ReadBigEndian(&data[16]);
+
+  // Parse the FlexFEC packet mask and remove the interleaved K-bits.
+  // (See FEC header schematic in flexfec_header_reader_writer.h.)
+  // We store the packed packet mask in-band, which "destroys" the standards
+  // compliance of the header. That is fine though, since the code that
+  // reads from the header (from this point and onwards) is aware of this.
+  // TODO(brandtr): When the FEC packet classes have been refactored, store
+  // the packed packet masks out-of-band, thus leaving the FlexFEC header as is.
+  //
+  // We treat the mask parts as unsigned integers with host order endianness
+  // in order to simplify the bit shifting between bytes.
+  if (fec_packet->pkt->data.size() < kHeaderSizes[0]) {
+    RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC03 packet.";
+    return false;
+  }
+  uint8_t* const packet_mask = data + kPacketMaskOffset;
+  bool k_bit0 = (packet_mask[0] & 0x80) != 0;
+  uint16_t mask_part0 = ByteReader<uint16_t>::ReadBigEndian(&packet_mask[0]);
+  // Shift away K-bit 0, implicitly clearing the last bit.
+  mask_part0 <<= 1;
+  ByteWriter<uint16_t>::WriteBigEndian(&packet_mask[0], mask_part0);
+  size_t packet_mask_size;
+  if (k_bit0) {
+    // The first K-bit is set, and the packet mask is thus only 2 bytes long.
+    // We have now read the entire FEC header, and the rest of the packet
+    // is payload.
+    packet_mask_size = kFlexfecPacketMaskSizes[0];
+  } else {
+    if (fec_packet->pkt->data.size() < kHeaderSizes[1]) {
+      return false;
+    }
+    bool k_bit1 = (packet_mask[2] & 0x80) != 0;
+    // We have already shifted the first two bytes of the packet mask one step
+    // to the left, thus removing K-bit 0. We will now shift the next four bytes
+    // of the packet mask two steps to the left. (One step for the removed
+    // K-bit 0, and one step for the to be removed K-bit 1).
+    uint8_t bit15 = (packet_mask[2] >> 6) & 0x01;
+    packet_mask[1] |= bit15;
+    uint32_t mask_part1 = ByteReader<uint32_t>::ReadBigEndian(&packet_mask[2]);
+    // Shift away K-bit 1 and bit 15, implicitly clearing the last two bits.
+    mask_part1 <<= 2;
+    ByteWriter<uint32_t>::WriteBigEndian(&packet_mask[2], mask_part1);
+    if (k_bit1) {
+      // The first K-bit is clear, but the second K-bit is set. The packet
+      // mask is thus 6 bytes long.  We have now read the entire FEC header,
+      // and the rest of the packet is payload.
+      packet_mask_size = kFlexfecPacketMaskSizes[1];
+    } else {
+      if (fec_packet->pkt->data.size() < kHeaderSizes[2]) {
+        RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC03 packet.";
+        return false;
+      }
+      bool k_bit2 = (packet_mask[6] & 0x80) != 0;
+      if (k_bit2) {
+        // The first and second K-bits are clear, but the third K-bit is set.
+        // The packet mask is thus 14 bytes long. We have now read the entire
+        // FEC header, and the rest of the packet is payload.
+        packet_mask_size = kFlexfecPacketMaskSizes[2];
+      } else {
+        RTC_LOG(LS_WARNING)
+            << "Discarding FlexFEC03 packet with malformed header.";
+        return false;
+      }
+      // At this point, K-bits 0 and 1 have been removed, and the front-most
+      // part of the FlexFEC packet mask has been packed accordingly. We will
+      // now shift the remaining part of the packet mask three steps to
+      // the left. This corresponds to the (in total) three K-bits, which
+      // have been removed.
+      uint8_t tail_bits = (packet_mask[6] >> 5) & 0x03;
+      packet_mask[5] |= tail_bits;
+      uint64_t mask_part2 =
+          ByteReader<uint64_t>::ReadBigEndian(&packet_mask[6]);
+      // Shift away K-bit 2, bit 46, and bit 47, implicitly clearing the last
+      // three bits.
+      mask_part2 <<= 3;
+      ByteWriter<uint64_t>::WriteBigEndian(&packet_mask[6], mask_part2);
+    }
+  }
+
+  // Store "ULPFECized" packet mask info.
+  fec_packet->fec_header_size = FlexfecHeaderSize(packet_mask_size);
+  fec_packet->protected_streams = {{.ssrc = protected_ssrc,
+                                    .seq_num_base = seq_num_base,
+                                    .packet_mask_offset = kPacketMaskOffset,
+                                    .packet_mask_size = packet_mask_size}};
+  // In FlexFEC, all media packets are protected in their entirety.
+  fec_packet->protection_length =
+      fec_packet->pkt->data.size() - fec_packet->fec_header_size;
+
+  return true;
+}
+
+Flexfec03HeaderWriter::Flexfec03HeaderWriter()
+    : FecHeaderWriter(kMaxMediaPackets, kMaxFecPackets, kHeaderSizes[2]) {}
+
+Flexfec03HeaderWriter::~Flexfec03HeaderWriter() = default;
+
+size_t Flexfec03HeaderWriter::MinPacketMaskSize(const uint8_t* packet_mask,
+                                                size_t packet_mask_size) const {
+  if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear &&
+      (packet_mask[1] & 0x01) == 0) {
+    // Packet mask is 16 bits long, with bit 15 clear.
+    // It can be used as is.
+    return kFlexfecPacketMaskSizes[0];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear) {
+    // Packet mask is 16 bits long, with bit 15 set.
+    // We must expand the packet mask with zeros in the FlexFEC header.
+    return kFlexfecPacketMaskSizes[1];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet &&
+             (packet_mask[5] & 0x03) == 0) {
+    // Packet mask is 48 bits long, with bits 46 and 47 clear.
+    // It can be used as is.
+    return kFlexfecPacketMaskSizes[1];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet) {
+    // Packet mask is 48 bits long, with at least one of bits 46 and 47 set.
+    // We must expand it with zeros.
+    return kFlexfecPacketMaskSizes[2];
+  }
+  RTC_DCHECK_NOTREACHED() << "Incorrect packet mask size: " << packet_mask_size
+                          << ".";
+  return kFlexfecPacketMaskSizes[2];
+}
+
+size_t Flexfec03HeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
+  return FlexfecHeaderSize(packet_mask_size);
+}
+
+// This function adapts the precomputed ULPFEC packet masks to the
+// FlexFEC header standard. Note that the header size is computed by
+// FecHeaderSize(), so in this function we can be sure that we are
+// writing in space that is intended for the header.
+void Flexfec03HeaderWriter::FinalizeFecHeader(
+    rtc::ArrayView<const ProtectedStream> protected_streams,
+    ForwardErrorCorrection::Packet& fec_packet) const {
+  RTC_CHECK_EQ(protected_streams.size(), 1);
+  uint32_t media_ssrc = protected_streams[0].ssrc;
+  uint16_t seq_num_base = protected_streams[0].seq_num_base;
+  const uint8_t* packet_mask = protected_streams[0].packet_mask.data();
+  size_t packet_mask_size = protected_streams[0].packet_mask.size();
+
+  uint8_t* data = fec_packet.data.MutableData();
+  data[0] &= 0x7f;  // Clear R bit.
+  data[0] &= 0xbf;  // Clear F bit.
+  ByteWriter<uint8_t>::WriteBigEndian(&data[8], kSsrcCount);
+  ByteWriter<uint32_t, 3>::WriteBigEndian(&data[9], kReservedBits);
+  ByteWriter<uint32_t>::WriteBigEndian(&data[12], media_ssrc);
+  ByteWriter<uint16_t>::WriteBigEndian(&data[16], seq_num_base);
+  // Adapt ULPFEC packet mask to FlexFEC header.
+  //
+  // We treat the mask parts as unsigned integers with host order endianness
+  // in order to simplify the bit shifting between bytes.
+  uint8_t* const written_packet_mask = data + kPacketMaskOffset;
+  if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet) {
+    // The packet mask is 48 bits long.
+    uint16_t tmp_mask_part0 =
+        ByteReader<uint16_t>::ReadBigEndian(&packet_mask[0]);
+    uint32_t tmp_mask_part1 =
+        ByteReader<uint32_t>::ReadBigEndian(&packet_mask[2]);
+
+    tmp_mask_part0 >>= 1;  // Shift, thus clearing K-bit 0.
+    ByteWriter<uint16_t>::WriteBigEndian(&written_packet_mask[0],
+                                         tmp_mask_part0);
+    tmp_mask_part1 >>= 2;  // Shift, thus clearing K-bit 1 and bit 15.
+    ByteWriter<uint32_t>::WriteBigEndian(&written_packet_mask[2],
+                                         tmp_mask_part1);
+    bool bit15 = (packet_mask[1] & 0x01) != 0;
+    if (bit15)
+      written_packet_mask[2] |= 0x40;  // Set bit 15.
+    bool bit46 = (packet_mask[5] & 0x02) != 0;
+    bool bit47 = (packet_mask[5] & 0x01) != 0;
+    if (!bit46 && !bit47) {
+      written_packet_mask[2] |= 0x80;  // Set K-bit 1.
+    } else {
+      memset(&written_packet_mask[6], 0, 8);  // Clear all trailing bits.
+      written_packet_mask[6] |= 0x80;         // Set K-bit 2.
+      if (bit46)
+        written_packet_mask[6] |= 0x40;  // Set bit 46.
+      if (bit47)
+        written_packet_mask[6] |= 0x20;  // Set bit 47.
+    }
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear) {
+    // The packet mask is 16 bits long.
+    uint16_t tmp_mask_part0 =
+        ByteReader<uint16_t>::ReadBigEndian(&packet_mask[0]);
+
+    tmp_mask_part0 >>= 1;  // Shift, thus clearing K-bit 0.
+    ByteWriter<uint16_t>::WriteBigEndian(&written_packet_mask[0],
+                                         tmp_mask_part0);
+    bool bit15 = (packet_mask[1] & 0x01) != 0;
+    if (!bit15) {
+      written_packet_mask[0] |= 0x80;  // Set K-bit 0.
+    } else {
+      memset(&written_packet_mask[2], 0U, 4);  // Clear all trailing bits.
+      written_packet_mask[2] |= 0x80;          // Set K-bit 1.
+      written_packet_mask[2] |= 0x40;          // Set bit 15.
+    }
+  } else {
+    RTC_DCHECK_NOTREACHED()
+        << "Incorrect packet mask size: " << packet_mask_size << ".";
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h

@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_FLEXFEC_03_HEADER_READER_WRITER_H_
+#define MODULES_RTP_RTCP_SOURCE_FLEXFEC_03_HEADER_READER_WRITER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+namespace webrtc {
+
+// FlexFEC header, minimum 20 bytes.
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |R|F|P|X|  CC   |M| PT recovery |        length recovery        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          TS recovery                          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |   SSRCCount   |                    reserved                   |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// 12 |                             SSRC_i                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |           SN base_i           |k|          Mask [0-14]        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |k|                   Mask [15-45] (optional)                   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 24 |k|                                                             |
+//    +-+                   Mask [46-108] (optional)                  |
+// 28 |                                                               |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//    :                     ... next in SSRC_i ...                    :
+//
+//
+// FlexFEC header in 'inflexible' mode (F = 1), 20 bytes.
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |0|1|P|X|  CC   |M| PT recovery |        length recovery        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          TS recovery                          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |   SSRCCount   |                    reserved                   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                             SSRC_i                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |           SN base_i           |  M (columns)  |    N (rows)   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+class Flexfec03HeaderReader : public FecHeaderReader {
+ public:
+  Flexfec03HeaderReader();
+  ~Flexfec03HeaderReader() override;
+
+  bool ReadFecHeader(
+      ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const override;
+};
+
+class Flexfec03HeaderWriter : public FecHeaderWriter {
+ public:
+  Flexfec03HeaderWriter();
+  ~Flexfec03HeaderWriter() override;
+
+  size_t MinPacketMaskSize(const uint8_t* packet_mask,
+                           size_t packet_mask_size) const override;
+
+  size_t FecHeaderSize(size_t packet_mask_row_size) const override;
+
+  void FinalizeFecHeader(
+      rtc::ArrayView<const ProtectedStream> protected_streams,
+      ForwardErrorCorrection::Packet& fec_packet) const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FLEXFEC_03_HEADER_READER_WRITER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc

@@ -0,0 +1,328 @@
+/*
+ *  Copyright (c) 2023 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 "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
+
+#include <string.h>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Maximum number of media packets that can be protected in one batch.
+constexpr size_t kMaxMediaPackets = 48;  // Since we are reusing ULPFEC masks.
+
+// Maximum number of media packets tracked by FEC decoder.
+// Maintain a sufficiently larger tracking window than `kMaxMediaPackets`
+// to account for packet reordering in pacer/ network.
+constexpr size_t kMaxTrackedMediaPackets = 4 * kMaxMediaPackets;
+
+// Maximum number of FEC packets stored inside ForwardErrorCorrection.
+constexpr size_t kMaxFecPackets = kMaxMediaPackets;
+
+// Size (in bytes) of packet masks, given number of K bits set.
+constexpr size_t kFlexfecPacketMaskSizes[] = {2, 6, 14};
+
+// Size (in bytes) of part of header which is not packet mask specific.
+constexpr size_t kBaseHeaderSize = 8;
+
+// Size (in bytes) of part of header which is stream specific.
+constexpr size_t kStreamSpecificHeaderSize = 2;
+
+// Size (in bytes) of header, given the single stream packet mask size, i.e.
+// the number of K-bits set.
+constexpr size_t kHeaderSizes[] = {
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[0],
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[1],
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[2]};
+
+// Here we count the K-bits as belonging to the packet mask.
+// This can be used in conjunction with FlexfecHeaderWriter::MinPacketMaskSize,
+// which calculates a bound on the needed packet mask size including K-bits,
+// given a packet mask without K-bits.
+size_t FlexfecHeaderSize(size_t packet_mask_size) {
+  RTC_DCHECK_LE(packet_mask_size, kFlexfecPacketMaskSizes[2]);
+  if (packet_mask_size <= kFlexfecPacketMaskSizes[0]) {
+    return kHeaderSizes[0];
+  } else if (packet_mask_size <= kFlexfecPacketMaskSizes[1]) {
+    return kHeaderSizes[1];
+  }
+  return kHeaderSizes[2];
+}
+
+}  // namespace
+
+FlexfecHeaderReader::FlexfecHeaderReader()
+    : FecHeaderReader(kMaxTrackedMediaPackets, kMaxFecPackets) {}
+
+FlexfecHeaderReader::~FlexfecHeaderReader() = default;
+
+// TODO(brandtr): Update this function when we support flexible masks,
+// and retransmissions.
+bool FlexfecHeaderReader::ReadFecHeader(
+    ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const {
+  // Protected ssrcs should already be populated from RTP header.
+  if (fec_packet->protected_streams.empty()) {
+    RTC_LOG(LS_WARNING)
+        << "Discarding FlexFEC packet with no protected sources.";
+    return false;
+  }
+  if (fec_packet->pkt->data.size() <=
+      kBaseHeaderSize + kStreamSpecificHeaderSize) {
+    RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+    return false;
+  }
+  uint8_t* const data = fec_packet->pkt->data.MutableData();
+  bool r_bit = (data[0] & 0x80) != 0;
+  if (r_bit) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC packet with retransmission bit set. We do not yet "
+           "support this, thus discarding the packet.";
+    return false;
+  }
+  bool f_bit = (data[0] & 0x40) != 0;
+  if (f_bit) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC packet with inflexible generator matrix. We do "
+           "not yet support this, thus discarding packet.";
+    return false;
+  }
+
+  // First seq_num will be in byte index 8
+  // (See FEC header schematic in flexfec_header_reader_writer.h.)
+  size_t byte_index = 8;
+  for (size_t i = 0; i < fec_packet->protected_streams.size(); ++i) {
+    if (fec_packet->pkt->data.size() < byte_index + kStreamSpecificHeaderSize) {
+      RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+      return false;
+    }
+
+    fec_packet->protected_streams[i].seq_num_base =
+        ByteReader<uint16_t>::ReadBigEndian(&data[byte_index]);
+    byte_index += kStreamSpecificHeaderSize;
+
+    // Parse the FlexFEC packet mask and remove the interleaved K-bits.
+    // (See FEC header schematic in flexfec_header_reader_writer.h.)
+    // We store the packed packet mask in-band, which "destroys" the standards
+    // compliance of the header. That is fine though, since the code that
+    // reads from the header (from this point and onwards) is aware of this.
+    // TODO(brandtr): When the FEC packet classes have been refactored, store
+    // the packed packet masks out-of-band, thus leaving the FlexFEC header as
+    // is.
+    //
+    // We treat the mask parts as unsigned integers with host order endianness
+    // in order to simplify the bit shifting between bytes.
+    if (fec_packet->pkt->data.size() <
+        (byte_index + kFlexfecPacketMaskSizes[0])) {
+      RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+      return false;
+    }
+    fec_packet->protected_streams[i].packet_mask_offset = byte_index;
+    bool k_bit0 = (data[byte_index] & 0x80) != 0;
+    uint16_t mask_part0 =
+        ByteReader<uint16_t>::ReadBigEndian(&data[byte_index]);
+    // Shift away K-bit 0, implicitly clearing the last bit.
+    mask_part0 <<= 1;
+    ByteWriter<uint16_t>::WriteBigEndian(&data[byte_index], mask_part0);
+    byte_index += kFlexfecPacketMaskSizes[0];
+    if (!k_bit0) {
+      // The first K-bit is clear, and the packet mask is thus only 2 bytes
+      // long. We have finished reading the properties for current ssrc.
+      fec_packet->protected_streams[i].packet_mask_size =
+          kFlexfecPacketMaskSizes[0];
+    } else {
+      if (fec_packet->pkt->data.size() <
+          (byte_index + kFlexfecPacketMaskSizes[1] -
+           kFlexfecPacketMaskSizes[0])) {
+        return false;
+      }
+      bool k_bit1 = (data[byte_index] & 0x80) != 0;
+      // We have already shifted the first two bytes of the packet mask one step
+      // to the left, thus removing K-bit 0. We will now shift the next four
+      // bytes of the packet mask two steps to the left. (One step for the
+      // removed K-bit 0, and one step for the to be removed K-bit 1).
+      uint8_t bit15 = (data[byte_index] >> 6) & 0x01;
+      data[byte_index - 1] |= bit15;
+      uint32_t mask_part1 =
+          ByteReader<uint32_t>::ReadBigEndian(&data[byte_index]);
+      // Shift away K-bit 1 and bit 15, implicitly clearing the last two bits.
+      mask_part1 <<= 2;
+      ByteWriter<uint32_t>::WriteBigEndian(&data[byte_index], mask_part1);
+      byte_index += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0];
+      if (!k_bit1) {
+        // The first K-bit is set, but the second K-bit is clear. The packet
+        // mask is thus 6 bytes long. We have finished reading the properties
+        // for current ssrc.
+        fec_packet->protected_streams[i].packet_mask_size =
+            kFlexfecPacketMaskSizes[1];
+      } else {
+        if (fec_packet->pkt->data.size() <
+            (byte_index + kFlexfecPacketMaskSizes[2] -
+             kFlexfecPacketMaskSizes[1])) {
+          RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+          return false;
+        }
+        fec_packet->protected_streams[i].packet_mask_size =
+            kFlexfecPacketMaskSizes[2];
+        // At this point, K-bits 0 and 1 have been removed, and the front-most
+        // part of the FlexFEC packet mask has been packed accordingly. We will
+        // now shift the remaining part of the packet mask two steps to
+        // the left. This corresponds to the (in total) two K-bits, which
+        // have been removed.
+        uint8_t tail_bits = (data[byte_index] >> 6) & 0x03;
+        data[byte_index - 1] |= tail_bits;
+        uint64_t mask_part2 =
+            ByteReader<uint64_t>::ReadBigEndian(&data[byte_index]);
+        // Shift away bit 46, and bit 47, which were copied to the previous
+        // part of the mask, implicitly clearing the last two bits.
+        mask_part2 <<= 2;
+        ByteWriter<uint64_t>::WriteBigEndian(&data[byte_index], mask_part2);
+        byte_index += kFlexfecPacketMaskSizes[2] - kFlexfecPacketMaskSizes[1];
+      }
+    }
+  }
+
+  fec_packet->fec_header_size = byte_index;
+
+  // In FlexFEC, all media packets are protected in their entirety.
+  fec_packet->protection_length =
+      fec_packet->pkt->data.size() - fec_packet->fec_header_size;
+
+  return true;
+}
+
+FlexfecHeaderWriter::FlexfecHeaderWriter()
+    : FecHeaderWriter(kMaxMediaPackets, kMaxFecPackets, kHeaderSizes[2]) {}
+
+FlexfecHeaderWriter::~FlexfecHeaderWriter() = default;
+
+size_t FlexfecHeaderWriter::MinPacketMaskSize(const uint8_t* packet_mask,
+                                              size_t packet_mask_size) const {
+  if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear &&
+      (packet_mask[1] & 0x01) == 0) {
+    // Packet mask is 16 bits long, with bit 15 clear.
+    // It can be used as is.
+    return kFlexfecPacketMaskSizes[0];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear) {
+    // Packet mask is 16 bits long, with bit 15 set.
+    // We must expand the packet mask with zeros in the FlexFEC header.
+    return kFlexfecPacketMaskSizes[1];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet &&
+             (packet_mask[5] & 0x03) == 0) {
+    // Packet mask is 48 bits long, with bits 46 and 47 clear.
+    // It can be used as is.
+    return kFlexfecPacketMaskSizes[1];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet) {
+    // Packet mask is 48 bits long, with at least one of bits 46 and 47 set.
+    // We must expand it with zeros.
+    return kFlexfecPacketMaskSizes[2];
+  }
+  RTC_DCHECK_NOTREACHED() << "Incorrect packet mask size: " << packet_mask_size
+                          << ".";
+  return kFlexfecPacketMaskSizes[2];
+}
+
+size_t FlexfecHeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
+  return FlexfecHeaderSize(packet_mask_size);
+}
+
+// This function adapts the precomputed ULPFEC packet masks to the
+// FlexFEC header standard. Note that the header size is computed by
+// FecHeaderSize(), so in this function we can be sure that we are
+// writing in space that is intended for the header.
+//
+// TODO(brandtr): Update this function when we support offset-based masks
+// and retransmissions.
+void FlexfecHeaderWriter::FinalizeFecHeader(
+    rtc::ArrayView<const ProtectedStream> protected_streams,
+    ForwardErrorCorrection::Packet& fec_packet) const {
+  uint8_t* data = fec_packet.data.MutableData();
+  *data &= 0x7f;  // Clear R bit.
+  *data &= 0xbf;  // Clear F bit.
+
+  // First seq_num will be in byte index 8
+  // (See FEC header schematic in flexfec_header_reader_writer.h.)
+  uint8_t* write_at = data + 8;
+  for (const ProtectedStream& protected_stream : protected_streams) {
+    ByteWriter<uint16_t>::WriteBigEndian(write_at,
+                                         protected_stream.seq_num_base);
+    write_at += kStreamSpecificHeaderSize;
+    // Adapt ULPFEC packet mask to FlexFEC header.
+    //
+    // We treat the mask parts as unsigned integers with host order endianness
+    // in order to simplify the bit shifting between bytes.
+    if (protected_stream.packet_mask.size() == kUlpfecPacketMaskSizeLBitSet) {
+      // The packet mask is 48 bits long.
+      uint16_t tmp_mask_part0 =
+          ByteReader<uint16_t>::ReadBigEndian(&protected_stream.packet_mask[0]);
+      uint32_t tmp_mask_part1 =
+          ByteReader<uint32_t>::ReadBigEndian(&protected_stream.packet_mask[2]);
+
+      tmp_mask_part0 >>= 1;  // Shift, thus clearing K-bit 0.
+      ByteWriter<uint16_t>::WriteBigEndian(write_at, tmp_mask_part0);
+      *write_at |= 0x80;  // Set K-bit 0.
+      write_at += kFlexfecPacketMaskSizes[0];
+      tmp_mask_part1 >>= 2;  // Shift twice, thus clearing K-bit 1 and bit 15.
+      ByteWriter<uint32_t>::WriteBigEndian(write_at, tmp_mask_part1);
+
+      bool bit15 = (protected_stream.packet_mask[1] & 0x01) != 0;
+      if (bit15)
+        *write_at |= 0x40;  // Set bit 15.
+
+      bool bit46 = (protected_stream.packet_mask[5] & 0x02) != 0;
+      bool bit47 = (protected_stream.packet_mask[5] & 0x01) != 0;
+      if (!bit46 && !bit47) {
+        write_at += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0];
+      } else {
+        *write_at |= 0x80;  // Set K-bit 1.
+        write_at += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0];
+        // Clear all trailing bits.
+        memset(write_at, 0,
+               kFlexfecPacketMaskSizes[2] - kFlexfecPacketMaskSizes[1]);
+        if (bit46)
+          *write_at |= 0x80;  // Set bit 46.
+        if (bit47)
+          *write_at |= 0x40;  // Set bit 47.
+        write_at += kFlexfecPacketMaskSizes[2] - kFlexfecPacketMaskSizes[1];
+      }
+    } else if (protected_stream.packet_mask.size() ==
+               kUlpfecPacketMaskSizeLBitClear) {
+      // The packet mask is 16 bits long.
+      uint16_t tmp_mask_part0 =
+          ByteReader<uint16_t>::ReadBigEndian(&protected_stream.packet_mask[0]);
+
+      tmp_mask_part0 >>= 1;  // Shift, thus clearing K-bit 0.
+      ByteWriter<uint16_t>::WriteBigEndian(write_at, tmp_mask_part0);
+      bool bit15 = (protected_stream.packet_mask[1] & 0x01) != 0;
+      if (!bit15) {
+        write_at += kFlexfecPacketMaskSizes[0];
+      } else {
+        *write_at |= 0x80;  // Set K-bit 0.
+        write_at += kFlexfecPacketMaskSizes[0];
+        // Clear all trailing bits.
+        memset(write_at, 0U,
+               kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0]);
+        *write_at |= 0x40;  // Set bit 15.
+        write_at += kFlexfecPacketMaskSizes[1] - kFlexfecPacketMaskSizes[0];
+      }
+    } else {
+      RTC_DCHECK_NOTREACHED() << "Incorrect packet mask size: "
+                              << protected_stream.packet_mask.size() << ".";
+    }
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h

@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2023 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_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
+#define MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+namespace webrtc {
+
+// FlexFEC header in flexible mode (R=0, F=0), minimum 12 bytes.
+// https://datatracker.ietf.org/doc/html/rfc8627#section-4.2.2.1
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |0|0|P|X|  CC   |M| PT recovery |        length recovery        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          TS recovery                          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |           SN base_i           |k|          Mask [0-14]        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |k|                   Mask [15-45] (optional)                   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                     Mask [46-109] (optional)                  |
+// 20 |                                                               |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |   ... next SN base and Mask for CSRC_i in CSRC list ...       |
+//
+
+class FlexfecHeaderReader : public FecHeaderReader {
+ public:
+  FlexfecHeaderReader();
+  ~FlexfecHeaderReader() override;
+
+  bool ReadFecHeader(
+      ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const override;
+};
+
+class FlexfecHeaderWriter : public FecHeaderWriter {
+ public:
+  FlexfecHeaderWriter();
+  ~FlexfecHeaderWriter() override;
+
+  size_t MinPacketMaskSize(const uint8_t* packet_mask,
+                           size_t packet_mask_size) const override;
+
+  size_t FecHeaderSize(size_t packet_mask_row_size) const override;
+
+  void FinalizeFecHeader(
+      rtc::ArrayView<const ProtectedStream> protected_streams,
+      ForwardErrorCorrection::Packet& fec_packet) const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc

@@ -0,0 +1,202 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/include/flexfec_receiver.h"
+
+#include <string.h>
+
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Minimum header size (in bytes) of a well-formed non-singular FlexFEC packet.
+constexpr size_t kMinFlexfecHeaderSize = 20;
+
+// How often to log the recovered packets to the text log.
+constexpr TimeDelta kPacketLogInterval = TimeDelta::Seconds(10);
+
+}  // namespace
+
+FlexfecReceiver::FlexfecReceiver(
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc,
+    RecoveredPacketReceiver* recovered_packet_receiver)
+    : FlexfecReceiver(Clock::GetRealTimeClock(),
+                      ssrc,
+                      protected_media_ssrc,
+                      recovered_packet_receiver) {}
+
+FlexfecReceiver::FlexfecReceiver(
+    Clock* clock,
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc,
+    RecoveredPacketReceiver* recovered_packet_receiver)
+    : ssrc_(ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      erasure_code_(
+          ForwardErrorCorrection::CreateFlexfec(ssrc, protected_media_ssrc)),
+      recovered_packet_receiver_(recovered_packet_receiver),
+      clock_(clock) {
+  // It's OK to create this object on a different thread/task queue than
+  // the one used during main operation.
+  sequence_checker_.Detach();
+}
+
+FlexfecReceiver::~FlexfecReceiver() = default;
+
+void FlexfecReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // If this packet was recovered, it might be originating from
+  // ProcessReceivedPacket in this object. To avoid lifetime issues with
+  // `recovered_packets_`, we therefore break the cycle here.
+  // This might reduce decoding efficiency a bit, since we can't disambiguate
+  // recovered packets by RTX from recovered packets by FlexFEC.
+  if (packet.recovered())
+    return;
+
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      AddReceivedPacket(packet);
+  if (!received_packet)
+    return;
+
+  ProcessReceivedPacket(*received_packet);
+}
+
+FecPacketCounter FlexfecReceiver::GetPacketCounter() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return packet_counter_;
+}
+
+// TODO(eladalon): Consider using packet.recovered() to avoid processing
+// recovered packets here.
+std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>
+FlexfecReceiver::AddReceivedPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // RTP packets with a full base header (12 bytes), but without payload,
+  // could conceivably be useful in the decoding. Therefore we check
+  // with a non-strict inequality here.
+  RTC_DCHECK_GE(packet.size(), kRtpHeaderSize);
+
+  // Demultiplex based on SSRC, and insert into erasure code decoder.
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
+      new ForwardErrorCorrection::ReceivedPacket());
+  received_packet->seq_num = packet.SequenceNumber();
+  received_packet->ssrc = packet.Ssrc();
+  received_packet->extensions = packet.extension_manager();
+  if (received_packet->ssrc == ssrc_) {
+    // This is a FlexFEC packet.
+    if (packet.payload_size() < kMinFlexfecHeaderSize) {
+      RTC_LOG(LS_WARNING) << "Truncated FlexFEC packet, discarding.";
+      return nullptr;
+    }
+    received_packet->is_fec = true;
+    ++packet_counter_.num_fec_packets;
+
+    // Insert packet payload into erasure code.
+    received_packet->pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
+        new ForwardErrorCorrection::Packet());
+    received_packet->pkt->data =
+        packet.Buffer().Slice(packet.headers_size(), packet.payload_size());
+  } else {
+    // This is a media packet, or a FlexFEC packet belonging to some
+    // other FlexFEC stream.
+    if (received_packet->ssrc != protected_media_ssrc_) {
+      return nullptr;
+    }
+    received_packet->is_fec = false;
+
+    // Insert entire packet into erasure code.
+    // Create a copy and fill with zeros all mutable extensions.
+    received_packet->pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
+        new ForwardErrorCorrection::Packet());
+    RtpPacketReceived packet_copy(packet);
+    packet_copy.ZeroMutableExtensions();
+    received_packet->pkt->data = packet_copy.Buffer();
+  }
+
+  ++packet_counter_.num_packets;
+
+  return received_packet;
+}
+
+// Note that the implementation of this member function and the implementation
+// in UlpfecReceiver::ProcessReceivedFec() are slightly different.
+// This implementation only returns _recovered_ media packets through the
+// callback, whereas the implementation in UlpfecReceiver returns _all inserted_
+// media packets through the callback. The latter behaviour makes sense
+// for ULPFEC, since the ULPFEC receiver is owned by the RtpVideoStreamReceiver.
+// Here, however, the received media pipeline is more decoupled from the
+// FlexFEC decoder, and we therefore do not interfere with the reception
+// of non-recovered media packets.
+void FlexfecReceiver::ProcessReceivedPacket(
+    const ForwardErrorCorrection::ReceivedPacket& received_packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // Decode.
+  ForwardErrorCorrection::DecodeFecResult decode_result =
+      erasure_code_->DecodeFec(received_packet, &recovered_packets_);
+
+  if (decode_result.num_recovered_packets == 0) {
+    return;
+  }
+
+  // Return recovered packets through callback.
+  for (const auto& recovered_packet : recovered_packets_) {
+    RTC_CHECK(recovered_packet);
+    if (recovered_packet->returned) {
+      continue;
+    }
+    ++packet_counter_.num_recovered_packets;
+    // Set this flag first, since OnRecoveredPacket may end up here
+    // again, with the same packet.
+    recovered_packet->returned = true;
+    RTC_CHECK_GE(recovered_packet->pkt->data.size(), kRtpHeaderSize);
+
+    RtpPacketReceived parsed_packet(&received_packet.extensions);
+    if (!parsed_packet.Parse(recovered_packet->pkt->data)) {
+      continue;
+    }
+    parsed_packet.set_recovered(true);
+
+    // TODO(brandtr): Update here when we support protecting audio packets too.
+    parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+    recovered_packet_receiver_->OnRecoveredPacket(parsed_packet);
+
+    // Periodically log the incoming packets at LS_INFO.
+    Timestamp now = clock_->CurrentTime();
+    bool should_log_periodically =
+        now - last_recovered_packet_ > kPacketLogInterval;
+    if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE) || should_log_periodically) {
+      rtc::LoggingSeverity level =
+          should_log_periodically ? rtc::LS_INFO : rtc::LS_VERBOSE;
+      RTC_LOG_V(level) << "Recovered media packet with SSRC: "
+                       << parsed_packet.Ssrc() << " seq "
+                       << parsed_packet.SequenceNumber() << " recovered length "
+                       << recovered_packet->pkt->data.size()
+                       << " received length "
+                       << received_packet.pkt->data.size()
+                       << " from FlexFEC stream with SSRC: " << ssrc_;
+      if (should_log_periodically) {
+        last_recovered_packet_ = now;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc

@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/include/flexfec_sender.h"
+
+#include <string.h>
+
+#include <list>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Let first sequence number be in the first half of the interval.
+constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff;
+
+// See breakdown in flexfec_header_reader_writer.cc.
+constexpr size_t kFlexfecMaxHeaderSize = 32;
+
+// Since we will mainly use FlexFEC to protect video streams, we use a 90 kHz
+// clock for the RTP timestamps. (This is according to the RFC, which states
+// that it is RECOMMENDED to use the same clock frequency for FlexFEC as for
+// the protected media stream.)
+// The constant converts from clock millisecond timestamps to the 90 kHz
+// RTP timestamp.
+const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
+
+// How often to log the generated FEC packets to the text log.
+constexpr TimeDelta kPacketLogInterval = TimeDelta::Seconds(10);
+
+RtpHeaderExtensionMap RegisterSupportedExtensions(
+    const std::vector<RtpExtension>& rtp_header_extensions) {
+  RtpHeaderExtensionMap map;
+  for (const auto& extension : rtp_header_extensions) {
+    if (extension.uri == TransportSequenceNumber::Uri()) {
+      map.Register<TransportSequenceNumber>(extension.id);
+    } else if (extension.uri == AbsoluteSendTime::Uri()) {
+      map.Register<AbsoluteSendTime>(extension.id);
+    } else if (extension.uri == TransmissionOffset::Uri()) {
+      map.Register<TransmissionOffset>(extension.id);
+    } else if (extension.uri == RtpMid::Uri()) {
+      map.Register<RtpMid>(extension.id);
+    } else {
+      RTC_LOG(LS_INFO)
+          << "FlexfecSender only supports RTP header extensions for "
+             "BWE and MID, so the extension "
+          << extension.ToString() << " will not be used.";
+    }
+  }
+  return map;
+}
+
+}  // namespace
+
+FlexfecSender::FlexfecSender(
+    int payload_type,
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc,
+    absl::string_view mid,
+    const std::vector<RtpExtension>& rtp_header_extensions,
+    rtc::ArrayView<const RtpExtensionSize> extension_sizes,
+    const RtpState* rtp_state,
+    Clock* clock)
+    : clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
+      payload_type_(payload_type),
+      // Reset RTP state if this is not the first time we are operating.
+      // Otherwise, randomize the initial timestamp offset and RTP sequence
+      // numbers. (This is not intended to be cryptographically strong.)
+      timestamp_offset_(rtp_state ? rtp_state->start_timestamp
+                                  : random_.Rand<uint32_t>()),
+      ssrc_(ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      mid_(mid),
+      seq_num_(rtp_state ? rtp_state->sequence_number
+                         : random_.Rand(1, kMaxInitRtpSeqNumber)),
+      ulpfec_generator_(
+          ForwardErrorCorrection::CreateFlexfec(ssrc, protected_media_ssrc),
+          clock_),
+      rtp_header_extension_map_(
+          RegisterSupportedExtensions(rtp_header_extensions)),
+      header_extensions_size_(
+          RtpHeaderExtensionSize(extension_sizes, rtp_header_extension_map_)),
+      fec_bitrate_(/*max_window_size=*/TimeDelta::Seconds(1)) {
+  // This object should not have been instantiated if FlexFEC is disabled.
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
+}
+
+FlexfecSender::~FlexfecSender() = default;
+
+// We are reusing the implementation from UlpfecGenerator for SetFecParameters,
+// AddRtpPacketAndGenerateFec, and FecAvailable.
+void FlexfecSender::SetProtectionParameters(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  ulpfec_generator_.SetProtectionParameters(delta_params, key_params);
+}
+
+void FlexfecSender::AddPacketAndGenerateFec(const RtpPacketToSend& packet) {
+  // TODO(brandtr): Generalize this SSRC check when we support multistream
+  // protection.
+  RTC_DCHECK_EQ(packet.Ssrc(), protected_media_ssrc_);
+  ulpfec_generator_.AddPacketAndGenerateFec(packet);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> FlexfecSender::GetFecPackets() {
+  RTC_CHECK_RUNS_SERIALIZED(&ulpfec_generator_.race_checker_);
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets_to_send;
+  fec_packets_to_send.reserve(ulpfec_generator_.generated_fec_packets_.size());
+  size_t total_fec_data_bytes = 0;
+  for (const auto* fec_packet : ulpfec_generator_.generated_fec_packets_) {
+    std::unique_ptr<RtpPacketToSend> fec_packet_to_send(
+        new RtpPacketToSend(&rtp_header_extension_map_));
+    fec_packet_to_send->set_packet_type(
+        RtpPacketMediaType::kForwardErrorCorrection);
+    fec_packet_to_send->set_allow_retransmission(false);
+
+    // RTP header.
+    fec_packet_to_send->SetMarker(false);
+    fec_packet_to_send->SetPayloadType(payload_type_);
+    fec_packet_to_send->SetSequenceNumber(seq_num_++);
+    fec_packet_to_send->SetTimestamp(
+        timestamp_offset_ +
+        static_cast<uint32_t>(kMsToRtpTimestamp *
+                              clock_->TimeInMilliseconds()));
+    // Set "capture time" so that the TransmissionOffset header extension
+    // can be set by the RTPSender.
+    fec_packet_to_send->set_capture_time(clock_->CurrentTime());
+    fec_packet_to_send->SetSsrc(ssrc_);
+    // Reserve extensions, if registered. These will be set by the RTPSender.
+    fec_packet_to_send->ReserveExtension<AbsoluteSendTime>();
+    fec_packet_to_send->ReserveExtension<TransmissionOffset>();
+    fec_packet_to_send->ReserveExtension<TransportSequenceNumber>();
+    // Possibly include the MID header extension.
+    if (!mid_.empty()) {
+      // This is a no-op if the MID header extension is not registered.
+      fec_packet_to_send->SetExtension<RtpMid>(mid_);
+    }
+
+    // RTP payload.
+    uint8_t* payload =
+        fec_packet_to_send->AllocatePayload(fec_packet->data.size());
+    memcpy(payload, fec_packet->data.cdata(), fec_packet->data.size());
+
+    total_fec_data_bytes += fec_packet_to_send->size();
+    fec_packets_to_send.push_back(std::move(fec_packet_to_send));
+  }
+
+  if (!fec_packets_to_send.empty()) {
+    ulpfec_generator_.ResetState();
+  }
+
+  Timestamp now = clock_->CurrentTime();
+  if (!fec_packets_to_send.empty() &&
+      now - last_generated_packet_ > kPacketLogInterval) {
+    RTC_LOG(LS_VERBOSE) << "Generated " << fec_packets_to_send.size()
+                        << " FlexFEC packets with payload type: "
+                        << payload_type_ << " and SSRC: " << ssrc_ << ".";
+    last_generated_packet_ = now;
+  }
+
+  MutexLock lock(&mutex_);
+  fec_bitrate_.Update(total_fec_data_bytes, now);
+
+  return fec_packets_to_send;
+}
+
+// The overhead is BWE RTP header extensions and FlexFEC header.
+size_t FlexfecSender::MaxPacketOverhead() const {
+  return header_extensions_size_ + kFlexfecMaxHeaderSize;
+}
+
+DataRate FlexfecSender::CurrentFecRate() const {
+  MutexLock lock(&mutex_);
+  return fec_bitrate_.Rate(clock_->CurrentTime()).value_or(DataRate::Zero());
+}
+
+absl::optional<RtpState> FlexfecSender::GetRtpState() {
+  RtpState rtp_state;
+  rtp_state.sequence_number = seq_num_;
+  rtp_state.start_timestamp = timestamp_offset_;
+  return rtp_state;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc

@@ -0,0 +1,838 @@
+/*
+ *  Copyright (c) 2012 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 "modules/rtp_rtcp/source/forward_error_correction.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/flexfec_03_header_reader_writer.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/mod_ops.h"
+
+namespace webrtc {
+
+namespace {
+// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
+constexpr size_t kTransportOverhead = 28;
+
+constexpr uint16_t kOldSequenceThreshold = 0x3fff;
+}  // namespace
+
+ForwardErrorCorrection::Packet::Packet() : data(0), ref_count_(0) {}
+ForwardErrorCorrection::Packet::~Packet() = default;
+
+int32_t ForwardErrorCorrection::Packet::AddRef() {
+  return ++ref_count_;
+}
+
+int32_t ForwardErrorCorrection::Packet::Release() {
+  int32_t ref_count;
+  ref_count = --ref_count_;
+  if (ref_count == 0)
+    delete this;
+  return ref_count;
+}
+
+// This comparator is used to compare std::unique_ptr's pointing to
+// subclasses of SortablePackets. It needs to be parametric since
+// the std::unique_ptr's are not covariant w.r.t. the types that
+// they are pointing to.
+template <typename S, typename T>
+bool ForwardErrorCorrection::SortablePacket::LessThan::operator()(
+    const S& first,
+    const T& second) {
+  RTC_DCHECK_EQ(first->ssrc, second->ssrc);
+  return IsNewerSequenceNumber(second->seq_num, first->seq_num);
+}
+
+ForwardErrorCorrection::ReceivedPacket::ReceivedPacket() = default;
+ForwardErrorCorrection::ReceivedPacket::~ReceivedPacket() = default;
+
+ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() = default;
+ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() = default;
+
+ForwardErrorCorrection::ProtectedPacket::ProtectedPacket() = default;
+ForwardErrorCorrection::ProtectedPacket::~ProtectedPacket() = default;
+
+ForwardErrorCorrection::ReceivedFecPacket::ReceivedFecPacket() = default;
+ForwardErrorCorrection::ReceivedFecPacket::~ReceivedFecPacket() = default;
+
+ForwardErrorCorrection::ForwardErrorCorrection(
+    std::unique_ptr<FecHeaderReader> fec_header_reader,
+    std::unique_ptr<FecHeaderWriter> fec_header_writer,
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc)
+    : ssrc_(ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      fec_header_reader_(std::move(fec_header_reader)),
+      fec_header_writer_(std::move(fec_header_writer)),
+      generated_fec_packets_(fec_header_writer_->MaxFecPackets()),
+      packet_mask_size_(0) {}
+
+ForwardErrorCorrection::~ForwardErrorCorrection() = default;
+
+std::unique_ptr<ForwardErrorCorrection> ForwardErrorCorrection::CreateUlpfec(
+    uint32_t ssrc) {
+  std::unique_ptr<FecHeaderReader> fec_header_reader(new UlpfecHeaderReader());
+  std::unique_ptr<FecHeaderWriter> fec_header_writer(new UlpfecHeaderWriter());
+  return std::unique_ptr<ForwardErrorCorrection>(new ForwardErrorCorrection(
+      std::move(fec_header_reader), std::move(fec_header_writer), ssrc, ssrc));
+}
+
+std::unique_ptr<ForwardErrorCorrection> ForwardErrorCorrection::CreateFlexfec(
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc) {
+  std::unique_ptr<FecHeaderReader> fec_header_reader(
+      new Flexfec03HeaderReader());
+  std::unique_ptr<FecHeaderWriter> fec_header_writer(
+      new Flexfec03HeaderWriter());
+  return std::unique_ptr<ForwardErrorCorrection>(new ForwardErrorCorrection(
+      std::move(fec_header_reader), std::move(fec_header_writer), ssrc,
+      protected_media_ssrc));
+}
+
+int ForwardErrorCorrection::EncodeFec(const PacketList& media_packets,
+                                      uint8_t protection_factor,
+                                      int num_important_packets,
+                                      bool use_unequal_protection,
+                                      FecMaskType fec_mask_type,
+                                      std::list<Packet*>* fec_packets) {
+  const size_t num_media_packets = media_packets.size();
+
+  // Sanity check arguments.
+  RTC_DCHECK_GT(num_media_packets, 0);
+  RTC_DCHECK_GE(num_important_packets, 0);
+  RTC_DCHECK_LE(num_important_packets, num_media_packets);
+  RTC_DCHECK(fec_packets->empty());
+  const size_t max_media_packets = fec_header_writer_->MaxMediaPackets();
+  if (num_media_packets > max_media_packets) {
+    RTC_LOG(LS_WARNING) << "Can't protect " << num_media_packets
+                        << " media packets per frame. Max is "
+                        << max_media_packets << ".";
+    return -1;
+  }
+
+  // Error check the media packets.
+  for (const auto& media_packet : media_packets) {
+    RTC_DCHECK(media_packet);
+    if (media_packet->data.size() < kRtpHeaderSize) {
+      RTC_LOG(LS_WARNING) << "Media packet " << media_packet->data.size()
+                          << " bytes "
+                             "is smaller than RTP header.";
+      return -1;
+    }
+    // Ensure the FEC packets will fit in a typical MTU.
+    if (media_packet->data.size() + MaxPacketOverhead() + kTransportOverhead >
+        IP_PACKET_SIZE) {
+      RTC_LOG(LS_WARNING) << "Media packet " << media_packet->data.size()
+                          << " bytes "
+                             "with overhead is larger than "
+                          << IP_PACKET_SIZE << " bytes.";
+    }
+  }
+
+  // Prepare generated FEC packets.
+  int num_fec_packets = NumFecPackets(num_media_packets, protection_factor);
+  if (num_fec_packets == 0) {
+    return 0;
+  }
+  for (int i = 0; i < num_fec_packets; ++i) {
+    generated_fec_packets_[i].data.EnsureCapacity(IP_PACKET_SIZE);
+    memset(generated_fec_packets_[i].data.MutableData(), 0, IP_PACKET_SIZE);
+    // Use this as a marker for untouched packets.
+    generated_fec_packets_[i].data.SetSize(0);
+    fec_packets->push_back(&generated_fec_packets_[i]);
+  }
+
+  internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
+  packet_mask_size_ = internal::PacketMaskSize(num_media_packets);
+  memset(packet_masks_, 0, num_fec_packets * packet_mask_size_);
+  internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
+                                num_important_packets, use_unequal_protection,
+                                &mask_table, packet_masks_);
+
+  // Adapt packet masks to missing media packets.
+  int num_mask_bits = InsertZerosInPacketMasks(media_packets, num_fec_packets);
+  if (num_mask_bits < 0) {
+    RTC_LOG(LS_INFO) << "Due to sequence number gaps, cannot protect media "
+                        "packets with a single block of FEC packets.";
+    fec_packets->clear();
+    return -1;
+  }
+  packet_mask_size_ = internal::PacketMaskSize(num_mask_bits);
+
+  // Write FEC packets to `generated_fec_packets_`.
+  GenerateFecPayloads(media_packets, num_fec_packets);
+  // TODO(brandtr): Generalize this when multistream protection support is
+  // added.
+  const uint32_t media_ssrc = ParseSsrc(media_packets.front()->data.data());
+  const uint16_t seq_num_base =
+      ParseSequenceNumber(media_packets.front()->data.data());
+  FinalizeFecHeaders(num_fec_packets, media_ssrc, seq_num_base);
+
+  return 0;
+}
+
+int ForwardErrorCorrection::NumFecPackets(int num_media_packets,
+                                          int protection_factor) {
+  // Result in Q0 with an unsigned round.
+  int num_fec_packets = (num_media_packets * protection_factor + (1 << 7)) >> 8;
+  // Generate at least one FEC packet if we need protection.
+  if (protection_factor > 0 && num_fec_packets == 0) {
+    num_fec_packets = 1;
+  }
+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
+  return num_fec_packets;
+}
+
+void ForwardErrorCorrection::GenerateFecPayloads(
+    const PacketList& media_packets,
+    size_t num_fec_packets) {
+  RTC_DCHECK(!media_packets.empty());
+  for (size_t i = 0; i < num_fec_packets; ++i) {
+    Packet* const fec_packet = &generated_fec_packets_[i];
+    size_t pkt_mask_idx = i * packet_mask_size_;
+    const size_t min_packet_mask_size = fec_header_writer_->MinPacketMaskSize(
+        &packet_masks_[pkt_mask_idx], packet_mask_size_);
+    const size_t fec_header_size =
+        fec_header_writer_->FecHeaderSize(min_packet_mask_size);
+
+    size_t media_pkt_idx = 0;
+    auto media_packets_it = media_packets.cbegin();
+    uint16_t prev_seq_num =
+        ParseSequenceNumber((*media_packets_it)->data.data());
+    while (media_packets_it != media_packets.end()) {
+      Packet* const media_packet = media_packets_it->get();
+      // Should `media_packet` be protected by `fec_packet`?
+      if (packet_masks_[pkt_mask_idx] & (1 << (7 - media_pkt_idx))) {
+        size_t media_payload_length =
+            media_packet->data.size() - kRtpHeaderSize;
+
+        size_t fec_packet_length = fec_header_size + media_payload_length;
+        if (fec_packet_length > fec_packet->data.size()) {
+          size_t old_size = fec_packet->data.size();
+          fec_packet->data.SetSize(fec_packet_length);
+          memset(fec_packet->data.MutableData() + old_size, 0,
+                 fec_packet_length - old_size);
+        }
+        XorHeaders(*media_packet, fec_packet);
+        XorPayloads(*media_packet, media_payload_length, fec_header_size,
+                    fec_packet);
+      }
+      media_packets_it++;
+      if (media_packets_it != media_packets.end()) {
+        uint16_t seq_num =
+            ParseSequenceNumber((*media_packets_it)->data.data());
+        media_pkt_idx += static_cast<uint16_t>(seq_num - prev_seq_num);
+        prev_seq_num = seq_num;
+      }
+      pkt_mask_idx += media_pkt_idx / 8;
+      media_pkt_idx %= 8;
+    }
+    RTC_DCHECK_GT(fec_packet->data.size(), 0)
+        << "Packet mask is wrong or poorly designed.";
+  }
+}
+
+int ForwardErrorCorrection::InsertZerosInPacketMasks(
+    const PacketList& media_packets,
+    size_t num_fec_packets) {
+  size_t num_media_packets = media_packets.size();
+  if (num_media_packets <= 1) {
+    return num_media_packets;
+  }
+  uint16_t last_seq_num =
+      ParseSequenceNumber(media_packets.back()->data.data());
+  uint16_t first_seq_num =
+      ParseSequenceNumber(media_packets.front()->data.data());
+  size_t total_missing_seq_nums =
+      static_cast<uint16_t>(last_seq_num - first_seq_num) - num_media_packets +
+      1;
+  if (total_missing_seq_nums == 0) {
+    // All sequence numbers are covered by the packet mask.
+    // No zero insertion required.
+    return num_media_packets;
+  }
+  const size_t max_media_packets = fec_header_writer_->MaxMediaPackets();
+  if (total_missing_seq_nums + num_media_packets > max_media_packets) {
+    return -1;
+  }
+  // Allocate the new mask.
+  size_t tmp_packet_mask_size =
+      internal::PacketMaskSize(total_missing_seq_nums + num_media_packets);
+  memset(tmp_packet_masks_, 0, num_fec_packets * tmp_packet_mask_size);
+
+  auto media_packets_it = media_packets.cbegin();
+  uint16_t prev_seq_num = first_seq_num;
+  ++media_packets_it;
+
+  // Insert the first column.
+  internal::CopyColumn(tmp_packet_masks_, tmp_packet_mask_size, packet_masks_,
+                       packet_mask_size_, num_fec_packets, 0, 0);
+  size_t new_bit_index = 1;
+  size_t old_bit_index = 1;
+  // Insert zeros in the bit mask for every hole in the sequence.
+  while (media_packets_it != media_packets.end()) {
+    if (new_bit_index == max_media_packets) {
+      // We can only cover up to 48 packets.
+      break;
+    }
+    uint16_t seq_num = ParseSequenceNumber((*media_packets_it)->data.data());
+    const int num_zeros_to_insert =
+        static_cast<uint16_t>(seq_num - prev_seq_num - 1);
+    if (num_zeros_to_insert > 0) {
+      internal::InsertZeroColumns(num_zeros_to_insert, tmp_packet_masks_,
+                                  tmp_packet_mask_size, num_fec_packets,
+                                  new_bit_index);
+    }
+    new_bit_index += num_zeros_to_insert;
+    internal::CopyColumn(tmp_packet_masks_, tmp_packet_mask_size, packet_masks_,
+                         packet_mask_size_, num_fec_packets, new_bit_index,
+                         old_bit_index);
+    ++new_bit_index;
+    ++old_bit_index;
+    prev_seq_num = seq_num;
+    ++media_packets_it;
+  }
+  if (new_bit_index % 8 != 0) {
+    // We didn't fill the last byte. Shift bits to correct position.
+    for (uint16_t row = 0; row < num_fec_packets; ++row) {
+      int new_byte_index = row * tmp_packet_mask_size + new_bit_index / 8;
+      tmp_packet_masks_[new_byte_index] <<= (7 - (new_bit_index % 8));
+    }
+  }
+  // Replace the old mask with the new.
+  memcpy(packet_masks_, tmp_packet_masks_,
+         num_fec_packets * tmp_packet_mask_size);
+  return new_bit_index;
+}
+
+void ForwardErrorCorrection::FinalizeFecHeaders(size_t num_fec_packets,
+                                                uint32_t media_ssrc,
+                                                uint16_t seq_num_base) {
+  for (size_t i = 0; i < num_fec_packets; ++i) {
+    const FecHeaderWriter::ProtectedStream protected_streams[] = {
+        {.ssrc = media_ssrc,
+         .seq_num_base = seq_num_base,
+         .packet_mask = {&packet_masks_[i * packet_mask_size_],
+                         packet_mask_size_}}};
+    fec_header_writer_->FinalizeFecHeader(protected_streams,
+                                          generated_fec_packets_[i]);
+  }
+}
+
+void ForwardErrorCorrection::ResetState(
+    RecoveredPacketList* recovered_packets) {
+  // Free the memory for any existing recovered packets, if the caller hasn't.
+  recovered_packets->clear();
+  received_fec_packets_.clear();
+}
+
+void ForwardErrorCorrection::InsertMediaPacket(
+    RecoveredPacketList* recovered_packets,
+    const ReceivedPacket& received_packet) {
+  RTC_DCHECK_EQ(received_packet.ssrc, protected_media_ssrc_);
+
+  // Search for duplicate packets.
+  for (const auto& recovered_packet : *recovered_packets) {
+    RTC_DCHECK_EQ(recovered_packet->ssrc, received_packet.ssrc);
+    if (recovered_packet->seq_num == received_packet.seq_num) {
+      // Duplicate packet, no need to add to list.
+      return;
+    }
+  }
+
+  std::unique_ptr<RecoveredPacket> recovered_packet(new RecoveredPacket());
+  // This "recovered packet" was not recovered using parity packets.
+  recovered_packet->was_recovered = false;
+  // This media packet has already been passed on.
+  recovered_packet->returned = true;
+  recovered_packet->ssrc = received_packet.ssrc;
+  recovered_packet->seq_num = received_packet.seq_num;
+  recovered_packet->pkt = received_packet.pkt;
+  // TODO(holmer): Consider replacing this with a binary search for the right
+  // position, and then just insert the new packet. Would get rid of the sort.
+  RecoveredPacket* recovered_packet_ptr = recovered_packet.get();
+  recovered_packets->push_back(std::move(recovered_packet));
+  recovered_packets->sort(SortablePacket::LessThan());
+  UpdateCoveringFecPackets(*recovered_packet_ptr);
+}
+
+void ForwardErrorCorrection::UpdateCoveringFecPackets(
+    const RecoveredPacket& packet) {
+  for (auto& fec_packet : received_fec_packets_) {
+    // Is this FEC packet protecting the media packet `packet`?
+    auto protected_it = absl::c_lower_bound(
+        fec_packet->protected_packets, &packet, SortablePacket::LessThan());
+    if (protected_it != fec_packet->protected_packets.end() &&
+        (*protected_it)->seq_num == packet.seq_num) {
+      // Found an FEC packet which is protecting `packet`.
+      (*protected_it)->pkt = packet.pkt;
+    }
+  }
+}
+
+void ForwardErrorCorrection::InsertFecPacket(
+    const RecoveredPacketList& recovered_packets,
+    const ReceivedPacket& received_packet) {
+  RTC_DCHECK_EQ(received_packet.ssrc, ssrc_);
+
+  // Check for duplicate.
+  for (const auto& existing_fec_packet : received_fec_packets_) {
+    RTC_DCHECK_EQ(existing_fec_packet->ssrc, received_packet.ssrc);
+    if (existing_fec_packet->seq_num == received_packet.seq_num) {
+      // Drop duplicate FEC packet data.
+      return;
+    }
+  }
+
+  std::unique_ptr<ReceivedFecPacket> fec_packet(new ReceivedFecPacket());
+  fec_packet->pkt = received_packet.pkt;
+  fec_packet->ssrc = received_packet.ssrc;
+  fec_packet->seq_num = received_packet.seq_num;
+  // Parse ULPFEC/FlexFEC header specific info.
+  bool ret = fec_header_reader_->ReadFecHeader(fec_packet.get());
+  if (!ret) {
+    return;
+  }
+
+  RTC_CHECK_EQ(fec_packet->protected_streams.size(), 1);
+
+  if (fec_packet->protected_streams[0].ssrc != protected_media_ssrc_) {
+    RTC_LOG(LS_INFO)
+        << "Received FEC packet is protecting an unknown media SSRC; dropping.";
+    return;
+  }
+
+  if (fec_packet->protected_streams[0].packet_mask_offset +
+          fec_packet->protected_streams[0].packet_mask_size >
+      fec_packet->pkt->data.size()) {
+    RTC_LOG(LS_INFO) << "Received corrupted FEC packet; dropping.";
+    return;
+  }
+
+  // Parse packet mask from header and represent as protected packets.
+  for (uint16_t byte_idx = 0;
+       byte_idx < fec_packet->protected_streams[0].packet_mask_size;
+       ++byte_idx) {
+    uint8_t packet_mask =
+        fec_packet->pkt
+            ->data[fec_packet->protected_streams[0].packet_mask_offset +
+                   byte_idx];
+    for (uint16_t bit_idx = 0; bit_idx < 8; ++bit_idx) {
+      if (packet_mask & (1 << (7 - bit_idx))) {
+        std::unique_ptr<ProtectedPacket> protected_packet(
+            new ProtectedPacket());
+        // This wraps naturally with the sequence number.
+        protected_packet->ssrc = protected_media_ssrc_;
+        protected_packet->seq_num = static_cast<uint16_t>(
+            fec_packet->protected_streams[0].seq_num_base + (byte_idx << 3) +
+            bit_idx);
+        protected_packet->pkt = nullptr;
+        fec_packet->protected_packets.push_back(std::move(protected_packet));
+      }
+    }
+  }
+
+  if (fec_packet->protected_packets.empty()) {
+    // All-zero packet mask; we can discard this FEC packet.
+    RTC_LOG(LS_WARNING) << "Received FEC packet has an all-zero packet mask.";
+  } else {
+    AssignRecoveredPackets(recovered_packets, fec_packet.get());
+    // TODO(holmer): Consider replacing this with a binary search for the right
+    // position, and then just insert the new packet. Would get rid of the sort.
+    received_fec_packets_.push_back(std::move(fec_packet));
+    received_fec_packets_.sort(SortablePacket::LessThan());
+    const size_t max_fec_packets = fec_header_reader_->MaxFecPackets();
+    if (received_fec_packets_.size() > max_fec_packets) {
+      received_fec_packets_.pop_front();
+    }
+    RTC_DCHECK_LE(received_fec_packets_.size(), max_fec_packets);
+  }
+}
+
+void ForwardErrorCorrection::AssignRecoveredPackets(
+    const RecoveredPacketList& recovered_packets,
+    ReceivedFecPacket* fec_packet) {
+  ProtectedPacketList* protected_packets = &fec_packet->protected_packets;
+  std::vector<RecoveredPacket*> recovered_protected_packets;
+
+  // Find intersection between the (sorted) containers `protected_packets`
+  // and `recovered_packets`, i.e. all protected packets that have already
+  // been recovered. Update the corresponding protected packets to point to
+  // the recovered packets.
+  auto it_p = protected_packets->cbegin();
+  auto it_r = recovered_packets.cbegin();
+  SortablePacket::LessThan less_than;
+  while (it_p != protected_packets->end() && it_r != recovered_packets.end()) {
+    if (less_than(*it_p, *it_r)) {
+      ++it_p;
+    } else if (less_than(*it_r, *it_p)) {
+      ++it_r;
+    } else {  // *it_p == *it_r.
+      // This protected packet has already been recovered.
+      (*it_p)->pkt = (*it_r)->pkt;
+      ++it_p;
+      ++it_r;
+    }
+  }
+}
+
+void ForwardErrorCorrection::InsertPacket(
+    const ReceivedPacket& received_packet,
+    RecoveredPacketList* recovered_packets) {
+  // Discard old FEC packets such that the sequence numbers in
+  // `received_fec_packets_` span at most 1/2 of the sequence number space.
+  // This is important for keeping `received_fec_packets_` sorted, and may
+  // also reduce the possibility of incorrect decoding due to sequence number
+  // wrap-around.
+  if (!received_fec_packets_.empty() &&
+      received_packet.ssrc == received_fec_packets_.front()->ssrc) {
+    // It only makes sense to detect wrap-around when `received_packet`
+    // and `front_received_fec_packet` belong to the same sequence number
+    // space, i.e., the same SSRC. This happens when `received_packet`
+    // is a FEC packet, or if `received_packet` is a media packet and
+    // RED+ULPFEC is used.
+    auto it = received_fec_packets_.begin();
+    while (it != received_fec_packets_.end()) {
+      uint16_t seq_num_diff = MinDiff(received_packet.seq_num, (*it)->seq_num);
+      if (seq_num_diff > kOldSequenceThreshold) {
+        it = received_fec_packets_.erase(it);
+      } else {
+        // No need to keep iterating, since `received_fec_packets_` is sorted.
+        break;
+      }
+    }
+  }
+
+  if (received_packet.is_fec) {
+    InsertFecPacket(*recovered_packets, received_packet);
+  } else {
+    InsertMediaPacket(recovered_packets, received_packet);
+  }
+
+  DiscardOldRecoveredPackets(recovered_packets);
+}
+
+bool ForwardErrorCorrection::StartPacketRecovery(
+    const ReceivedFecPacket& fec_packet,
+    RecoveredPacket* recovered_packet) {
+  // Ensure pkt is initialized.
+  recovered_packet->pkt = new Packet();
+  // Sanity check packet length.
+  if (fec_packet.pkt->data.size() <
+      fec_packet.fec_header_size + fec_packet.protection_length) {
+    RTC_LOG(LS_WARNING)
+        << "The FEC packet is truncated: it does not contain enough room "
+           "for its own header.";
+    return false;
+  }
+  if (fec_packet.protection_length >
+      std::min(size_t{IP_PACKET_SIZE - kRtpHeaderSize},
+               IP_PACKET_SIZE - fec_packet.fec_header_size)) {
+    RTC_LOG(LS_WARNING) << "Incorrect protection length, dropping FEC packet.";
+    return false;
+  }
+  // Initialize recovered packet data.
+  recovered_packet->pkt->data.EnsureCapacity(IP_PACKET_SIZE);
+  recovered_packet->pkt->data.SetSize(fec_packet.protection_length +
+                                      kRtpHeaderSize);
+  recovered_packet->returned = false;
+  recovered_packet->was_recovered = true;
+  // Copy bytes corresponding to minimum RTP header size.
+  // Note that the sequence number and SSRC fields will be overwritten
+  // at the end of packet recovery.
+  memcpy(recovered_packet->pkt->data.MutableData(),
+         fec_packet.pkt->data.cdata(), kRtpHeaderSize);
+  // Copy remaining FEC payload.
+  if (fec_packet.protection_length > 0) {
+    memcpy(recovered_packet->pkt->data.MutableData() + kRtpHeaderSize,
+           fec_packet.pkt->data.cdata() + fec_packet.fec_header_size,
+           fec_packet.protection_length);
+  }
+  return true;
+}
+
+bool ForwardErrorCorrection::FinishPacketRecovery(
+    const ReceivedFecPacket& fec_packet,
+    RecoveredPacket* recovered_packet) {
+  uint8_t* data = recovered_packet->pkt->data.MutableData();
+  // Set the RTP version to 2.
+  data[0] |= 0x80;  // Set the 1st bit.
+  data[0] &= 0xbf;  // Clear the 2nd bit.
+  // Recover the packet length, from temporary location.
+  const size_t new_size =
+      ByteReader<uint16_t>::ReadBigEndian(&data[2]) + kRtpHeaderSize;
+  if (new_size > size_t{IP_PACKET_SIZE - kRtpHeaderSize}) {
+    RTC_LOG(LS_WARNING) << "The recovered packet had a length larger than a "
+                           "typical IP packet, and is thus dropped.";
+    return false;
+  }
+  size_t old_size = recovered_packet->pkt->data.size();
+  recovered_packet->pkt->data.SetSize(new_size);
+  data = recovered_packet->pkt->data.MutableData();
+  if (new_size > old_size) {
+    memset(data + old_size, 0, new_size - old_size);
+  }
+
+  // Set the SN field.
+  ByteWriter<uint16_t>::WriteBigEndian(&data[2], recovered_packet->seq_num);
+  // Set the SSRC field.
+  ByteWriter<uint32_t>::WriteBigEndian(&data[8], recovered_packet->ssrc);
+  return true;
+}
+
+void ForwardErrorCorrection::XorHeaders(const Packet& src, Packet* dst) {
+  uint8_t* dst_data = dst->data.MutableData();
+  const uint8_t* src_data = src.data.cdata();
+  // XOR the first 2 bytes of the header: V, P, X, CC, M, PT fields.
+  dst_data[0] ^= src_data[0];
+  dst_data[1] ^= src_data[1];
+
+  // XOR the length recovery field.
+  uint8_t src_payload_length_network_order[2];
+  ByteWriter<uint16_t>::WriteBigEndian(src_payload_length_network_order,
+                                       src.data.size() - kRtpHeaderSize);
+  dst_data[2] ^= src_payload_length_network_order[0];
+  dst_data[3] ^= src_payload_length_network_order[1];
+
+  // XOR the 5th to 8th bytes of the header: the timestamp field.
+  dst_data[4] ^= src_data[4];
+  dst_data[5] ^= src_data[5];
+  dst_data[6] ^= src_data[6];
+  dst_data[7] ^= src_data[7];
+
+  // Skip the 9th to 12th bytes of the header.
+}
+
+void ForwardErrorCorrection::XorPayloads(const Packet& src,
+                                         size_t payload_length,
+                                         size_t dst_offset,
+                                         Packet* dst) {
+  // XOR the payload.
+  RTC_DCHECK_LE(kRtpHeaderSize + payload_length, src.data.size());
+  RTC_DCHECK_LE(dst_offset + payload_length, dst->data.capacity());
+  if (dst_offset + payload_length > dst->data.size()) {
+    size_t old_size = dst->data.size();
+    size_t new_size = dst_offset + payload_length;
+    dst->data.SetSize(new_size);
+    memset(dst->data.MutableData() + old_size, 0, new_size - old_size);
+  }
+  uint8_t* dst_data = dst->data.MutableData();
+  const uint8_t* src_data = src.data.cdata();
+  for (size_t i = 0; i < payload_length; ++i) {
+    dst_data[dst_offset + i] ^= src_data[kRtpHeaderSize + i];
+  }
+}
+
+bool ForwardErrorCorrection::RecoverPacket(const ReceivedFecPacket& fec_packet,
+                                           RecoveredPacket* recovered_packet) {
+  if (!StartPacketRecovery(fec_packet, recovered_packet)) {
+    return false;
+  }
+  for (const auto& protected_packet : fec_packet.protected_packets) {
+    if (protected_packet->pkt == nullptr) {
+      // This is the packet we're recovering.
+      recovered_packet->seq_num = protected_packet->seq_num;
+      recovered_packet->ssrc = protected_packet->ssrc;
+    } else {
+      XorHeaders(*protected_packet->pkt, recovered_packet->pkt.get());
+      XorPayloads(*protected_packet->pkt,
+                  protected_packet->pkt->data.size() - kRtpHeaderSize,
+                  kRtpHeaderSize, recovered_packet->pkt.get());
+    }
+  }
+  if (!FinishPacketRecovery(fec_packet, recovered_packet)) {
+    return false;
+  }
+  return true;
+}
+
+size_t ForwardErrorCorrection::AttemptRecovery(
+    RecoveredPacketList* recovered_packets) {
+  size_t num_recovered_packets = 0;
+
+  auto fec_packet_it = received_fec_packets_.begin();
+  while (fec_packet_it != received_fec_packets_.end()) {
+    // Search for each FEC packet's protected media packets.
+    int packets_missing = NumCoveredPacketsMissing(**fec_packet_it);
+
+    // We can only recover one packet with an FEC packet.
+    if (packets_missing == 1) {
+      // Recovery possible.
+      std::unique_ptr<RecoveredPacket> recovered_packet(new RecoveredPacket());
+      recovered_packet->pkt = nullptr;
+      if (!RecoverPacket(**fec_packet_it, recovered_packet.get())) {
+        // Can't recover using this packet, drop it.
+        fec_packet_it = received_fec_packets_.erase(fec_packet_it);
+        continue;
+      }
+
+      ++num_recovered_packets;
+
+      auto* recovered_packet_ptr = recovered_packet.get();
+      // Add recovered packet to the list of recovered packets and update any
+      // FEC packets covering this packet with a pointer to the data.
+      // TODO(holmer): Consider replacing this with a binary search for the
+      // right position, and then just insert the new packet. Would get rid of
+      // the sort.
+      recovered_packets->push_back(std::move(recovered_packet));
+      recovered_packets->sort(SortablePacket::LessThan());
+      UpdateCoveringFecPackets(*recovered_packet_ptr);
+      DiscardOldRecoveredPackets(recovered_packets);
+      fec_packet_it = received_fec_packets_.erase(fec_packet_it);
+
+      // A packet has been recovered. We need to check the FEC list again, as
+      // this may allow additional packets to be recovered.
+      // Restart for first FEC packet.
+      fec_packet_it = received_fec_packets_.begin();
+    } else if (packets_missing == 0 ||
+               IsOldFecPacket(**fec_packet_it, recovered_packets)) {
+      // Either all protected packets arrived or have been recovered, or the FEC
+      // packet is old. We can discard this FEC packet.
+      fec_packet_it = received_fec_packets_.erase(fec_packet_it);
+    } else {
+      fec_packet_it++;
+    }
+  }
+
+  return num_recovered_packets;
+}
+
+int ForwardErrorCorrection::NumCoveredPacketsMissing(
+    const ReceivedFecPacket& fec_packet) {
+  int packets_missing = 0;
+  for (const auto& protected_packet : fec_packet.protected_packets) {
+    if (protected_packet->pkt == nullptr) {
+      ++packets_missing;
+      if (packets_missing > 1) {
+        break;  // We can't recover more than one packet.
+      }
+    }
+  }
+  return packets_missing;
+}
+
+void ForwardErrorCorrection::DiscardOldRecoveredPackets(
+    RecoveredPacketList* recovered_packets) {
+  const size_t max_media_packets = fec_header_reader_->MaxMediaPackets();
+  while (recovered_packets->size() > max_media_packets) {
+    recovered_packets->pop_front();
+  }
+  RTC_DCHECK_LE(recovered_packets->size(), max_media_packets);
+}
+
+bool ForwardErrorCorrection::IsOldFecPacket(
+    const ReceivedFecPacket& fec_packet,
+    const RecoveredPacketList* recovered_packets) {
+  if (recovered_packets->empty()) {
+    return false;
+  }
+
+  const uint16_t back_recovered_seq_num = recovered_packets->back()->seq_num;
+  const uint16_t last_protected_seq_num =
+      fec_packet.protected_packets.back()->seq_num;
+
+  // FEC packet is old if its last protected sequence number is much
+  // older than the latest protected sequence number received.
+  return (MinDiff(back_recovered_seq_num, last_protected_seq_num) >
+          kOldSequenceThreshold);
+}
+
+uint16_t ForwardErrorCorrection::ParseSequenceNumber(const uint8_t* packet) {
+  return (packet[2] << 8) + packet[3];
+}
+
+uint32_t ForwardErrorCorrection::ParseSsrc(const uint8_t* packet) {
+  return (packet[8] << 24) + (packet[9] << 16) + (packet[10] << 8) + packet[11];
+}
+
+ForwardErrorCorrection::DecodeFecResult ForwardErrorCorrection::DecodeFec(
+    const ReceivedPacket& received_packet,
+    RecoveredPacketList* recovered_packets) {
+  RTC_DCHECK(recovered_packets);
+
+  const size_t max_media_packets = fec_header_reader_->MaxMediaPackets();
+  if (recovered_packets->size() == max_media_packets) {
+    const RecoveredPacket* back_recovered_packet =
+        recovered_packets->back().get();
+
+    if (received_packet.ssrc == back_recovered_packet->ssrc) {
+      const unsigned int seq_num_diff =
+          MinDiff(received_packet.seq_num, back_recovered_packet->seq_num);
+      if (seq_num_diff > max_media_packets) {
+        // A big gap in sequence numbers. The old recovered packets
+        // are now useless, so it's safe to do a reset.
+        RTC_LOG(LS_INFO) << "Big gap in media/ULPFEC sequence numbers. No need "
+                            "to keep the old packets in the FEC buffers, thus "
+                            "resetting them.";
+        ResetState(recovered_packets);
+      }
+    }
+  }
+
+  InsertPacket(received_packet, recovered_packets);
+
+  DecodeFecResult decode_result;
+  decode_result.num_recovered_packets = AttemptRecovery(recovered_packets);
+  return decode_result;
+}
+
+size_t ForwardErrorCorrection::MaxPacketOverhead() const {
+  return fec_header_writer_->MaxPacketOverhead();
+}
+
+FecHeaderReader::FecHeaderReader(size_t max_media_packets,
+                                 size_t max_fec_packets)
+    : max_media_packets_(max_media_packets),
+      max_fec_packets_(max_fec_packets) {}
+
+FecHeaderReader::~FecHeaderReader() = default;
+
+size_t FecHeaderReader::MaxMediaPackets() const {
+  return max_media_packets_;
+}
+
+size_t FecHeaderReader::MaxFecPackets() const {
+  return max_fec_packets_;
+}
+
+FecHeaderWriter::FecHeaderWriter(size_t max_media_packets,
+                                 size_t max_fec_packets,
+                                 size_t max_packet_overhead)
+    : max_media_packets_(max_media_packets),
+      max_fec_packets_(max_fec_packets),
+      max_packet_overhead_(max_packet_overhead) {}
+
+FecHeaderWriter::~FecHeaderWriter() = default;
+
+size_t FecHeaderWriter::MaxMediaPackets() const {
+  return max_media_packets_;
+}
+
+size_t FecHeaderWriter::MaxFecPackets() const {
+  return max_fec_packets_;
+}
+
+size_t FecHeaderWriter::MaxPacketOverhead() const {
+  return max_packet_overhead_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/forward_error_correction.h

@@ -0,0 +1,443 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
+#define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/scoped_refptr.h"
+#include "api/units/timestamp.h"
+#include "modules/include/module_fec_types.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class FecHeaderReader;
+class FecHeaderWriter;
+
+// Performs codec-independent forward error correction (FEC), based on RFC 5109.
+// Option exists to enable unequal protection (UEP) across packets.
+// This is not to be confused with protection within packets
+// (referred to as uneven level protection (ULP) in RFC 5109).
+// TODO(brandtr): Split this class into a separate encoder
+// and a separate decoder.
+class ForwardErrorCorrection {
+ public:
+  // TODO(holmer): As a next step all these struct-like packet classes should be
+  // refactored into proper classes, and their members should be made private.
+  // This will require parts of the functionality in forward_error_correction.cc
+  // and receiver_fec.cc to be refactored into the packet classes.
+  class Packet {
+   public:
+    Packet();
+    virtual ~Packet();
+
+    // Add a reference.
+    virtual int32_t AddRef();
+
+    // Release a reference. Will delete the object if the reference count
+    // reaches zero.
+    virtual int32_t Release();
+
+    rtc::CopyOnWriteBuffer data;  // Packet data.
+
+   private:
+    int32_t ref_count_;  // Counts the number of references to a packet.
+  };
+
+  // TODO(holmer): Refactor into a proper class.
+  class SortablePacket {
+   public:
+    // Functor which returns true if the sequence number of `first`
+    // is < the sequence number of `second`. Should only ever be called for
+    // packets belonging to the same SSRC.
+    struct LessThan {
+      template <typename S, typename T>
+      bool operator()(const S& first, const T& second);
+    };
+
+    uint32_t ssrc;
+    uint16_t seq_num;
+  };
+
+  // Used for the input to DecodeFec().
+  class ReceivedPacket : public SortablePacket {
+   public:
+    ReceivedPacket();
+    ~ReceivedPacket();
+
+    bool is_fec;  // Set to true if this is an FEC packet and false
+                  // otherwise.
+    bool is_recovered;
+    RtpHeaderExtensionMap extensions;
+    rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
+  };
+
+  // The recovered list parameter of DecodeFec() references structs of
+  // this type.
+  // TODO(holmer): Refactor into a proper class.
+  class RecoveredPacket : public SortablePacket {
+   public:
+    RecoveredPacket();
+    ~RecoveredPacket();
+
+    bool was_recovered;  // Will be true if this packet was recovered by
+                         // the FEC. Otherwise it was a media packet passed in
+                         // through the received packet list.
+    bool returned;  // True when the packet already has been returned to the
+                    // caller through the callback.
+    rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
+  };
+
+  // Used to link media packets to their protecting FEC packets.
+  //
+  // TODO(holmer): Refactor into a proper class.
+  class ProtectedPacket : public SortablePacket {
+   public:
+    ProtectedPacket();
+    ~ProtectedPacket();
+
+    rtc::scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+  };
+
+  using ProtectedPacketList = std::list<std::unique_ptr<ProtectedPacket>>;
+
+  struct ProtectedStream {
+    uint32_t ssrc = 0;
+    uint16_t seq_num_base = 0;
+    size_t packet_mask_offset = 0;  // Relative start of FEC header.
+    size_t packet_mask_size = 0;
+  };
+
+  // Used for internal storage of received FEC packets in a list.
+  //
+  // TODO(holmer): Refactor into a proper class.
+  class ReceivedFecPacket : public SortablePacket {
+   public:
+    // SSRC count is limited by 4 bits of CSRC count in RTP header (max 15).
+    // Since most of the time number of SSRCs will be low (probably 1 most of
+    // the time) setting this value to 4 for optimization.
+    static constexpr size_t kInlinedSsrcsVectorSize = 4;
+
+    ReceivedFecPacket();
+    ~ReceivedFecPacket();
+
+    // List of media packets that this FEC packet protects.
+    ProtectedPacketList protected_packets;
+    // RTP header fields.
+    uint32_t ssrc;
+    // FEC header fields.
+    size_t fec_header_size;
+    absl::InlinedVector<ProtectedStream, kInlinedSsrcsVectorSize>
+        protected_streams;
+    size_t protection_length;
+    // Raw data.
+    rtc::scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+  };
+
+  using PacketList = std::list<std::unique_ptr<Packet>>;
+  using RecoveredPacketList = std::list<std::unique_ptr<RecoveredPacket>>;
+  using ReceivedFecPacketList = std::list<std::unique_ptr<ReceivedFecPacket>>;
+
+  ~ForwardErrorCorrection();
+
+  // Creates a ForwardErrorCorrection tailored for a specific FEC scheme.
+  static std::unique_ptr<ForwardErrorCorrection> CreateUlpfec(uint32_t ssrc);
+  static std::unique_ptr<ForwardErrorCorrection> CreateFlexfec(
+      uint32_t ssrc,
+      uint32_t protected_media_ssrc);
+
+  // Generates a list of FEC packets from supplied media packets.
+  //
+  // Input:  media_packets          List of media packets to protect, of type
+  //                                Packet. All packets must belong to the
+  //                                same frame and the list must not be empty.
+  // Input:  protection_factor      FEC protection overhead in the [0, 255]
+  //                                domain. To obtain 100% overhead, or an
+  //                                equal number of FEC packets as
+  //                                media packets, use 255.
+  // Input:  num_important_packets  The number of "important" packets in the
+  //                                frame. These packets may receive greater
+  //                                protection than the remaining packets.
+  //                                The important packets must be located at the
+  //                                start of the media packet list. For codecs
+  //                                with data partitioning, the important
+  //                                packets may correspond to first partition
+  //                                packets.
+  // Input:  use_unequal_protection Parameter to enable/disable unequal
+  //                                protection (UEP) across packets. Enabling
+  //                                UEP will allocate more protection to the
+  //                                num_important_packets from the start of the
+  //                                media_packets.
+  // Input:  fec_mask_type          The type of packet mask used in the FEC.
+  //                                Random or bursty type may be selected. The
+  //                                bursty type is only defined up to 12 media
+  //                                packets. If the number of media packets is
+  //                                above 12, the packet masks from the random
+  //                                table will be selected.
+  // Output: fec_packets            List of pointers to generated FEC packets,
+  //                                of type Packet. Must be empty on entry.
+  //                                The memory available through the list will
+  //                                be valid until the next call to
+  //                                EncodeFec().
+  //
+  // Returns 0 on success, -1 on failure.
+  //
+  int EncodeFec(const PacketList& media_packets,
+                uint8_t protection_factor,
+                int num_important_packets,
+                bool use_unequal_protection,
+                FecMaskType fec_mask_type,
+                std::list<Packet*>* fec_packets);
+
+  // Decodes a list of received media and FEC packets. It will parse the
+  // `received_packets`, storing FEC packets internally, and move
+  // media packets to `recovered_packets`. The recovered list will be
+  // sorted by ascending sequence number and have duplicates removed.
+  // The function should be called as new packets arrive, and
+  // `recovered_packets` will be progressively assembled with each call.
+  // When the function returns, `received_packets` will be empty.
+  //
+  // The caller will allocate packets submitted through `received_packets`.
+  // The function will handle allocation of recovered packets.
+  //
+  // Input:  received_packets   List of new received packets, of type
+  //                            ReceivedPacket, belonging to a single
+  //                            frame. At output the list will be empty,
+  //                            with packets either stored internally,
+  //                            or accessible through the recovered list.
+  // Output: recovered_packets  List of recovered media packets, of type
+  //                            RecoveredPacket, belonging to a single
+  //                            frame. The memory available through the
+  //                            list will be valid until the next call to
+  //                            DecodeFec().
+  //
+  struct DecodeFecResult {
+    // Number of recovered media packets using FEC.
+    size_t num_recovered_packets = 0;
+  };
+
+  DecodeFecResult DecodeFec(const ReceivedPacket& received_packet,
+                            RecoveredPacketList* recovered_packets);
+
+  // Get the number of generated FEC packets, given the number of media packets
+  // and the protection factor.
+  static int NumFecPackets(int num_media_packets, int protection_factor);
+
+  // Gets the maximum size of the FEC headers in bytes, which must be
+  // accounted for as packet overhead.
+  size_t MaxPacketOverhead() const;
+
+  // Reset internal states from last frame and clear `recovered_packets`.
+  // Frees all memory allocated by this class.
+  void ResetState(RecoveredPacketList* recovered_packets);
+
+  // TODO(brandtr): Remove these functions when the Packet classes
+  // have been refactored.
+  static uint16_t ParseSequenceNumber(const uint8_t* packet);
+  static uint32_t ParseSsrc(const uint8_t* packet);
+
+ protected:
+  ForwardErrorCorrection(std::unique_ptr<FecHeaderReader> fec_header_reader,
+                         std::unique_ptr<FecHeaderWriter> fec_header_writer,
+                         uint32_t ssrc,
+                         uint32_t protected_media_ssrc);
+
+ private:
+  // Analyzes `media_packets` for holes in the sequence and inserts zero columns
+  // into the `packet_mask` where those holes are found. Zero columns means that
+  // those packets will have no protection.
+  // Returns the number of bits used for one row of the new packet mask.
+  // Requires that `packet_mask` has at least 6 * `num_fec_packets` bytes
+  // allocated.
+  int InsertZerosInPacketMasks(const PacketList& media_packets,
+                               size_t num_fec_packets);
+
+  // Writes FEC payloads and some recovery fields in the FEC headers.
+  void GenerateFecPayloads(const PacketList& media_packets,
+                           size_t num_fec_packets);
+
+  // Writes the FEC header fields that are not written by GenerateFecPayloads.
+  // This includes writing the packet masks.
+  void FinalizeFecHeaders(size_t num_fec_packets,
+                          uint32_t media_ssrc,
+                          uint16_t seq_num_base);
+
+  // Inserts the `received_packet` into the internal received FEC packet list
+  // or into `recovered_packets`.
+  void InsertPacket(const ReceivedPacket& received_packet,
+                    RecoveredPacketList* recovered_packets);
+
+  // Inserts the `received_packet` into `recovered_packets`. Deletes duplicates.
+  void InsertMediaPacket(RecoveredPacketList* recovered_packets,
+                         const ReceivedPacket& received_packet);
+
+  // Assigns pointers to the recovered packet from all FEC packets which cover
+  // it.
+  // Note: This reduces the complexity when we want to try to recover a packet
+  // since we don't have to find the intersection between recovered packets and
+  // packets covered by the FEC packet.
+  void UpdateCoveringFecPackets(const RecoveredPacket& packet);
+
+  // Insert `received_packet` into internal FEC list. Deletes duplicates.
+  void InsertFecPacket(const RecoveredPacketList& recovered_packets,
+                       const ReceivedPacket& received_packet);
+
+  // Assigns pointers to already recovered packets covered by `fec_packet`.
+  static void AssignRecoveredPackets(
+      const RecoveredPacketList& recovered_packets,
+      ReceivedFecPacket* fec_packet);
+
+  // Attempt to recover missing packets, using the internally stored
+  // received FEC packets.
+  size_t AttemptRecovery(RecoveredPacketList* recovered_packets);
+
+  // Initializes headers and payload before the XOR operation
+  // that recovers a packet.
+  static bool StartPacketRecovery(const ReceivedFecPacket& fec_packet,
+                                  RecoveredPacket* recovered_packet);
+
+  // Performs XOR between the first 8 bytes of `src` and `dst` and stores
+  // the result in `dst`. The 3rd and 4th bytes are used for storing
+  // the length recovery field.
+  static void XorHeaders(const Packet& src, Packet* dst);
+
+  // Performs XOR between the payloads of `src` and `dst` and stores the result
+  // in `dst`. The parameter `dst_offset` determines at  what byte the
+  // XOR operation starts in `dst`. In total, `payload_length` bytes are XORed.
+  static void XorPayloads(const Packet& src,
+                          size_t payload_length,
+                          size_t dst_offset,
+                          Packet* dst);
+
+  // Finalizes recovery of packet by setting RTP header fields.
+  // This is not specific to the FEC scheme used.
+  static bool FinishPacketRecovery(const ReceivedFecPacket& fec_packet,
+                                   RecoveredPacket* recovered_packet);
+
+  // Recover a missing packet.
+  static bool RecoverPacket(const ReceivedFecPacket& fec_packet,
+                            RecoveredPacket* recovered_packet);
+
+  // Get the number of missing media packets which are covered by `fec_packet`.
+  // An FEC packet can recover at most one packet, and if zero packets are
+  // missing the FEC packet can be discarded. This function returns 2 when two
+  // or more packets are missing.
+  static int NumCoveredPacketsMissing(const ReceivedFecPacket& fec_packet);
+
+  // Discards old packets in `recovered_packets`, which are no longer relevant
+  // for recovering lost packets.
+  void DiscardOldRecoveredPackets(RecoveredPacketList* recovered_packets);
+
+  // Checks if the FEC packet is old enough and no longer relevant for
+  // recovering lost media packets.
+  bool IsOldFecPacket(const ReceivedFecPacket& fec_packet,
+                      const RecoveredPacketList* recovered_packets);
+
+  // These SSRCs are only used by the decoder.
+  const uint32_t ssrc_;
+  const uint32_t protected_media_ssrc_;
+
+  std::unique_ptr<FecHeaderReader> fec_header_reader_;
+  std::unique_ptr<FecHeaderWriter> fec_header_writer_;
+
+  std::vector<Packet> generated_fec_packets_;
+  ReceivedFecPacketList received_fec_packets_;
+
+  // Arrays used to avoid dynamically allocating memory when generating
+  // the packet masks.
+  // (There are never more than `kUlpfecMaxMediaPackets` FEC packets generated.)
+  uint8_t packet_masks_[kUlpfecMaxMediaPackets * kUlpfecMaxPacketMaskSize];
+  uint8_t tmp_packet_masks_[kUlpfecMaxMediaPackets * kUlpfecMaxPacketMaskSize];
+  size_t packet_mask_size_;
+};
+
+// Classes derived from FecHeader{Reader,Writer} encapsulate the
+// specifics of reading and writing FEC header for, e.g., ULPFEC
+// and FlexFEC.
+class FecHeaderReader {
+ public:
+  virtual ~FecHeaderReader();
+
+  // The maximum number of media packets that can be covered by one FEC packet.
+  size_t MaxMediaPackets() const;
+
+  // The maximum number of FEC packets that is supported, per call
+  // to ForwardErrorCorrection::EncodeFec().
+  size_t MaxFecPackets() const;
+
+  // Parses FEC header and stores information in ReceivedFecPacket members.
+  virtual bool ReadFecHeader(
+      ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const = 0;
+
+ protected:
+  FecHeaderReader(size_t max_media_packets, size_t max_fec_packets);
+
+  const size_t max_media_packets_;
+  const size_t max_fec_packets_;
+};
+
+class FecHeaderWriter {
+ public:
+  struct ProtectedStream {
+    uint32_t ssrc = 0;
+    uint16_t seq_num_base = 0;
+    rtc::ArrayView<const uint8_t> packet_mask;
+  };
+
+  virtual ~FecHeaderWriter();
+
+  // The maximum number of media packets that can be covered by one FEC packet.
+  size_t MaxMediaPackets() const;
+
+  // The maximum number of FEC packets that is supported, per call
+  // to ForwardErrorCorrection::EncodeFec().
+  size_t MaxFecPackets() const;
+
+  // The maximum overhead (in bytes) per packet, due to FEC headers.
+  size_t MaxPacketOverhead() const;
+
+  // Calculates the minimum packet mask size needed (in bytes),
+  // given the discrete options of the ULPFEC masks and the bits
+  // set in the current packet mask.
+  virtual size_t MinPacketMaskSize(const uint8_t* packet_mask,
+                                   size_t packet_mask_size) const = 0;
+
+  // The header size (in bytes), given the packet mask size.
+  virtual size_t FecHeaderSize(size_t packet_mask_size) const = 0;
+
+  // Writes FEC header.
+  virtual void FinalizeFecHeader(
+      rtc::ArrayView<const ProtectedStream> protected_streams,
+      ForwardErrorCorrection::Packet& fec_packet) const = 0;
+
+ protected:
+  FecHeaderWriter(size_t max_media_packets,
+                  size_t max_fec_packets,
+                  size_t max_packet_overhead);
+
+  const size_t max_media_packets_;
+  const size_t max_fec_packets_;
+  const size_t max_packet_overhead_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc

@@ -0,0 +1,519 @@
+/*
+ *  Copyright (c) 2012 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 "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+
+#include <string.h>
+
+#include <algorithm>
+
+#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+#include "modules/rtp_rtcp/source/fec_private_tables_random.h"
+#include "rtc_base/checks.h"
+
+namespace {
+// Allow for different modes of protection for packets in UEP case.
+enum ProtectionMode {
+  kModeNoOverlap,
+  kModeOverlap,
+  kModeBiasFirstPacket,
+};
+
+// Fits an input mask (sub_mask) to an output mask.
+// The mask is a matrix where the rows are the FEC packets,
+// and the columns are the source packets the FEC is applied to.
+// Each row of the mask is represented by a number of mask bytes.
+//
+// \param[in]  num_mask_bytes     The number of mask bytes of output mask.
+// \param[in]  num_sub_mask_bytes The number of mask bytes of input mask.
+// \param[in]  num_rows           The number of rows of the input mask.
+// \param[in]  sub_mask           A pointer to hold the input mask, of size
+//                                [0, num_rows * num_sub_mask_bytes]
+// \param[out] packet_mask        A pointer to hold the output mask, of size
+//                                [0, x * num_mask_bytes], where x >= num_rows.
+void FitSubMask(int num_mask_bytes,
+                int num_sub_mask_bytes,
+                int num_rows,
+                const uint8_t* sub_mask,
+                uint8_t* packet_mask) {
+  if (num_mask_bytes == num_sub_mask_bytes) {
+    memcpy(packet_mask, sub_mask, num_rows * num_sub_mask_bytes);
+  } else {
+    for (int i = 0; i < num_rows; ++i) {
+      int pkt_mask_idx = i * num_mask_bytes;
+      int pkt_mask_idx2 = i * num_sub_mask_bytes;
+      for (int j = 0; j < num_sub_mask_bytes; ++j) {
+        packet_mask[pkt_mask_idx] = sub_mask[pkt_mask_idx2];
+        pkt_mask_idx++;
+        pkt_mask_idx2++;
+      }
+    }
+  }
+}
+
+// Shifts a mask by number of columns (bits), and fits it to an output mask.
+// The mask is a matrix where the rows are the FEC packets,
+// and the columns are the source packets the FEC is applied to.
+// Each row of the mask is represented by a number of mask bytes.
+//
+// \param[in]  num_mask_bytes     The number of mask bytes of output mask.
+// \param[in]  num_sub_mask_bytes The number of mask bytes of input mask.
+// \param[in]  num_column_shift   The number columns to be shifted, and
+//                                the starting row for the output mask.
+// \param[in]  end_row            The ending row for the output mask.
+// \param[in]  sub_mask           A pointer to hold the input mask, of size
+//                                [0, (end_row_fec - start_row_fec) *
+//                                    num_sub_mask_bytes]
+// \param[out] packet_mask        A pointer to hold the output mask, of size
+//                                [0, x * num_mask_bytes],
+//                                where x >= end_row_fec.
+// TODO(marpan): This function is doing three things at the same time:
+// shift within a byte, byte shift and resizing.
+// Split up into subroutines.
+void ShiftFitSubMask(int num_mask_bytes,
+                     int res_mask_bytes,
+                     int num_column_shift,
+                     int end_row,
+                     const uint8_t* sub_mask,
+                     uint8_t* packet_mask) {
+  // Number of bit shifts within a byte
+  const int num_bit_shifts = (num_column_shift % 8);
+  const int num_byte_shifts = num_column_shift >> 3;
+
+  // Modify new mask with sub-mask21.
+
+  // Loop over the remaining FEC packets.
+  for (int i = num_column_shift; i < end_row; ++i) {
+    // Byte index of new mask, for row i and column res_mask_bytes,
+    // offset by the number of bytes shifts
+    int pkt_mask_idx =
+        i * num_mask_bytes + res_mask_bytes - 1 + num_byte_shifts;
+    // Byte index of sub_mask, for row i and column res_mask_bytes
+    int pkt_mask_idx2 =
+        (i - num_column_shift) * res_mask_bytes + res_mask_bytes - 1;
+
+    uint8_t shift_right_curr_byte = 0;
+    uint8_t shift_left_prev_byte = 0;
+    uint8_t comb_new_byte = 0;
+
+    // Handle case of num_mask_bytes > res_mask_bytes:
+    // For a given row, copy the rightmost "numBitShifts" bits
+    // of the last byte of sub_mask into output mask.
+    if (num_mask_bytes > res_mask_bytes) {
+      shift_left_prev_byte = (sub_mask[pkt_mask_idx2] << (8 - num_bit_shifts));
+      packet_mask[pkt_mask_idx + 1] = shift_left_prev_byte;
+    }
+
+    // For each row i (FEC packet), shift the bit-mask of the sub_mask.
+    // Each row of the mask contains "resMaskBytes" of bytes.
+    // We start from the last byte of the sub_mask and move to first one.
+    for (int j = res_mask_bytes - 1; j > 0; j--) {
+      // Shift current byte of sub21 to the right by "numBitShifts".
+      shift_right_curr_byte = sub_mask[pkt_mask_idx2] >> num_bit_shifts;
+
+      // Fill in shifted bits with bits from the previous (left) byte:
+      // First shift the previous byte to the left by "8-numBitShifts".
+      shift_left_prev_byte =
+          (sub_mask[pkt_mask_idx2 - 1] << (8 - num_bit_shifts));
+
+      // Then combine both shifted bytes into new mask byte.
+      comb_new_byte = shift_right_curr_byte | shift_left_prev_byte;
+
+      // Assign to new mask.
+      packet_mask[pkt_mask_idx] = comb_new_byte;
+      pkt_mask_idx--;
+      pkt_mask_idx2--;
+    }
+    // For the first byte in the row (j=0 case).
+    shift_right_curr_byte = sub_mask[pkt_mask_idx2] >> num_bit_shifts;
+    packet_mask[pkt_mask_idx] = shift_right_curr_byte;
+  }
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace internal {
+
+PacketMaskTable::PacketMaskTable(FecMaskType fec_mask_type,
+                                 int num_media_packets)
+    : table_(PickTable(fec_mask_type, num_media_packets)) {}
+
+PacketMaskTable::~PacketMaskTable() = default;
+
+rtc::ArrayView<const uint8_t> PacketMaskTable::LookUp(int num_media_packets,
+                                                      int num_fec_packets) {
+  RTC_DCHECK_GT(num_media_packets, 0);
+  RTC_DCHECK_GT(num_fec_packets, 0);
+  RTC_DCHECK_LE(num_media_packets, kUlpfecMaxMediaPackets);
+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
+
+  if (num_media_packets <= 12) {
+    return LookUpInFecTable(table_, num_media_packets - 1, num_fec_packets - 1);
+  }
+  int mask_length =
+      static_cast<int>(PacketMaskSize(static_cast<size_t>(num_media_packets)));
+
+  // Generate FEC code mask for {num_media_packets(M), num_fec_packets(N)} (use
+  // N FEC packets to protect M media packets) In the mask, each FEC packet
+  // occupies one row, each bit / column represent one media packet. E.g. Row
+  // A, Col/Bit B is set to 1, means FEC packet A will have protection for media
+  // packet B.
+
+  // Loop through each fec packet.
+  for (int row = 0; row < num_fec_packets; row++) {
+    // Loop through each fec code in a row, one code has 8 bits.
+    // Bit X will be set to 1 if media packet X shall be protected by current
+    // FEC packet. In this implementation, the protection is interleaved, thus
+    // media packet X will be protected by FEC packet (X % N)
+    for (int col = 0; col < mask_length; col++) {
+      fec_packet_mask_[row * mask_length + col] =
+          ((col * 8) % num_fec_packets == row && (col * 8) < num_media_packets
+               ? 0x80
+               : 0x00) |
+          ((col * 8 + 1) % num_fec_packets == row &&
+                   (col * 8 + 1) < num_media_packets
+               ? 0x40
+               : 0x00) |
+          ((col * 8 + 2) % num_fec_packets == row &&
+                   (col * 8 + 2) < num_media_packets
+               ? 0x20
+               : 0x00) |
+          ((col * 8 + 3) % num_fec_packets == row &&
+                   (col * 8 + 3) < num_media_packets
+               ? 0x10
+               : 0x00) |
+          ((col * 8 + 4) % num_fec_packets == row &&
+                   (col * 8 + 4) < num_media_packets
+               ? 0x08
+               : 0x00) |
+          ((col * 8 + 5) % num_fec_packets == row &&
+                   (col * 8 + 5) < num_media_packets
+               ? 0x04
+               : 0x00) |
+          ((col * 8 + 6) % num_fec_packets == row &&
+                   (col * 8 + 6) < num_media_packets
+               ? 0x02
+               : 0x00) |
+          ((col * 8 + 7) % num_fec_packets == row &&
+                   (col * 8 + 7) < num_media_packets
+               ? 0x01
+               : 0x00);
+    }
+  }
+  return {&fec_packet_mask_[0],
+          static_cast<size_t>(num_fec_packets * mask_length)};
+}
+
+// If `num_media_packets` is larger than the maximum allowed by `fec_mask_type`
+// for the bursty type, or the random table is explicitly asked for, then the
+// random type is selected. Otherwise the bursty table callback is returned.
+const uint8_t* PacketMaskTable::PickTable(FecMaskType fec_mask_type,
+                                          int num_media_packets) {
+  RTC_DCHECK_GE(num_media_packets, 0);
+  RTC_DCHECK_LE(static_cast<size_t>(num_media_packets), kUlpfecMaxMediaPackets);
+
+  if (fec_mask_type != kFecMaskRandom &&
+      num_media_packets <=
+          static_cast<int>(fec_private_tables::kPacketMaskBurstyTbl[0])) {
+    return &fec_private_tables::kPacketMaskBurstyTbl[0];
+  }
+
+  return &fec_private_tables::kPacketMaskRandomTbl[0];
+}
+
+// Remaining protection after important (first partition) packet protection
+void RemainingPacketProtection(int num_media_packets,
+                               int num_fec_remaining,
+                               int num_fec_for_imp_packets,
+                               int num_mask_bytes,
+                               ProtectionMode mode,
+                               uint8_t* packet_mask,
+                               PacketMaskTable* mask_table) {
+  if (mode == kModeNoOverlap) {
+    // sub_mask21
+
+    const int res_mask_bytes =
+        PacketMaskSize(num_media_packets - num_fec_for_imp_packets);
+
+    auto end_row = (num_fec_for_imp_packets + num_fec_remaining);
+    rtc::ArrayView<const uint8_t> packet_mask_sub_21 = mask_table->LookUp(
+        num_media_packets - num_fec_for_imp_packets, num_fec_remaining);
+
+    ShiftFitSubMask(num_mask_bytes, res_mask_bytes, num_fec_for_imp_packets,
+                    end_row, &packet_mask_sub_21[0], packet_mask);
+
+  } else if (mode == kModeOverlap || mode == kModeBiasFirstPacket) {
+    // sub_mask22
+    rtc::ArrayView<const uint8_t> packet_mask_sub_22 =
+        mask_table->LookUp(num_media_packets, num_fec_remaining);
+
+    FitSubMask(num_mask_bytes, num_mask_bytes, num_fec_remaining,
+               &packet_mask_sub_22[0],
+               &packet_mask[num_fec_for_imp_packets * num_mask_bytes]);
+
+    if (mode == kModeBiasFirstPacket) {
+      for (int i = 0; i < num_fec_remaining; ++i) {
+        int pkt_mask_idx = i * num_mask_bytes;
+        packet_mask[pkt_mask_idx] = packet_mask[pkt_mask_idx] | (1 << 7);
+      }
+    }
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+}
+
+// Protection for important (first partition) packets
+void ImportantPacketProtection(int num_fec_for_imp_packets,
+                               int num_imp_packets,
+                               int num_mask_bytes,
+                               uint8_t* packet_mask,
+                               PacketMaskTable* mask_table) {
+  const int num_imp_mask_bytes = PacketMaskSize(num_imp_packets);
+
+  // Get sub_mask1 from table
+  rtc::ArrayView<const uint8_t> packet_mask_sub_1 =
+      mask_table->LookUp(num_imp_packets, num_fec_for_imp_packets);
+
+  FitSubMask(num_mask_bytes, num_imp_mask_bytes, num_fec_for_imp_packets,
+             &packet_mask_sub_1[0], packet_mask);
+}
+
+// This function sets the protection allocation: i.e., how many FEC packets
+// to use for num_imp (1st partition) packets, given the: number of media
+// packets, number of FEC packets, and number of 1st partition packets.
+int SetProtectionAllocation(int num_media_packets,
+                            int num_fec_packets,
+                            int num_imp_packets) {
+  // TODO(marpan): test different cases for protection allocation:
+
+  // Use at most (alloc_par * num_fec_packets) for important packets.
+  float alloc_par = 0.5;
+  int max_num_fec_for_imp = alloc_par * num_fec_packets;
+
+  int num_fec_for_imp_packets = (num_imp_packets < max_num_fec_for_imp)
+                                    ? num_imp_packets
+                                    : max_num_fec_for_imp;
+
+  // Fall back to equal protection in this case
+  if (num_fec_packets == 1 && (num_media_packets > 2 * num_imp_packets)) {
+    num_fec_for_imp_packets = 0;
+  }
+
+  return num_fec_for_imp_packets;
+}
+
+// Modification for UEP: reuse the off-line tables for the packet masks.
+// Note: these masks were designed for equal packet protection case,
+// assuming random packet loss.
+
+// Current version has 3 modes (options) to build UEP mask from existing ones.
+// Various other combinations may be added in future versions.
+// Longer-term, we may add another set of tables specifically for UEP cases.
+// TODO(marpan): also consider modification of masks for bursty loss cases.
+
+// Mask is characterized as (#packets_to_protect, #fec_for_protection).
+// Protection factor defined as: (#fec_for_protection / #packets_to_protect).
+
+// Let k=num_media_packets, n=total#packets, (n-k)=num_fec_packets,
+// m=num_imp_packets.
+
+// For ProtectionMode 0 and 1:
+// one mask (sub_mask1) is used for 1st partition packets,
+// the other mask (sub_mask21/22, for 0/1) is for the remaining FEC packets.
+
+// In both mode 0 and 1, the packets of 1st partition (num_imp_packets) are
+// treated equally important, and are afforded more protection than the
+// residual partition packets.
+
+// For num_imp_packets:
+// sub_mask1 = (m, t): protection = t/(m), where t=F(k,n-k,m).
+// t=F(k,n-k,m) is the number of packets used to protect first partition in
+// sub_mask1. This is determined from the function SetProtectionAllocation().
+
+// For the left-over protection:
+// Mode 0: sub_mask21 = (k-m,n-k-t): protection = (n-k-t)/(k-m)
+// mode 0 has no protection overlap between the two partitions.
+// For mode 0, we would typically set t = min(m, n-k).
+
+// Mode 1: sub_mask22 = (k, n-k-t), with protection (n-k-t)/(k)
+// mode 1 has protection overlap between the two partitions (preferred).
+
+// For ProtectionMode 2:
+// This gives 1st packet of list (which is 1st packet of 1st partition) more
+// protection. In mode 2, the equal protection mask (which is obtained from
+// mode 1 for t=0) is modified (more "1s" added in 1st column of packet mask)
+// to bias higher protection for the 1st source packet.
+
+// Protection Mode 2 may be extended for a sort of sliding protection
+// (i.e., vary the number/density of "1s" across columns) across packets.
+
+void UnequalProtectionMask(int num_media_packets,
+                           int num_fec_packets,
+                           int num_imp_packets,
+                           int num_mask_bytes,
+                           uint8_t* packet_mask,
+                           PacketMaskTable* mask_table) {
+  // Set Protection type and allocation
+  // TODO(marpan): test/update for best mode and some combinations thereof.
+
+  ProtectionMode mode = kModeOverlap;
+  int num_fec_for_imp_packets = 0;
+
+  if (mode != kModeBiasFirstPacket) {
+    num_fec_for_imp_packets = SetProtectionAllocation(
+        num_media_packets, num_fec_packets, num_imp_packets);
+  }
+
+  int num_fec_remaining = num_fec_packets - num_fec_for_imp_packets;
+  // Done with setting protection type and allocation
+
+  //
+  // Generate sub_mask1
+  //
+  if (num_fec_for_imp_packets > 0) {
+    ImportantPacketProtection(num_fec_for_imp_packets, num_imp_packets,
+                              num_mask_bytes, packet_mask, mask_table);
+  }
+
+  //
+  // Generate sub_mask2
+  //
+  if (num_fec_remaining > 0) {
+    RemainingPacketProtection(num_media_packets, num_fec_remaining,
+                              num_fec_for_imp_packets, num_mask_bytes, mode,
+                              packet_mask, mask_table);
+  }
+}
+
+// This algorithm is tailored to look up data in the `kPacketMaskRandomTbl` and
+// `kPacketMaskBurstyTbl` tables. These tables only cover fec code for up to 12
+// media packets. Starting from 13 media packets, the fec code will be generated
+// at runtime. The format of those arrays is that they're essentially a 3
+// dimensional array with the following dimensions: * media packet
+//   * Size for kPacketMaskRandomTbl: 12
+//   * Size for kPacketMaskBurstyTbl: 12
+// * fec index
+//   * Size for both random and bursty table increases from 1 to number of rows.
+//     (i.e. 1-48, or 1-12 respectively).
+// * Fec data (what actually gets returned)
+//   * Size for kPacketMaskRandomTbl: 2 bytes.
+//     * For all entries: 2 * fec index (1 based)
+//   * Size for kPacketMaskBurstyTbl: 2 bytes.
+//     * For all entries: 2 * fec index (1 based)
+rtc::ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
+                                               int media_packet_index,
+                                               int fec_index) {
+  RTC_DCHECK_LT(media_packet_index, table[0]);
+
+  // Skip over the table size.
+  const uint8_t* entry = &table[1];
+
+  uint8_t entry_size_increment = 2;  // 0-16 are 2 byte wide, then changes to 6.
+
+  // Hop over un-interesting array entries.
+  for (int i = 0; i < media_packet_index; ++i) {
+    if (i == 16)
+      entry_size_increment = 6;
+    uint8_t count = entry[0];
+    ++entry;  // skip over the count.
+    for (int j = 0; j < count; ++j) {
+      entry += entry_size_increment * (j + 1);  // skip over the data.
+    }
+  }
+
+  if (media_packet_index == 16)
+    entry_size_increment = 6;
+
+  RTC_DCHECK_LT(fec_index, entry[0]);
+  ++entry;  // Skip over the size.
+
+  // Find the appropriate data in the second dimension.
+
+  // Find the specific data we're looking for.
+  for (int i = 0; i < fec_index; ++i)
+    entry += entry_size_increment * (i + 1);  // skip over the data.
+
+  size_t size = entry_size_increment * (fec_index + 1);
+  return {&entry[0], size};
+}
+
+void GeneratePacketMasks(int num_media_packets,
+                         int num_fec_packets,
+                         int num_imp_packets,
+                         bool use_unequal_protection,
+                         PacketMaskTable* mask_table,
+                         uint8_t* packet_mask) {
+  RTC_DCHECK_GT(num_media_packets, 0);
+  RTC_DCHECK_GT(num_fec_packets, 0);
+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
+  RTC_DCHECK_LE(num_imp_packets, num_media_packets);
+  RTC_DCHECK_GE(num_imp_packets, 0);
+
+  const int num_mask_bytes = PacketMaskSize(num_media_packets);
+
+  // Equal-protection for these cases.
+  if (!use_unequal_protection || num_imp_packets == 0) {
+    // Retrieve corresponding mask table directly:for equal-protection case.
+    // Mask = (k,n-k), with protection factor = (n-k)/k,
+    // where k = num_media_packets, n=total#packets, (n-k)=num_fec_packets.
+    rtc::ArrayView<const uint8_t> mask =
+        mask_table->LookUp(num_media_packets, num_fec_packets);
+    memcpy(packet_mask, &mask[0], mask.size());
+  } else {  // UEP case
+    UnequalProtectionMask(num_media_packets, num_fec_packets, num_imp_packets,
+                          num_mask_bytes, packet_mask, mask_table);
+  }  // End of UEP modification
+}  // End of GetPacketMasks
+
+size_t PacketMaskSize(size_t num_sequence_numbers) {
+  RTC_DCHECK_LE(num_sequence_numbers, 8 * kUlpfecPacketMaskSizeLBitSet);
+  if (num_sequence_numbers > 8 * kUlpfecPacketMaskSizeLBitClear) {
+    return kUlpfecPacketMaskSizeLBitSet;
+  }
+  return kUlpfecPacketMaskSizeLBitClear;
+}
+
+void InsertZeroColumns(int num_zeros,
+                       uint8_t* new_mask,
+                       int new_mask_bytes,
+                       int num_fec_packets,
+                       int new_bit_index) {
+  for (uint16_t row = 0; row < num_fec_packets; ++row) {
+    const int new_byte_index = row * new_mask_bytes + new_bit_index / 8;
+    const int max_shifts = (7 - (new_bit_index % 8));
+    new_mask[new_byte_index] <<= std::min(num_zeros, max_shifts);
+  }
+}
+
+void CopyColumn(uint8_t* new_mask,
+                int new_mask_bytes,
+                uint8_t* old_mask,
+                int old_mask_bytes,
+                int num_fec_packets,
+                int new_bit_index,
+                int old_bit_index) {
+  RTC_CHECK_LT(new_bit_index, 8 * new_mask_bytes);
+
+  // Copy column from the old mask to the beginning of the new mask and shift it
+  // out from the old mask.
+  for (uint16_t row = 0; row < num_fec_packets; ++row) {
+    int new_byte_index = row * new_mask_bytes + new_bit_index / 8;
+    int old_byte_index = row * old_mask_bytes + old_bit_index / 8;
+    new_mask[new_byte_index] |= ((old_mask[old_byte_index] & 0x80) >> 7);
+    if (new_bit_index % 8 != 7) {
+      new_mask[new_byte_index] <<= 1;
+    }
+    old_mask[old_byte_index] <<= 1;
+  }
+}
+
+}  // namespace internal
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h

@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2012 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_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
+#define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "modules/include/module_fec_types.h"
+
+namespace webrtc {
+
+// Maximum number of media packets that can be protected
+// by these packet masks.
+constexpr size_t kUlpfecMaxMediaPackets = 48;
+
+// Packet mask size in bytes (given L bit).
+constexpr size_t kUlpfecPacketMaskSizeLBitClear = 2;
+constexpr size_t kUlpfecPacketMaskSizeLBitSet = 6;
+
+// Packet code mask maximum length. kFECPacketMaskMaxSize = MaxNumFECPackets *
+// (kUlpfecMaxMediaPackets / 8), and MaxNumFECPackets is equal to maximum number
+// of media packets (kUlpfecMaxMediaPackets)
+constexpr size_t kFECPacketMaskMaxSize = 288;
+
+// Convenience constants.
+constexpr size_t kUlpfecMinPacketMaskSize = kUlpfecPacketMaskSizeLBitClear;
+constexpr size_t kUlpfecMaxPacketMaskSize = kUlpfecPacketMaskSizeLBitSet;
+
+namespace internal {
+
+class PacketMaskTable {
+ public:
+  PacketMaskTable(FecMaskType fec_mask_type, int num_media_packets);
+  ~PacketMaskTable();
+
+  rtc::ArrayView<const uint8_t> LookUp(int num_media_packets,
+                                       int num_fec_packets);
+
+ private:
+  static const uint8_t* PickTable(FecMaskType fec_mask_type,
+                                  int num_media_packets);
+  const uint8_t* table_;
+  uint8_t fec_packet_mask_[kFECPacketMaskMaxSize];
+};
+
+rtc::ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
+                                               int media_packet_index,
+                                               int fec_index);
+
+// Returns an array of packet masks. The mask of a single FEC packet
+// corresponds to a number of mask bytes. The mask indicates which
+// media packets should be protected by the FEC packet.
+
+// \param[in]  num_media_packets       The number of media packets to protect.
+//                                     [1, max_media_packets].
+// \param[in]  num_fec_packets         The number of FEC packets which will
+//                                     be generated. [1, num_media_packets].
+// \param[in]  num_imp_packets         The number of important packets.
+//                                     [0, num_media_packets].
+//                                     num_imp_packets = 0 is the equal
+//                                     protection scenario.
+// \param[in]  use_unequal_protection  Enables unequal protection: allocates
+//                                     more protection to the num_imp_packets.
+// \param[in]  mask_table              An instance of the `PacketMaskTable`
+//                                     class, which contains the type of FEC
+//                                     packet mask used, and a pointer to the
+//                                     corresponding packet masks.
+// \param[out] packet_mask             A pointer to hold the packet mask array,
+//                                     of size: num_fec_packets *
+//                                     "number of mask bytes".
+void GeneratePacketMasks(int num_media_packets,
+                         int num_fec_packets,
+                         int num_imp_packets,
+                         bool use_unequal_protection,
+                         PacketMaskTable* mask_table,
+                         uint8_t* packet_mask);
+
+// Returns the required packet mask size, given the number of sequence numbers
+// that will be covered.
+size_t PacketMaskSize(size_t num_sequence_numbers);
+
+// Inserts `num_zeros` zero columns into `new_mask` at position
+// `new_bit_index`. If the current byte of `new_mask` can't fit all zeros, the
+// byte will be filled with zeros from `new_bit_index`, but the next byte will
+// be untouched.
+void InsertZeroColumns(int num_zeros,
+                       uint8_t* new_mask,
+                       int new_mask_bytes,
+                       int num_fec_packets,
+                       int new_bit_index);
+
+// Copies the left most bit column from the byte pointed to by
+// `old_bit_index` in `old_mask` to the right most column of the byte pointed
+// to by `new_bit_index` in `new_mask`. `old_mask_bytes` and `new_mask_bytes`
+// represent the number of bytes used per row for each mask. `num_fec_packets`
+// represent the number of rows of the masks.
+// The copied bit is shifted out from `old_mask` and is shifted one step to
+// the left in `new_mask`. `new_mask` will contain "xxxx xxn0" after this
+// operation, where x are previously inserted bits and n is the new bit.
+void CopyColumn(uint8_t* new_mask,
+                int new_mask_bytes,
+                uint8_t* old_mask,
+                int old_mask_bytes,
+                int num_fec_packets,
+                int new_bit_index,
+                int old_bit_index);
+
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/frame_object.cc

@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/frame_object.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "api/video/encoded_image.h"
+#include "api/video/video_timing.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+RtpFrameObject::RtpFrameObject(
+    uint16_t first_seq_num,
+    uint16_t last_seq_num,
+    bool markerBit,
+    int times_nacked,
+    int64_t first_packet_received_time,
+    int64_t last_packet_received_time,
+    uint32_t rtp_timestamp,
+    int64_t ntp_time_ms,
+    const VideoSendTiming& timing,
+    uint8_t payload_type,
+    VideoCodecType codec,
+    VideoRotation rotation,
+    VideoContentType content_type,
+    const RTPVideoHeader& video_header,
+    const absl::optional<webrtc::ColorSpace>& color_space,
+    RtpPacketInfos packet_infos,
+    rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
+    : image_buffer_(image_buffer),
+      first_seq_num_(first_seq_num),
+      last_seq_num_(last_seq_num),
+      last_packet_received_time_(last_packet_received_time),
+      times_nacked_(times_nacked) {
+  rtp_video_header_ = video_header;
+
+  // EncodedFrame members
+  codec_type_ = codec;
+
+  // TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
+  // VCMEncodedFrame members
+  CopyCodecSpecific(&rtp_video_header_);
+  _payloadType = payload_type;
+  SetRtpTimestamp(rtp_timestamp);
+  ntp_time_ms_ = ntp_time_ms;
+  _frameType = rtp_video_header_.frame_type;
+
+  // Setting frame's playout delays to the same values
+  // as of the first packet's.
+  SetPlayoutDelay(rtp_video_header_.playout_delay);
+
+  SetEncodedData(image_buffer_);
+  _encodedWidth = rtp_video_header_.width;
+  _encodedHeight = rtp_video_header_.height;
+
+  if (packet_infos.begin() != packet_infos.end()) {
+    csrcs_ = packet_infos.begin()->csrcs();
+  }
+
+  // EncodedFrame members
+  SetPacketInfos(std::move(packet_infos));
+
+  rotation_ = rotation;
+  SetColorSpace(color_space);
+  SetVideoFrameTrackingId(rtp_video_header_.video_frame_tracking_id);
+  content_type_ = content_type;
+  if (timing.flags != VideoSendTiming::kInvalid) {
+    // ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
+    // as this will be dealt with at the time of reporting.
+    timing_.encode_start_ms = ntp_time_ms_ + timing.encode_start_delta_ms;
+    timing_.encode_finish_ms = ntp_time_ms_ + timing.encode_finish_delta_ms;
+    timing_.packetization_finish_ms =
+        ntp_time_ms_ + timing.packetization_finish_delta_ms;
+    timing_.pacer_exit_ms = ntp_time_ms_ + timing.pacer_exit_delta_ms;
+    timing_.network_timestamp_ms =
+        ntp_time_ms_ + timing.network_timestamp_delta_ms;
+    timing_.network2_timestamp_ms =
+        ntp_time_ms_ + timing.network2_timestamp_delta_ms;
+  }
+  timing_.receive_start_ms = first_packet_received_time;
+  timing_.receive_finish_ms = last_packet_received_time;
+  timing_.flags = timing.flags;
+  is_last_spatial_layer = markerBit;
+}
+
+RtpFrameObject::~RtpFrameObject() {}
+
+uint16_t RtpFrameObject::first_seq_num() const {
+  return first_seq_num_;
+}
+
+uint16_t RtpFrameObject::last_seq_num() const {
+  return last_seq_num_;
+}
+
+int RtpFrameObject::times_nacked() const {
+  return times_nacked_;
+}
+
+VideoFrameType RtpFrameObject::frame_type() const {
+  return rtp_video_header_.frame_type;
+}
+
+VideoCodecType RtpFrameObject::codec_type() const {
+  return codec_type_;
+}
+
+int64_t RtpFrameObject::ReceivedTime() const {
+  return last_packet_received_time_;
+}
+
+int64_t RtpFrameObject::RenderTime() const {
+  return _renderTimeMs;
+}
+
+bool RtpFrameObject::delayed_by_retransmission() const {
+  return times_nacked() > 0;
+}
+
+const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
+  return rtp_video_header_;
+}
+
+void RtpFrameObject::SetHeaderFromMetadata(const VideoFrameMetadata& metadata) {
+  rtp_video_header_.SetFromMetadata(metadata);
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/frame_object.h

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
+#define MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_frame.h"
+#include "api/video/video_frame_metadata.h"
+
+namespace webrtc {
+
+class RtpFrameObject : public EncodedFrame {
+ public:
+  RtpFrameObject(uint16_t first_seq_num,
+                 uint16_t last_seq_num,
+                 bool markerBit,
+                 int times_nacked,
+                 int64_t first_packet_received_time,
+                 int64_t last_packet_received_time,
+                 uint32_t rtp_timestamp,
+                 int64_t ntp_time_ms,
+                 const VideoSendTiming& timing,
+                 uint8_t payload_type,
+                 VideoCodecType codec,
+                 VideoRotation rotation,
+                 VideoContentType content_type,
+                 const RTPVideoHeader& video_header,
+                 const absl::optional<webrtc::ColorSpace>& color_space,
+                 RtpPacketInfos packet_infos,
+                 rtc::scoped_refptr<EncodedImageBuffer> image_buffer);
+
+  ~RtpFrameObject() override;
+  uint16_t first_seq_num() const;
+  uint16_t last_seq_num() const;
+  int times_nacked() const;
+  VideoFrameType frame_type() const;
+  VideoCodecType codec_type() const;
+  int64_t ReceivedTime() const override;
+  int64_t RenderTime() const override;
+  bool delayed_by_retransmission() const override;
+  const RTPVideoHeader& GetRtpVideoHeader() const;
+
+  uint8_t* mutable_data() { return image_buffer_->data(); }
+
+  const std::vector<uint32_t>& Csrcs() const { return csrcs_; }
+
+  void SetFirstSeqNum(uint16_t first_seq_num) {
+    first_seq_num_ = first_seq_num;
+  }
+  void SetLastSeqNum(uint16_t last_seq_num) { last_seq_num_ = last_seq_num; }
+  void SetHeaderFromMetadata(const VideoFrameMetadata& metadata);
+
+ private:
+  // Reference for mutable access.
+  rtc::scoped_refptr<EncodedImageBuffer> image_buffer_;
+  RTPVideoHeader rtp_video_header_;
+  VideoCodecType codec_type_;
+  uint16_t first_seq_num_;
+  uint16_t last_seq_num_;
+  int64_t last_packet_received_time_;
+  std::vector<uint32_t> csrcs_;
+
+  // Equal to times nacked of the packet with the highet times nacked
+  // belonging to this frame.
+  int times_nacked_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/leb128.cc

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2023 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 "modules/rtp_rtcp/source/leb128.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+int Leb128Size(uint64_t value) {
+  int size = 0;
+  while (value >= 0x80) {
+    ++size;
+    value >>= 7;
+  }
+  return size + 1;
+}
+
+uint64_t ReadLeb128(const uint8_t*& read_at, const uint8_t* end) {
+  uint64_t value = 0;
+  int fill_bits = 0;
+  while (read_at != end && fill_bits < 64 - 7) {
+    uint8_t leb128_byte = *read_at;
+    value |= uint64_t{leb128_byte & 0x7Fu} << fill_bits;
+    ++read_at;
+    fill_bits += 7;
+    if ((leb128_byte & 0x80) == 0) {
+      return value;
+    }
+  }
+  // Read 9 bytes and didn't find the terminator byte. Check if 10th byte
+  // is that terminator, however to fit result into uint64_t it may carry only
+  // single bit.
+  if (read_at != end && *read_at <= 1) {
+    value |= uint64_t{*read_at} << fill_bits;
+    ++read_at;
+    return value;
+  }
+  // Failed to find terminator leb128 byte.
+  read_at = nullptr;
+  return 0;
+}
+
+int WriteLeb128(uint64_t value, uint8_t* buffer) {
+  int size = 0;
+  while (value >= 0x80) {
+    buffer[size] = 0x80 | (value & 0x7F);
+    ++size;
+    value >>= 7;
+  }
+  buffer[size] = value;
+  ++size;
+  return size;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/leb128.h

@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2023 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_RTP_RTCP_SOURCE_LEB128_H_
+#define MODULES_RTP_RTCP_SOURCE_LEB128_H_
+
+#include <cstdint>
+
+namespace webrtc {
+
+// Returns number of bytes needed to store `value` in leb128 format.
+int Leb128Size(uint64_t value);
+
+// Reads leb128 encoded value and advance read_at by number of bytes consumed.
+// Sets read_at to nullptr on error.
+uint64_t ReadLeb128(const uint8_t*& read_at, const uint8_t* end);
+
+// Encodes `value` in leb128 format. Assumes buffer has size of at least
+// Leb128Size(value). Returns number of bytes consumed.
+int WriteLeb128(uint64_t value, uint8_t* buffer);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_LEB128_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc

@@ -0,0 +1,141 @@
+/*
+ *  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 "modules/rtp_rtcp/source/packet_loss_stats.h"
+
+#include <cstdint>
+#include <iterator>
+#include <vector>
+
+#include "rtc_base/checks.h"
+
+// After this many packets are added, adding additional packets will cause the
+// oldest packets to be pruned from the buffer.
+static const int kBufferSize = 100;
+
+namespace webrtc {
+
+PacketLossStats::PacketLossStats()
+    : single_loss_historic_count_(0),
+      multiple_loss_historic_event_count_(0),
+      multiple_loss_historic_packet_count_(0) {}
+
+PacketLossStats::~PacketLossStats() = default;
+
+void PacketLossStats::AddLostPacket(uint16_t sequence_number) {
+  // Detect sequence number wrap around.
+  if (!lost_packets_buffer_.empty() &&
+      static_cast<int>(*(lost_packets_buffer_.rbegin())) - sequence_number >
+          0x8000) {
+    // The buffer contains large numbers and this is a small number.
+    lost_packets_wrapped_buffer_.insert(sequence_number);
+  } else {
+    lost_packets_buffer_.insert(sequence_number);
+  }
+  if (lost_packets_wrapped_buffer_.size() + lost_packets_buffer_.size() >
+          kBufferSize ||
+      (!lost_packets_wrapped_buffer_.empty() &&
+       *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000)) {
+    PruneBuffer();
+  }
+}
+
+int PacketLossStats::GetSingleLossCount() const {
+  int single_loss_count, unused1, unused2;
+  ComputeLossCounts(&single_loss_count, &unused1, &unused2);
+  return single_loss_count;
+}
+
+int PacketLossStats::GetMultipleLossEventCount() const {
+  int event_count, unused1, unused2;
+  ComputeLossCounts(&unused1, &event_count, &unused2);
+  return event_count;
+}
+
+int PacketLossStats::GetMultipleLossPacketCount() const {
+  int packet_count, unused1, unused2;
+  ComputeLossCounts(&unused1, &unused2, &packet_count);
+  return packet_count;
+}
+
+void PacketLossStats::ComputeLossCounts(
+    int* out_single_loss_count,
+    int* out_multiple_loss_event_count,
+    int* out_multiple_loss_packet_count) const {
+  *out_single_loss_count = single_loss_historic_count_;
+  *out_multiple_loss_event_count = multiple_loss_historic_event_count_;
+  *out_multiple_loss_packet_count = multiple_loss_historic_packet_count_;
+  if (lost_packets_buffer_.empty()) {
+    RTC_DCHECK(lost_packets_wrapped_buffer_.empty());
+    return;
+  }
+  uint16_t last_num = 0;
+  int sequential_count = 0;
+  std::vector<const std::set<uint16_t>*> buffers;
+  buffers.push_back(&lost_packets_buffer_);
+  buffers.push_back(&lost_packets_wrapped_buffer_);
+  for (const auto* buffer : buffers) {
+    for (auto it = buffer->begin(); it != buffer->end(); ++it) {
+      uint16_t current_num = *it;
+      if (sequential_count > 0 && current_num != ((last_num + 1) & 0xFFFF)) {
+        if (sequential_count == 1) {
+          (*out_single_loss_count)++;
+        } else {
+          (*out_multiple_loss_event_count)++;
+          *out_multiple_loss_packet_count += sequential_count;
+        }
+        sequential_count = 0;
+      }
+      sequential_count++;
+      last_num = current_num;
+    }
+  }
+  if (sequential_count == 1) {
+    (*out_single_loss_count)++;
+  } else if (sequential_count > 1) {
+    (*out_multiple_loss_event_count)++;
+    *out_multiple_loss_packet_count += sequential_count;
+  }
+}
+
+void PacketLossStats::PruneBuffer() {
+  // Remove the oldest lost packet and any contiguous packets and move them
+  // into the historic counts.
+  auto it = lost_packets_buffer_.begin();
+  uint16_t last_removed = 0;
+  int remove_count = 0;
+  // Count adjacent packets and continue counting if it is wrap around by
+  // swapping in the wrapped buffer and letting our value wrap as well.
+  while (remove_count == 0 || (!lost_packets_buffer_.empty() &&
+                               *it == ((last_removed + 1) & 0xFFFF))) {
+    last_removed = *it;
+    remove_count++;
+    auto to_erase = it++;
+    lost_packets_buffer_.erase(to_erase);
+    if (lost_packets_buffer_.empty()) {
+      lost_packets_buffer_.swap(lost_packets_wrapped_buffer_);
+      it = lost_packets_buffer_.begin();
+    }
+  }
+  if (remove_count > 1) {
+    multiple_loss_historic_event_count_++;
+    multiple_loss_historic_packet_count_ += remove_count;
+  } else {
+    single_loss_historic_count_++;
+  }
+  // Continue pruning if the wrapped buffer is beyond a threshold and there are
+  // things left in the pre-wrapped buffer.
+  if (!lost_packets_wrapped_buffer_.empty() &&
+      *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000) {
+    PruneBuffer();
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h

@@ -0,0 +1,58 @@
+/*
+ *  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_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
+#define MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
+
+#include <stdint.h>
+
+#include <set>
+
+namespace webrtc {
+
+// Keeps track of statistics of packet loss including whether losses are a
+// single packet or multiple packets in a row.
+class PacketLossStats {
+ public:
+  PacketLossStats();
+  ~PacketLossStats();
+
+  // Adds a lost packet to the stats by sequence number.
+  void AddLostPacket(uint16_t sequence_number);
+
+  // Queries the number of packets that were lost by themselves, no neighboring
+  // packets were lost.
+  int GetSingleLossCount() const;
+
+  // Queries the number of times that multiple packets with sequential numbers
+  // were lost. This is the number of events with more than one packet lost,
+  // regardless of the size of the event;
+  int GetMultipleLossEventCount() const;
+
+  // Queries the number of packets lost in multiple packet loss events. Combined
+  // with the event count, this can be used to determine the average event size.
+  int GetMultipleLossPacketCount() const;
+
+ private:
+  std::set<uint16_t> lost_packets_buffer_;
+  std::set<uint16_t> lost_packets_wrapped_buffer_;
+  int single_loss_historic_count_;
+  int multiple_loss_historic_event_count_;
+  int multiple_loss_historic_packet_count_;
+
+  void ComputeLossCounts(int* out_single_loss_count,
+                         int* out_multiple_loss_event_count,
+                         int* out_multiple_loss_packet_count) const;
+  void PruneBuffer();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/packet_sequencer.cc

@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2021 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 "modules/rtp_rtcp/source/packet_sequencer.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+
+namespace {
+// RED header is first byte of payload, if present.
+constexpr size_t kRedForFecHeaderLength = 1;
+
+// Timestamps use a 90kHz clock.
+constexpr uint32_t kTimestampTicksPerMs = 90;
+}  // namespace
+
+PacketSequencer::PacketSequencer(uint32_t media_ssrc,
+                                 absl::optional<uint32_t> rtx_ssrc,
+                                 bool require_marker_before_media_padding,
+                                 Clock* clock)
+    : media_ssrc_(media_ssrc),
+      rtx_ssrc_(rtx_ssrc),
+      require_marker_before_media_padding_(require_marker_before_media_padding),
+      clock_(clock),
+      media_sequence_number_(0),
+      rtx_sequence_number_(0),
+      last_payload_type_(-1),
+      last_rtp_timestamp_(0),
+      last_packet_marker_bit_(false) {
+  Random random(clock_->TimeInMicroseconds());
+  // Random start, 16 bits. Upper half of range is avoided in order to prevent
+  // wraparound issues during startup. Sequence number 0 is avoided for
+  // historical reasons, presumably to avoid debugability or test usage
+  // conflicts.
+  constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff;  // 2^15 - 1.
+  media_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
+  rtx_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
+}
+
+void PacketSequencer::Sequence(RtpPacketToSend& packet) {
+  if (packet.Ssrc() == media_ssrc_) {
+    if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
+      // Retransmission of an already sequenced packet, ignore.
+      return;
+    } else if (packet.packet_type() == RtpPacketMediaType::kPadding) {
+      PopulatePaddingFields(packet);
+    }
+    packet.SetSequenceNumber(media_sequence_number_++);
+    if (packet.packet_type() != RtpPacketMediaType::kPadding) {
+      UpdateLastPacketState(packet);
+    }
+  } else if (packet.Ssrc() == rtx_ssrc_) {
+    if (packet.packet_type() == RtpPacketMediaType::kPadding) {
+      PopulatePaddingFields(packet);
+    }
+    packet.SetSequenceNumber(rtx_sequence_number_++);
+  } else {
+    RTC_DCHECK_NOTREACHED() << "Unexpected ssrc " << packet.Ssrc();
+  }
+}
+
+void PacketSequencer::SetRtpState(const RtpState& state) {
+  media_sequence_number_ = state.sequence_number;
+  last_rtp_timestamp_ = state.timestamp;
+  last_capture_time_ = state.capture_time;
+  last_timestamp_time_ = state.last_timestamp_time;
+}
+
+void PacketSequencer::PopulateRtpState(RtpState& state) const {
+  state.sequence_number = media_sequence_number_;
+  state.timestamp = last_rtp_timestamp_;
+  state.capture_time = last_capture_time_;
+  state.last_timestamp_time = last_timestamp_time_;
+}
+
+void PacketSequencer::UpdateLastPacketState(const RtpPacketToSend& packet) {
+  // Remember marker bit to determine if padding can be inserted with
+  // sequence number following `packet`.
+  last_packet_marker_bit_ = packet.Marker();
+  // Remember media payload type to use in the padding packet if rtx is
+  // disabled.
+  if (packet.is_red()) {
+    RTC_DCHECK_GE(packet.payload_size(), kRedForFecHeaderLength);
+    last_payload_type_ = packet.PayloadBuffer()[0];
+  } else {
+    last_payload_type_ = packet.PayloadType();
+  }
+  // Save timestamps to generate timestamp field and extensions for the padding.
+  last_rtp_timestamp_ = packet.Timestamp();
+  last_timestamp_time_ = clock_->CurrentTime();
+  last_capture_time_ = packet.capture_time();
+}
+
+void PacketSequencer::PopulatePaddingFields(RtpPacketToSend& packet) {
+  if (packet.Ssrc() == media_ssrc_) {
+    RTC_DCHECK(CanSendPaddingOnMediaSsrc());
+
+    packet.SetTimestamp(last_rtp_timestamp_);
+    packet.set_capture_time(last_capture_time_);
+    packet.SetPayloadType(last_payload_type_);
+    return;
+  }
+
+  RTC_DCHECK(packet.Ssrc() == rtx_ssrc_);
+  if (packet.payload_size() > 0) {
+    // This is payload padding packet, don't update timestamp fields.
+    return;
+  }
+
+  packet.SetTimestamp(last_rtp_timestamp_);
+  packet.set_capture_time(last_capture_time_);
+
+  // Only change the timestamp of padding packets sent over RTX.
+  // Padding only packets over RTP has to be sent as part of a media
+  // frame (and therefore the same timestamp).
+  if (last_timestamp_time_ > Timestamp::Zero()) {
+    TimeDelta since_last_media = clock_->CurrentTime() - last_timestamp_time_;
+    packet.SetTimestamp(packet.Timestamp() +
+                        since_last_media.ms() * kTimestampTicksPerMs);
+    if (packet.capture_time() > Timestamp::Zero()) {
+      packet.set_capture_time(packet.capture_time() + since_last_media);
+    }
+  }
+}
+
+bool PacketSequencer::CanSendPaddingOnMediaSsrc() const {
+  if (last_payload_type_ == -1) {
+    return false;
+  }
+
+  // Without RTX we can't send padding in the middle of frames.
+  // For audio marker bits doesn't mark the end of a frame and frames
+  // are usually a single packet, so for now we don't apply this rule
+  // for audio.
+  if (require_marker_before_media_padding_ && !last_packet_marker_bit_) {
+    return false;
+  }
+
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/packet_sequencer.h

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2021 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_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
+#define MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+// Helper class used to assign RTP sequence numbers and populate some fields for
+// padding packets based on the last sequenced packets.
+// This class is not thread safe, the caller must provide that.
+class PacketSequencer {
+ public:
+  // If `require_marker_before_media_padding_` is true, padding packets on the
+  // media ssrc is not allowed unless the last sequenced media packet had the
+  // marker bit set (i.e. don't insert padding packets between the first and
+  // last packets of a video frame).
+  // Packets with unknown SSRCs will be ignored.
+  PacketSequencer(uint32_t media_ssrc,
+                  absl::optional<uint32_t> rtx_ssrc,
+                  bool require_marker_before_media_padding,
+                  Clock* clock);
+
+  // Assigns sequence number, and in the case of non-RTX padding also timestamps
+  // and payload type.
+  void Sequence(RtpPacketToSend& packet);
+
+  void set_media_sequence_number(uint16_t sequence_number) {
+    media_sequence_number_ = sequence_number;
+  }
+  void set_rtx_sequence_number(uint16_t sequence_number) {
+    rtx_sequence_number_ = sequence_number;
+  }
+
+  void SetRtpState(const RtpState& state);
+  void PopulateRtpState(RtpState& state) const;
+
+  uint16_t media_sequence_number() const { return media_sequence_number_; }
+  uint16_t rtx_sequence_number() const { return rtx_sequence_number_; }
+
+  // Checks whether it is allowed to send padding on the media SSRC at this
+  // time, e.g. that we don't send padding in the middle of a video frame.
+  bool CanSendPaddingOnMediaSsrc() const;
+
+ private:
+  void UpdateLastPacketState(const RtpPacketToSend& packet);
+  void PopulatePaddingFields(RtpPacketToSend& packet);
+
+  const uint32_t media_ssrc_;
+  const absl::optional<uint32_t> rtx_ssrc_;
+  const bool require_marker_before_media_padding_;
+  Clock* const clock_;
+
+  uint16_t media_sequence_number_;
+  uint16_t rtx_sequence_number_;
+
+  int8_t last_payload_type_;
+  uint32_t last_rtp_timestamp_;
+  Timestamp last_capture_time_ = Timestamp::MinusInfinity();
+  Timestamp last_timestamp_time_ = Timestamp::MinusInfinity();
+  bool last_packet_marker_bit_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc

@@ -0,0 +1,433 @@
+/*
+ *  Copyright (c) 2013 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 "modules/rtp_rtcp/source/receive_statistics_impl.h"
+
+#include <cmath>
+#include <cstdlib>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/units/time_delta.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kStatisticsTimeout = TimeDelta::Seconds(8);
+constexpr TimeDelta kStatisticsProcessInterval = TimeDelta::Seconds(1);
+
+TimeDelta UnixEpochDelta(Clock& clock) {
+  Timestamp now = clock.CurrentTime();
+  NtpTime ntp_now = clock.ConvertTimestampToNtpTime(now);
+  return TimeDelta::Millis(ntp_now.ToMs() - now.ms() -
+                           rtc::kNtpJan1970Millisecs);
+}
+
+}  // namespace
+
+StreamStatistician::~StreamStatistician() {}
+
+StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc,
+                                               Clock* clock,
+                                               int max_reordering_threshold)
+    : ssrc_(ssrc),
+      clock_(clock),
+      delta_internal_unix_epoch_(UnixEpochDelta(*clock_)),
+      incoming_bitrate_(/*max_window_size=*/kStatisticsProcessInterval),
+      max_reordering_threshold_(max_reordering_threshold),
+      enable_retransmit_detection_(false),
+      cumulative_loss_is_capped_(false),
+      jitter_q4_(0),
+      cumulative_loss_(0),
+      cumulative_loss_rtcp_offset_(0),
+      last_received_timestamp_(0),
+      received_seq_first_(-1),
+      received_seq_max_(-1),
+      last_report_cumulative_loss_(0),
+      last_report_seq_max_(-1),
+      last_payload_type_frequency_(0) {}
+
+StreamStatisticianImpl::~StreamStatisticianImpl() = default;
+
+bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
+                                              int64_t sequence_number,
+                                              Timestamp now) {
+  // Check if `packet` is second packet of a stream restart.
+  if (received_seq_out_of_order_) {
+    // Count the previous packet as a received; it was postponed below.
+    --cumulative_loss_;
+
+    uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1;
+    received_seq_out_of_order_ = absl::nullopt;
+    if (packet.SequenceNumber() == expected_sequence_number) {
+      // Ignore sequence number gap caused by stream restart for packet loss
+      // calculation, by setting received_seq_max_ to the sequence number just
+      // before the out-of-order seqno. This gives a net zero change of
+      // `cumulative_loss_`, for the two packets interpreted as a stream reset.
+      //
+      // Fraction loss for the next report may get a bit off, since we don't
+      // update last_report_seq_max_ and last_report_cumulative_loss_ in a
+      // consistent way.
+      last_report_seq_max_ = sequence_number - 2;
+      received_seq_max_ = sequence_number - 2;
+      return false;
+    }
+  }
+
+  if (std::abs(sequence_number - received_seq_max_) >
+      max_reordering_threshold_) {
+    // Sequence number gap looks too large, wait until next packet to check
+    // for a stream restart.
+    received_seq_out_of_order_ = packet.SequenceNumber();
+    // Postpone counting this as a received packet until we know how to update
+    // `received_seq_max_`, otherwise we temporarily decrement
+    // `cumulative_loss_`. The
+    // ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
+    // `cumulative_loss_` to be unchanged by the reception of the first packet
+    // after stream reset.
+    ++cumulative_loss_;
+    return true;
+  }
+
+  if (sequence_number > received_seq_max_)
+    return false;
+
+  // Old out of order packet, may be retransmit.
+  if (enable_retransmit_detection_ && IsRetransmitOfOldPacket(packet, now))
+    receive_counters_.retransmitted.AddPacket(packet);
+  return true;
+}
+
+void StreamStatisticianImpl::UpdateCounters(const RtpPacketReceived& packet) {
+  RTC_DCHECK_EQ(ssrc_, packet.Ssrc());
+  Timestamp now = clock_->CurrentTime();
+
+  incoming_bitrate_.Update(packet.size(), now);
+  receive_counters_.transmitted.AddPacket(packet);
+  --cumulative_loss_;
+
+  // Use PeekUnwrap and later update the state to avoid updating the state for
+  // out of order packets.
+  int64_t sequence_number = seq_unwrapper_.PeekUnwrap(packet.SequenceNumber());
+
+  if (!ReceivedRtpPacket()) {
+    received_seq_first_ = sequence_number;
+    last_report_seq_max_ = sequence_number - 1;
+    received_seq_max_ = sequence_number - 1;
+    receive_counters_.first_packet_time = now;
+  } else if (UpdateOutOfOrder(packet, sequence_number, now)) {
+    return;
+  }
+  // In order packet.
+  cumulative_loss_ += sequence_number - received_seq_max_;
+  received_seq_max_ = sequence_number;
+  // Update the internal state of `seq_unwrapper_`.
+  seq_unwrapper_.Unwrap(packet.SequenceNumber());
+
+  // If new time stamp and more than one in-order packet received, calculate
+  // new jitter statistics.
+  if (packet.Timestamp() != last_received_timestamp_ &&
+      (receive_counters_.transmitted.packets -
+       receive_counters_.retransmitted.packets) > 1) {
+    UpdateJitter(packet, now);
+  }
+  last_received_timestamp_ = packet.Timestamp();
+  last_receive_time_ = now;
+}
+
+void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
+                                          Timestamp receive_time) {
+  RTC_DCHECK(last_receive_time_.has_value());
+  TimeDelta receive_diff = receive_time - *last_receive_time_;
+  RTC_DCHECK_GE(receive_diff, TimeDelta::Zero());
+  uint32_t receive_diff_rtp =
+      (receive_diff * packet.payload_type_frequency()).seconds<uint32_t>();
+  int32_t time_diff_samples =
+      receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
+
+  ReviseFrequencyAndJitter(packet.payload_type_frequency());
+
+  // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
+  // If this happens, don't update jitter value. Use 5 secs video frequency
+  // as the threshold.
+  if (time_diff_samples < 5 * kVideoPayloadTypeFrequency &&
+      time_diff_samples > -5 * kVideoPayloadTypeFrequency) {
+    // Note we calculate in Q4 to avoid using float.
+    int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_;
+    jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
+  }
+}
+
+void StreamStatisticianImpl::ReviseFrequencyAndJitter(
+    int payload_type_frequency) {
+  if (payload_type_frequency == last_payload_type_frequency_) {
+    return;
+  }
+
+  if (payload_type_frequency != 0) {
+    if (last_payload_type_frequency_ != 0) {
+      // Value in "jitter_q4_" variable is a number of samples.
+      // I.e. jitter = timestamp (s) * frequency (Hz).
+      // Since the frequency has changed we have to update the number of samples
+      // accordingly. The new value should rely on a new frequency.
+
+      // If we don't do such procedure we end up with the number of samples that
+      // cannot be converted into TimeDelta correctly
+      // (i.e. jitter = jitter_q4_ >> 4 / payload_type_frequency).
+      // In such case, the number of samples has a "mix".
+
+      // Doing so we pretend that everything prior and including the current
+      // packet were computed on packet's frequency.
+      jitter_q4_ = static_cast<int>(static_cast<uint64_t>(jitter_q4_) *
+                                    payload_type_frequency /
+                                    last_payload_type_frequency_);
+    }
+    // If last_payload_type_frequency_ is not present, the jitter_q4_
+    // variable has its initial value.
+
+    // Keep last_payload_type_frequency_ up to date and non-zero (set).
+    last_payload_type_frequency_ = payload_type_frequency;
+  }
+}
+
+void StreamStatisticianImpl::SetMaxReorderingThreshold(
+    int max_reordering_threshold) {
+  max_reordering_threshold_ = max_reordering_threshold;
+}
+
+void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) {
+  enable_retransmit_detection_ = enable;
+}
+
+RtpReceiveStats StreamStatisticianImpl::GetStats() const {
+  RtpReceiveStats stats;
+  stats.packets_lost = cumulative_loss_;
+  // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
+  stats.jitter = jitter_q4_ >> 4;
+  if (last_payload_type_frequency_ > 0) {
+    // Divide value in fractional seconds by frequency to get jitter in
+    // fractional seconds.
+    stats.interarrival_jitter =
+        TimeDelta::Seconds(stats.jitter) / last_payload_type_frequency_;
+  }
+  if (last_receive_time_.has_value()) {
+    stats.last_packet_received =
+        *last_receive_time_ + delta_internal_unix_epoch_;
+  }
+  stats.packet_counter = receive_counters_.transmitted;
+  return stats;
+}
+
+void StreamStatisticianImpl::MaybeAppendReportBlockAndReset(
+    std::vector<rtcp::ReportBlock>& report_blocks) {
+  if (!ReceivedRtpPacket()) {
+    return;
+  }
+  Timestamp now = clock_->CurrentTime();
+  if (now - *last_receive_time_ >= kStatisticsTimeout) {
+    // Not active.
+    return;
+  }
+
+  report_blocks.emplace_back();
+  rtcp::ReportBlock& stats = report_blocks.back();
+  stats.SetMediaSsrc(ssrc_);
+  // Calculate fraction lost.
+  int64_t exp_since_last = received_seq_max_ - last_report_seq_max_;
+  RTC_DCHECK_GE(exp_since_last, 0);
+
+  int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
+  if (exp_since_last > 0 && lost_since_last > 0) {
+    // Scale 0 to 255, where 255 is 100% loss.
+    stats.SetFractionLost(255 * lost_since_last / exp_since_last);
+  }
+
+  int packets_lost = cumulative_loss_ + cumulative_loss_rtcp_offset_;
+  if (packets_lost < 0) {
+    // Clamp to zero. Work around to accommodate for senders that misbehave with
+    // negative cumulative loss.
+    packets_lost = 0;
+    cumulative_loss_rtcp_offset_ = -cumulative_loss_;
+  }
+  if (packets_lost > 0x7fffff) {
+    // Packets lost is a 24 bit signed field, and thus should be clamped, as
+    // described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3
+    if (!cumulative_loss_is_capped_) {
+      cumulative_loss_is_capped_ = true;
+      RTC_LOG(LS_WARNING) << "Cumulative loss reached maximum value for ssrc "
+                          << ssrc_;
+    }
+    packets_lost = 0x7fffff;
+  }
+  stats.SetCumulativeLost(packets_lost);
+  stats.SetExtHighestSeqNum(received_seq_max_);
+  // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
+  stats.SetJitter(jitter_q4_ >> 4);
+
+  // Only for report blocks in RTCP SR and RR.
+  last_report_cumulative_loss_ = cumulative_loss_;
+  last_report_seq_max_ = received_seq_max_;
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now.ms(),
+                                  cumulative_loss_, ssrc_);
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now.ms(),
+                                  (received_seq_max_ - received_seq_first_),
+                                  ssrc_);
+}
+
+absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
+  if (!ReceivedRtpPacket()) {
+    return absl::nullopt;
+  }
+  int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_;
+  if (expected_packets <= 0) {
+    return absl::nullopt;
+  }
+  if (cumulative_loss_ <= 0) {
+    return 0;
+  }
+  return 100 * static_cast<int64_t>(cumulative_loss_) / expected_packets;
+}
+
+StreamDataCounters StreamStatisticianImpl::GetReceiveStreamDataCounters()
+    const {
+  return receive_counters_;
+}
+
+uint32_t StreamStatisticianImpl::BitrateReceived() const {
+  return incoming_bitrate_.Rate(clock_->CurrentTime())
+      .value_or(DataRate::Zero())
+      .bps<uint32_t>();
+}
+
+bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
+    const RtpPacketReceived& packet,
+    Timestamp now) const {
+  int frequency_hz = packet.payload_type_frequency();
+  RTC_DCHECK(last_receive_time_.has_value());
+  RTC_CHECK_GT(frequency_hz, 0);
+  TimeDelta time_diff = now - *last_receive_time_;
+
+  // Diff in time stamp since last received in order.
+  uint32_t timestamp_diff = packet.Timestamp() - last_received_timestamp_;
+  TimeDelta rtp_time_stamp_diff =
+      TimeDelta::Seconds(timestamp_diff) / frequency_hz;
+
+  // Jitter standard deviation in samples.
+  float jitter_std = std::sqrt(static_cast<float>(jitter_q4_ >> 4));
+
+  // 2 times the standard deviation => 95% confidence.
+  // Min max_delay is 1ms.
+  TimeDelta max_delay = std::max(
+      TimeDelta::Seconds(2 * jitter_std / frequency_hz), TimeDelta::Millis(1));
+
+  return time_diff > rtp_time_stamp_diff + max_delay;
+}
+
+std::unique_ptr<ReceiveStatistics> ReceiveStatistics::Create(Clock* clock) {
+  return std::make_unique<ReceiveStatisticsLocked>(
+      clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
+        return std::make_unique<StreamStatisticianLocked>(
+            ssrc, clock, max_reordering_threshold);
+      });
+}
+
+std::unique_ptr<ReceiveStatistics> ReceiveStatistics::CreateThreadCompatible(
+    Clock* clock) {
+  return std::make_unique<ReceiveStatisticsImpl>(
+      clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
+        return std::make_unique<StreamStatisticianImpl>(
+            ssrc, clock, max_reordering_threshold);
+      });
+}
+
+ReceiveStatisticsImpl::ReceiveStatisticsImpl(
+    Clock* clock,
+    std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+        uint32_t ssrc,
+        Clock* clock,
+        int max_reordering_threshold)> stream_statistician_factory)
+    : clock_(clock),
+      stream_statistician_factory_(std::move(stream_statistician_factory)),
+      last_returned_ssrc_idx_(0),
+      max_reordering_threshold_(kDefaultMaxReorderingThreshold) {}
+
+void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
+  // StreamStatisticianImpl instance is created once and only destroyed when
+  // this whole ReceiveStatisticsImpl is destroyed. StreamStatisticianImpl has
+  // it's own locking so don't hold receive_statistics_lock_ (potential
+  // deadlock).
+  GetOrCreateStatistician(packet.Ssrc())->UpdateCounters(packet);
+}
+
+StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
+    uint32_t ssrc) const {
+  const auto& it = statisticians_.find(ssrc);
+  if (it == statisticians_.end())
+    return nullptr;
+  return it->second.get();
+}
+
+StreamStatisticianImplInterface* ReceiveStatisticsImpl::GetOrCreateStatistician(
+    uint32_t ssrc) {
+  std::unique_ptr<StreamStatisticianImplInterface>& impl = statisticians_[ssrc];
+  if (impl == nullptr) {  // new element
+    impl =
+        stream_statistician_factory_(ssrc, clock_, max_reordering_threshold_);
+    all_ssrcs_.push_back(ssrc);
+  }
+  return impl.get();
+}
+
+void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
+    int max_reordering_threshold) {
+  max_reordering_threshold_ = max_reordering_threshold;
+  for (auto& statistician : statisticians_) {
+    statistician.second->SetMaxReorderingThreshold(max_reordering_threshold);
+  }
+}
+
+void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
+    uint32_t ssrc,
+    int max_reordering_threshold) {
+  GetOrCreateStatistician(ssrc)->SetMaxReorderingThreshold(
+      max_reordering_threshold);
+}
+
+void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
+                                                      bool enable) {
+  GetOrCreateStatistician(ssrc)->EnableRetransmitDetection(enable);
+}
+
+std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
+    size_t max_blocks) {
+  std::vector<rtcp::ReportBlock> result;
+  result.reserve(std::min(max_blocks, all_ssrcs_.size()));
+
+  size_t ssrc_idx = 0;
+  for (size_t i = 0; i < all_ssrcs_.size() && result.size() < max_blocks; ++i) {
+    ssrc_idx = (last_returned_ssrc_idx_ + i + 1) % all_ssrcs_.size();
+    const uint32_t media_ssrc = all_ssrcs_[ssrc_idx];
+    auto statistician_it = statisticians_.find(media_ssrc);
+    RTC_DCHECK(statistician_it != statisticians_.end());
+    statistician_it->second->MaybeAppendReportBlockAndReset(result);
+  }
+  last_returned_ssrc_idx_ = ssrc_idx;
+  return result;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h

@@ -0,0 +1,252 @@
+/*
+ *  Copyright (c) 2013 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_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
+#define MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
+
+#include <algorithm>
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "rtc_base/bitrate_tracker.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Extends StreamStatistician with methods needed by the implementation.
+class StreamStatisticianImplInterface : public StreamStatistician {
+ public:
+  virtual ~StreamStatisticianImplInterface() = default;
+  virtual void MaybeAppendReportBlockAndReset(
+      std::vector<rtcp::ReportBlock>& report_blocks) = 0;
+  virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
+  virtual void EnableRetransmitDetection(bool enable) = 0;
+  virtual void UpdateCounters(const RtpPacketReceived& packet) = 0;
+};
+
+// Thread-compatible implementation of StreamStatisticianImplInterface.
+class StreamStatisticianImpl : public StreamStatisticianImplInterface {
+ public:
+  StreamStatisticianImpl(uint32_t ssrc,
+                         Clock* clock,
+                         int max_reordering_threshold);
+  ~StreamStatisticianImpl() override;
+
+  // Implements StreamStatistician
+  RtpReceiveStats GetStats() const override;
+  absl::optional<int> GetFractionLostInPercent() const override;
+  StreamDataCounters GetReceiveStreamDataCounters() const override;
+  uint32_t BitrateReceived() const override;
+
+  // Implements StreamStatisticianImplInterface
+  void MaybeAppendReportBlockAndReset(
+      std::vector<rtcp::ReportBlock>& report_blocks) override;
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override;
+  void EnableRetransmitDetection(bool enable) override;
+  // Updates StreamStatistician for incoming packets.
+  void UpdateCounters(const RtpPacketReceived& packet) override;
+
+ private:
+  bool IsRetransmitOfOldPacket(const RtpPacketReceived& packet,
+                               Timestamp now) const;
+  void UpdateJitter(const RtpPacketReceived& packet, Timestamp receive_time);
+  void ReviseFrequencyAndJitter(int payload_type_frequency);
+  // Updates StreamStatistician for out of order packets.
+  // Returns true if packet considered to be out of order.
+  bool UpdateOutOfOrder(const RtpPacketReceived& packet,
+                        int64_t sequence_number,
+                        Timestamp now);
+  // Checks if this StreamStatistician received any rtp packets.
+  bool ReceivedRtpPacket() const { return last_receive_time_.has_value(); }
+
+  const uint32_t ssrc_;
+  Clock* const clock_;
+  // Delta used to map internal timestamps to Unix epoch ones.
+  const TimeDelta delta_internal_unix_epoch_;
+  BitrateTracker incoming_bitrate_;
+  // In number of packets or sequence numbers.
+  int max_reordering_threshold_;
+  bool enable_retransmit_detection_;
+  bool cumulative_loss_is_capped_;
+
+  // Stats on received RTP packets.
+  uint32_t jitter_q4_;
+  // Cumulative loss according to RFC 3550, which may be negative (and often is,
+  // if packets are reordered and there are non-RTX retransmissions).
+  int32_t cumulative_loss_;
+  // Offset added to outgoing rtcp reports, to make ensure that the reported
+  // cumulative loss is non-negative. Reports with negative values confuse some
+  // senders, in particular, our own loss-based bandwidth estimator.
+  int32_t cumulative_loss_rtcp_offset_;
+
+  absl::optional<Timestamp> last_receive_time_;
+  uint32_t last_received_timestamp_;
+  RtpSequenceNumberUnwrapper seq_unwrapper_;
+  int64_t received_seq_first_;
+  int64_t received_seq_max_;
+  // Assume that the other side restarted when there are two sequential packets
+  // with large jump from received_seq_max_.
+  absl::optional<uint16_t> received_seq_out_of_order_;
+
+  // Current counter values.
+  StreamDataCounters receive_counters_;
+
+  // Counter values when we sent the last report.
+  int32_t last_report_cumulative_loss_;
+  int64_t last_report_seq_max_;
+
+  // The sample frequency of the last received packet.
+  int last_payload_type_frequency_;
+};
+
+// Thread-safe implementation of StreamStatisticianImplInterface.
+class StreamStatisticianLocked : public StreamStatisticianImplInterface {
+ public:
+  StreamStatisticianLocked(uint32_t ssrc,
+                           Clock* clock,
+                           int max_reordering_threshold)
+      : impl_(ssrc, clock, max_reordering_threshold) {}
+  ~StreamStatisticianLocked() override = default;
+
+  RtpReceiveStats GetStats() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.GetStats();
+  }
+  absl::optional<int> GetFractionLostInPercent() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.GetFractionLostInPercent();
+  }
+  StreamDataCounters GetReceiveStreamDataCounters() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.GetReceiveStreamDataCounters();
+  }
+  uint32_t BitrateReceived() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.BitrateReceived();
+  }
+  void MaybeAppendReportBlockAndReset(
+      std::vector<rtcp::ReportBlock>& report_blocks) override {
+    MutexLock lock(&stream_lock_);
+    impl_.MaybeAppendReportBlockAndReset(report_blocks);
+  }
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override {
+    MutexLock lock(&stream_lock_);
+    return impl_.SetMaxReorderingThreshold(max_reordering_threshold);
+  }
+  void EnableRetransmitDetection(bool enable) override {
+    MutexLock lock(&stream_lock_);
+    return impl_.EnableRetransmitDetection(enable);
+  }
+  void UpdateCounters(const RtpPacketReceived& packet) override {
+    MutexLock lock(&stream_lock_);
+    return impl_.UpdateCounters(packet);
+  }
+
+ private:
+  mutable Mutex stream_lock_;
+  StreamStatisticianImpl impl_ RTC_GUARDED_BY(&stream_lock_);
+};
+
+// Thread-compatible implementation.
+class ReceiveStatisticsImpl : public ReceiveStatistics {
+ public:
+  ReceiveStatisticsImpl(
+      Clock* clock,
+      std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+          uint32_t ssrc,
+          Clock* clock,
+          int max_reordering_threshold)> stream_statistician_factory);
+  ~ReceiveStatisticsImpl() override = default;
+
+  // Implements ReceiveStatisticsProvider.
+  std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override;
+
+  // Implements RtpPacketSinkInterface
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+  // Implements ReceiveStatistics.
+  StreamStatistician* GetStatistician(uint32_t ssrc) const override;
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override;
+  void SetMaxReorderingThreshold(uint32_t ssrc,
+                                 int max_reordering_threshold) override;
+  void EnableRetransmitDetection(uint32_t ssrc, bool enable) override;
+
+ private:
+  StreamStatisticianImplInterface* GetOrCreateStatistician(uint32_t ssrc);
+
+  Clock* const clock_;
+  std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+      uint32_t ssrc,
+      Clock* clock,
+      int max_reordering_threshold)>
+      stream_statistician_factory_;
+  // The index within `all_ssrcs_` that was last returned.
+  size_t last_returned_ssrc_idx_;
+  std::vector<uint32_t> all_ssrcs_;
+  int max_reordering_threshold_;
+  flat_map<uint32_t /*ssrc*/, std::unique_ptr<StreamStatisticianImplInterface>>
+      statisticians_;
+};
+
+// Thread-safe implementation wrapping access to ReceiveStatisticsImpl with a
+// mutex.
+class ReceiveStatisticsLocked : public ReceiveStatistics {
+ public:
+  explicit ReceiveStatisticsLocked(
+      Clock* clock,
+      std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+          uint32_t ssrc,
+          Clock* clock,
+          int max_reordering_threshold)> stream_statitician_factory)
+      : impl_(clock, std::move(stream_statitician_factory)) {}
+  ~ReceiveStatisticsLocked() override = default;
+  std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.RtcpReportBlocks(max_blocks);
+  }
+  void OnRtpPacket(const RtpPacketReceived& packet) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.OnRtpPacket(packet);
+  }
+  StreamStatistician* GetStatistician(uint32_t ssrc) const override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.GetStatistician(ssrc);
+  }
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.SetMaxReorderingThreshold(max_reordering_threshold);
+  }
+  void SetMaxReorderingThreshold(uint32_t ssrc,
+                                 int max_reordering_threshold) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.SetMaxReorderingThreshold(ssrc, max_reordering_threshold);
+  }
+  void EnableRetransmitDetection(uint32_t ssrc, bool enable) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.EnableRetransmitDetection(ssrc, enable);
+  }
+
+ private:
+  mutable Mutex receive_statistics_lock_;
+  ReceiveStatisticsImpl impl_ RTC_GUARDED_BY(&receive_statistics_lock_);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc

@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2014 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 "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+
+#include <cstdint>
+
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kMinimumNumberOfSamples = 2;
+constexpr TimeDelta kTimingLogInterval = TimeDelta::Seconds(10);
+constexpr int kClocksOffsetSmoothingWindow = 100;
+
+// Subtracts two NtpTime values keeping maximum precision.
+int64_t Subtract(NtpTime minuend, NtpTime subtrahend) {
+  uint64_t a = static_cast<uint64_t>(minuend);
+  uint64_t b = static_cast<uint64_t>(subtrahend);
+  return a >= b ? static_cast<int64_t>(a - b) : -static_cast<int64_t>(b - a);
+}
+
+NtpTime Add(NtpTime lhs, int64_t rhs) {
+  uint64_t result = static_cast<uint64_t>(lhs);
+  if (rhs >= 0) {
+    result += static_cast<uint64_t>(rhs);
+  } else {
+    result -= static_cast<uint64_t>(-rhs);
+  }
+  return NtpTime(result);
+}
+
+}  // namespace
+
+// TODO(wu): Refactor this class so that it can be shared with
+// vie_sync_module.cc.
+RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock)
+    : clock_(clock),
+      ntp_clocks_offset_estimator_(kClocksOffsetSmoothingWindow) {}
+
+bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(TimeDelta rtt,
+                                                 NtpTime sender_send_time,
+                                                 uint32_t rtp_timestamp) {
+  switch (rtp_to_ntp_.UpdateMeasurements(sender_send_time, rtp_timestamp)) {
+    case RtpToNtpEstimator::kInvalidMeasurement:
+      return false;
+    case RtpToNtpEstimator::kSameMeasurement:
+      // No new RTCP SR since last time this function was called.
+      return true;
+    case RtpToNtpEstimator::kNewMeasurement:
+      break;
+  }
+
+  // Assume connection is symmetric and thus time to deliver the packet is half
+  // the round trip time.
+  int64_t deliver_time_ntp = ToNtpUnits(rtt) / 2;
+
+  // Update extrapolator with the new arrival time.
+  NtpTime receiver_arrival_time = clock_->CurrentNtpTime();
+  int64_t remote_to_local_clocks_offset =
+      Subtract(receiver_arrival_time, sender_send_time) - deliver_time_ntp;
+  ntp_clocks_offset_estimator_.Insert(remote_to_local_clocks_offset);
+  return true;
+}
+
+NtpTime RemoteNtpTimeEstimator::EstimateNtp(uint32_t rtp_timestamp) {
+  NtpTime sender_capture = rtp_to_ntp_.Estimate(rtp_timestamp);
+  if (!sender_capture.Valid()) {
+    return sender_capture;
+  }
+
+  int64_t remote_to_local_clocks_offset =
+      ntp_clocks_offset_estimator_.GetFilteredValue();
+  NtpTime receiver_capture = Add(sender_capture, remote_to_local_clocks_offset);
+
+  Timestamp now = clock_->CurrentTime();
+  if (now - last_timing_log_ > kTimingLogInterval) {
+    RTC_LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
+                     << " in NTP clock: " << sender_capture.ToMs()
+                     << " estimated time in receiver NTP clock: "
+                     << receiver_capture.ToMs();
+    last_timing_log_ = now;
+  }
+
+  return receiver_capture;
+}
+
+absl::optional<int64_t>
+RemoteNtpTimeEstimator::EstimateRemoteToLocalClockOffset() {
+  if (ntp_clocks_offset_estimator_.GetNumberOfSamplesStored() <
+      kMinimumNumberOfSamples) {
+    return absl::nullopt;
+  }
+  return ntp_clocks_offset_estimator_.GetFilteredValue();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2017 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 "modules/rtp_rtcp/source/rtcp_nack_stats.h"
+
+#include "modules/include/module_common_types_public.h"
+
+namespace webrtc {
+
+RtcpNackStats::RtcpNackStats()
+    : max_sequence_number_(0), requests_(0), unique_requests_(0) {}
+
+void RtcpNackStats::ReportRequest(uint16_t sequence_number) {
+  if (requests_ == 0 ||
+      IsNewerSequenceNumber(sequence_number, max_sequence_number_)) {
+    max_sequence_number_ = sequence_number;
+    ++unique_requests_;
+  }
+  ++requests_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2017 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_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+class RtcpNackStats {
+ public:
+  RtcpNackStats();
+
+  // Updates stats with requested sequence number.
+  // This function should be called for each NACK request to calculate the
+  // number of unique NACKed RTP packets.
+  void ReportRequest(uint16_t sequence_number);
+
+  // Gets the number of NACKed RTP packets.
+  uint32_t requests() const { return requests_; }
+
+  // Gets the number of unique NACKed RTP packets.
+  uint32_t unique_requests() const { return unique_requests_; }
+
+ private:
+  uint16_t max_sequence_number_;
+  uint32_t requests_;
+  uint32_t unique_requests_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc

@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2014 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 "modules/rtp_rtcp/source/rtcp_packet.h"
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr size_t RtcpPacket::kHeaderLength;
+
+rtc::Buffer RtcpPacket::Build() const {
+  rtc::Buffer packet(BlockLength());
+
+  size_t length = 0;
+  bool created = Create(packet.data(), &length, packet.capacity(), nullptr);
+  RTC_DCHECK(created) << "Invalid packet is not supported.";
+  RTC_DCHECK_EQ(length, packet.size())
+      << "BlockLength mispredicted size used by Create";
+
+  return packet;
+}
+
+bool RtcpPacket::Build(size_t max_length, PacketReadyCallback callback) const {
+  RTC_CHECK_LE(max_length, IP_PACKET_SIZE);
+  uint8_t buffer[IP_PACKET_SIZE];
+  size_t index = 0;
+  if (!Create(buffer, &index, max_length, callback))
+    return false;
+  return OnBufferFull(buffer, &index, callback);
+}
+
+bool RtcpPacket::OnBufferFull(uint8_t* packet,
+                              size_t* index,
+                              PacketReadyCallback callback) const {
+  if (*index == 0)
+    return false;
+  RTC_DCHECK(callback) << "Fragmentation not supported.";
+  callback(rtc::ArrayView<const uint8_t>(packet, *index));
+  *index = 0;
+  return true;
+}
+
+size_t RtcpPacket::HeaderLength() const {
+  size_t length_in_bytes = BlockLength();
+  RTC_DCHECK_GT(length_in_bytes, 0);
+  RTC_DCHECK_EQ(length_in_bytes % 4, 0)
+      << "Padding must be handled by each subclass.";
+  // Length in 32-bit words without common header.
+  return (length_in_bytes - kHeaderLength) / 4;
+}
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTP header format.
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P| RC/FMT  |      PT       |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+void RtcpPacket::CreateHeader(
+    size_t count_or_format,  // Depends on packet type.
+    uint8_t packet_type,
+    size_t length,
+    uint8_t* buffer,
+    size_t* pos) {
+  CreateHeader(count_or_format, packet_type, length, /*padding=*/false, buffer,
+               pos);
+}
+
+void RtcpPacket::CreateHeader(
+    size_t count_or_format,  // Depends on packet type.
+    uint8_t packet_type,
+    size_t length,
+    bool padding,
+    uint8_t* buffer,
+    size_t* pos) {
+  RTC_DCHECK_LE(length, 0xffffU);
+  RTC_DCHECK_LE(count_or_format, 0x1f);
+  constexpr uint8_t kVersionBits = 2 << 6;
+  uint8_t padding_bit = padding ? 1 << 5 : 0;
+  buffer[*pos + 0] =
+      kVersionBits | padding_bit | static_cast<uint8_t>(count_or_format);
+  buffer[*pos + 1] = packet_type;
+  buffer[*pos + 2] = (length >> 8) & 0xff;
+  buffer[*pos + 3] = length & 0xff;
+  *pos += kHeaderLength;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet.h

@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2014 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_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/function_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace rtcp {
+// Class for building RTCP packets.
+//
+//  Example:
+//  ReportBlock report_block;
+//  report_block.SetMediaSsrc(234);
+//  report_block.SetFractionLost(10);
+//
+//  ReceiverReport rr;
+//  rr.SetSenderSsrc(123);
+//  rr.AddReportBlock(report_block);
+//
+//  Fir fir;
+//  fir.SetSenderSsrc(123);
+//  fir.AddRequestTo(234, 56);
+//
+//  size_t length = 0;                     // Builds an intra frame request
+//  uint8_t packet[kPacketSize];           // with sequence number 56.
+//  fir.Build(packet, &length, kPacketSize);
+//
+//  rtc::Buffer packet = fir.Build();      // Returns a RawPacket holding
+//                                         // the built rtcp packet.
+//
+//  CompoundPacket compound;               // Builds a compound RTCP packet with
+//  compound.Append(&rr);                  // a receiver report, report block
+//  compound.Append(&fir);                 // and fir message.
+//  rtc::Buffer packet = compound.Build();
+
+class RtcpPacket {
+ public:
+  // Callback used to signal that an RTCP packet is ready. Note that this may
+  // not contain all data in this RtcpPacket; if a packet cannot fit in
+  // max_length bytes, it will be fragmented and multiple calls to this
+  // callback will be made.
+  using PacketReadyCallback =
+      rtc::FunctionView<void(rtc::ArrayView<const uint8_t> packet)>;
+
+  virtual ~RtcpPacket() = default;
+
+  void SetSenderSsrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+
+  // Convenience method mostly used for test. Creates packet without
+  // fragmentation using BlockLength() to allocate big enough buffer.
+  rtc::Buffer Build() const;
+
+  // Returns true if call to Create succeeded.
+  bool Build(size_t max_length, PacketReadyCallback callback) const;
+
+  // Size of this packet in bytes (including headers).
+  virtual size_t BlockLength() const = 0;
+
+  // Creates packet in the given buffer at the given position.
+  // Calls PacketReadyCallback::OnPacketReady if remaining buffer is too small
+  // and assume buffer can be reused after OnPacketReady returns.
+  virtual bool Create(uint8_t* packet,
+                      size_t* index,
+                      size_t max_length,
+                      PacketReadyCallback callback) const = 0;
+
+ protected:
+  // Size of the rtcp common header.
+  static constexpr size_t kHeaderLength = 4;
+  RtcpPacket() {}
+
+  static void CreateHeader(size_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Payload size in 32bit words.
+                           uint8_t* buffer,
+                           size_t* pos);
+
+  static void CreateHeader(size_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Payload size in 32bit words.
+                           bool padding,  // True if there are padding bytes.
+                           uint8_t* buffer,
+                           size_t* pos);
+
+  bool OnBufferFull(uint8_t* packet,
+                    size_t* index,
+                    PacketReadyCallback callback) const;
+  // Size of the rtcp packet as written in header.
+  size_t HeaderLength() const;
+
+ private:
+  uint32_t sender_ssrc_ = 0;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc

@@ -0,0 +1,103 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include <string.h>
+
+#include <cstdint>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t App::kPacketType;
+constexpr size_t App::kMaxDataSize;
+// Application-Defined packet (APP) (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| subtype |   PT=APP=204  |             length            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                           SSRC/CSRC                           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          name (ASCII)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |                   application-dependent data                ...
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+App::App() : sub_type_(0), name_(0) {}
+
+App::~App() = default;
+
+bool App::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  if (packet.payload_size_bytes() < kAppBaseLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to be a valid APP packet";
+    return false;
+  }
+  if (packet.payload_size_bytes() % 4 != 0) {
+    RTC_LOG(LS_WARNING)
+        << "Packet payload must be 32 bits aligned to make a valid APP packet";
+    return false;
+  }
+  sub_type_ = packet.fmt();
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[0]));
+  name_ = ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[4]);
+  data_.SetData(packet.payload() + kAppBaseLength,
+                packet.payload_size_bytes() - kAppBaseLength);
+  return true;
+}
+
+void App::SetSubType(uint8_t subtype) {
+  RTC_DCHECK_LE(subtype, 0x1f);
+  sub_type_ = subtype;
+}
+
+void App::SetData(const uint8_t* data, size_t data_length) {
+  RTC_DCHECK(data);
+  RTC_DCHECK_EQ(data_length % 4, 0) << "Data must be 32 bits aligned.";
+  RTC_DCHECK_LE(data_length, kMaxDataSize)
+      << "App data size " << data_length << " exceed maximum of "
+      << kMaxDataSize << " bytes.";
+  data_.SetData(data, data_length);
+}
+
+size_t App::BlockLength() const {
+  return kHeaderLength + kAppBaseLength + data_.size();
+}
+
+bool App::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
+
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], sender_ssrc());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
+  if (!data_.empty()) {
+    memcpy(&packet[*index + 8], data_.data(), data_.size());
+  }
+  *index += (8 + data_.size());
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h

@@ -0,0 +1,67 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class App : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 204;
+  App();
+  App(App&&) = default;
+  ~App() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetSubType(uint8_t subtype);
+  void SetName(uint32_t name) { name_ = name; }
+  void SetData(const uint8_t* data, size_t data_length);
+
+  uint8_t sub_type() const { return sub_type_; }
+  uint32_t name() const { return name_; }
+  size_t data_size() const { return data_.size(); }
+  const uint8_t* data() const { return data_.data(); }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+  static inline constexpr uint32_t NameToInt(const char name[5]) {
+    return static_cast<uint32_t>(name[0]) << 24 |
+           static_cast<uint32_t>(name[1]) << 16 |
+           static_cast<uint32_t>(name[2]) << 8 | static_cast<uint32_t>(name[3]);
+  }
+
+ private:
+  static constexpr size_t kAppBaseLength = 8;  // Ssrc and Name.
+  static constexpr size_t kMaxDataSize = 0xffff * 4 - kAppBaseLength;
+
+  uint8_t sub_type_;
+  uint32_t name_;
+  rtc::Buffer data_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc

@@ -0,0 +1,141 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Bye::kPacketType;
+// Bye packet (BYE) (RFC 3550).
+//
+//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |                           SSRC/CSRC                           |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       :                              ...                              :
+//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// (opt) |     length    |               reason for leaving            ...
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Bye::Bye() = default;
+
+Bye::~Bye() = default;
+
+bool Bye::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  const uint8_t src_count = packet.count();
+  // Validate packet.
+  if (packet.payload_size_bytes() < 4u * src_count) {
+    RTC_LOG(LS_WARNING)
+        << "Packet is too small to contain CSRCs it promise to have.";
+    return false;
+  }
+  const uint8_t* const payload = packet.payload();
+  bool has_reason = packet.payload_size_bytes() > 4u * src_count;
+  uint8_t reason_length = 0;
+  if (has_reason) {
+    reason_length = payload[4u * src_count];
+    if (packet.payload_size_bytes() - 4u * src_count < 1u + reason_length) {
+      RTC_LOG(LS_WARNING) << "Invalid reason length: " << reason_length;
+      return false;
+    }
+  }
+  // Once sure packet is valid, copy values.
+  if (src_count == 0) {  // A count value of zero is valid, but useless.
+    SetSenderSsrc(0);
+    csrcs_.clear();
+  } else {
+    SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(payload));
+    csrcs_.resize(src_count - 1);
+    for (size_t i = 1; i < src_count; ++i)
+      csrcs_[i - 1] = ByteReader<uint32_t>::ReadBigEndian(&payload[4 * i]);
+  }
+
+  if (has_reason) {
+    reason_.assign(reinterpret_cast<const char*>(&payload[4u * src_count + 1]),
+                   reason_length);
+  } else {
+    reason_.clear();
+  }
+
+  return true;
+}
+
+bool Bye::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(1 + csrcs_.size(), kPacketType, HeaderLength(), packet, index);
+  // Store srcs of the leaving clients.
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], sender_ssrc());
+  *index += sizeof(uint32_t);
+  for (uint32_t csrc : csrcs_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], csrc);
+    *index += sizeof(uint32_t);
+  }
+  // Store the reason to leave.
+  if (!reason_.empty()) {
+    uint8_t reason_length = static_cast<uint8_t>(reason_.size());
+    packet[(*index)++] = reason_length;
+    memcpy(&packet[*index], reason_.data(), reason_length);
+    *index += reason_length;
+    // Add padding bytes if needed.
+    size_t bytes_to_pad = index_end - *index;
+    RTC_DCHECK_LE(bytes_to_pad, 3);
+    if (bytes_to_pad > 0) {
+      memset(&packet[*index], 0, bytes_to_pad);
+      *index += bytes_to_pad;
+    }
+  }
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+bool Bye::SetCsrcs(std::vector<uint32_t> csrcs) {
+  if (csrcs.size() > kMaxNumberOfCsrcs) {
+    RTC_LOG(LS_WARNING) << "Too many CSRCs for Bye packet.";
+    return false;
+  }
+  csrcs_ = std::move(csrcs);
+  return true;
+}
+
+void Bye::SetReason(absl::string_view reason) {
+  RTC_DCHECK_LE(reason.size(), 0xffu);
+  reason_ = std::string(reason);
+}
+
+size_t Bye::BlockLength() const {
+  size_t src_count = (1 + csrcs_.size());
+  size_t reason_size_in_32bits = reason_.empty() ? 0 : (reason_.size() / 4 + 1);
+  return kHeaderLength + 4 * (src_count + reason_size_in_32bits);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h

@@ -0,0 +1,57 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class Bye : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 203;
+
+  Bye();
+  ~Bye() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool SetCsrcs(std::vector<uint32_t> csrcs);
+  void SetReason(absl::string_view reason);
+
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  const std::string& reason() const { return reason_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static const int kMaxNumberOfCsrcs = 0x1f - 1;  // First item is sender SSRC.
+
+  std::vector<uint32_t> csrcs_;
+  std::string reason_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.cc

@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr size_t CommonHeader::kHeaderSizeBytes;
+//    0                   1           1       2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |V=2|P|   C/F   |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 1                 |  Packet Type  |
+//   ----------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 2                                 |             length            |
+//   --------------------------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Common header for all RTCP packets, 4 octets.
+bool CommonHeader::Parse(const uint8_t* buffer, size_t size_bytes) {
+  const uint8_t kVersion = 2;
+
+  if (size_bytes < kHeaderSizeBytes) {
+    RTC_LOG(LS_WARNING)
+        << "Too little data (" << size_bytes << " byte"
+        << (size_bytes != 1 ? "s" : "")
+        << ") remaining in buffer to parse RTCP header (4 bytes).";
+    return false;
+  }
+
+  uint8_t version = buffer[0] >> 6;
+  if (version != kVersion) {
+    RTC_LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
+                        << static_cast<int>(kVersion) << " but was "
+                        << static_cast<int>(version);
+    return false;
+  }
+
+  bool has_padding = (buffer[0] & 0x20) != 0;
+  count_or_format_ = buffer[0] & 0x1F;
+  packet_type_ = buffer[1];
+  payload_size_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) * 4;
+  payload_ = buffer + kHeaderSizeBytes;
+  padding_size_ = 0;
+
+  if (size_bytes < kHeaderSizeBytes + payload_size_) {
+    RTC_LOG(LS_WARNING) << "Buffer too small (" << size_bytes
+                        << " bytes) to fit an RtcpPacket with a header and "
+                        << payload_size_ << " bytes.";
+    return false;
+  }
+
+  if (has_padding) {
+    if (payload_size_ == 0) {
+      RTC_LOG(LS_WARNING)
+          << "Invalid RTCP header: Padding bit set but 0 payload "
+             "size specified.";
+      return false;
+    }
+
+    padding_size_ = payload_[payload_size_ - 1];
+    if (padding_size_ == 0) {
+      RTC_LOG(LS_WARNING)
+          << "Invalid RTCP header: Padding bit set but 0 padding "
+             "size specified.";
+      return false;
+    }
+    if (padding_size_ > payload_size_) {
+      RTC_LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
+                          << padding_size_ << ") for a packet payload size of "
+                          << payload_size_ << " bytes.";
+      return false;
+    }
+    payload_size_ -= padding_size_;
+  }
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h

@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader {
+ public:
+  static constexpr size_t kHeaderSizeBytes = 4;
+
+  CommonHeader() {}
+  CommonHeader(const CommonHeader&) = default;
+  CommonHeader& operator=(const CommonHeader&) = default;
+
+  bool Parse(const uint8_t* buffer, size_t size_bytes);
+
+  uint8_t type() const { return packet_type_; }
+  // Depending on packet type same header field can be used either as count or
+  // as feedback message type (fmt). Caller expected to know how it is used.
+  uint8_t fmt() const { return count_or_format_; }
+  uint8_t count() const { return count_or_format_; }
+  size_t payload_size_bytes() const { return payload_size_; }
+  const uint8_t* payload() const { return payload_; }
+  size_t packet_size() const {
+    return kHeaderSizeBytes + payload_size_ + padding_size_;
+  }
+  // Returns pointer to the next RTCP packet in compound packet.
+  const uint8_t* NextPacket() const {
+    return payload_ + payload_size_ + padding_size_;
+  }
+
+ private:
+  uint8_t packet_type_ = 0;
+  uint8_t count_or_format_ = 0;
+  uint8_t padding_size_ = 0;
+  uint32_t payload_size_ = 0;
+  const uint8_t* payload_ = nullptr;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc

@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+
+CompoundPacket::CompoundPacket() = default;
+
+CompoundPacket::~CompoundPacket() = default;
+
+void CompoundPacket::Append(std::unique_ptr<RtcpPacket> packet) {
+  RTC_CHECK(packet);
+  appended_packets_.push_back(std::move(packet));
+}
+
+bool CompoundPacket::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            PacketReadyCallback callback) const {
+  for (const auto& appended : appended_packets_) {
+    if (!appended->Create(packet, index, max_length, callback))
+      return false;
+  }
+  return true;
+}
+
+size_t CompoundPacket::BlockLength() const {
+  size_t block_length = 0;
+  for (const auto& appended : appended_packets_) {
+    block_length += appended->BlockLength();
+  }
+  return block_length;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class CompoundPacket : public RtcpPacket {
+ public:
+  CompoundPacket();
+  ~CompoundPacket() override;
+
+  CompoundPacket(const CompoundPacket&) = delete;
+  CompoundPacket& operator=(const CompoundPacket&) = delete;
+
+  void Append(std::unique_ptr<RtcpPacket> packet);
+
+  // Size of this packet in bytes (i.e. total size of nested packets).
+  size_t BlockLength() const override;
+  // Returns true if all calls to Create succeeded.
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ protected:
+  std::vector<std::unique_ptr<RtcpPacket>> appended_packets_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc

@@ -0,0 +1,94 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace rtcp {
+// DLRR Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=5      |   reserved    |         block length          |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_1 (SSRC of first receiver)               | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  |                         last RR (LRR)                         |   1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                   delay since last RR (DLRR)                  |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_2 (SSRC of second receiver)              | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  :                               ...                             :   2
+
+Dlrr::Dlrr() = default;
+
+Dlrr::Dlrr(const Dlrr& other) = default;
+
+Dlrr::~Dlrr() = default;
+
+bool Dlrr::Parse(const uint8_t* buffer, uint16_t block_length_32bits) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // kReserved = buffer[1];
+  RTC_DCHECK_EQ(block_length_32bits,
+                ByteReader<uint16_t>::ReadBigEndian(&buffer[2]));
+  if (block_length_32bits % 3 != 0) {
+    RTC_LOG(LS_WARNING) << "Invalid size for dlrr block.";
+    return false;
+  }
+
+  size_t blocks_count = block_length_32bits / 3;
+  const uint8_t* read_at = buffer + kBlockHeaderLength;
+  sub_blocks_.resize(blocks_count);
+  for (ReceiveTimeInfo& sub_block : sub_blocks_) {
+    sub_block.ssrc = ByteReader<uint32_t>::ReadBigEndian(&read_at[0]);
+    sub_block.last_rr = ByteReader<uint32_t>::ReadBigEndian(&read_at[4]);
+    sub_block.delay_since_last_rr =
+        ByteReader<uint32_t>::ReadBigEndian(&read_at[8]);
+    read_at += kSubBlockLength;
+  }
+  return true;
+}
+
+size_t Dlrr::BlockLength() const {
+  if (sub_blocks_.empty())
+    return 0;
+  return kBlockHeaderLength + kSubBlockLength * sub_blocks_.size();
+}
+
+void Dlrr::Create(uint8_t* buffer) const {
+  if (sub_blocks_.empty())  // No subblocks, no need to write header either.
+    return;
+  // Create block header.
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &buffer[2], rtc::dchecked_cast<uint16_t>(3 * sub_blocks_.size()));
+  // Create sub blocks.
+  uint8_t* write_at = buffer + kBlockHeaderLength;
+  for (const ReceiveTimeInfo& sub_block : sub_blocks_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[0], sub_block.ssrc);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[4], sub_block.last_rr);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[8],
+                                         sub_block.delay_since_last_rr);
+    write_at += kSubBlockLength;
+  }
+  RTC_DCHECK_EQ(buffer + BlockLength(), write_at);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h

@@ -0,0 +1,80 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+namespace webrtc {
+namespace rtcp {
+struct ReceiveTimeInfo {
+  // RFC 3611 4.5
+  ReceiveTimeInfo() : ssrc(0), last_rr(0), delay_since_last_rr(0) {}
+  ReceiveTimeInfo(uint32_t ssrc, uint32_t last_rr, uint32_t delay)
+      : ssrc(ssrc), last_rr(last_rr), delay_since_last_rr(delay) {}
+
+  uint32_t ssrc;
+  uint32_t last_rr;
+  uint32_t delay_since_last_rr;
+};
+
+inline bool operator==(const ReceiveTimeInfo& lhs, const ReceiveTimeInfo& rhs) {
+  return lhs.ssrc == rhs.ssrc && lhs.last_rr == rhs.last_rr &&
+         lhs.delay_since_last_rr == rhs.delay_since_last_rr;
+}
+
+inline bool operator!=(const ReceiveTimeInfo& lhs, const ReceiveTimeInfo& rhs) {
+  return !(lhs == rhs);
+}
+
+// DLRR Report Block: Delay since the Last Receiver Report (RFC 3611).
+class Dlrr {
+ public:
+  static const uint8_t kBlockType = 5;
+
+  Dlrr();
+  Dlrr(const Dlrr& other);
+  ~Dlrr();
+
+  Dlrr& operator=(const Dlrr& other) = default;
+
+  // Dlrr without items treated same as no dlrr block.
+  explicit operator bool() const { return !sub_blocks_.empty(); }
+
+  // Second parameter is value read from block header,
+  // i.e. size of block in 32bits excluding block header itself.
+  bool Parse(const uint8_t* buffer, uint16_t block_length_32bits);
+
+  size_t BlockLength() const;
+  // Fills buffer with the Dlrr.
+  // Consumes BlockLength() bytes.
+  void Create(uint8_t* buffer) const;
+
+  void ClearItems() { sub_blocks_.clear(); }
+  void AddDlrrItem(const ReceiveTimeInfo& time_info) {
+    sub_blocks_.push_back(time_info);
+  }
+
+  const std::vector<ReceiveTimeInfo>& sub_blocks() const { return sub_blocks_; }
+
+ private:
+  static const size_t kBlockHeaderLength = 4;
+  static const size_t kSubBlockLength = 12;
+
+  std::vector<ReceiveTimeInfo> sub_blocks_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc

@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t ExtendedReports::kPacketType;
+constexpr size_t ExtendedReports::kMaxNumberOfDlrrItems;
+// From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
+//
+// Format for XR packets:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|reserved |   PT=XR=207   |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              SSRC                             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :                         report blocks                         :
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Extended report block:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Block Type   |   reserved    |         block length          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :             type-specific block contents                      :
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ExtendedReports::ExtendedReports() = default;
+ExtendedReports::ExtendedReports(const ExtendedReports& xr) = default;
+ExtendedReports::~ExtendedReports() = default;
+
+bool ExtendedReports::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  if (packet.payload_size_bytes() < kXrBaseLength) {
+    RTC_LOG(LS_WARNING)
+        << "Packet is too small to be an ExtendedReports packet.";
+    return false;
+  }
+
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(packet.payload()));
+  rrtr_block_.reset();
+  dlrr_block_.ClearItems();
+  target_bitrate_ = absl::nullopt;
+
+  const uint8_t* current_block = packet.payload() + kXrBaseLength;
+  const uint8_t* const packet_end =
+      packet.payload() + packet.payload_size_bytes();
+  constexpr size_t kBlockHeaderSizeBytes = 4;
+  while (current_block + kBlockHeaderSizeBytes <= packet_end) {
+    uint8_t block_type = ByteReader<uint8_t>::ReadBigEndian(current_block);
+    uint16_t block_length =
+        ByteReader<uint16_t>::ReadBigEndian(current_block + 2);
+    const uint8_t* next_block =
+        current_block + kBlockHeaderSizeBytes + block_length * 4;
+    if (next_block > packet_end) {
+      RTC_LOG(LS_WARNING)
+          << "Report block in extended report packet is too big.";
+      return false;
+    }
+    switch (block_type) {
+      case Rrtr::kBlockType:
+        ParseRrtrBlock(current_block, block_length);
+        break;
+      case Dlrr::kBlockType:
+        ParseDlrrBlock(current_block, block_length);
+        break;
+      case TargetBitrate::kBlockType:
+        ParseTargetBitrateBlock(current_block, block_length);
+        break;
+      default:
+        // Unknown block, ignore.
+        RTC_LOG(LS_WARNING)
+            << "Unknown extended report block type " << block_type;
+        break;
+    }
+    current_block = next_block;
+  }
+
+  return true;
+}
+
+void ExtendedReports::SetRrtr(const Rrtr& rrtr) {
+  if (rrtr_block_)
+    RTC_LOG(LS_WARNING) << "Rrtr already set, overwriting.";
+  rrtr_block_.emplace(rrtr);
+}
+
+bool ExtendedReports::AddDlrrItem(const ReceiveTimeInfo& time_info) {
+  if (dlrr_block_.sub_blocks().size() >= kMaxNumberOfDlrrItems) {
+    RTC_LOG(LS_WARNING) << "Reached maximum number of DLRR items.";
+    return false;
+  }
+  dlrr_block_.AddDlrrItem(time_info);
+  return true;
+}
+
+void ExtendedReports::SetTargetBitrate(const TargetBitrate& bitrate) {
+  if (target_bitrate_)
+    RTC_LOG(LS_WARNING) << "TargetBitrate already set, overwriting.";
+
+  target_bitrate_ = bitrate;
+}
+
+size_t ExtendedReports::BlockLength() const {
+  return kHeaderLength + kXrBaseLength + RrtrLength() + DlrrLength() +
+         TargetBitrateLength();
+}
+
+bool ExtendedReports::Create(uint8_t* packet,
+                             size_t* index,
+                             size_t max_length,
+                             PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  size_t index_end = *index + BlockLength();
+  const uint8_t kReserved = 0;
+  CreateHeader(kReserved, kPacketType, HeaderLength(), packet, index);
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc());
+  *index += sizeof(uint32_t);
+  if (rrtr_block_) {
+    rrtr_block_->Create(packet + *index);
+    *index += Rrtr::kLength;
+  }
+  if (dlrr_block_) {
+    dlrr_block_.Create(packet + *index);
+    *index += dlrr_block_.BlockLength();
+  }
+  if (target_bitrate_) {
+    target_bitrate_->Create(packet + *index);
+    *index += target_bitrate_->BlockLength();
+  }
+  RTC_CHECK_EQ(*index, index_end);
+  return true;
+}
+
+size_t ExtendedReports::TargetBitrateLength() const {
+  if (target_bitrate_)
+    return target_bitrate_->BlockLength();
+  return 0;
+}
+
+void ExtendedReports::ParseRrtrBlock(const uint8_t* block,
+                                     uint16_t block_length) {
+  if (block_length != Rrtr::kBlockLength) {
+    RTC_LOG(LS_WARNING) << "Incorrect rrtr block size " << block_length
+                        << " Should be " << Rrtr::kBlockLength;
+    return;
+  }
+  if (rrtr_block_) {
+    RTC_LOG(LS_WARNING)
+        << "Two rrtr blocks found in same Extended Report packet";
+    return;
+  }
+  rrtr_block_.emplace();
+  rrtr_block_->Parse(block);
+}
+
+void ExtendedReports::ParseDlrrBlock(const uint8_t* block,
+                                     uint16_t block_length) {
+  if (dlrr_block_) {
+    RTC_LOG(LS_WARNING)
+        << "Two Dlrr blocks found in same Extended Report packet";
+    return;
+  }
+  dlrr_block_.Parse(block, block_length);
+}
+
+void ExtendedReports::ParseTargetBitrateBlock(const uint8_t* block,
+                                              uint16_t block_length) {
+  target_bitrate_.emplace();
+  target_bitrate_->Parse(block, block_length);
+}
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h

@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
+class ExtendedReports : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 207;
+  static constexpr size_t kMaxNumberOfDlrrItems = 50;
+
+  ExtendedReports();
+  ExtendedReports(const ExtendedReports& xr);
+  ~ExtendedReports() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetRrtr(const Rrtr& rrtr);
+  bool AddDlrrItem(const ReceiveTimeInfo& time_info);
+  void SetTargetBitrate(const TargetBitrate& target_bitrate);
+
+  const absl::optional<Rrtr>& rrtr() const { return rrtr_block_; }
+  const Dlrr& dlrr() const { return dlrr_block_; }
+  const absl::optional<TargetBitrate>& target_bitrate() const {
+    return target_bitrate_;
+  }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kXrBaseLength = 4;
+
+  size_t RrtrLength() const { return rrtr_block_ ? Rrtr::kLength : 0; }
+  size_t DlrrLength() const { return dlrr_block_.BlockLength(); }
+  size_t TargetBitrateLength() const;
+
+  void ParseRrtrBlock(const uint8_t* block, uint16_t block_length);
+  void ParseDlrrBlock(const uint8_t* block, uint16_t block_length);
+  void ParseTargetBitrateBlock(const uint8_t* block, uint16_t block_length);
+
+  absl::optional<Rrtr> rrtr_block_;
+  Dlrr dlrr_block_;  // Dlrr without items treated same as no dlrr block.
+  absl::optional<TargetBitrate> target_bitrate_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc

@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Fir::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             SSRC of media source (unused) = 0                 |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+// Full intra request (FIR) (RFC 5104).
+// The Feedback Control Information (FCI) for the Full Intra Request
+// consists of one or more FCI entries.
+// FCI:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              SSRC                             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | Seq nr.       |    Reserved = 0                               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+Fir::Fir() = default;
+
+Fir::Fir(const Fir& fir) = default;
+
+Fir::~Fir() = default;
+
+bool Fir::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  // The FCI field MUST contain one or more FIR entries.
+  if (packet.payload_size_bytes() < kCommonFeedbackLength + kFciLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to be a valid FIR packet.";
+    return false;
+  }
+
+  if ((packet.payload_size_bytes() - kCommonFeedbackLength) % kFciLength != 0) {
+    RTC_LOG(LS_WARNING) << "Invalid size for a valid FIR packet.";
+    return false;
+  }
+
+  ParseCommonFeedback(packet.payload());
+
+  size_t number_of_fci_items =
+      (packet.payload_size_bytes() - kCommonFeedbackLength) / kFciLength;
+  const uint8_t* next_fci = packet.payload() + kCommonFeedbackLength;
+  items_.resize(number_of_fci_items);
+  for (Request& request : items_) {
+    request.ssrc = ByteReader<uint32_t>::ReadBigEndian(next_fci);
+    request.seq_nr = ByteReader<uint8_t>::ReadBigEndian(next_fci + 4);
+    next_fci += kFciLength;
+  }
+  return true;
+}
+
+size_t Fir::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength + kFciLength * items_.size();
+}
+
+bool Fir::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  RTC_DCHECK(!items_.empty());
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  size_t index_end = *index + BlockLength();
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  RTC_DCHECK_EQ(Psfb::media_ssrc(), 0);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+
+  constexpr uint32_t kReserved = 0;
+  for (const Request& request : items_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, request.ssrc);
+    ByteWriter<uint8_t>::WriteBigEndian(packet + *index + 4, request.seq_nr);
+    ByteWriter<uint32_t, 3>::WriteBigEndian(packet + *index + 5, kReserved);
+    *index += kFciLength;
+  }
+  RTC_CHECK_EQ(*index, index_end);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTCP_PACKET_FIR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_FIR_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+// Full intra request (FIR) (RFC 5104).
+class Fir : public Psfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 4;
+  struct Request {
+    Request() : ssrc(0), seq_nr(0) {}
+    Request(uint32_t ssrc, uint8_t seq_nr) : ssrc(ssrc), seq_nr(seq_nr) {}
+    uint32_t ssrc;
+    uint8_t seq_nr;
+  };
+
+  Fir();
+  Fir(const Fir& fir);
+  ~Fir() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void AddRequestTo(uint32_t ssrc, uint8_t seq_num) {
+    items_.emplace_back(ssrc, seq_num);
+  }
+  const std::vector<Request>& requests() const { return items_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kFciLength = 8;
+
+  // SSRC of media source is not used in FIR packet. Shadow base functions.
+  void SetMediaSsrc(uint32_t ssrc);
+  uint32_t media_ssrc() const;
+
+  std::vector<Request> items_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_FIR_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc

@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Loss Notification
+// -----------------
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| FMT=15  |   PT=206      |             length            |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                  SSRC of packet sender                        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                  SSRC of media source                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |  Unique identifier 'L' 'N' 'T' 'F'                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 | Last Decoded Sequence Number  | Last Received SeqNum Delta  |D|
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+LossNotification::LossNotification()
+    : last_decoded_(0), last_received_(0), decodability_flag_(false) {}
+
+LossNotification::LossNotification(uint16_t last_decoded,
+                                   uint16_t last_received,
+                                   bool decodability_flag)
+    : last_decoded_(last_decoded),
+      last_received_(last_received),
+      decodability_flag_(decodability_flag) {}
+
+LossNotification::LossNotification(const LossNotification& rhs) = default;
+
+LossNotification::~LossNotification() = default;
+
+size_t LossNotification::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength + kLossNotificationPayloadLength;
+}
+
+bool LossNotification::Create(uint8_t* packet,
+                              size_t* index,
+                              size_t max_length,
+                              PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  const size_t index_end = *index + BlockLength();
+
+  // Note: `index` updated by the function below.
+  CreateHeader(Psfb::kAfbMessageType, kPacketType, HeaderLength(), packet,
+               index);
+
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, kUniqueIdentifier);
+  *index += sizeof(uint32_t);
+
+  ByteWriter<uint16_t>::WriteBigEndian(packet + *index, last_decoded_);
+  *index += sizeof(uint16_t);
+
+  const uint16_t last_received_delta = last_received_ - last_decoded_;
+  RTC_DCHECK_LE(last_received_delta, 0x7fff);
+  const uint16_t last_received_delta_and_decodability =
+      (last_received_delta << 1) | (decodability_flag_ ? 0x0001 : 0x0000);
+
+  ByteWriter<uint16_t>::WriteBigEndian(packet + *index,
+                                       last_received_delta_and_decodability);
+  *index += sizeof(uint16_t);
+
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+bool LossNotification::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), Psfb::kAfbMessageType);
+
+  if (packet.payload_size_bytes() <
+      kCommonFeedbackLength + kLossNotificationPayloadLength) {
+    return false;
+  }
+
+  const uint8_t* const payload = packet.payload();
+
+  if (ByteReader<uint32_t>::ReadBigEndian(&payload[8]) != kUniqueIdentifier) {
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+
+  last_decoded_ = ByteReader<uint16_t>::ReadBigEndian(&payload[12]);
+
+  const uint16_t last_received_delta_and_decodability =
+      ByteReader<uint16_t>::ReadBigEndian(&payload[14]);
+  last_received_ = last_decoded_ + (last_received_delta_and_decodability >> 1);
+  decodability_flag_ = (last_received_delta_and_decodability & 0x0001);
+
+  return true;
+}
+
+bool LossNotification::Set(uint16_t last_decoded,
+                           uint16_t last_received,
+                           bool decodability_flag) {
+  const uint16_t delta = last_received - last_decoded;
+  if (delta > 0x7fff) {
+    return false;
+  }
+  last_received_ = last_received;
+  last_decoded_ = last_decoded;
+  decodability_flag_ = decodability_flag;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h

@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2019 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_RTP_RTCP_SOURCE_RTCP_PACKET_LOSS_NOTIFICATION_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_LOSS_NOTIFICATION_H_
+
+#include "absl/base/attributes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class LossNotification : public Psfb {
+ public:
+  LossNotification();
+  LossNotification(uint16_t last_decoded,
+                   uint16_t last_received,
+                   bool decodability_flag);
+  LossNotification(const LossNotification& other);
+  ~LossNotification() override;
+
+  size_t BlockLength() const override;
+
+  ABSL_MUST_USE_RESULT
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+  // Parse assumes header is already parsed and validated.
+  ABSL_MUST_USE_RESULT
+  bool Parse(const CommonHeader& packet);
+
+  // Set all of the values transmitted by the loss notification message.
+  // If the values may not be represented by a loss notification message,
+  // false is returned, and no change is made to the object; this happens
+  // when `last_received` is ahead of `last_decoded` by more than 0x7fff.
+  // This is because `last_received` is represented on the wire as a delta,
+  // and only 15 bits are available for that delta.
+  ABSL_MUST_USE_RESULT
+  bool Set(uint16_t last_decoded,
+           uint16_t last_received,
+           bool decodability_flag);
+
+  // RTP sequence number of the first packet belong to the last decoded
+  // non-discardable frame.
+  uint16_t last_decoded() const { return last_decoded_; }
+
+  // RTP sequence number of the last received packet.
+  uint16_t last_received() const { return last_received_; }
+
+  // A decodability flag, whose specific meaning depends on the last-received
+  // RTP sequence number. The decodability flag is true if and only if all of
+  // the frame's dependencies are known to be decodable, and the frame itself
+  // is not yet known to be unassemblable.
+  // * Clarification #1: In a multi-packet frame, the first packet's
+  //   dependencies are known, but it is not yet known whether all parts
+  //   of the current frame will be received.
+  // * Clarification #2: In a multi-packet frame, the dependencies would be
+  //   unknown if the first packet was not received. Then, the packet will
+  //   be known-unassemblable.
+  bool decodability_flag() const { return decodability_flag_; }
+
+ private:
+  static constexpr uint32_t kUniqueIdentifier = 0x4C4E5446;  // 'L' 'N' 'T' 'F'.
+  static constexpr size_t kLossNotificationPayloadLength = 8;
+
+  uint16_t last_decoded_;
+  uint16_t last_received_;
+  bool decodability_flag_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_LOSS_NOTIFICATION_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc

@@ -0,0 +1,176 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Nack::kFeedbackMessageType;
+constexpr size_t Nack::kNackItemLength;
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+//
+// Generic NACK (RFC 4585).
+//
+// FCI:
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |            PID                |             BLP               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Nack::Nack() = default;
+Nack::Nack(const Nack& rhs) = default;
+Nack::~Nack() = default;
+
+bool Nack::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() < kCommonFeedbackLength + kNackItemLength) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is too small for a Nack.";
+    return false;
+  }
+  size_t nack_items =
+      (packet.payload_size_bytes() - kCommonFeedbackLength) / kNackItemLength;
+
+  ParseCommonFeedback(packet.payload());
+  const uint8_t* next_nack = packet.payload() + kCommonFeedbackLength;
+
+  packet_ids_.clear();
+  packed_.resize(nack_items);
+  for (size_t index = 0; index < nack_items; ++index) {
+    packed_[index].first_pid = ByteReader<uint16_t>::ReadBigEndian(next_nack);
+    packed_[index].bitmask = ByteReader<uint16_t>::ReadBigEndian(next_nack + 2);
+    next_nack += kNackItemLength;
+  }
+  Unpack();
+
+  return true;
+}
+
+size_t Nack::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength +
+         packed_.size() * kNackItemLength;
+}
+
+bool Nack::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  PacketReadyCallback callback) const {
+  RTC_DCHECK(!packed_.empty());
+  // If nack list can't fit in packet, try to fragment.
+  constexpr size_t kNackHeaderLength = kHeaderLength + kCommonFeedbackLength;
+  for (size_t nack_index = 0; nack_index < packed_.size();) {
+    size_t bytes_left_in_buffer = max_length - *index;
+    if (bytes_left_in_buffer < kNackHeaderLength + kNackItemLength) {
+      if (!OnBufferFull(packet, index, callback))
+        return false;
+      continue;
+    }
+    size_t num_nack_fields =
+        std::min((bytes_left_in_buffer - kNackHeaderLength) / kNackItemLength,
+                 packed_.size() - nack_index);
+
+    size_t payload_size_bytes =
+        kCommonFeedbackLength + (num_nack_fields * kNackItemLength);
+    size_t payload_size_32bits =
+        rtc::CheckedDivExact<size_t>(payload_size_bytes, 4);
+    CreateHeader(kFeedbackMessageType, kPacketType, payload_size_32bits, packet,
+                 index);
+
+    CreateCommonFeedback(packet + *index);
+    *index += kCommonFeedbackLength;
+
+    size_t nack_end_index = nack_index + num_nack_fields;
+    for (; nack_index < nack_end_index; ++nack_index) {
+      const PackedNack& item = packed_[nack_index];
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 0, item.first_pid);
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 2, item.bitmask);
+      *index += kNackItemLength;
+    }
+    RTC_DCHECK_LE(*index, max_length);
+  }
+
+  return true;
+}
+
+void Nack::SetPacketIds(const uint16_t* nack_list, size_t length) {
+  RTC_DCHECK(nack_list);
+  SetPacketIds(std::vector<uint16_t>(nack_list, nack_list + length));
+}
+
+void Nack::SetPacketIds(std::vector<uint16_t> nack_list) {
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  packet_ids_ = std::move(nack_list);
+  Pack();
+}
+
+void Nack::Pack() {
+  RTC_DCHECK(!packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  auto it = packet_ids_.begin();
+  const auto end = packet_ids_.end();
+  while (it != end) {
+    PackedNack item;
+    item.first_pid = *it++;
+    // Bitmask specifies losses in any of the 16 packets following the pid.
+    item.bitmask = 0;
+    while (it != end) {
+      uint16_t shift = static_cast<uint16_t>(*it - item.first_pid - 1);
+      if (shift <= 15) {
+        item.bitmask |= (1 << shift);
+        ++it;
+      } else {
+        break;
+      }
+    }
+    packed_.push_back(item);
+  }
+}
+
+void Nack::Unpack() {
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(!packed_.empty());
+  for (const PackedNack& item : packed_) {
+    packet_ids_.push_back(item.first_pid);
+    uint16_t pid = item.first_pid + 1;
+    for (uint16_t bitmask = item.bitmask; bitmask != 0; bitmask >>= 1, ++pid) {
+      if (bitmask & 1)
+        packet_ids_.push_back(pid);
+    }
+  }
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h

@@ -0,0 +1,59 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class Nack : public Rtpfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 1;
+  Nack();
+  Nack(const Nack&);
+  ~Nack() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetPacketIds(const uint16_t* nack_list, size_t length);
+  void SetPacketIds(std::vector<uint16_t> nack_list);
+  const std::vector<uint16_t>& packet_ids() const { return packet_ids_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kNackItemLength = 4;
+  struct PackedNack {
+    uint16_t first_pid;
+    uint16_t bitmask;
+  };
+
+  void Pack();    // Fills packed_ using packed_ids_. (used in SetPacketIds).
+  void Unpack();  // Fills packet_ids_ using packed_. (used in Parse).
+
+  std::vector<PackedNack> packed_;
+  std::vector<uint16_t> packet_ids_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc

@@ -0,0 +1,79 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Pli::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+
+Pli::Pli() = default;
+
+Pli::Pli(const Pli& pli) = default;
+
+Pli::~Pli() = default;
+
+//
+// Picture loss indication (PLI) (RFC 4585).
+// FCI: no feedback control information.
+bool Pli::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() < kCommonFeedbackLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to be a valid PLI packet";
+    return false;
+  }
+
+  ParseCommonFeedback(packet.payload());
+  return true;
+}
+
+size_t Pli::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength;
+}
+
+bool Pli::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h

@@ -0,0 +1,39 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+// Picture loss indication (PLI) (RFC 4585).
+class Pli : public Psfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 1;
+
+  Pli();
+  Pli(const Pli& pli);
+  ~Pli() override;
+
+  bool Parse(const CommonHeader& packet);
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc

@@ -0,0 +1,47 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Psfb::kPacketType;
+constexpr uint8_t Psfb::kAfbMessageType;
+constexpr size_t Psfb::kCommonFeedbackLength;
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Psfb::ParseCommonFeedback(const uint8_t* payload) {
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[0]));
+  SetMediaSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[4]));
+}
+
+void Psfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc());
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h

@@ -0,0 +1,48 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// PSFB: Payload-specific feedback message.
+// RFC 4585, Section 6.3.
+class Psfb : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 206;
+  static constexpr uint8_t kAfbMessageType = 15;
+
+  Psfb() = default;
+  ~Psfb() override = default;
+
+  void SetMediaSsrc(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static constexpr size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t media_ssrc_ = 0;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc

@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t RapidResyncRequest::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+// Rapid Resynchronisation Request (draft-perkins-avt-rapid-rtp-sync-03).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|  FMT=5  |     PT=205    |         length=2              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool RapidResyncRequest::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() != kCommonFeedbackLength) {
+    RTC_LOG(LS_WARNING) << "Packet payload size should be "
+                        << kCommonFeedbackLength << " instead of "
+                        << packet.payload_size_bytes()
+                        << " to be a valid Rapid Resynchronisation Request";
+    return false;
+  }
+
+  ParseCommonFeedback(packet.payload());
+  return true;
+}
+
+size_t RapidResyncRequest::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength;
+}
+
+bool RapidResyncRequest::Create(uint8_t* packet,
+                                size_t* index,
+                                size_t max_length,
+                                PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// draft-perkins-avt-rapid-rtp-sync-03
+class RapidResyncRequest : public Rtpfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 5;
+
+  RapidResyncRequest() {}
+  ~RapidResyncRequest() override {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& header);
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc

@@ -0,0 +1,112 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t ReceiverReport::kPacketType;
+constexpr size_t ReceiverReport::kMaxNumberOfReportBlocks;
+// RTCP receiver report (RFC 3550).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|    RC   |   PT=RR=201   |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                     SSRC of packet sender                     |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                         report block(s)                       |
+//  |                            ....                               |
+
+ReceiverReport::ReceiverReport() = default;
+
+ReceiverReport::ReceiverReport(const ReceiverReport& rhs) = default;
+
+ReceiverReport::~ReceiverReport() = default;
+
+bool ReceiverReport::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  const uint8_t report_blocks_count = packet.count();
+
+  if (packet.payload_size_bytes() <
+      kRrBaseLength + report_blocks_count * ReportBlock::kLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+    return false;
+  }
+
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(packet.payload()));
+
+  const uint8_t* next_report_block = packet.payload() + kRrBaseLength;
+
+  report_blocks_.resize(report_blocks_count);
+  for (ReportBlock& block : report_blocks_) {
+    block.Parse(next_report_block, ReportBlock::kLength);
+    next_report_block += ReportBlock::kLength;
+  }
+
+  RTC_DCHECK_LE(next_report_block - packet.payload(),
+                static_cast<ptrdiff_t>(packet.payload_size_bytes()));
+  return true;
+}
+
+size_t ReceiverReport::BlockLength() const {
+  return kHeaderLength + kRrBaseLength +
+         report_blocks_.size() * ReportBlock::kLength;
+}
+
+bool ReceiverReport::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+               index);
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc());
+  *index += kRrBaseLength;
+  for (const ReportBlock& block : report_blocks_) {
+    block.Create(packet + *index);
+    *index += ReportBlock::kLength;
+  }
+  return true;
+}
+
+bool ReceiverReport::AddReportBlock(const ReportBlock& block) {
+  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+    RTC_LOG(LS_WARNING) << "Max report blocks reached.";
+    return false;
+  }
+  report_blocks_.push_back(block);
+  return true;
+}
+
+bool ReceiverReport::SetReportBlocks(std::vector<ReportBlock> blocks) {
+  if (blocks.size() > kMaxNumberOfReportBlocks) {
+    RTC_LOG(LS_WARNING) << "Too many report blocks (" << blocks.size()
+                        << ") for receiver report.";
+    return false;
+  }
+  report_blocks_ = std::move(blocks);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h

@@ -0,0 +1,60 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class ReceiverReport : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 201;
+  static constexpr size_t kMaxNumberOfReportBlocks = 0x1f;
+
+  ReceiverReport();
+  ReceiverReport(const ReceiverReport&);
+  ~ReceiverReport() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool AddReportBlock(const ReportBlock& block);
+  bool SetReportBlocks(std::vector<ReportBlock> blocks);
+
+  const std::vector<ReportBlock>& report_blocks() const {
+    return report_blocks_;
+  }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static const size_t kRrBaseLength = 4;
+
+  std::vector<ReportBlock> report_blocks_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc

@@ -0,0 +1,143 @@
+/*
+ *  Copyright (c) 2016 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 "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+
+#include <cstdint>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+// Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| FMT=15  |   PT=206      |             length            |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                  SSRC of packet sender                        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                       Unused = 0                              |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |  Unique identifier 'R' 'E' 'M' 'B'                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |  Num SSRC     | BR Exp    |  BR Mantissa                      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |   SSRC feedback                                               |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    :  ...                                                          :
+
+Remb::Remb() : bitrate_bps_(0) {}
+
+Remb::Remb(const Remb& rhs) = default;
+
+Remb::~Remb() = default;
+
+bool Remb::Parse(const CommonHeader& packet) {
+  RTC_DCHECK(packet.type() == kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), Psfb::kAfbMessageType);
+
+  if (packet.payload_size_bytes() < 16) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is too small for Remb packet.";
+    return false;
+  }
+  const uint8_t* const payload = packet.payload();
+  if (kUniqueIdentifier != ByteReader<uint32_t>::ReadBigEndian(&payload[8])) {
+    return false;
+  }
+  uint8_t number_of_ssrcs = payload[12];
+  if (packet.payload_size_bytes() !=
+      kCommonFeedbackLength + (2 + number_of_ssrcs) * 4) {
+    RTC_LOG(LS_WARNING) << "Payload size " << packet.payload_size_bytes()
+                        << " does not match " << number_of_ssrcs << " ssrcs.";
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+  uint8_t exponent = payload[13] >> 2;
+  uint64_t mantissa = (static_cast<uint32_t>(payload[13] & 0x03) << 16) |
+                      ByteReader<uint16_t>::ReadBigEndian(&payload[14]);
+  bitrate_bps_ = (mantissa << exponent);
+  bool shift_overflow =
+      (static_cast<uint64_t>(bitrate_bps_) >> exponent) != mantissa;
+  if (bitrate_bps_ < 0 || shift_overflow) {
+    RTC_LOG(LS_ERROR) << "Invalid remb bitrate value : " << mantissa << "*2^"
+                      << static_cast<int>(exponent);
+    return false;
+  }
+
+  const uint8_t* next_ssrc = payload + 16;
+  ssrcs_.clear();
+  ssrcs_.reserve(number_of_ssrcs);
+  for (uint8_t i = 0; i < number_of_ssrcs; ++i) {
+    ssrcs_.push_back(ByteReader<uint32_t>::ReadBigEndian(next_ssrc));
+    next_ssrc += sizeof(uint32_t);
+  }
+
+  return true;
+}
+
+bool Remb::SetSsrcs(std::vector<uint32_t> ssrcs) {
+  if (ssrcs.size() > kMaxNumberOfSsrcs) {
+    RTC_LOG(LS_WARNING) << "Not enough space for all given SSRCs.";
+    return false;
+  }
+  ssrcs_ = std::move(ssrcs);
+  return true;
+}
+
+size_t Remb::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength + (2 + ssrcs_.size()) * 4;
+}
+
+bool Remb::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  size_t index_end = *index + BlockLength();
+  CreateHeader(Psfb::kAfbMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  RTC_DCHECK_EQ(0, Psfb::media_ssrc());
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, kUniqueIdentifier);
+  *index += sizeof(uint32_t);
+  const uint32_t kMaxMantissa = 0x3ffff;  // 18 bits.
+  uint64_t mantissa = bitrate_bps_;
+  uint8_t exponenta = 0;
+  while (mantissa > kMaxMantissa) {
+    mantissa >>= 1;
+    ++exponenta;
+  }
+  packet[(*index)++] = static_cast<uint8_t>(ssrcs_.size());
+  packet[(*index)++] = (exponenta << 2) | (mantissa >> 16);
+  ByteWriter<uint16_t>::WriteBigEndian(packet + *index, mantissa & 0xffff);
+  *index += sizeof(uint16_t);
+
+  for (uint32_t ssrc : ssrcs_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, ssrc);
+    *index += sizeof(uint32_t);
+  }
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
+class Remb : public Psfb {
+ public:
+  static constexpr size_t kMaxNumberOfSsrcs = 0xff;
+
+  Remb();
+  Remb(const Remb&);
+  ~Remb() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool SetSsrcs(std::vector<uint32_t> ssrcs);
+  void SetBitrateBps(int64_t bitrate_bps) { bitrate_bps_ = bitrate_bps; }
+
+  int64_t bitrate_bps() const { return bitrate_bps_; }
+  const std::vector<uint32_t>& ssrcs() const { return ssrcs_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr uint32_t kUniqueIdentifier = 0x52454D42;  // 'R' 'E' 'M' 'B'.
+
+  // Media ssrc is unused, shadow base class setter and getter.
+  void SetMediaSsrc(uint32_t);
+  uint32_t media_ssrc() const;
+
+  int64_t bitrate_bps_;
+  std::vector<uint32_t> ssrcs_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc

@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+
+#include <algorithm>
+#include <cmath>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+
+static constexpr int kFieldValueSize = 3;
+static constexpr int kFieldSize = 1 + kFieldValueSize;
+static constexpr DataRate kDataRateResolution = DataRate::KilobitsPerSec(1);
+constexpr int64_t kMaxEncoded = (1 << (kFieldValueSize * 8)) - 1;
+
+class DataRateSerializer {
+ public:
+  DataRateSerializer(
+      uint8_t id,
+      std::function<DataRate*(NetworkStateEstimate*)> field_getter)
+      : id_(id), field_getter_(field_getter) {}
+
+  uint8_t id() const { return id_; }
+
+  void Read(const uint8_t* src, NetworkStateEstimate* target) const {
+    int64_t scaled = ByteReader<uint32_t, kFieldValueSize>::ReadBigEndian(src);
+    if (scaled == kMaxEncoded) {
+      *field_getter_(target) = DataRate::PlusInfinity();
+    } else {
+      *field_getter_(target) = kDataRateResolution * scaled;
+    }
+  }
+
+  bool Write(const NetworkStateEstimate& src, uint8_t* target) const {
+    auto value = *field_getter_(const_cast<NetworkStateEstimate*>(&src));
+    if (value.IsMinusInfinity()) {
+      RTC_LOG(LS_WARNING) << "Trying to serialize MinusInfinity";
+      return false;
+    }
+    ByteWriter<uint8_t>::WriteBigEndian(target++, id_);
+    int64_t scaled;
+    if (value.IsPlusInfinity()) {
+      scaled = kMaxEncoded;
+    } else {
+      scaled = value / kDataRateResolution;
+      if (scaled >= kMaxEncoded) {
+        scaled = kMaxEncoded;
+        RTC_LOG(LS_WARNING) << ToString(value) << " is larger than max ("
+                            << ToString(kMaxEncoded * kDataRateResolution)
+                            << "), encoded as PlusInfinity.";
+      }
+    }
+    ByteWriter<uint32_t, kFieldValueSize>::WriteBigEndian(target, scaled);
+    return true;
+  }
+
+ private:
+  const uint8_t id_;
+  const std::function<DataRate*(NetworkStateEstimate*)> field_getter_;
+};
+
+class RemoteEstimateSerializerImpl : public RemoteEstimateSerializer {
+ public:
+  explicit RemoteEstimateSerializerImpl(std::vector<DataRateSerializer> fields)
+      : fields_(fields) {}
+
+  rtc::Buffer Serialize(const NetworkStateEstimate& src) const override {
+    size_t max_size = fields_.size() * kFieldSize;
+    size_t size = 0;
+    rtc::Buffer buf(max_size);
+    for (const auto& field : fields_) {
+      if (field.Write(src, buf.data() + size)) {
+        size += kFieldSize;
+      }
+    }
+    buf.SetSize(size);
+    return buf;
+  }
+
+  bool Parse(rtc::ArrayView<const uint8_t> src,
+             NetworkStateEstimate* target) const override {
+    if (src.size() % kFieldSize != 0)
+      return false;
+    RTC_DCHECK_EQ(src.size() % kFieldSize, 0);
+    for (const uint8_t* data_ptr = src.data(); data_ptr < src.end();
+         data_ptr += kFieldSize) {
+      uint8_t field_id = ByteReader<uint8_t>::ReadBigEndian(data_ptr);
+      for (const auto& field : fields_) {
+        if (field.id() == field_id) {
+          field.Read(data_ptr + 1, target);
+          break;
+        }
+      }
+    }
+    return true;
+  }
+
+ private:
+  const std::vector<DataRateSerializer> fields_;
+};
+
+}  // namespace
+
+const RemoteEstimateSerializer* GetRemoteEstimateSerializer() {
+  using E = NetworkStateEstimate;
+  static auto* serializer = new RemoteEstimateSerializerImpl({
+      {1, [](E* e) { return &e->link_capacity_lower; }},
+      {2, [](E* e) { return &e->link_capacity_upper; }},
+  });
+  return serializer;
+}
+
+RemoteEstimate::RemoteEstimate() : serializer_(GetRemoteEstimateSerializer()) {
+  SetSubType(kSubType);
+  SetName(kName);
+  SetSenderSsrc(0);
+}
+
+RemoteEstimate::RemoteEstimate(App&& app)
+    : App(std::move(app)), serializer_(GetRemoteEstimateSerializer()) {}
+
+bool RemoteEstimate::ParseData() {
+  return serializer_->Parse({data(), data_size()}, &estimate_);
+}
+
+void RemoteEstimate::SetEstimate(NetworkStateEstimate estimate) {
+  estimate_ = estimate;
+  auto buf = serializer_->Serialize(estimate);
+  SetData(buf.data(), buf.size());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2019 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_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class CommonHeader;
+class RemoteEstimateSerializer {
+ public:
+  virtual bool Parse(rtc::ArrayView<const uint8_t> src,
+                     NetworkStateEstimate* target) const = 0;
+  virtual rtc::Buffer Serialize(const NetworkStateEstimate& src) const = 0;
+  virtual ~RemoteEstimateSerializer() = default;
+};
+
+// Using a static global implementation to avoid incurring initialization
+// overhead of the serializer every time RemoteEstimate is created.
+const RemoteEstimateSerializer* GetRemoteEstimateSerializer();
+
+// The RemoteEstimate packet provides network estimation results from the
+// receive side. This functionality is experimental and subject to change
+// without notice.
+class RemoteEstimate : public App {
+ public:
+  RemoteEstimate();
+  explicit RemoteEstimate(App&& app);
+  // Note, sub type must be unique among all app messages with "goog" name.
+  static constexpr uint8_t kSubType = 13;
+  static constexpr uint32_t kName = NameToInt("goog");
+  static TimeDelta GetTimestampPeriod();
+
+  bool ParseData();
+  void SetEstimate(NetworkStateEstimate estimate);
+  NetworkStateEstimate estimate() const { return estimate_; }
+
+ private:
+  NetworkStateEstimate estimate_;
+  const RemoteEstimateSerializer* const serializer_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc

@@ -0,0 +1,91 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTCP report block (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                 SSRC_1 (SSRC of first source)                 |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 | fraction lost |       cumulative number of packets lost       |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |           extended highest sequence number received           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                      interarrival jitter                      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                         last SR (LSR)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |                   delay since last SR (DLSR)                  |
+// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+ReportBlock::ReportBlock()
+    : source_ssrc_(0),
+      fraction_lost_(0),
+      cumulative_lost_(0),
+      extended_high_seq_num_(0),
+      jitter_(0),
+      last_sr_(0),
+      delay_since_last_sr_(0) {}
+
+bool ReportBlock::Parse(const uint8_t* buffer, size_t length) {
+  RTC_DCHECK(buffer != nullptr);
+  if (length < ReportBlock::kLength) {
+    RTC_LOG(LS_ERROR) << "Report Block should be 24 bytes long";
+    return false;
+  }
+
+  source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
+  fraction_lost_ = buffer[4];
+  cumulative_lost_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[5]);
+  extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
+  last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
+  delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);
+
+  return true;
+}
+
+void ReportBlock::Create(uint8_t* buffer) const {
+  // Runtime check should be done while setting cumulative_lost.
+  RTC_DCHECK_LT(cumulative_lost(), (1 << 23));  // Have only 3 bytes for it.
+
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc());
+  ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost());
+  ByteWriter<int32_t, 3>::WriteBigEndian(&buffer[5], cumulative_lost());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], extended_high_seq_num());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[16], last_sr());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[20], delay_since_last_sr());
+}
+
+bool ReportBlock::SetCumulativeLost(int32_t cumulative_lost) {
+  // We have only 3 bytes to store it, and it's a signed value.
+  if (cumulative_lost >= (1 << 23) || cumulative_lost < -(1 << 23)) {
+    RTC_LOG(LS_WARNING)
+        << "Cumulative lost is too big to fit into Report Block";
+    return false;
+  }
+  cumulative_lost_ = cumulative_lost;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h

@@ -0,0 +1,70 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+namespace rtcp {
+
+// A ReportBlock represents the Sender Report packet from
+// RFC 3550 section 6.4.1.
+class ReportBlock {
+ public:
+  static const size_t kLength = 24;
+
+  ReportBlock();
+  ~ReportBlock() {}
+
+  bool Parse(const uint8_t* buffer, size_t length);
+
+  // Fills buffer with the ReportBlock.
+  // Consumes ReportBlock::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void SetMediaSsrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
+  void SetFractionLost(uint8_t fraction_lost) {
+    fraction_lost_ = fraction_lost;
+  }
+  bool SetCumulativeLost(int32_t cumulative_lost);
+  void SetExtHighestSeqNum(uint32_t ext_highest_seq_num) {
+    extended_high_seq_num_ = ext_highest_seq_num;
+  }
+  void SetJitter(uint32_t jitter) { jitter_ = jitter; }
+  void SetLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
+  void SetDelayLastSr(uint32_t delay_last_sr) {
+    delay_since_last_sr_ = delay_last_sr;
+  }
+
+  uint32_t source_ssrc() const { return source_ssrc_; }
+  uint8_t fraction_lost() const { return fraction_lost_; }
+  int32_t cumulative_lost() const { return cumulative_lost_; }
+  uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
+  uint32_t jitter() const { return jitter_; }
+  uint32_t last_sr() const { return last_sr_; }
+  uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
+
+ private:
+  uint32_t source_ssrc_;     // 32 bits
+  uint8_t fraction_lost_;    // 8 bits representing a fixed point value 0..1
+  int32_t cumulative_lost_;  // Signed 24-bit value
+  uint32_t extended_high_seq_num_;  // 32 bits
+  uint32_t jitter_;                 // 32 bits
+  uint32_t last_sr_;                // 32 bits
+  uint32_t delay_since_last_sr_;    // 32 bits, units of 1/65536 seconds
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc

@@ -0,0 +1,49 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+// Receiver Reference Time Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=4      |   reserved    |       block length = 2        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |              NTP timestamp, most significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             NTP timestamp, least significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void Rrtr::Parse(const uint8_t* buffer) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // reserved = buffer[1];
+  RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
+  uint32_t seconds = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  uint32_t fraction = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  ntp_.Set(seconds, fraction);
+}
+
+void Rrtr::Create(uint8_t* buffer) const {
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ntp_.seconds());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], ntp_.fractions());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h

@@ -0,0 +1,59 @@
+/*
+ *  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_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Rrtr {
+ public:
+  static const uint8_t kBlockType = 4;
+  static const uint16_t kBlockLength = 2;
+  static const size_t kLength = 4 * (kBlockLength + 1);  // 12
+
+  Rrtr() {}
+  Rrtr(const Rrtr&) = default;
+  ~Rrtr() {}
+
+  Rrtr& operator=(const Rrtr&) = default;
+
+  void Parse(const uint8_t* buffer);
+
+  // Fills buffer with the Rrtr.
+  // Consumes Rrtr::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void SetNtp(NtpTime ntp) { ntp_ = ntp; }
+
+  NtpTime ntp() const { return ntp_; }
+
+ private:
+  NtpTime ntp_;
+};
+
+inline bool operator==(const Rrtr& rrtr1, const Rrtr& rrtr2) {
+  return rrtr1.ntp() == rrtr2.ntp();
+}
+
+inline bool operator!=(const Rrtr& rrtr1, const Rrtr& rrtr2) {
+  return !(rrtr1 == rrtr2);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_

TMessagesProj/jni/voip/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc

@@ -0,0 +1,50 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "test/gtest.h"
+
+using webrtc::rtcp::Rrtr;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kNtpSec = 0x12345678;
+const uint32_t kNtpFrac = 0x23456789;
+const uint8_t kBlock[] = {0x04, 0x00, 0x00, 0x02, 0x12, 0x34,
+                          0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+static_assert(
+    kBlockSizeBytes == Rrtr::kLength,
+    "Size of manually created Rrtr block should match class constant");
+
+TEST(RtcpPacketRrtrTest, Create) {
+  uint8_t buffer[Rrtr::kLength];
+  Rrtr rrtr;
+  rrtr.SetNtp(NtpTime(kNtpSec, kNtpFrac));
+
+  rrtr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketRrtrTest, Parse) {
+  Rrtr read_rrtr;
+  read_rrtr.Parse(kBlock);
+
+  // Run checks on const object to ensure all accessors have const modifier.
+  const Rrtr& parsed = read_rrtr;
+
+  EXPECT_EQ(kNtpSec, parsed.ntp().seconds());
+  EXPECT_EQ(kNtpFrac, parsed.ntp().fractions());
+}
+
+}  // namespace
+}  // namespace webrtc