Repo created

TMessagesProj/jni/voip/webrtc/media/OWNERS

@@ -0,0 +1,13 @@
+brandtr@webrtc.org
+ilnik@webrtc.org
+sprang@webrtc.org
+magjed@webrtc.org
+mflodman@webrtc.org
+perkj@webrtc.org
+
+# Audio-related changes:
+peah@webrtc.org
+saza@webrtc.org
+
+# Datachannel-related changes:
+orphis@webrtc.org

TMessagesProj/jni/voip/webrtc/media/base/adapted_video_track_source.cc

@@ -0,0 +1,126 @@
+/*
+ *  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 "media/base/adapted_video_track_source.h"
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+
+namespace rtc {
+
+AdaptedVideoTrackSource::AdaptedVideoTrackSource() = default;
+
+AdaptedVideoTrackSource::AdaptedVideoTrackSource(int required_alignment)
+    : video_adapter_(required_alignment) {}
+
+AdaptedVideoTrackSource::~AdaptedVideoTrackSource() = default;
+
+bool AdaptedVideoTrackSource::GetStats(Stats* stats) {
+  webrtc::MutexLock lock(&stats_mutex_);
+
+  if (!stats_) {
+    return false;
+  }
+
+  *stats = *stats_;
+  return true;
+}
+
+void AdaptedVideoTrackSource::OnFrame(const webrtc::VideoFrame& frame) {
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer(
+      frame.video_frame_buffer());
+  /* Note that this is a "best effort" approach to
+     wants.rotation_applied; apply_rotation_ can change from false to
+     true between the check of apply_rotation() and the call to
+     broadcaster_.OnFrame(), in which case we generate a frame with
+     pending rotation despite some sink with wants.rotation_applied ==
+     true was just added. The VideoBroadcaster enforces
+     synchronization for us in this case, by not passing the frame on
+     to sinks which don't want it. */
+  if (apply_rotation() && frame.rotation() != webrtc::kVideoRotation_0 &&
+      buffer->type() == webrtc::VideoFrameBuffer::Type::kI420) {
+    /* Apply pending rotation. */
+    webrtc::VideoFrame rotated_frame(frame);
+    rotated_frame.set_video_frame_buffer(
+        webrtc::I420Buffer::Rotate(*buffer->GetI420(), frame.rotation()));
+    rotated_frame.set_rotation(webrtc::kVideoRotation_0);
+    broadcaster_.OnFrame(rotated_frame);
+  } else {
+    broadcaster_.OnFrame(frame);
+  }
+}
+
+void AdaptedVideoTrackSource::OnFrameDropped() {
+  broadcaster_.OnDiscardedFrame();
+}
+
+void AdaptedVideoTrackSource::AddOrUpdateSink(
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+    const rtc::VideoSinkWants& wants) {
+  broadcaster_.AddOrUpdateSink(sink, wants);
+  OnSinkWantsChanged(broadcaster_.wants());
+}
+
+void AdaptedVideoTrackSource::RemoveSink(
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  broadcaster_.RemoveSink(sink);
+  OnSinkWantsChanged(broadcaster_.wants());
+}
+
+bool AdaptedVideoTrackSource::apply_rotation() {
+  return broadcaster_.wants().rotation_applied;
+}
+
+void AdaptedVideoTrackSource::OnSinkWantsChanged(
+    const rtc::VideoSinkWants& wants) {
+  video_adapter_.OnSinkWants(wants);
+}
+
+bool AdaptedVideoTrackSource::AdaptFrame(int width,
+                                         int height,
+                                         int64_t time_us,
+                                         int* out_width,
+                                         int* out_height,
+                                         int* crop_width,
+                                         int* crop_height,
+                                         int* crop_x,
+                                         int* crop_y) {
+  {
+    webrtc::MutexLock lock(&stats_mutex_);
+    stats_ = Stats{width, height};
+  }
+
+  if (!broadcaster_.frame_wanted()) {
+    return false;
+  }
+
+  if (!video_adapter_.AdaptFrameResolution(
+          width, height, time_us * rtc::kNumNanosecsPerMicrosec, crop_width,
+          crop_height, out_width, out_height)) {
+    broadcaster_.OnDiscardedFrame();
+    // VideoAdapter dropped the frame.
+    return false;
+  }
+
+  *crop_x = (width - *crop_width) / 2;
+  *crop_y = (height - *crop_height) / 2;
+  return true;
+}
+
+void AdaptedVideoTrackSource::ProcessConstraints(
+    const webrtc::VideoTrackSourceConstraints& constraints) {
+  broadcaster_.ProcessConstraints(constraints);
+}
+
+}  // namespace rtc

TMessagesProj/jni/voip/webrtc/media/base/adapted_video_track_source.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 MEDIA_BASE_ADAPTED_VIDEO_TRACK_SOURCE_H_
+#define MEDIA_BASE_ADAPTED_VIDEO_TRACK_SOURCE_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/media_stream_interface.h"
+#include "api/notifier.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "media/base/video_adapter.h"
+#include "media/base/video_broadcaster.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace rtc {
+
+// Base class for sources which needs video adaptation, e.g., video
+// capture sources. Sinks must be added and removed on one and only
+// one thread, while AdaptFrame and OnFrame may be called on any
+// thread.
+class RTC_EXPORT AdaptedVideoTrackSource
+    : public webrtc::Notifier<webrtc::VideoTrackSourceInterface> {
+ public:
+  AdaptedVideoTrackSource();
+  ~AdaptedVideoTrackSource() override;
+
+ protected:
+  // Allows derived classes to initialize `video_adapter_` with a custom
+  // alignment.
+  explicit AdaptedVideoTrackSource(int required_alignment);
+  // Checks the apply_rotation() flag. If the frame needs rotation, and it is a
+  // plain memory frame, it is rotated. Subclasses producing native frames must
+  // handle apply_rotation() themselves.
+  void OnFrame(const webrtc::VideoFrame& frame);
+  // Indication from source that a frame was dropped.
+  void OnFrameDropped();
+
+  // Reports the appropriate frame size after adaptation. Returns true
+  // if a frame is wanted. Returns false if there are no interested
+  // sinks, or if the VideoAdapter decides to drop the frame.
+  bool AdaptFrame(int width,
+                  int height,
+                  int64_t time_us,
+                  int* out_width,
+                  int* out_height,
+                  int* crop_width,
+                  int* crop_height,
+                  int* crop_x,
+                  int* crop_y);
+
+  // Returns the current value of the apply_rotation flag, derived
+  // from the VideoSinkWants of registered sinks. The value is derived
+  // from sinks' wants, in AddOrUpdateSink and RemoveSink. Beware that
+  // when using this method from a different thread, the value may
+  // become stale before it is used.
+  bool apply_rotation();
+
+  cricket::VideoAdapter* video_adapter() { return &video_adapter_; }
+
+ private:
+  // Implements rtc::VideoSourceInterface.
+  void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants) override;
+  void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
+  // Part of VideoTrackSourceInterface.
+  bool GetStats(Stats* stats) override;
+
+  void OnSinkWantsChanged(const rtc::VideoSinkWants& wants);
+
+  // Encoded sinks not implemented for AdaptedVideoTrackSource.
+  bool SupportsEncodedOutput() const override { return false; }
+  void GenerateKeyFrame() override {}
+  void AddEncodedSink(
+      rtc::VideoSinkInterface<webrtc::RecordableEncodedFrame>* sink) override {}
+  void RemoveEncodedSink(
+      rtc::VideoSinkInterface<webrtc::RecordableEncodedFrame>* sink) override {}
+  void ProcessConstraints(
+      const webrtc::VideoTrackSourceConstraints& constraints) override;
+
+  cricket::VideoAdapter video_adapter_;
+
+  webrtc::Mutex stats_mutex_;
+  absl::optional<Stats> stats_ RTC_GUARDED_BY(stats_mutex_);
+
+  VideoBroadcaster broadcaster_;
+};
+
+}  // namespace rtc
+
+#endif  // MEDIA_BASE_ADAPTED_VIDEO_TRACK_SOURCE_H_

TMessagesProj/jni/voip/webrtc/media/base/audio_source.h

@@ -0,0 +1,58 @@
+/*
+ *  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 MEDIA_BASE_AUDIO_SOURCE_H_
+#define MEDIA_BASE_AUDIO_SOURCE_H_
+
+#include <cstddef>
+
+#include "absl/types/optional.h"
+
+namespace cricket {
+
+// Abstract interface for providing the audio data.
+// TODO(deadbeef): Rename this to AudioSourceInterface, and rename
+// webrtc::AudioSourceInterface to AudioTrackSourceInterface.
+class AudioSource {
+ public:
+  class Sink {
+   public:
+    // Callback to receive data from the AudioSource.
+    virtual void OnData(
+        const void* audio_data,
+        int bits_per_sample,
+        int sample_rate,
+        size_t number_of_channels,
+        size_t number_of_frames,
+        absl::optional<int64_t> absolute_capture_timestamp_ms) = 0;
+
+    // Called when the AudioSource is going away.
+    virtual void OnClose() = 0;
+
+    // Returns the number of channels encoded by the sink. This can be less than
+    // the number_of_channels if down-mixing occur. A value of -1 means an
+    // unknown number.
+    virtual int NumPreferredChannels() const = 0;
+
+   protected:
+    virtual ~Sink() {}
+  };
+
+  // Sets a sink to the AudioSource. There can be only one sink connected
+  // to the source at a time.
+  virtual void SetSink(Sink* sink) = 0;
+
+ protected:
+  virtual ~AudioSource() {}
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_AUDIO_SOURCE_H_

TMessagesProj/jni/voip/webrtc/media/base/codec.cc

@@ -0,0 +1,522 @@
+/*
+ *  Copyright (c) 2004 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 "media/base/codec.h"
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/match.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/video_codecs/av1_profile.h"
+#include "api/video_codecs/h264_profile_level_id.h"
+#ifdef RTC_ENABLE_H265
+#include "api/video_codecs/h265_profile_tier_level.h"
+#endif
+#include "api/video_codecs/vp9_profile.h"
+#include "media/base/media_constants.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace cricket {
+namespace {
+
+std::string GetH264PacketizationModeOrDefault(
+    const webrtc::CodecParameterMap& params) {
+  auto it = params.find(kH264FmtpPacketizationMode);
+  if (it != params.end()) {
+    return it->second;
+  }
+  // If packetization-mode is not present, default to "0".
+  // https://tools.ietf.org/html/rfc6184#section-6.2
+  return "0";
+}
+
+bool IsSameH264PacketizationMode(const webrtc::CodecParameterMap& left,
+                                 const webrtc::CodecParameterMap& right) {
+  return GetH264PacketizationModeOrDefault(left) ==
+         GetH264PacketizationModeOrDefault(right);
+}
+
+#ifdef RTC_ENABLE_H265
+std::string GetH265TxModeOrDefault(const webrtc::CodecParameterMap& params) {
+  auto it = params.find(kH265FmtpTxMode);
+  if (it != params.end()) {
+    return it->second;
+  }
+  // If TxMode is not present, a value of "SRST" must be inferred.
+  // https://tools.ietf.org/html/rfc7798@section-7.1
+  return "SRST";
+}
+
+bool IsSameH265TxMode(const webrtc::CodecParameterMap& left,
+                      const webrtc::CodecParameterMap& right) {
+  return absl::EqualsIgnoreCase(GetH265TxModeOrDefault(left),
+                                GetH265TxModeOrDefault(right));
+}
+#endif
+
+// Some (video) codecs are actually families of codecs and rely on parameters
+// to distinguish different incompatible family members.
+bool IsSameCodecSpecific(const std::string& name1,
+                         const webrtc::CodecParameterMap& params1,
+                         const std::string& name2,
+                         const webrtc::CodecParameterMap& params2) {
+  // The names might not necessarily match, so check both.
+  auto either_name_matches = [&](const std::string name) {
+    return absl::EqualsIgnoreCase(name, name1) ||
+           absl::EqualsIgnoreCase(name, name2);
+  };
+  if (either_name_matches(kH264CodecName))
+    return webrtc::H264IsSameProfile(params1, params2) &&
+           IsSameH264PacketizationMode(params1, params2);
+  if (either_name_matches(kVp9CodecName))
+    return webrtc::VP9IsSameProfile(params1, params2);
+  if (either_name_matches(kAv1CodecName))
+    return webrtc::AV1IsSameProfile(params1, params2);
+#ifdef RTC_ENABLE_H265
+  if (either_name_matches(kH265CodecName)) {
+    return webrtc::H265IsSameProfileTierLevel(params1, params2) &&
+           IsSameH265TxMode(params1, params2);
+  }
+#endif
+  return true;
+}
+
+}  // namespace
+
+FeedbackParams::FeedbackParams() = default;
+FeedbackParams::~FeedbackParams() = default;
+
+bool FeedbackParam::operator==(const FeedbackParam& other) const {
+  return absl::EqualsIgnoreCase(other.id(), id()) &&
+         absl::EqualsIgnoreCase(other.param(), param());
+}
+
+bool FeedbackParams::operator==(const FeedbackParams& other) const {
+  return params_ == other.params_;
+}
+
+bool FeedbackParams::Has(const FeedbackParam& param) const {
+  return absl::c_linear_search(params_, param);
+}
+
+void FeedbackParams::Add(const FeedbackParam& param) {
+  if (param.id().empty()) {
+    return;
+  }
+  if (Has(param)) {
+    // Param already in `this`.
+    return;
+  }
+  params_.push_back(param);
+  RTC_CHECK(!HasDuplicateEntries());
+}
+
+void FeedbackParams::Intersect(const FeedbackParams& from) {
+  std::vector<FeedbackParam>::iterator iter_to = params_.begin();
+  while (iter_to != params_.end()) {
+    if (!from.Has(*iter_to)) {
+      iter_to = params_.erase(iter_to);
+    } else {
+      ++iter_to;
+    }
+  }
+}
+
+bool FeedbackParams::HasDuplicateEntries() const {
+  for (std::vector<FeedbackParam>::const_iterator iter = params_.begin();
+       iter != params_.end(); ++iter) {
+    for (std::vector<FeedbackParam>::const_iterator found = iter + 1;
+         found != params_.end(); ++found) {
+      if (*found == *iter) {
+        return true;
+      }
+    }
+  }
+  return false;
+}
+
+Codec::Codec(Type type, int id, const std::string& name, int clockrate)
+    : Codec(type, id, name, clockrate, 0) {}
+Codec::Codec(Type type,
+             int id,
+             const std::string& name,
+             int clockrate,
+             size_t channels)
+    : type(type),
+      id(id),
+      name(name),
+      clockrate(clockrate),
+      bitrate(0),
+      channels(channels) {}
+
+Codec::Codec(Type type) : Codec(type, 0, "", 0) {}
+
+Codec::Codec(const webrtc::SdpAudioFormat& c)
+    : Codec(Type::kAudio, 0, c.name, c.clockrate_hz, c.num_channels) {
+  params = c.parameters;
+}
+
+Codec::Codec(const webrtc::SdpVideoFormat& c)
+    : Codec(Type::kVideo, 0, c.name, kVideoCodecClockrate) {
+  params = c.parameters;
+  scalability_modes = c.scalability_modes;
+}
+
+Codec::Codec(const Codec& c) = default;
+Codec::Codec(Codec&& c) = default;
+Codec::~Codec() = default;
+Codec& Codec::operator=(const Codec& c) = default;
+Codec& Codec::operator=(Codec&& c) = default;
+
+bool Codec::operator==(const Codec& c) const {
+  return type == c.type && this->id == c.id &&  // id is reserved in objective-c
+         name == c.name && clockrate == c.clockrate && params == c.params &&
+         feedback_params == c.feedback_params &&
+         (type == Type::kAudio
+              ? (bitrate == c.bitrate && channels == c.channels)
+              : (packetization == c.packetization));
+}
+
+bool Codec::Matches(const Codec& codec) const {
+  // Match the codec id/name based on the typical static/dynamic name rules.
+  // Matching is case-insensitive.
+
+  // We support the ranges [96, 127] and more recently [35, 65].
+  // https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-1
+  // Within those ranges we match by codec name, outside by codec id.
+  // Since no codecs are assigned an id in the range [66, 95] by us, these will
+  // never match.
+  const int kLowerDynamicRangeMin = 35;
+  const int kLowerDynamicRangeMax = 65;
+  const int kUpperDynamicRangeMin = 96;
+  const int kUpperDynamicRangeMax = 127;
+  const bool is_id_in_dynamic_range =
+      (id >= kLowerDynamicRangeMin && id <= kLowerDynamicRangeMax) ||
+      (id >= kUpperDynamicRangeMin && id <= kUpperDynamicRangeMax);
+  const bool is_codec_id_in_dynamic_range =
+      (codec.id >= kLowerDynamicRangeMin &&
+       codec.id <= kLowerDynamicRangeMax) ||
+      (codec.id >= kUpperDynamicRangeMin && codec.id <= kUpperDynamicRangeMax);
+  bool matches_id = is_id_in_dynamic_range && is_codec_id_in_dynamic_range
+                        ? (absl::EqualsIgnoreCase(name, codec.name))
+                        : (id == codec.id);
+
+  auto matches_type_specific = [&]() {
+    switch (type) {
+      case Type::kAudio:
+        // If a nonzero clockrate is specified, it must match the actual
+        // clockrate. If a nonzero bitrate is specified, it must match the
+        // actual bitrate, unless the codec is VBR (0), where we just force the
+        // supplied value. The number of channels must match exactly, with the
+        // exception that channels=0 is treated synonymously as channels=1, per
+        // RFC 4566 section 6: " [The channels] parameter is OPTIONAL and may be
+        // omitted if the number of channels is one."
+        // Preference is ignored.
+        // TODO(juberti): Treat a zero clockrate as 8000Hz, the RTP default
+        // clockrate.
+        return ((codec.clockrate == 0 /*&& clockrate == 8000*/) ||
+                clockrate == codec.clockrate) &&
+               (codec.bitrate == 0 || bitrate <= 0 ||
+                bitrate == codec.bitrate) &&
+               ((codec.channels < 2 && channels < 2) ||
+                channels == codec.channels);
+
+      case Type::kVideo:
+        return IsSameCodecSpecific(name, params, codec.name, codec.params);
+    }
+  };
+
+  return matches_id && matches_type_specific();
+}
+
+bool Codec::MatchesRtpCodec(const webrtc::RtpCodec& codec_capability) const {
+  webrtc::RtpCodecParameters codec_parameters = ToCodecParameters();
+
+  return codec_parameters.name == codec_capability.name &&
+         codec_parameters.kind == codec_capability.kind &&
+         (codec_parameters.name == cricket::kRtxCodecName ||
+          (codec_parameters.num_channels == codec_capability.num_channels &&
+           codec_parameters.clock_rate == codec_capability.clock_rate &&
+           codec_parameters.parameters == codec_capability.parameters));
+}
+
+bool Codec::GetParam(const std::string& name, std::string* out) const {
+  webrtc::CodecParameterMap::const_iterator iter = params.find(name);
+  if (iter == params.end())
+    return false;
+  *out = iter->second;
+  return true;
+}
+
+bool Codec::GetParam(const std::string& name, int* out) const {
+  webrtc::CodecParameterMap::const_iterator iter = params.find(name);
+  if (iter == params.end())
+    return false;
+  return rtc::FromString(iter->second, out);
+}
+
+void Codec::SetParam(const std::string& name, const std::string& value) {
+  params[name] = value;
+}
+
+void Codec::SetParam(const std::string& name, int value) {
+  params[name] = rtc::ToString(value);
+}
+
+bool Codec::RemoveParam(const std::string& name) {
+  return params.erase(name) == 1;
+}
+
+void Codec::AddFeedbackParam(const FeedbackParam& param) {
+  feedback_params.Add(param);
+}
+
+bool Codec::HasFeedbackParam(const FeedbackParam& param) const {
+  return feedback_params.Has(param);
+}
+
+void Codec::IntersectFeedbackParams(const Codec& other) {
+  feedback_params.Intersect(other.feedback_params);
+}
+
+webrtc::RtpCodecParameters Codec::ToCodecParameters() const {
+  webrtc::RtpCodecParameters codec_params;
+  codec_params.payload_type = id;
+  codec_params.name = name;
+  codec_params.clock_rate = clockrate;
+  codec_params.parameters.insert(params.begin(), params.end());
+
+  switch (type) {
+    case Type::kAudio: {
+      codec_params.num_channels = static_cast<int>(channels);
+      codec_params.kind = MEDIA_TYPE_AUDIO;
+      break;
+    }
+    case Type::kVideo: {
+      codec_params.kind = MEDIA_TYPE_VIDEO;
+      break;
+    }
+  }
+
+  return codec_params;
+}
+
+bool Codec::IsMediaCodec() const {
+  return !IsResiliencyCodec() &&
+         !absl::EqualsIgnoreCase(name, kComfortNoiseCodecName);
+}
+
+bool Codec::IsResiliencyCodec() const {
+  return GetResiliencyType() != ResiliencyType::kNone;
+}
+
+Codec::ResiliencyType Codec::GetResiliencyType() const {
+  if (absl::EqualsIgnoreCase(name, kRedCodecName)) {
+    return ResiliencyType::kRed;
+  }
+  if (absl::EqualsIgnoreCase(name, kUlpfecCodecName)) {
+    return ResiliencyType::kUlpfec;
+  }
+  if (absl::EqualsIgnoreCase(name, kFlexfecCodecName)) {
+    return ResiliencyType::kFlexfec;
+  }
+  if (absl::EqualsIgnoreCase(name, kRtxCodecName)) {
+    return ResiliencyType::kRtx;
+  }
+  return ResiliencyType::kNone;
+}
+
+bool Codec::ValidateCodecFormat() const {
+  if (id < 0 || id > 127) {
+    RTC_LOG(LS_ERROR) << "Codec with invalid payload type: " << ToString();
+    return false;
+  }
+  if (IsResiliencyCodec()) {
+    return true;
+  }
+
+  int min_bitrate = -1;
+  int max_bitrate = -1;
+  if (GetParam(kCodecParamMinBitrate, &min_bitrate) &&
+      GetParam(kCodecParamMaxBitrate, &max_bitrate)) {
+    if (max_bitrate < min_bitrate) {
+      RTC_LOG(LS_ERROR) << "Codec with max < min bitrate: " << ToString();
+      return false;
+    }
+  }
+  return true;
+}
+
+std::string Codec::ToString() const {
+  char buf[256];
+
+  rtc::SimpleStringBuilder sb(buf);
+  switch (type) {
+    case Type::kAudio: {
+      sb << "AudioCodec[" << id << ":" << name << ":" << clockrate << ":"
+         << bitrate << ":" << channels << "]";
+      break;
+    }
+    case Type::kVideo: {
+      sb << "VideoCodec[" << id << ":" << name;
+      if (packetization.has_value()) {
+        sb << ":" << *packetization;
+      }
+      sb << "]";
+      break;
+    }
+  }
+  return sb.str();
+}
+
+Codec CreateAudioRtxCodec(int rtx_payload_type, int associated_payload_type) {
+  Codec rtx_codec = CreateAudioCodec(rtx_payload_type, kRtxCodecName, 0, 1);
+  rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type);
+  return rtx_codec;
+}
+
+Codec CreateVideoRtxCodec(int rtx_payload_type, int associated_payload_type) {
+  Codec rtx_codec = CreateVideoCodec(rtx_payload_type, kRtxCodecName);
+  rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type);
+  return rtx_codec;
+}
+
+const Codec* FindCodecById(const std::vector<Codec>& codecs, int payload_type) {
+  for (const auto& codec : codecs) {
+    if (codec.id == payload_type)
+      return &codec;
+  }
+  return nullptr;
+}
+
+bool HasLntf(const Codec& codec) {
+  return codec.HasFeedbackParam(
+      FeedbackParam(kRtcpFbParamLntf, kParamValueEmpty));
+}
+
+bool HasNack(const Codec& codec) {
+  return codec.HasFeedbackParam(
+      FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
+}
+
+bool HasRemb(const Codec& codec) {
+  return codec.HasFeedbackParam(
+      FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
+}
+
+bool HasRrtr(const Codec& codec) {
+  return codec.HasFeedbackParam(
+      FeedbackParam(kRtcpFbParamRrtr, kParamValueEmpty));
+}
+
+bool HasTransportCc(const Codec& codec) {
+  return codec.HasFeedbackParam(
+      FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
+}
+
+const Codec* FindMatchingVideoCodec(const std::vector<Codec>& supported_codecs,
+                                    const Codec& codec) {
+  webrtc::SdpVideoFormat sdp_video_format{codec.name, codec.params};
+  for (const Codec& supported_codec : supported_codecs) {
+    if (sdp_video_format.IsSameCodec(
+            {supported_codec.name, supported_codec.params})) {
+      return &supported_codec;
+    }
+  }
+  return nullptr;
+}
+
+std::vector<const Codec*> FindAllMatchingCodecs(
+    const std::vector<Codec>& supported_codecs,
+    const Codec& codec) {
+  std::vector<const Codec*> result;
+  webrtc::SdpVideoFormat sdp(codec.name, codec.params);
+  for (const Codec& supported_codec : supported_codecs) {
+    if (sdp.IsSameCodec({supported_codec.name, supported_codec.params})) {
+      result.push_back(&supported_codec);
+    }
+  }
+  return result;
+}
+
+// If a decoder supports any H264 profile, it is implicitly assumed to also
+// support constrained base line even though it's not explicitly listed.
+void AddH264ConstrainedBaselineProfileToSupportedFormats(
+    std::vector<webrtc::SdpVideoFormat>* supported_formats) {
+  std::vector<webrtc::SdpVideoFormat> cbr_supported_formats;
+
+  // For any H264 supported profile, add the corresponding constrained baseline
+  // profile.
+  for (auto it = supported_formats->cbegin(); it != supported_formats->cend();
+       ++it) {
+    if (it->name == cricket::kH264CodecName) {
+      const absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
+          webrtc::ParseSdpForH264ProfileLevelId(it->parameters);
+      if (profile_level_id &&
+          profile_level_id->profile !=
+              webrtc::H264Profile::kProfileConstrainedBaseline) {
+        webrtc::SdpVideoFormat cbp_format = *it;
+        webrtc::H264ProfileLevelId cbp_profile = *profile_level_id;
+        cbp_profile.profile = webrtc::H264Profile::kProfileConstrainedBaseline;
+        cbp_format.parameters[cricket::kH264FmtpProfileLevelId] =
+            *webrtc::H264ProfileLevelIdToString(cbp_profile);
+        cbr_supported_formats.push_back(cbp_format);
+      }
+    }
+  }
+
+  size_t original_size = supported_formats->size();
+  // ...if it's not already in the list.
+  std::copy_if(cbr_supported_formats.begin(), cbr_supported_formats.end(),
+               std::back_inserter(*supported_formats),
+               [supported_formats](const webrtc::SdpVideoFormat& format) {
+                 return !format.IsCodecInList(*supported_formats);
+               });
+
+  if (supported_formats->size() > original_size) {
+    RTC_LOG(LS_WARNING) << "Explicitly added H264 constrained baseline to list "
+                           "of supported formats.";
+  }
+}
+
+Codec CreateAudioCodec(int id,
+                       const std::string& name,
+                       int clockrate,
+                       size_t channels) {
+  return Codec(Codec::Type::kAudio, id, name, clockrate, channels);
+}
+
+Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c) {
+  return Codec(c);
+}
+
+Codec CreateVideoCodec(const std::string& name) {
+  return CreateVideoCodec(0, name);
+}
+
+Codec CreateVideoCodec(int id, const std::string& name) {
+  Codec c(Codec::Type::kVideo, id, name, kVideoCodecClockrate);
+  if (absl::EqualsIgnoreCase(kH264CodecName, name)) {
+    // This default is set for all H.264 codecs created because
+    // that was the default before packetization mode support was added.
+    // TODO(hta): Move this to the places that create VideoCodecs from
+    // SDP or from knowledge of implementation capabilities.
+    c.SetParam(kH264FmtpPacketizationMode, "1");
+  }
+  return c;
+}
+
+Codec CreateVideoCodec(const webrtc::SdpVideoFormat& c) {
+  return Codec(c);
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/codec.h

@@ -0,0 +1,230 @@
+/*
+ *  Copyright (c) 2004 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 MEDIA_BASE_CODEC_H_
+#define MEDIA_BASE_CODEC_H_
+
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/rtp_parameters.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "media/base/media_constants.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace cricket {
+
+class FeedbackParam {
+ public:
+  FeedbackParam() = default;
+  FeedbackParam(absl::string_view id, const std::string& param)
+      : id_(id), param_(param) {}
+  explicit FeedbackParam(absl::string_view id)
+      : id_(id), param_(kParamValueEmpty) {}
+
+  bool operator==(const FeedbackParam& other) const;
+  bool operator!=(const FeedbackParam& c) const { return !(*this == c); }
+
+  const std::string& id() const { return id_; }
+  const std::string& param() const { return param_; }
+
+ private:
+  std::string id_;     // e.g. "nack", "ccm"
+  std::string param_;  // e.g. "", "rpsi", "fir"
+};
+
+class FeedbackParams {
+ public:
+  FeedbackParams();
+  ~FeedbackParams();
+  bool operator==(const FeedbackParams& other) const;
+  bool operator!=(const FeedbackParams& c) const { return !(*this == c); }
+
+  bool Has(const FeedbackParam& param) const;
+  void Add(const FeedbackParam& param);
+
+  void Intersect(const FeedbackParams& from);
+
+  const std::vector<FeedbackParam>& params() const { return params_; }
+
+ private:
+  bool HasDuplicateEntries() const;
+
+  std::vector<FeedbackParam> params_;
+};
+
+struct RTC_EXPORT Codec {
+  enum class Type {
+    kAudio,
+    kVideo,
+  };
+
+  enum class ResiliencyType {
+    kNone,
+    kRed,
+    kUlpfec,
+    kFlexfec,
+    kRtx,
+  };
+
+  Type type;
+  int id;
+  std::string name;
+  int clockrate;
+
+  // Audio only
+  // Can be used to override the target bitrate in the encoder.
+  // TODO(orphis): Remove in favor of alternative APIs
+  int bitrate;
+  size_t channels;
+
+  // Video only
+  absl::optional<std::string> packetization;
+  absl::InlinedVector<webrtc::ScalabilityMode, webrtc::kScalabilityModeCount>
+      scalability_modes;
+
+  // H.265 only
+  absl::optional<std::string> tx_mode;
+
+  // Non key-value parameters such as the telephone-event "0‐15" are
+  // represented using an empty string as key, i.e. {"": "0-15"}.
+  webrtc::CodecParameterMap params;
+  FeedbackParams feedback_params;
+
+  Codec(const Codec& c);
+  Codec(Codec&& c);
+
+  virtual ~Codec();
+
+  // Indicates if this codec is compatible with the specified codec by
+  // checking the assigned id and profile values for the relevant video codecs.
+  // For H.264, packetization modes will be compared; If H.265 is enabled,
+  // TxModes will be compared.
+  // H.264(and H.265, if enabled) levels are not compared.
+  bool Matches(const Codec& codec) const;
+  bool MatchesRtpCodec(const webrtc::RtpCodec& capability) const;
+
+  // Find the parameter for `name` and write the value to `out`.
+  bool GetParam(const std::string& name, std::string* out) const;
+  bool GetParam(const std::string& name, int* out) const;
+
+  void SetParam(const std::string& name, const std::string& value);
+  void SetParam(const std::string& name, int value);
+
+  // It is safe to input a non-existent parameter.
+  // Returns true if the parameter existed, false if it did not exist.
+  bool RemoveParam(const std::string& name);
+
+  bool HasFeedbackParam(const FeedbackParam& param) const;
+  void AddFeedbackParam(const FeedbackParam& param);
+
+  // Filter `this` feedbacks params such that only those shared by both `this`
+  // and `other` are kept.
+  void IntersectFeedbackParams(const Codec& other);
+
+  virtual webrtc::RtpCodecParameters ToCodecParameters() const;
+
+  // The codec represent an actual media codec, and not a resiliency codec.
+  bool IsMediaCodec() const;
+  // The codec represent a resiliency codec such as RED, RTX or FEC variants.
+  bool IsResiliencyCodec() const;
+  ResiliencyType GetResiliencyType() const;
+
+  // Validates a VideoCodec's payload type, dimensions and bitrates etc. If they
+  // don't make sense (such as max < min bitrate), and error is logged and
+  // ValidateCodecFormat returns false.
+  bool ValidateCodecFormat() const;
+
+  std::string ToString() const;
+
+  Codec& operator=(const Codec& c);
+  Codec& operator=(Codec&& c);
+
+  bool operator==(const Codec& c) const;
+
+  bool operator!=(const Codec& c) const { return !(*this == c); }
+
+ protected:
+  // Creates an empty codec.
+  explicit Codec(Type type);
+  // Creates a codec with the given parameters.
+  Codec(Type type, int id, const std::string& name, int clockrate);
+  Codec(Type type,
+        int id,
+        const std::string& name,
+        int clockrate,
+        size_t channels);
+
+  explicit Codec(const webrtc::SdpAudioFormat& c);
+  explicit Codec(const webrtc::SdpVideoFormat& c);
+
+  friend Codec CreateAudioCodec(int id,
+                                const std::string& name,
+                                int clockrate,
+                                size_t channels);
+  friend Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c);
+  friend Codec CreateAudioRtxCodec(int rtx_payload_type,
+                                   int associated_payload_type);
+  friend Codec CreateVideoCodec(int id, const std::string& name);
+  friend Codec CreateVideoCodec(const webrtc::SdpVideoFormat& c);
+  friend Codec CreateVideoRtxCodec(int rtx_payload_type,
+                                   int associated_payload_type);
+};
+
+// TODO(webrtc:15214): Compatibility names, to be migrated away and removed.
+using VideoCodec = Codec;
+using AudioCodec = Codec;
+
+using VideoCodecs = std::vector<Codec>;
+using AudioCodecs = std::vector<Codec>;
+
+Codec CreateAudioCodec(int id,
+                       const std::string& name,
+                       int clockrate,
+                       size_t channels);
+Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c);
+Codec CreateAudioRtxCodec(int rtx_payload_type, int associated_payload_type);
+Codec CreateVideoCodec(const std::string& name);
+Codec CreateVideoCodec(int id, const std::string& name);
+Codec CreateVideoCodec(const webrtc::SdpVideoFormat& c);
+Codec CreateVideoRtxCodec(int rtx_payload_type, int associated_payload_type);
+
+// Get the codec setting associated with `payload_type`. If there
+// is no codec associated with that payload type it returns nullptr.
+const Codec* FindCodecById(const std::vector<Codec>& codecs, int payload_type);
+
+bool HasLntf(const Codec& codec);
+bool HasNack(const Codec& codec);
+bool HasRemb(const Codec& codec);
+bool HasRrtr(const Codec& codec);
+bool HasTransportCc(const Codec& codec);
+
+// Returns the first codec in `supported_codecs` that matches `codec`, or
+// nullptr if no codec matches.
+const Codec* FindMatchingVideoCodec(const std::vector<Codec>& supported_codecs,
+                                    const Codec& codec);
+
+// Returns all codecs in `supported_codecs` that matches `codec`.
+std::vector<const Codec*> FindAllMatchingCodecs(
+    const std::vector<Codec>& supported_codecs,
+    const Codec& codec);
+
+RTC_EXPORT void AddH264ConstrainedBaselineProfileToSupportedFormats(
+    std::vector<webrtc::SdpVideoFormat>* supported_formats);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_CODEC_H_

TMessagesProj/jni/voip/webrtc/media/base/fake_frame_source.cc

@@ -0,0 +1,88 @@
+/*
+ *  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 "media/base/fake_frame_source.h"
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+
+namespace cricket {
+
+FakeFrameSource::FakeFrameSource(int width,
+                                 int height,
+                                 int interval_us,
+                                 int64_t timestamp_offset_us)
+    : width_(width),
+      height_(height),
+      interval_us_(interval_us),
+      next_timestamp_us_(timestamp_offset_us) {
+  RTC_CHECK_GT(width_, 0);
+  RTC_CHECK_GT(height_, 0);
+  RTC_CHECK_GT(interval_us_, 0);
+  RTC_CHECK_GE(next_timestamp_us_, 0);
+}
+
+FakeFrameSource::FakeFrameSource(int width, int height, int interval_us)
+    : FakeFrameSource(width, height, interval_us, rtc::TimeMicros()) {}
+
+webrtc::VideoRotation FakeFrameSource::GetRotation() const {
+  return rotation_;
+}
+
+void FakeFrameSource::SetRotation(webrtc::VideoRotation rotation) {
+  rotation_ = rotation;
+}
+
+webrtc::VideoFrame FakeFrameSource::GetFrameRotationApplied() {
+  switch (rotation_) {
+    case webrtc::kVideoRotation_0:
+    case webrtc::kVideoRotation_180:
+      return GetFrame(width_, height_, webrtc::kVideoRotation_0, interval_us_);
+    case webrtc::kVideoRotation_90:
+    case webrtc::kVideoRotation_270:
+      return GetFrame(height_, width_, webrtc::kVideoRotation_0, interval_us_);
+  }
+  RTC_DCHECK_NOTREACHED() << "Invalid rotation value: "
+                          << static_cast<int>(rotation_);
+  // Without this return, the Windows Visual Studio compiler complains
+  // "not all control paths return a value".
+  return GetFrame();
+}
+
+webrtc::VideoFrame FakeFrameSource::GetFrame() {
+  return GetFrame(width_, height_, rotation_, interval_us_);
+}
+
+webrtc::VideoFrame FakeFrameSource::GetFrame(int width,
+                                             int height,
+                                             webrtc::VideoRotation rotation,
+                                             int interval_us) {
+  RTC_CHECK_GT(width, 0);
+  RTC_CHECK_GT(height, 0);
+  RTC_CHECK_GT(interval_us, 0);
+
+  rtc::scoped_refptr<webrtc::I420Buffer> buffer(
+      webrtc::I420Buffer::Create(width, height));
+
+  buffer->InitializeData();
+  webrtc::VideoFrame frame = webrtc::VideoFrame::Builder()
+                                 .set_video_frame_buffer(buffer)
+                                 .set_rotation(rotation)
+                                 .set_timestamp_us(next_timestamp_us_)
+                                 .build();
+
+  next_timestamp_us_ += interval_us;
+  return frame;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/fake_frame_source.h

@@ -0,0 +1,50 @@
+/*
+ *  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 MEDIA_BASE_FAKE_FRAME_SOURCE_H_
+#define MEDIA_BASE_FAKE_FRAME_SOURCE_H_
+
+#include "api/video/video_frame.h"
+#include "rtc_base/time_utils.h"
+
+namespace cricket {
+
+class FakeFrameSource {
+ public:
+  FakeFrameSource(int width,
+                  int height,
+                  int interval_us,
+                  int64_t timestamp_offset_us);
+  FakeFrameSource(int width, int height, int interval_us);
+
+  webrtc::VideoRotation GetRotation() const;
+  void SetRotation(webrtc::VideoRotation rotation);
+
+  webrtc::VideoFrame GetFrame();
+  webrtc::VideoFrame GetFrameRotationApplied();
+
+  // Override configuration.
+  webrtc::VideoFrame GetFrame(int width,
+                              int height,
+                              webrtc::VideoRotation rotation,
+                              int interval_us);
+
+ private:
+  const int width_;
+  const int height_;
+  const int interval_us_;
+
+  webrtc::VideoRotation rotation_ = webrtc::kVideoRotation_0;
+  int64_t next_timestamp_us_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_FAKE_FRAME_SOURCE_H_

TMessagesProj/jni/voip/webrtc/media/base/fake_media_engine.cc

@@ -0,0 +1,705 @@
+/*
+ *  Copyright 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 "media/base/fake_media_engine.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "media/base/media_channel.h"
+#include "rtc_base/checks.h"
+
+namespace cricket {
+using webrtc::TaskQueueBase;
+
+FakeVoiceMediaReceiveChannel::DtmfInfo::DtmfInfo(uint32_t ssrc,
+                                                 int event_code,
+                                                 int duration)
+    : ssrc(ssrc), event_code(event_code), duration(duration) {}
+
+FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::VoiceChannelAudioSink(
+    AudioSource* source)
+    : source_(source) {
+  source_->SetSink(this);
+}
+FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() {
+  if (source_) {
+    source_->SetSink(nullptr);
+  }
+}
+void FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::OnData(
+    const void* audio_data,
+    int bits_per_sample,
+    int sample_rate,
+    size_t number_of_channels,
+    size_t number_of_frames,
+    absl::optional<int64_t> absolute_capture_timestamp_ms) {}
+void FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::OnClose() {
+  source_ = nullptr;
+}
+AudioSource* FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::source()
+    const {
+  return source_;
+}
+
+FakeVoiceMediaReceiveChannel::FakeVoiceMediaReceiveChannel(
+    const AudioOptions& options,
+    TaskQueueBase* network_thread)
+    : RtpReceiveChannelHelper<VoiceMediaReceiveChannelInterface>(
+          network_thread),
+      max_bps_(-1) {
+  output_scalings_[0] = 1.0;  // For default channel.
+  SetOptions(options);
+}
+FakeVoiceMediaReceiveChannel::~FakeVoiceMediaReceiveChannel() = default;
+const std::vector<AudioCodec>& FakeVoiceMediaReceiveChannel::recv_codecs()
+    const {
+  return recv_codecs_;
+}
+const std::vector<FakeVoiceMediaReceiveChannel::DtmfInfo>&
+FakeVoiceMediaReceiveChannel::dtmf_info_queue() const {
+  return dtmf_info_queue_;
+}
+const AudioOptions& FakeVoiceMediaReceiveChannel::options() const {
+  return options_;
+}
+int FakeVoiceMediaReceiveChannel::max_bps() const {
+  return max_bps_;
+}
+bool FakeVoiceMediaReceiveChannel::SetReceiverParameters(
+    const AudioReceiverParameters& params) {
+  set_recv_rtcp_parameters(params.rtcp);
+  return (SetRecvCodecs(params.codecs) &&
+          SetRecvRtpHeaderExtensions(params.extensions));
+}
+void FakeVoiceMediaReceiveChannel::SetPlayout(bool playout) {
+  set_playout(playout);
+}
+bool FakeVoiceMediaReceiveChannel::HasSource(uint32_t ssrc) const {
+  return local_sinks_.find(ssrc) != local_sinks_.end();
+}
+bool FakeVoiceMediaReceiveChannel::AddRecvStream(const StreamParams& sp) {
+  if (!RtpReceiveChannelHelper<
+          VoiceMediaReceiveChannelInterface>::AddRecvStream(sp))
+    return false;
+  output_scalings_[sp.first_ssrc()] = 1.0;
+  output_delays_[sp.first_ssrc()] = 0;
+  return true;
+}
+bool FakeVoiceMediaReceiveChannel::RemoveRecvStream(uint32_t ssrc) {
+  if (!RtpReceiveChannelHelper<
+          VoiceMediaReceiveChannelInterface>::RemoveRecvStream(ssrc))
+    return false;
+  output_scalings_.erase(ssrc);
+  output_delays_.erase(ssrc);
+  return true;
+}
+bool FakeVoiceMediaReceiveChannel::SetOutputVolume(uint32_t ssrc,
+                                                   double volume) {
+  if (output_scalings_.find(ssrc) != output_scalings_.end()) {
+    output_scalings_[ssrc] = volume;
+    return true;
+  }
+  return false;
+}
+bool FakeVoiceMediaReceiveChannel::SetDefaultOutputVolume(double volume) {
+  for (auto& entry : output_scalings_) {
+    entry.second = volume;
+  }
+  return true;
+}
+bool FakeVoiceMediaReceiveChannel::GetOutputVolume(uint32_t ssrc,
+                                                   double* volume) {
+  if (output_scalings_.find(ssrc) == output_scalings_.end())
+    return false;
+  *volume = output_scalings_[ssrc];
+  return true;
+}
+bool FakeVoiceMediaReceiveChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
+                                                                int delay_ms) {
+  if (output_delays_.find(ssrc) == output_delays_.end()) {
+    return false;
+  } else {
+    output_delays_[ssrc] = delay_ms;
+    return true;
+  }
+}
+absl::optional<int> FakeVoiceMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
+    uint32_t ssrc) const {
+  const auto it = output_delays_.find(ssrc);
+  if (it != output_delays_.end()) {
+    return it->second;
+  }
+  return absl::nullopt;
+}
+bool FakeVoiceMediaReceiveChannel::GetStats(VoiceMediaReceiveInfo* info,
+                                            bool get_and_clear_legacy_stats) {
+  return false;
+}
+void FakeVoiceMediaReceiveChannel::SetRawAudioSink(
+    uint32_t ssrc,
+    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
+  sink_ = std::move(sink);
+}
+void FakeVoiceMediaReceiveChannel::SetDefaultRawAudioSink(
+    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
+  sink_ = std::move(sink);
+}
+std::vector<webrtc::RtpSource> FakeVoiceMediaReceiveChannel::GetSources(
+    uint32_t ssrc) const {
+  return std::vector<webrtc::RtpSource>();
+}
+bool FakeVoiceMediaReceiveChannel::SetRecvCodecs(
+    const std::vector<AudioCodec>& codecs) {
+  if (fail_set_recv_codecs()) {
+    // Fake the failure in SetRecvCodecs.
+    return false;
+  }
+  recv_codecs_ = codecs;
+  return true;
+}
+bool FakeVoiceMediaReceiveChannel::SetMaxSendBandwidth(int bps) {
+  max_bps_ = bps;
+  return true;
+}
+bool FakeVoiceMediaReceiveChannel::SetOptions(const AudioOptions& options) {
+  // Does a "merge" of current options and set options.
+  options_.SetAll(options);
+  return true;
+}
+
+FakeVoiceMediaSendChannel::DtmfInfo::DtmfInfo(uint32_t ssrc,
+                                              int event_code,
+                                              int duration)
+    : ssrc(ssrc), event_code(event_code), duration(duration) {}
+
+FakeVoiceMediaSendChannel::VoiceChannelAudioSink::VoiceChannelAudioSink(
+    AudioSource* source)
+    : source_(source) {
+  source_->SetSink(this);
+}
+FakeVoiceMediaSendChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() {
+  if (source_) {
+    source_->SetSink(nullptr);
+  }
+}
+void FakeVoiceMediaSendChannel::VoiceChannelAudioSink::OnData(
+    const void* audio_data,
+    int bits_per_sample,
+    int sample_rate,
+    size_t number_of_channels,
+    size_t number_of_frames,
+    absl::optional<int64_t> absolute_capture_timestamp_ms) {}
+void FakeVoiceMediaSendChannel::VoiceChannelAudioSink::OnClose() {
+  source_ = nullptr;
+}
+AudioSource* FakeVoiceMediaSendChannel::VoiceChannelAudioSink::source() const {
+  return source_;
+}
+
+FakeVoiceMediaSendChannel::FakeVoiceMediaSendChannel(
+    const AudioOptions& options,
+    TaskQueueBase* network_thread)
+    : RtpSendChannelHelper<VoiceMediaSendChannelInterface>(network_thread),
+      max_bps_(-1) {
+  output_scalings_[0] = 1.0;  // For default channel.
+  SetOptions(options);
+}
+FakeVoiceMediaSendChannel::~FakeVoiceMediaSendChannel() = default;
+const std::vector<AudioCodec>& FakeVoiceMediaSendChannel::send_codecs() const {
+  return send_codecs_;
+}
+absl::optional<Codec> FakeVoiceMediaSendChannel::GetSendCodec() const {
+  if (!send_codecs_.empty()) {
+    return send_codecs_.front();
+  }
+  return absl::nullopt;
+}
+const std::vector<FakeVoiceMediaSendChannel::DtmfInfo>&
+FakeVoiceMediaSendChannel::dtmf_info_queue() const {
+  return dtmf_info_queue_;
+}
+const AudioOptions& FakeVoiceMediaSendChannel::options() const {
+  return options_;
+}
+int FakeVoiceMediaSendChannel::max_bps() const {
+  return max_bps_;
+}
+bool FakeVoiceMediaSendChannel::SetSenderParameters(
+    const AudioSenderParameter& params) {
+  set_send_rtcp_parameters(params.rtcp);
+  SetExtmapAllowMixed(params.extmap_allow_mixed);
+  return (SetSendCodecs(params.codecs) &&
+          SetSendRtpHeaderExtensions(params.extensions) &&
+          SetMaxSendBandwidth(params.max_bandwidth_bps) &&
+          SetOptions(params.options));
+}
+void FakeVoiceMediaSendChannel::SetSend(bool send) {
+  set_sending(send);
+}
+bool FakeVoiceMediaSendChannel::SetAudioSend(uint32_t ssrc,
+                                             bool enable,
+                                             const AudioOptions* options,
+                                             AudioSource* source) {
+  if (!SetLocalSource(ssrc, source)) {
+    return false;
+  }
+  if (!RtpSendChannelHelper<VoiceMediaSendChannelInterface>::MuteStream(
+          ssrc, !enable)) {
+    return false;
+  }
+  if (enable && options) {
+    return SetOptions(*options);
+  }
+  return true;
+}
+bool FakeVoiceMediaSendChannel::HasSource(uint32_t ssrc) const {
+  return local_sinks_.find(ssrc) != local_sinks_.end();
+}
+bool FakeVoiceMediaSendChannel::CanInsertDtmf() {
+  for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
+       it != send_codecs_.end(); ++it) {
+    // Find the DTMF telephone event "codec".
+    if (absl::EqualsIgnoreCase(it->name, "telephone-event")) {
+      return true;
+    }
+  }
+  return false;
+}
+bool FakeVoiceMediaSendChannel::InsertDtmf(uint32_t ssrc,
+                                           int event_code,
+                                           int duration) {
+  dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
+  return true;
+}
+bool FakeVoiceMediaSendChannel::GetOutputVolume(uint32_t ssrc, double* volume) {
+  if (output_scalings_.find(ssrc) == output_scalings_.end())
+    return false;
+  *volume = output_scalings_[ssrc];
+  return true;
+}
+bool FakeVoiceMediaSendChannel::GetStats(VoiceMediaSendInfo* info) {
+  return false;
+}
+bool FakeVoiceMediaSendChannel::SetSendCodecs(
+    const std::vector<AudioCodec>& codecs) {
+  if (fail_set_send_codecs()) {
+    // Fake the failure in SetSendCodecs.
+    return false;
+  }
+  send_codecs_ = codecs;
+  return true;
+}
+bool FakeVoiceMediaSendChannel::SetMaxSendBandwidth(int bps) {
+  max_bps_ = bps;
+  return true;
+}
+bool FakeVoiceMediaSendChannel::SetOptions(const AudioOptions& options) {
+  // Does a "merge" of current options and set options.
+  options_.SetAll(options);
+  return true;
+}
+bool FakeVoiceMediaSendChannel::SetLocalSource(uint32_t ssrc,
+                                               AudioSource* source) {
+  auto it = local_sinks_.find(ssrc);
+  if (source) {
+    if (it != local_sinks_.end()) {
+      RTC_CHECK(it->second->source() == source);
+    } else {
+      local_sinks_.insert(std::make_pair(
+          ssrc, std::make_unique<VoiceChannelAudioSink>(source)));
+    }
+  } else {
+    if (it != local_sinks_.end()) {
+      local_sinks_.erase(it);
+    }
+  }
+  return true;
+}
+
+bool CompareDtmfInfo(const FakeVoiceMediaSendChannel::DtmfInfo& info,
+                     uint32_t ssrc,
+                     int event_code,
+                     int duration) {
+  return (info.duration == duration && info.event_code == event_code &&
+          info.ssrc == ssrc);
+}
+
+FakeVideoMediaSendChannel::FakeVideoMediaSendChannel(
+    const VideoOptions& options,
+    TaskQueueBase* network_thread)
+    : RtpSendChannelHelper<VideoMediaSendChannelInterface>(network_thread),
+      max_bps_(-1) {
+  SetOptions(options);
+}
+FakeVideoMediaSendChannel::~FakeVideoMediaSendChannel() = default;
+const std::vector<VideoCodec>& FakeVideoMediaSendChannel::send_codecs() const {
+  return send_codecs_;
+}
+const std::vector<VideoCodec>& FakeVideoMediaSendChannel::codecs() const {
+  return send_codecs();
+}
+const VideoOptions& FakeVideoMediaSendChannel::options() const {
+  return options_;
+}
+int FakeVideoMediaSendChannel::max_bps() const {
+  return max_bps_;
+}
+bool FakeVideoMediaSendChannel::SetSenderParameters(
+    const VideoSenderParameters& params) {
+  set_send_rtcp_parameters(params.rtcp);
+  SetExtmapAllowMixed(params.extmap_allow_mixed);
+  return (SetSendCodecs(params.codecs) &&
+          SetSendRtpHeaderExtensions(params.extensions) &&
+          SetMaxSendBandwidth(params.max_bandwidth_bps));
+}
+absl::optional<Codec> FakeVideoMediaSendChannel::GetSendCodec() const {
+  if (send_codecs_.empty()) {
+    return absl::nullopt;
+  }
+  return send_codecs_[0];
+}
+bool FakeVideoMediaSendChannel::SetSend(bool send) {
+  return set_sending(send);
+}
+bool FakeVideoMediaSendChannel::SetVideoSend(
+    uint32_t ssrc,
+    const VideoOptions* options,
+    rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
+  if (options) {
+    if (!SetOptions(*options)) {
+      return false;
+    }
+  }
+  sources_[ssrc] = source;
+  return true;
+}
+bool FakeVideoMediaSendChannel::HasSource(uint32_t ssrc) const {
+  return sources_.find(ssrc) != sources_.end() && sources_.at(ssrc) != nullptr;
+}
+void FakeVideoMediaSendChannel::FillBitrateInfo(
+    BandwidthEstimationInfo* bwe_info) {}
+bool FakeVideoMediaSendChannel::GetStats(VideoMediaSendInfo* info) {
+  return false;
+}
+bool FakeVideoMediaSendChannel::SetSendCodecs(
+    const std::vector<VideoCodec>& codecs) {
+  if (fail_set_send_codecs()) {
+    // Fake the failure in SetSendCodecs.
+    return false;
+  }
+  send_codecs_ = codecs;
+
+  return true;
+}
+bool FakeVideoMediaSendChannel::SetOptions(const VideoOptions& options) {
+  options_ = options;
+  return true;
+}
+
+bool FakeVideoMediaSendChannel::SetMaxSendBandwidth(int bps) {
+  max_bps_ = bps;
+  return true;
+}
+void FakeVideoMediaSendChannel::GenerateSendKeyFrame(
+    uint32_t ssrc,
+    const std::vector<std::string>& rids) {}
+
+FakeVideoMediaReceiveChannel::FakeVideoMediaReceiveChannel(
+    const VideoOptions& options,
+    TaskQueueBase* network_thread)
+    : RtpReceiveChannelHelper<VideoMediaReceiveChannelInterface>(
+          network_thread),
+      max_bps_(-1) {
+  SetOptions(options);
+}
+FakeVideoMediaReceiveChannel::~FakeVideoMediaReceiveChannel() = default;
+const std::vector<VideoCodec>& FakeVideoMediaReceiveChannel::recv_codecs()
+    const {
+  return recv_codecs_;
+}
+bool FakeVideoMediaReceiveChannel::rendering() const {
+  return playout();
+}
+const VideoOptions& FakeVideoMediaReceiveChannel::options() const {
+  return options_;
+}
+const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
+FakeVideoMediaReceiveChannel::sinks() const {
+  return sinks_;
+}
+int FakeVideoMediaReceiveChannel::max_bps() const {
+  return max_bps_;
+}
+bool FakeVideoMediaReceiveChannel::SetReceiverParameters(
+    const VideoReceiverParameters& params) {
+  set_recv_rtcp_parameters(params.rtcp);
+  return (SetRecvCodecs(params.codecs) &&
+          SetRecvRtpHeaderExtensions(params.extensions));
+}
+bool FakeVideoMediaReceiveChannel::SetSink(
+    uint32_t ssrc,
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  auto it = sinks_.find(ssrc);
+  if (it == sinks_.end()) {
+    return false;
+  }
+  it->second = sink;
+  return true;
+}
+void FakeVideoMediaReceiveChannel::SetDefaultSink(
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {}
+bool FakeVideoMediaReceiveChannel::HasSink(uint32_t ssrc) const {
+  return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
+}
+bool FakeVideoMediaReceiveChannel::HasSource(uint32_t ssrc) const {
+  return sources_.find(ssrc) != sources_.end() && sources_.at(ssrc) != nullptr;
+}
+bool FakeVideoMediaReceiveChannel::AddRecvStream(const StreamParams& sp) {
+  if (!RtpReceiveChannelHelper<
+          VideoMediaReceiveChannelInterface>::AddRecvStream(sp))
+    return false;
+  sinks_[sp.first_ssrc()] = NULL;
+  output_delays_[sp.first_ssrc()] = 0;
+  return true;
+}
+bool FakeVideoMediaReceiveChannel::RemoveRecvStream(uint32_t ssrc) {
+  if (!RtpReceiveChannelHelper<
+          VideoMediaReceiveChannelInterface>::RemoveRecvStream(ssrc))
+    return false;
+  sinks_.erase(ssrc);
+  output_delays_.erase(ssrc);
+  return true;
+}
+std::vector<webrtc::RtpSource> FakeVideoMediaReceiveChannel::GetSources(
+    uint32_t ssrc) const {
+  return {};
+}
+bool FakeVideoMediaReceiveChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
+                                                                int delay_ms) {
+  if (output_delays_.find(ssrc) == output_delays_.end()) {
+    return false;
+  } else {
+    output_delays_[ssrc] = delay_ms;
+    return true;
+  }
+}
+absl::optional<int> FakeVideoMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
+    uint32_t ssrc) const {
+  const auto it = output_delays_.find(ssrc);
+  if (it != output_delays_.end()) {
+    return it->second;
+  }
+  return absl::nullopt;
+}
+bool FakeVideoMediaReceiveChannel::SetRecvCodecs(
+    const std::vector<VideoCodec>& codecs) {
+  if (fail_set_recv_codecs()) {
+    // Fake the failure in SetRecvCodecs.
+    return false;
+  }
+  recv_codecs_ = codecs;
+  return true;
+}
+bool FakeVideoMediaReceiveChannel::SetOptions(const VideoOptions& options) {
+  options_ = options;
+  return true;
+}
+
+bool FakeVideoMediaReceiveChannel::SetMaxSendBandwidth(int bps) {
+  max_bps_ = bps;
+  return true;
+}
+
+void FakeVideoMediaReceiveChannel::SetRecordableEncodedFrameCallback(
+    uint32_t ssrc,
+    std::function<void(const webrtc::RecordableEncodedFrame&)> callback) {}
+
+void FakeVideoMediaReceiveChannel::ClearRecordableEncodedFrameCallback(
+    uint32_t ssrc) {}
+
+void FakeVideoMediaReceiveChannel::RequestRecvKeyFrame(uint32_t ssrc) {}
+
+bool FakeVideoMediaReceiveChannel::GetStats(VideoMediaReceiveInfo* info) {
+  return false;
+}
+
+FakeVoiceEngine::FakeVoiceEngine() : fail_create_channel_(false) {
+  // Add a fake audio codec. Note that the name must not be "" as there are
+  // sanity checks against that.
+  SetCodecs({cricket::CreateAudioCodec(101, "fake_audio_codec", 8000, 1)});
+}
+void FakeVoiceEngine::Init() {}
+rtc::scoped_refptr<webrtc::AudioState> FakeVoiceEngine::GetAudioState() const {
+  return rtc::scoped_refptr<webrtc::AudioState>();
+}
+std::unique_ptr<VoiceMediaSendChannelInterface>
+FakeVoiceEngine::CreateSendChannel(webrtc::Call* call,
+                                   const MediaConfig& config,
+                                   const AudioOptions& options,
+                                   const webrtc::CryptoOptions& crypto_options,
+                                   webrtc::AudioCodecPairId codec_pair_id) {
+  std::unique_ptr<FakeVoiceMediaSendChannel> ch =
+      std::make_unique<FakeVoiceMediaSendChannel>(options,
+                                                  call->network_thread());
+  return ch;
+}
+std::unique_ptr<VoiceMediaReceiveChannelInterface>
+FakeVoiceEngine::CreateReceiveChannel(
+    webrtc::Call* call,
+    const MediaConfig& config,
+    const AudioOptions& options,
+    const webrtc::CryptoOptions& crypto_options,
+    webrtc::AudioCodecPairId codec_pair_id) {
+  std::unique_ptr<FakeVoiceMediaReceiveChannel> ch =
+      std::make_unique<FakeVoiceMediaReceiveChannel>(options,
+                                                     call->network_thread());
+  return ch;
+}
+const std::vector<AudioCodec>& FakeVoiceEngine::send_codecs() const {
+  return send_codecs_;
+}
+const std::vector<AudioCodec>& FakeVoiceEngine::recv_codecs() const {
+  return recv_codecs_;
+}
+void FakeVoiceEngine::SetCodecs(const std::vector<AudioCodec>& codecs) {
+  send_codecs_ = codecs;
+  recv_codecs_ = codecs;
+}
+void FakeVoiceEngine::SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
+  recv_codecs_ = codecs;
+}
+void FakeVoiceEngine::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
+  send_codecs_ = codecs;
+}
+int FakeVoiceEngine::GetInputLevel() {
+  return 0;
+}
+bool FakeVoiceEngine::StartAecDump(webrtc::FileWrapper file,
+                                   int64_t max_size_bytes) {
+  return false;
+}
+absl::optional<webrtc::AudioDeviceModule::Stats>
+FakeVoiceEngine::GetAudioDeviceStats() {
+  return absl::nullopt;
+}
+void FakeVoiceEngine::StopAecDump() {}
+
+std::vector<webrtc::RtpHeaderExtensionCapability>
+FakeVoiceEngine::GetRtpHeaderExtensions() const {
+  return header_extensions_;
+}
+
+void FakeVoiceEngine::SetRtpHeaderExtensions(
+    std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions) {
+  header_extensions_ = std::move(header_extensions);
+}
+
+FakeVideoEngine::FakeVideoEngine()
+    : capture_(false), fail_create_channel_(false) {
+  // Add a fake video codec. Note that the name must not be "" as there are
+  // sanity checks against that.
+  send_codecs_.push_back(cricket::CreateVideoCodec(111, "fake_video_codec"));
+  recv_codecs_.push_back(cricket::CreateVideoCodec(111, "fake_video_codec"));
+}
+bool FakeVideoEngine::SetOptions(const VideoOptions& options) {
+  options_ = options;
+  return true;
+}
+std::unique_ptr<VideoMediaSendChannelInterface>
+FakeVideoEngine::CreateSendChannel(
+    webrtc::Call* call,
+    const MediaConfig& config,
+    const VideoOptions& options,
+    const webrtc::CryptoOptions& crypto_options,
+    webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
+  if (fail_create_channel_) {
+    return nullptr;
+  }
+
+  std::unique_ptr<FakeVideoMediaSendChannel> ch =
+      std::make_unique<FakeVideoMediaSendChannel>(options,
+                                                  call->network_thread());
+  return ch;
+}
+std::unique_ptr<VideoMediaReceiveChannelInterface>
+FakeVideoEngine::CreateReceiveChannel(
+    webrtc::Call* call,
+    const MediaConfig& config,
+    const VideoOptions& options,
+    const webrtc::CryptoOptions& crypto_options) {
+  if (fail_create_channel_) {
+    return nullptr;
+  }
+
+  std::unique_ptr<FakeVideoMediaReceiveChannel> ch =
+      std::make_unique<FakeVideoMediaReceiveChannel>(options,
+                                                     call->network_thread());
+  return ch;
+}
+std::vector<VideoCodec> FakeVideoEngine::send_codecs(bool use_rtx) const {
+  return send_codecs_;
+}
+
+std::vector<VideoCodec> FakeVideoEngine::recv_codecs(bool use_rtx) const {
+  return recv_codecs_;
+}
+
+void FakeVideoEngine::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
+  send_codecs_ = codecs;
+}
+
+void FakeVideoEngine::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
+  recv_codecs_ = codecs;
+}
+
+bool FakeVideoEngine::SetCapture(bool capture) {
+  capture_ = capture;
+  return true;
+}
+std::vector<webrtc::RtpHeaderExtensionCapability>
+FakeVideoEngine::GetRtpHeaderExtensions() const {
+  return header_extensions_;
+}
+void FakeVideoEngine::SetRtpHeaderExtensions(
+    std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions) {
+  header_extensions_ = std::move(header_extensions);
+}
+
+FakeMediaEngine::FakeMediaEngine()
+    : CompositeMediaEngine(std::make_unique<FakeVoiceEngine>(),
+                           std::make_unique<FakeVideoEngine>()),
+      voice_(static_cast<FakeVoiceEngine*>(&voice())),
+      video_(static_cast<FakeVideoEngine*>(&video())) {}
+FakeMediaEngine::~FakeMediaEngine() {}
+void FakeMediaEngine::SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
+  voice_->SetCodecs(codecs);
+}
+void FakeMediaEngine::SetAudioRecvCodecs(
+    const std::vector<AudioCodec>& codecs) {
+  voice_->SetRecvCodecs(codecs);
+}
+void FakeMediaEngine::SetAudioSendCodecs(
+    const std::vector<AudioCodec>& codecs) {
+  voice_->SetSendCodecs(codecs);
+}
+void FakeMediaEngine::SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
+  video_->SetSendCodecs(codecs);
+  video_->SetRecvCodecs(codecs);
+}
+void FakeMediaEngine::set_fail_create_channel(bool fail) {
+  voice_->fail_create_channel_ = fail;
+  video_->fail_create_channel_ = fail;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/fake_media_engine.h

@@ -0,0 +1,876 @@
+/*
+ *  Copyright (c) 2004 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 MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
+#define MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
+
+#include <atomic>
+#include <list>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/functional/any_invocable.h"
+#include "api/call/audio_sink.h"
+#include "api/media_types.h"
+#include "media/base/audio_source.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_engine.h"
+#include "media/base/rtp_utils.h"
+#include "media/base/stream_params.h"
+#include "media/engine/webrtc_video_engine.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/thread.h"
+
+using webrtc::RtpExtension;
+
+namespace cricket {
+
+class FakeMediaEngine;
+class FakeVideoEngine;
+class FakeVoiceEngine;
+
+// A common helper class that handles sending and receiving RTP/RTCP packets.
+template <class Base>
+class RtpReceiveChannelHelper : public Base, public MediaChannelUtil {
+ public:
+  explicit RtpReceiveChannelHelper(webrtc::TaskQueueBase* network_thread)
+      : MediaChannelUtil(network_thread),
+        playout_(false),
+        fail_set_recv_codecs_(false),
+        transport_overhead_per_packet_(0),
+        num_network_route_changes_(0) {}
+  virtual ~RtpReceiveChannelHelper() = default;
+  const std::vector<RtpExtension>& recv_extensions() {
+    return recv_extensions_;
+  }
+  bool playout() const { return playout_; }
+  const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
+  const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
+
+  bool SendRtcp(const void* data, size_t len) {
+    rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
+                                  kMaxRtpPacketLen);
+    return Base::SendRtcp(&packet, rtc::PacketOptions());
+  }
+
+  bool CheckRtp(const void* data, size_t len) {
+    bool success = !rtp_packets_.empty();
+    if (success) {
+      std::string packet = rtp_packets_.front();
+      rtp_packets_.pop_front();
+      success = (packet == std::string(static_cast<const char*>(data), len));
+    }
+    return success;
+  }
+  bool CheckRtcp(const void* data, size_t len) {
+    bool success = !rtcp_packets_.empty();
+    if (success) {
+      std::string packet = rtcp_packets_.front();
+      rtcp_packets_.pop_front();
+      success = (packet == std::string(static_cast<const char*>(data), len));
+    }
+    return success;
+  }
+  bool CheckNoRtp() { return rtp_packets_.empty(); }
+  bool CheckNoRtcp() { return rtcp_packets_.empty(); }
+  void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
+  void ResetUnsignaledRecvStream() override {}
+  absl::optional<uint32_t> GetUnsignaledSsrc() const override {
+    return absl::nullopt;
+  }
+  void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {}
+
+  virtual bool SetLocalSsrc(const StreamParams& sp) { return true; }
+  void OnDemuxerCriteriaUpdatePending() override {}
+  void OnDemuxerCriteriaUpdateComplete() override {}
+
+  bool AddRecvStream(const StreamParams& sp) override {
+    if (absl::c_linear_search(receive_streams_, sp)) {
+      return false;
+    }
+    receive_streams_.push_back(sp);
+    rtp_receive_parameters_[sp.first_ssrc()] =
+        CreateRtpParametersWithEncodings(sp);
+    return true;
+  }
+  bool RemoveRecvStream(uint32_t ssrc) override {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
+      rtp_receive_parameters_.erase(parameters_iterator);
+    }
+    return RemoveStreamBySsrc(&receive_streams_, ssrc);
+  }
+
+  webrtc::RtpParameters GetRtpReceiverParameters(uint32_t ssrc) const override {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
+      return parameters_iterator->second;
+    }
+    return webrtc::RtpParameters();
+  }
+  webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override {
+    return webrtc::RtpParameters();
+  }
+
+  const std::vector<StreamParams>& recv_streams() const {
+    return receive_streams_;
+  }
+  bool HasRecvStream(uint32_t ssrc) const {
+    return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
+  }
+
+  const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; }
+
+  int transport_overhead_per_packet() const {
+    return transport_overhead_per_packet_;
+  }
+
+  rtc::NetworkRoute last_network_route() const { return last_network_route_; }
+  int num_network_route_changes() const { return num_network_route_changes_; }
+  void set_num_network_route_changes(int changes) {
+    num_network_route_changes_ = changes;
+  }
+
+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
+                            int64_t packet_time_us) {
+    rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
+  }
+
+  void SetFrameDecryptor(uint32_t ssrc,
+                         rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override {}
+
+  void SetDepacketizerToDecoderFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override {}
+
+  void SetInterface(MediaChannelNetworkInterface* iface) override {
+    network_interface_ = iface;
+    MediaChannelUtil::SetInterface(iface);
+  }
+
+ protected:
+  void set_playout(bool playout) { playout_ = playout; }
+  bool SetRecvRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
+    recv_extensions_ = extensions;
+    return true;
+  }
+  void set_recv_rtcp_parameters(const RtcpParameters& params) {
+    recv_rtcp_parameters_ = params;
+  }
+  void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override {
+    rtp_packets_.push_back(
+        std::string(packet.Buffer().cdata<char>(), packet.size()));
+  }
+  bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
+
+ private:
+  bool playout_;
+  std::vector<RtpExtension> recv_extensions_;
+  std::list<std::string> rtp_packets_;
+  std::list<std::string> rtcp_packets_;
+  std::vector<StreamParams> receive_streams_;
+  RtcpParameters recv_rtcp_parameters_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
+  bool fail_set_recv_codecs_;
+  std::string rtcp_cname_;
+  int transport_overhead_per_packet_;
+  rtc::NetworkRoute last_network_route_;
+  int num_network_route_changes_;
+  MediaChannelNetworkInterface* network_interface_ = nullptr;
+};
+
+// A common helper class that handles sending and receiving RTP/RTCP packets.
+template <class Base>
+class RtpSendChannelHelper : public Base, public MediaChannelUtil {
+ public:
+  explicit RtpSendChannelHelper(webrtc::TaskQueueBase* network_thread)
+      : MediaChannelUtil(network_thread),
+        sending_(false),
+        fail_set_send_codecs_(false),
+        send_ssrc_(0),
+        ready_to_send_(false),
+        transport_overhead_per_packet_(0),
+        num_network_route_changes_(0) {}
+  virtual ~RtpSendChannelHelper() = default;
+  const std::vector<RtpExtension>& send_extensions() {
+    return send_extensions_;
+  }
+  bool sending() const { return sending_; }
+  const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
+  const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
+
+  bool SendPacket(const void* data,
+                  size_t len,
+                  const rtc::PacketOptions& options) {
+    if (!sending_) {
+      return false;
+    }
+    rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
+                                  kMaxRtpPacketLen);
+    return MediaChannelUtil::SendPacket(&packet, options);
+  }
+  bool SendRtcp(const void* data, size_t len) {
+    rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
+                                  kMaxRtpPacketLen);
+    return MediaChannelUtil::SendRtcp(&packet, rtc::PacketOptions());
+  }
+
+  bool CheckRtp(const void* data, size_t len) {
+    bool success = !rtp_packets_.empty();
+    if (success) {
+      std::string packet = rtp_packets_.front();
+      rtp_packets_.pop_front();
+      success = (packet == std::string(static_cast<const char*>(data), len));
+    }
+    return success;
+  }
+  bool CheckRtcp(const void* data, size_t len) {
+    bool success = !rtcp_packets_.empty();
+    if (success) {
+      std::string packet = rtcp_packets_.front();
+      rtcp_packets_.pop_front();
+      success = (packet == std::string(static_cast<const char*>(data), len));
+    }
+    return success;
+  }
+  bool CheckNoRtp() { return rtp_packets_.empty(); }
+  bool CheckNoRtcp() { return rtcp_packets_.empty(); }
+  void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
+  bool AddSendStream(const StreamParams& sp) override {
+    if (absl::c_linear_search(send_streams_, sp)) {
+      return false;
+    }
+    send_streams_.push_back(sp);
+    rtp_send_parameters_[sp.first_ssrc()] =
+        CreateRtpParametersWithEncodings(sp);
+
+    if (ssrc_list_changed_callback_) {
+      std::set<uint32_t> ssrcs_in_use;
+      for (const auto& send_stream : send_streams_) {
+        ssrcs_in_use.insert(send_stream.first_ssrc());
+      }
+      ssrc_list_changed_callback_(ssrcs_in_use);
+    }
+
+    return true;
+  }
+  bool RemoveSendStream(uint32_t ssrc) override {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      rtp_send_parameters_.erase(parameters_iterator);
+    }
+    return RemoveStreamBySsrc(&send_streams_, ssrc);
+  }
+  void SetSsrcListChangedCallback(
+      absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
+    ssrc_list_changed_callback_ = std::move(callback);
+  }
+
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
+    return MediaChannelUtil::SetExtmapAllowMixed(extmap_allow_mixed);
+  }
+  bool ExtmapAllowMixed() const override {
+    return MediaChannelUtil::ExtmapAllowMixed();
+  }
+
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      return parameters_iterator->second;
+    }
+    return webrtc::RtpParameters();
+  }
+  webrtc::RTCError SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters,
+      webrtc::SetParametersCallback callback) override {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      auto result = CheckRtpParametersInvalidModificationAndValues(
+          parameters_iterator->second, parameters);
+      if (!result.ok()) {
+        return webrtc::InvokeSetParametersCallback(callback, result);
+      }
+
+      parameters_iterator->second = parameters;
+
+      return webrtc::InvokeSetParametersCallback(callback,
+                                                 webrtc::RTCError::OK());
+    }
+    // Replicate the behavior of the real media channel: return false
+    // when setting parameters for unknown SSRCs.
+    return InvokeSetParametersCallback(
+        callback, webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR));
+  }
+
+  bool IsStreamMuted(uint32_t ssrc) const {
+    bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
+    // If |ssrc = 0| check if the first send stream is muted.
+    if (!ret && ssrc == 0 && !send_streams_.empty()) {
+      return muted_streams_.find(send_streams_[0].first_ssrc()) !=
+             muted_streams_.end();
+    }
+    return ret;
+  }
+  const std::vector<StreamParams>& send_streams() const {
+    return send_streams_;
+  }
+  bool HasSendStream(uint32_t ssrc) const {
+    return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
+  }
+  // TODO(perkj): This is to support legacy unit test that only check one
+  // sending stream.
+  uint32_t send_ssrc() const {
+    if (send_streams_.empty())
+      return 0;
+    return send_streams_[0].first_ssrc();
+  }
+
+  const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; }
+
+  bool ready_to_send() const { return ready_to_send_; }
+
+  int transport_overhead_per_packet() const {
+    return transport_overhead_per_packet_;
+  }
+
+  rtc::NetworkRoute last_network_route() const { return last_network_route_; }
+  int num_network_route_changes() const { return num_network_route_changes_; }
+  void set_num_network_route_changes(int changes) {
+    num_network_route_changes_ = changes;
+  }
+
+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
+                            int64_t packet_time_us) {
+    rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
+  }
+
+  // Stuff that deals with encryptors, transformers and the like
+  void SetFrameEncryptor(uint32_t ssrc,
+                         rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
+                             frame_encryptor) override {}
+  void SetEncoderToPacketizerFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override {}
+
+  void SetInterface(MediaChannelNetworkInterface* iface) override {
+    network_interface_ = iface;
+    MediaChannelUtil::SetInterface(iface);
+  }
+  bool HasNetworkInterface() const override {
+    return network_interface_ != nullptr;
+  }
+
+ protected:
+  bool MuteStream(uint32_t ssrc, bool mute) {
+    if (!HasSendStream(ssrc) && ssrc != 0) {
+      return false;
+    }
+    if (mute) {
+      muted_streams_.insert(ssrc);
+    } else {
+      muted_streams_.erase(ssrc);
+    }
+    return true;
+  }
+  bool set_sending(bool send) {
+    sending_ = send;
+    return true;
+  }
+  bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
+    send_extensions_ = extensions;
+    return true;
+  }
+  void set_send_rtcp_parameters(const RtcpParameters& params) {
+    send_rtcp_parameters_ = params;
+  }
+  void OnPacketSent(const rtc::SentPacket& sent_packet) override {}
+  void OnReadyToSend(bool ready) override { ready_to_send_ = ready; }
+  void OnNetworkRouteChanged(absl::string_view transport_name,
+                             const rtc::NetworkRoute& network_route) override {
+    last_network_route_ = network_route;
+    ++num_network_route_changes_;
+    transport_overhead_per_packet_ = network_route.packet_overhead;
+  }
+  bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
+
+ private:
+  // TODO(bugs.webrtc.org/12783): This flag is used from more than one thread.
+  // As a workaround for tsan, it's currently std::atomic but that might not
+  // be the appropriate fix.
+  std::atomic<bool> sending_;
+  std::vector<RtpExtension> send_extensions_;
+  std::list<std::string> rtp_packets_;
+  std::list<std::string> rtcp_packets_;
+  std::vector<StreamParams> send_streams_;
+  RtcpParameters send_rtcp_parameters_;
+  std::set<uint32_t> muted_streams_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
+  bool fail_set_send_codecs_;
+  uint32_t send_ssrc_;
+  std::string rtcp_cname_;
+  bool ready_to_send_;
+  int transport_overhead_per_packet_;
+  rtc::NetworkRoute last_network_route_;
+  int num_network_route_changes_;
+  MediaChannelNetworkInterface* network_interface_ = nullptr;
+  absl::AnyInvocable<void(const std::set<uint32_t>&)>
+      ssrc_list_changed_callback_ = nullptr;
+};
+
+class FakeVoiceMediaReceiveChannel
+    : public RtpReceiveChannelHelper<VoiceMediaReceiveChannelInterface> {
+ public:
+  struct DtmfInfo {
+    DtmfInfo(uint32_t ssrc, int event_code, int duration);
+    uint32_t ssrc;
+    int event_code;
+    int duration;
+  };
+  FakeVoiceMediaReceiveChannel(const AudioOptions& options,
+                               webrtc::TaskQueueBase* network_thread);
+  virtual ~FakeVoiceMediaReceiveChannel();
+
+  // Test methods
+  const std::vector<AudioCodec>& recv_codecs() const;
+  const std::vector<DtmfInfo>& dtmf_info_queue() const;
+  const AudioOptions& options() const;
+  int max_bps() const;
+  bool HasSource(uint32_t ssrc) const;
+
+  // Overrides
+  VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
+    return nullptr;
+  }
+  VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
+    return this;
+  }
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_AUDIO;
+  }
+
+  bool SetReceiverParameters(const AudioReceiverParameters& params) override;
+  void SetPlayout(bool playout) override;
+
+  bool AddRecvStream(const StreamParams& sp) override;
+  bool RemoveRecvStream(uint32_t ssrc) override;
+
+  bool SetOutputVolume(uint32_t ssrc, double volume) override;
+  bool SetDefaultOutputVolume(double volume) override;
+
+  bool GetOutputVolume(uint32_t ssrc, double* volume);
+
+  bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
+  absl::optional<int> GetBaseMinimumPlayoutDelayMs(
+      uint32_t ssrc) const override;
+
+  bool GetStats(VoiceMediaReceiveInfo* info,
+                bool get_and_clear_legacy_stats) override;
+
+  void SetRawAudioSink(
+      uint32_t ssrc,
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+  void SetDefaultRawAudioSink(
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+
+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
+  void SetReceiveNackEnabled(bool enabled) override {}
+  void SetReceiveNonSenderRttEnabled(bool enabled) override {}
+
+ private:
+  class VoiceChannelAudioSink : public AudioSource::Sink {
+   public:
+    explicit VoiceChannelAudioSink(AudioSource* source);
+    ~VoiceChannelAudioSink() override;
+    void OnData(const void* audio_data,
+                int bits_per_sample,
+                int sample_rate,
+                size_t number_of_channels,
+                size_t number_of_frames,
+                absl::optional<int64_t> absolute_capture_timestamp_ms) override;
+    void OnClose() override;
+    int NumPreferredChannels() const override { return -1; }
+    AudioSource* source() const;
+
+   private:
+    AudioSource* source_;
+  };
+
+  bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
+  bool SetMaxSendBandwidth(int bps);
+  bool SetOptions(const AudioOptions& options);
+
+  std::vector<AudioCodec> recv_codecs_;
+  std::map<uint32_t, double> output_scalings_;
+  std::map<uint32_t, int> output_delays_;
+  std::vector<DtmfInfo> dtmf_info_queue_;
+  AudioOptions options_;
+  std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
+  std::unique_ptr<webrtc::AudioSinkInterface> sink_;
+  int max_bps_;
+};
+
+class FakeVoiceMediaSendChannel
+    : public RtpSendChannelHelper<VoiceMediaSendChannelInterface> {
+ public:
+  struct DtmfInfo {
+    DtmfInfo(uint32_t ssrc, int event_code, int duration);
+    uint32_t ssrc;
+    int event_code;
+    int duration;
+  };
+  FakeVoiceMediaSendChannel(const AudioOptions& options,
+                            webrtc::TaskQueueBase* network_thread);
+  ~FakeVoiceMediaSendChannel() override;
+
+  const std::vector<AudioCodec>& send_codecs() const;
+  const std::vector<DtmfInfo>& dtmf_info_queue() const;
+  const AudioOptions& options() const;
+  int max_bps() const;
+  bool HasSource(uint32_t ssrc) const;
+  bool GetOutputVolume(uint32_t ssrc, double* volume);
+
+  // Overrides
+  VideoMediaSendChannelInterface* AsVideoSendChannel() override {
+    return nullptr;
+  }
+  VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_AUDIO;
+  }
+
+  bool SetSenderParameters(const AudioSenderParameter& params) override;
+  void SetSend(bool send) override;
+  bool SetAudioSend(uint32_t ssrc,
+                    bool enable,
+                    const AudioOptions* options,
+                    AudioSource* source) override;
+
+  bool CanInsertDtmf() override;
+  bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
+
+  bool SenderNackEnabled() const override { return false; }
+  bool SenderNonSenderRttEnabled() const override { return false; }
+  void SetReceiveNackEnabled(bool enabled) {}
+  void SetReceiveNonSenderRttEnabled(bool enabled) {}
+  bool SendCodecHasNack() const override { return false; }
+  void SetSendCodecChangedCallback(
+      absl::AnyInvocable<void()> callback) override {}
+  absl::optional<Codec> GetSendCodec() const override;
+
+  bool GetStats(VoiceMediaSendInfo* stats) override;
+
+ private:
+  class VoiceChannelAudioSink : public AudioSource::Sink {
+   public:
+    explicit VoiceChannelAudioSink(AudioSource* source);
+    ~VoiceChannelAudioSink() override;
+    void OnData(const void* audio_data,
+                int bits_per_sample,
+                int sample_rate,
+                size_t number_of_channels,
+                size_t number_of_frames,
+                absl::optional<int64_t> absolute_capture_timestamp_ms) override;
+    void OnClose() override;
+    int NumPreferredChannels() const override { return -1; }
+    AudioSource* source() const;
+
+   private:
+    AudioSource* source_;
+  };
+
+  bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
+  bool SetMaxSendBandwidth(int bps);
+  bool SetOptions(const AudioOptions& options);
+  bool SetLocalSource(uint32_t ssrc, AudioSource* source);
+
+  std::vector<AudioCodec> send_codecs_;
+  std::map<uint32_t, double> output_scalings_;
+  std::map<uint32_t, int> output_delays_;
+  std::vector<DtmfInfo> dtmf_info_queue_;
+  AudioOptions options_;
+  std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
+  int max_bps_;
+};
+
+// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
+bool CompareDtmfInfo(const FakeVoiceMediaSendChannel::DtmfInfo& info,
+                     uint32_t ssrc,
+                     int event_code,
+                     int duration);
+
+class FakeVideoMediaReceiveChannel
+    : public RtpReceiveChannelHelper<VideoMediaReceiveChannelInterface> {
+ public:
+  FakeVideoMediaReceiveChannel(const VideoOptions& options,
+                               webrtc::TaskQueueBase* network_thread);
+
+  virtual ~FakeVideoMediaReceiveChannel();
+
+  VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
+    return this;
+  }
+  VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
+    return nullptr;
+  }
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_VIDEO;
+  }
+
+  const std::vector<VideoCodec>& recv_codecs() const;
+  const std::vector<VideoCodec>& send_codecs() const;
+  bool rendering() const;
+  const VideoOptions& options() const;
+  const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
+  sinks() const;
+  int max_bps() const;
+  bool SetReceiverParameters(const VideoReceiverParameters& params) override;
+
+  bool SetSink(uint32_t ssrc,
+               rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+  void SetDefaultSink(
+      rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+  bool HasSink(uint32_t ssrc) const;
+
+  void SetReceive(bool receive) override {}
+
+  bool HasSource(uint32_t ssrc) const;
+  bool AddRecvStream(const StreamParams& sp) override;
+  bool RemoveRecvStream(uint32_t ssrc) override;
+
+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
+
+  bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
+  absl::optional<int> GetBaseMinimumPlayoutDelayMs(
+      uint32_t ssrc) const override;
+
+  void SetRecordableEncodedFrameCallback(
+      uint32_t ssrc,
+      std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
+      override;
+  void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
+  void RequestRecvKeyFrame(uint32_t ssrc) override;
+  void SetReceiverFeedbackParameters(bool lntf_enabled,
+                                     bool nack_enabled,
+                                     webrtc::RtcpMode rtcp_mode,
+                                     absl::optional<int> rtx_time) override {}
+  bool GetStats(VideoMediaReceiveInfo* info) override;
+
+  bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
+    RTC_CHECK_NOTREACHED();
+    return false;
+  }
+
+ private:
+  bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
+  bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
+  bool SetOptions(const VideoOptions& options);
+  bool SetMaxSendBandwidth(int bps);
+
+  std::vector<VideoCodec> recv_codecs_;
+  std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*> sinks_;
+  std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
+  std::map<uint32_t, int> output_delays_;
+  VideoOptions options_;
+  int max_bps_;
+};
+
+class FakeVideoMediaSendChannel
+    : public RtpSendChannelHelper<VideoMediaSendChannelInterface> {
+ public:
+  FakeVideoMediaSendChannel(const VideoOptions& options,
+                            webrtc::TaskQueueBase* network_thread);
+
+  virtual ~FakeVideoMediaSendChannel();
+
+  VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
+  VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
+    return nullptr;
+  }
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_VIDEO;
+  }
+
+  const std::vector<VideoCodec>& send_codecs() const;
+  const std::vector<VideoCodec>& codecs() const;
+  const VideoOptions& options() const;
+  const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
+  sinks() const;
+  int max_bps() const;
+  bool SetSenderParameters(const VideoSenderParameters& params) override;
+
+  absl::optional<Codec> GetSendCodec() const override;
+
+  bool SetSend(bool send) override;
+  bool SetVideoSend(
+      uint32_t ssrc,
+      const VideoOptions* options,
+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
+
+  bool HasSource(uint32_t ssrc) const;
+
+  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
+
+  void GenerateSendKeyFrame(uint32_t ssrc,
+                            const std::vector<std::string>& rids) override;
+  webrtc::RtcpMode SendCodecRtcpMode() const override {
+    return webrtc::RtcpMode::kCompound;
+  }
+  void SetSendCodecChangedCallback(
+      absl::AnyInvocable<void()> callback) override {}
+  void SetSsrcListChangedCallback(
+      absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {}
+
+  bool SendCodecHasLntf() const override { return false; }
+  bool SendCodecHasNack() const override { return false; }
+  absl::optional<int> SendCodecRtxTime() const override {
+    return absl::nullopt;
+  }
+  bool GetStats(VideoMediaSendInfo* info) override;
+
+ private:
+  bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
+  bool SetOptions(const VideoOptions& options);
+  bool SetMaxSendBandwidth(int bps);
+
+  std::vector<VideoCodec> send_codecs_;
+  std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
+  VideoOptions options_;
+  int max_bps_;
+};
+
+class FakeVoiceEngine : public VoiceEngineInterface {
+ public:
+  FakeVoiceEngine();
+  void Init() override;
+  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
+
+  std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) override;
+  std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) override;
+
+  // TODO(ossu): For proper testing, These should either individually settable
+  //             or the voice engine should reference mockable factories.
+  const std::vector<AudioCodec>& send_codecs() const override;
+  const std::vector<AudioCodec>& recv_codecs() const override;
+  void SetCodecs(const std::vector<AudioCodec>& codecs);
+  void SetRecvCodecs(const std::vector<AudioCodec>& codecs);
+  void SetSendCodecs(const std::vector<AudioCodec>& codecs);
+  int GetInputLevel();
+  bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
+  void StopAecDump() override;
+  absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
+      override;
+  std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
+      const override;
+  void SetRtpHeaderExtensions(
+      std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
+
+ private:
+  std::vector<AudioCodec> recv_codecs_;
+  std::vector<AudioCodec> send_codecs_;
+  bool fail_create_channel_;
+  std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
+
+  friend class FakeMediaEngine;
+};
+
+class FakeVideoEngine : public VideoEngineInterface {
+ public:
+  FakeVideoEngine();
+  bool SetOptions(const VideoOptions& options);
+  std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
+      override;
+  std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options) override;
+  FakeVideoMediaSendChannel* GetSendChannel(size_t index);
+  FakeVideoMediaReceiveChannel* GetReceiveChannel(size_t index);
+
+  std::vector<VideoCodec> send_codecs() const override {
+    return send_codecs(true);
+  }
+  std::vector<VideoCodec> recv_codecs() const override {
+    return recv_codecs(true);
+  }
+  std::vector<VideoCodec> send_codecs(bool include_rtx) const override;
+  std::vector<VideoCodec> recv_codecs(bool include_rtx) const override;
+  void SetSendCodecs(const std::vector<VideoCodec>& codecs);
+  void SetRecvCodecs(const std::vector<VideoCodec>& codecs);
+  bool SetCapture(bool capture);
+  std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
+      const override;
+  void SetRtpHeaderExtensions(
+      std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
+
+ private:
+  std::vector<VideoCodec> send_codecs_;
+  std::vector<VideoCodec> recv_codecs_;
+  bool capture_;
+  VideoOptions options_;
+  bool fail_create_channel_;
+  std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
+
+  friend class FakeMediaEngine;
+};
+
+class FakeMediaEngine : public CompositeMediaEngine {
+ public:
+  FakeMediaEngine();
+
+  ~FakeMediaEngine() override;
+
+  void SetAudioCodecs(const std::vector<AudioCodec>& codecs);
+  void SetAudioRecvCodecs(const std::vector<AudioCodec>& codecs);
+  void SetAudioSendCodecs(const std::vector<AudioCodec>& codecs);
+  void SetVideoCodecs(const std::vector<VideoCodec>& codecs);
+
+  void set_fail_create_channel(bool fail);
+
+  FakeVoiceEngine* fake_voice_engine() { return voice_; }
+  FakeVideoEngine* fake_video_engine() { return video_; }
+
+ private:
+  FakeVoiceEngine* const voice_;
+  FakeVideoEngine* const video_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_FAKE_MEDIA_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/base/fake_network_interface.h

@@ -0,0 +1,232 @@
+/*
+ *  Copyright (c) 2004 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 MEDIA_BASE_FAKE_NETWORK_INTERFACE_H_
+#define MEDIA_BASE_FAKE_NETWORK_INTERFACE_H_
+
+#include <map>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "media/base/media_channel.h"
+#include "media/base/rtp_utils.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/byte_order.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/dscp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/time_utils.h"
+
+namespace cricket {
+
+// Fake NetworkInterface that sends/receives RTP/RTCP packets.
+class FakeNetworkInterface : public MediaChannelNetworkInterface {
+ public:
+  FakeNetworkInterface()
+      : thread_(rtc::Thread::Current()),
+        dest_(NULL),
+        conf_(false),
+        sendbuf_size_(-1),
+        recvbuf_size_(-1),
+        dscp_(rtc::DSCP_NO_CHANGE) {}
+
+  void SetDestination(MediaReceiveChannelInterface* dest) { dest_ = dest; }
+
+  // Conference mode is a mode where instead of simply forwarding the packets,
+  // the transport will send multiple copies of the packet with the specified
+  // SSRCs. This allows us to simulate receiving media from multiple sources.
+  void SetConferenceMode(bool conf, const std::vector<uint32_t>& ssrcs)
+      RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    conf_ = conf;
+    conf_sent_ssrcs_ = ssrcs;
+  }
+
+  int NumRtpBytes() RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    int bytes = 0;
+    for (size_t i = 0; i < rtp_packets_.size(); ++i) {
+      bytes += static_cast<int>(rtp_packets_[i].size());
+    }
+    return bytes;
+  }
+
+  int NumRtpBytes(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    int bytes = 0;
+    GetNumRtpBytesAndPackets(ssrc, &bytes, NULL);
+    return bytes;
+  }
+
+  int NumRtpPackets() RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    return static_cast<int>(rtp_packets_.size());
+  }
+
+  int NumRtpPackets(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    int packets = 0;
+    GetNumRtpBytesAndPackets(ssrc, NULL, &packets);
+    return packets;
+  }
+
+  int NumSentSsrcs() RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    return static_cast<int>(sent_ssrcs_.size());
+  }
+
+  rtc::CopyOnWriteBuffer GetRtpPacket(int index) RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    if (index >= static_cast<int>(rtp_packets_.size())) {
+      return {};
+    }
+    return rtp_packets_[index];
+  }
+
+  int NumRtcpPackets() RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    return static_cast<int>(rtcp_packets_.size());
+  }
+
+  // Note: callers are responsible for deleting the returned buffer.
+  const rtc::CopyOnWriteBuffer* GetRtcpPacket(int index)
+      RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    if (index >= static_cast<int>(rtcp_packets_.size())) {
+      return NULL;
+    }
+    return new rtc::CopyOnWriteBuffer(rtcp_packets_[index]);
+  }
+
+  int sendbuf_size() const { return sendbuf_size_; }
+  int recvbuf_size() const { return recvbuf_size_; }
+  rtc::DiffServCodePoint dscp() const { return dscp_; }
+  rtc::PacketOptions options() const { return options_; }
+
+ protected:
+  virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
+                          const rtc::PacketOptions& options)
+      RTC_LOCKS_EXCLUDED(mutex_) {
+    if (!webrtc::IsRtpPacket(*packet)) {
+      return false;
+    }
+
+    webrtc::MutexLock lock(&mutex_);
+    sent_ssrcs_[webrtc::ParseRtpSsrc(*packet)]++;
+    options_ = options;
+
+    rtp_packets_.push_back(*packet);
+    if (conf_) {
+      for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
+        SetRtpSsrc(conf_sent_ssrcs_[i], *packet);
+        PostPacket(*packet);
+      }
+    } else {
+      PostPacket(*packet);
+    }
+    return true;
+  }
+
+  virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
+                        const rtc::PacketOptions& options)
+      RTC_LOCKS_EXCLUDED(mutex_) {
+    webrtc::MutexLock lock(&mutex_);
+    rtcp_packets_.push_back(*packet);
+    options_ = options;
+    if (!conf_) {
+      // don't worry about RTCP in conf mode for now
+      RTC_LOG(LS_VERBOSE) << "Dropping RTCP packet, they are not handled by "
+                             "MediaChannel anymore.";
+    }
+    return true;
+  }
+
+  virtual int SetOption(SocketType type, rtc::Socket::Option opt, int option) {
+    if (opt == rtc::Socket::OPT_SNDBUF) {
+      sendbuf_size_ = option;
+    } else if (opt == rtc::Socket::OPT_RCVBUF) {
+      recvbuf_size_ = option;
+    } else if (opt == rtc::Socket::OPT_DSCP) {
+      dscp_ = static_cast<rtc::DiffServCodePoint>(option);
+    }
+    return 0;
+  }
+
+  void PostPacket(rtc::CopyOnWriteBuffer packet) {
+    thread_->PostTask(
+        SafeTask(safety_.flag(), [this, packet = std::move(packet)]() mutable {
+          if (dest_) {
+            webrtc::RtpPacketReceived parsed_packet;
+            if (parsed_packet.Parse(packet)) {
+              parsed_packet.set_arrival_time(
+                  webrtc::Timestamp::Micros(rtc::TimeMicros()));
+              dest_->OnPacketReceived(std::move(parsed_packet));
+            } else {
+              RTC_DCHECK_NOTREACHED();
+            }
+          }
+        }));
+  }
+
+ private:
+  void SetRtpSsrc(uint32_t ssrc, rtc::CopyOnWriteBuffer& buffer) {
+    RTC_CHECK_GE(buffer.size(), 12);
+    rtc::SetBE32(buffer.MutableData() + 8, ssrc);
+  }
+
+  void GetNumRtpBytesAndPackets(uint32_t ssrc, int* bytes, int* packets) {
+    if (bytes) {
+      *bytes = 0;
+    }
+    if (packets) {
+      *packets = 0;
+    }
+    for (size_t i = 0; i < rtp_packets_.size(); ++i) {
+      if (ssrc == webrtc::ParseRtpSsrc(rtp_packets_[i])) {
+        if (bytes) {
+          *bytes += static_cast<int>(rtp_packets_[i].size());
+        }
+        if (packets) {
+          ++(*packets);
+        }
+      }
+    }
+  }
+
+  webrtc::TaskQueueBase* thread_;
+  MediaReceiveChannelInterface* dest_;
+  bool conf_;
+  // The ssrcs used in sending out packets in conference mode.
+  std::vector<uint32_t> conf_sent_ssrcs_;
+  // Map to track counts of packets that have been sent per ssrc.
+  // This includes packets that are dropped.
+  std::map<uint32_t, uint32_t> sent_ssrcs_;
+  // Map to track packet-number that needs to be dropped per ssrc.
+  std::map<uint32_t, std::set<uint32_t> > drop_map_;
+  webrtc::Mutex mutex_;
+  std::vector<rtc::CopyOnWriteBuffer> rtp_packets_;
+  std::vector<rtc::CopyOnWriteBuffer> rtcp_packets_;
+  int sendbuf_size_;
+  int recvbuf_size_;
+  rtc::DiffServCodePoint dscp_;
+  // Options of the most recently sent packet.
+  rtc::PacketOptions options_;
+  webrtc::ScopedTaskSafety safety_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_FAKE_NETWORK_INTERFACE_H_

TMessagesProj/jni/voip/webrtc/media/base/fake_rtp.cc

@@ -0,0 +1,71 @@
+/*
+ *  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 "media/base/fake_rtp.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include "absl/algorithm/container.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+void CompareHeaderExtensions(const char* packet1,
+                             size_t packet1_size,
+                             const char* packet2,
+                             size_t packet2_size,
+                             const std::vector<int>& encrypted_headers,
+                             bool expect_equal) {
+  // Sanity check: packets must be large enough to contain the RTP header and
+  // extensions header.
+  RTC_CHECK_GE(packet1_size, 12 + 4);
+  RTC_CHECK_GE(packet2_size, 12 + 4);
+  // RTP extension headers are the same.
+  EXPECT_EQ(0, memcmp(packet1 + 12, packet2 + 12, 4));
+  // Check for one-byte header extensions.
+  EXPECT_EQ('\xBE', packet1[12]);
+  EXPECT_EQ('\xDE', packet1[13]);
+  // Determine position and size of extension headers.
+  size_t extension_words = packet1[14] << 8 | packet1[15];
+  const char* extension_data1 = packet1 + 12 + 4;
+  const char* extension_end1 = extension_data1 + extension_words * 4;
+  const char* extension_data2 = packet2 + 12 + 4;
+  // Sanity check: packets must be large enough to contain the RTP header
+  // extensions.
+  RTC_CHECK_GE(packet1_size, 12 + 4 + extension_words * 4);
+  RTC_CHECK_GE(packet2_size, 12 + 4 + extension_words * 4);
+  while (extension_data1 < extension_end1) {
+    uint8_t id = (*extension_data1 & 0xf0) >> 4;
+    uint8_t len = (*extension_data1 & 0x0f) + 1;
+    extension_data1++;
+    extension_data2++;
+    EXPECT_LE(extension_data1, extension_end1);
+    if (id == 15) {
+      // Finished parsing.
+      break;
+    }
+
+    // The header extension doesn't get encrypted if the id is not in the
+    // list of header extensions to encrypt.
+    if (expect_equal || !absl::c_linear_search(encrypted_headers, id)) {
+      EXPECT_EQ(0, memcmp(extension_data1, extension_data2, len));
+    } else {
+      EXPECT_NE(0, memcmp(extension_data1, extension_data2, len));
+    }
+
+    extension_data1 += len;
+    extension_data2 += len;
+    // Skip padding.
+    while (extension_data1 < extension_end1 && *extension_data1 == 0) {
+      extension_data1++;
+      extension_data2++;
+    }
+  }
+}

TMessagesProj/jni/voip/webrtc/media/base/fake_rtp.h

@@ -0,0 +1,301 @@
+/*
+ *  Copyright (c) 2004 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.
+ */
+
+// Fake RTP and RTCP packets to use in unit tests.
+
+#ifndef MEDIA_BASE_FAKE_RTP_H_
+#define MEDIA_BASE_FAKE_RTP_H_
+
+#include <cstddef>  // size_t
+#include <vector>
+
+// A typical PCMU RTP packet.
+// PT=0, SN=1, TS=0, SSRC=1
+// all data FF
+static const unsigned char kPcmuFrame[] = {
+    0x80, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF,
+};
+
+static const int kHeaderExtensionIDs[] = {1, 4};
+
+// A typical PCMU RTP packet with header extensions.
+// PT=0, SN=1, TS=0, SSRC=1
+// all data FF
+static const unsigned char kPcmuFrameWithExtensions[] = {
+    0x90,
+    0x00,
+    0x00,
+    0x01,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x01,
+    // RFC 5285, section 4.2. One-Byte Header.
+    0xBE,
+    0xDE,
+    // Header extension length 6 * 32 bits.
+    0x00,
+    0x06,
+    // 8 bytes header id 1.
+    0x17,
+    0x41,
+    0x42,
+    0x73,
+    0xA4,
+    0x75,
+    0x26,
+    0x27,
+    0x48,
+    // 3 bytes header id 2.
+    0x22,
+    0x00,
+    0x00,
+    0xC8,
+    // 1 byte header id 3.
+    0x30,
+    0x8E,
+    // 7 bytes header id 4.
+    0x46,
+    0x55,
+    0x99,
+    0x63,
+    0x86,
+    0xB3,
+    0x95,
+    0xFB,
+    // 1 byte header padding.
+    0x00,
+    // Payload data.
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+    0xFF,
+};
+
+// A typical Receiver Report RTCP packet.
+// PT=RR, LN=1, SSRC=1
+// send SSRC=2, all other fields 0
+static const unsigned char kRtcpReport[] = {
+    0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00,
+    0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+
+// PT = 97, TS = 0, Seq = 1, SSRC = 2
+// H264 - NRI = 1, Type = 1, bit stream = FF
+
+static const unsigned char kH264Packet[] = {
+    0x80, 0x61, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02,
+    0x21, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF,
+};
+
+// PT= 101, SN=2, TS=3, SSRC = 4
+static const unsigned char kDataPacket[] = {
+    0x80, 0x65, 0x00, 0x02, 0x00, 0x00, 0x00, 0x03, 0x00, 0x00, 0x00, 0x04,
+    0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08,
+};
+
+// This expects both packets to be based on kPcmuFrameWithExtensions.
+// Header extensions with an id in "encrypted_headers" are expected to be
+// different in the packets unless "expect_equal" is set to "true".
+void CompareHeaderExtensions(const char* packet1,
+                             size_t packet1_size,
+                             const char* packet2,
+                             size_t packet2_size,
+                             const std::vector<int>& encrypted_headers,
+                             bool expect_equal);
+
+#endif  // MEDIA_BASE_FAKE_RTP_H_

TMessagesProj/jni/voip/webrtc/media/base/fake_video_renderer.cc

@@ -0,0 +1,87 @@
+/*
+ *  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 "media/base/fake_video_renderer.h"
+
+namespace cricket {
+namespace {
+bool CheckFrameColorYuv(const webrtc::VideoFrame& frame) {
+  // TODO(zhurunz) Check with VP8 team to see if we can remove this
+  // tolerance on Y values. Some unit tests produce Y values close
+  // to 16 rather than close to zero, for supposedly black frames.
+  // Largest value observed is 34, e.g., running
+  // PeerConnectionIntegrationTest.SendAndReceive16To9AspectRatio.
+  static constexpr uint8_t y_min = 0;
+  static constexpr uint8_t y_max = 48;
+  static constexpr uint8_t u_min = 128;
+  static constexpr uint8_t u_max = 128;
+  static constexpr uint8_t v_min = 128;
+  static constexpr uint8_t v_max = 128;
+
+  if (!frame.video_frame_buffer()) {
+    return false;
+  }
+  rtc::scoped_refptr<const webrtc::I420BufferInterface> i420_buffer =
+      frame.video_frame_buffer()->ToI420();
+  // Y
+  int y_width = frame.width();
+  int y_height = frame.height();
+  const uint8_t* y_plane = i420_buffer->DataY();
+  const uint8_t* y_pos = y_plane;
+  int32_t y_pitch = i420_buffer->StrideY();
+  for (int i = 0; i < y_height; ++i) {
+    for (int j = 0; j < y_width; ++j) {
+      uint8_t y_value = *(y_pos + j);
+      if (y_value < y_min || y_value > y_max) {
+        return false;
+      }
+    }
+    y_pos += y_pitch;
+  }
+  // U and V
+  int chroma_width = i420_buffer->ChromaWidth();
+  int chroma_height = i420_buffer->ChromaHeight();
+  const uint8_t* u_plane = i420_buffer->DataU();
+  const uint8_t* v_plane = i420_buffer->DataV();
+  const uint8_t* u_pos = u_plane;
+  const uint8_t* v_pos = v_plane;
+  int32_t u_pitch = i420_buffer->StrideU();
+  int32_t v_pitch = i420_buffer->StrideV();
+  for (int i = 0; i < chroma_height; ++i) {
+    for (int j = 0; j < chroma_width; ++j) {
+      uint8_t u_value = *(u_pos + j);
+      if (u_value < u_min || u_value > u_max) {
+        return false;
+      }
+      uint8_t v_value = *(v_pos + j);
+      if (v_value < v_min || v_value > v_max) {
+        return false;
+      }
+    }
+    u_pos += u_pitch;
+    v_pos += v_pitch;
+  }
+  return true;
+}
+}  // namespace
+
+FakeVideoRenderer::FakeVideoRenderer() = default;
+
+void FakeVideoRenderer::OnFrame(const webrtc::VideoFrame& frame) {
+  webrtc::MutexLock lock(&mutex_);
+  black_frame_ = CheckFrameColorYuv(frame);
+  ++num_rendered_frames_;
+  width_ = frame.width();
+  height_ = frame.height();
+  rotation_ = frame.rotation();
+  timestamp_us_ = frame.timestamp_us();
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/fake_video_renderer.h

@@ -0,0 +1,73 @@
+/*
+ *  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 MEDIA_BASE_FAKE_VIDEO_RENDERER_H_
+#define MEDIA_BASE_FAKE_VIDEO_RENDERER_H_
+
+#include <stdint.h>
+
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_sink_interface.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace cricket {
+
+// Faked video renderer that has a callback for actions on rendering.
+class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  FakeVideoRenderer();
+
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+  int width() const {
+    webrtc::MutexLock lock(&mutex_);
+    return width_;
+  }
+  int height() const {
+    webrtc::MutexLock lock(&mutex_);
+    return height_;
+  }
+
+  webrtc::VideoRotation rotation() const {
+    webrtc::MutexLock lock(&mutex_);
+    return rotation_;
+  }
+
+  int64_t timestamp_us() const {
+    webrtc::MutexLock lock(&mutex_);
+    return timestamp_us_;
+  }
+
+  int num_rendered_frames() const {
+    webrtc::MutexLock lock(&mutex_);
+    return num_rendered_frames_;
+  }
+
+  bool black_frame() const {
+    webrtc::MutexLock lock(&mutex_);
+    return black_frame_;
+  }
+
+ private:
+  int width_ = 0;
+  int height_ = 0;
+  webrtc::VideoRotation rotation_ = webrtc::kVideoRotation_0;
+  int64_t timestamp_us_ = 0;
+  int num_rendered_frames_ = 0;
+  bool black_frame_ = false;
+  mutable webrtc::Mutex mutex_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_FAKE_VIDEO_RENDERER_H_

TMessagesProj/jni/voip/webrtc/media/base/media_channel_impl.cc

@@ -0,0 +1,303 @@
+/*
+ *  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 "media/base/media_channel_impl.h"
+
+#include <map>
+#include <string>
+#include <type_traits>
+#include <utility>
+
+#include "absl/functional/any_invocable.h"
+#include "api/audio_options.h"
+#include "api/media_stream_interface.h"
+#include "api/rtc_error.h"
+#include "api/rtp_sender_interface.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_timing.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "common_video/include/quality_limitation_reason.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/rtp_utils.h"
+#include "media/base/stream_params.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+                                             RTCError error) {
+  if (callback) {
+    std::move(callback)(error);
+    callback = nullptr;
+  }
+  return error;
+}
+
+}  // namespace webrtc
+
+namespace cricket {
+using webrtc::FrameDecryptorInterface;
+using webrtc::FrameEncryptorInterface;
+using webrtc::FrameTransformerInterface;
+using webrtc::PendingTaskSafetyFlag;
+using webrtc::SafeTask;
+using webrtc::TaskQueueBase;
+using webrtc::VideoTrackInterface;
+
+VideoOptions::VideoOptions()
+    : content_hint(VideoTrackInterface::ContentHint::kNone) {}
+VideoOptions::~VideoOptions() = default;
+
+MediaChannelUtil::MediaChannelUtil(TaskQueueBase* network_thread,
+                                   bool enable_dscp)
+    : transport_(network_thread, enable_dscp) {}
+
+MediaChannelUtil::~MediaChannelUtil() {}
+
+void MediaChannelUtil::SetInterface(MediaChannelNetworkInterface* iface) {
+  transport_.SetInterface(iface);
+}
+
+int MediaChannelUtil::GetRtpSendTimeExtnId() const {
+  return -1;
+}
+
+bool MediaChannelUtil::SendPacket(rtc::CopyOnWriteBuffer* packet,
+                                  const rtc::PacketOptions& options) {
+  return transport_.DoSendPacket(packet, false, options);
+}
+
+bool MediaChannelUtil::SendRtcp(rtc::CopyOnWriteBuffer* packet,
+                                const rtc::PacketOptions& options) {
+  return transport_.DoSendPacket(packet, true, options);
+}
+
+int MediaChannelUtil::SetOption(MediaChannelNetworkInterface::SocketType type,
+                                rtc::Socket::Option opt,
+                                int option) {
+  return transport_.SetOption(type, opt, option);
+}
+
+// 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. The setter and getter must only be called from
+// worker_thread.
+void MediaChannelUtil::SetExtmapAllowMixed(bool extmap_allow_mixed) {
+  extmap_allow_mixed_ = extmap_allow_mixed;
+}
+
+bool MediaChannelUtil::ExtmapAllowMixed() const {
+  return extmap_allow_mixed_;
+}
+
+bool MediaChannelUtil::HasNetworkInterface() const {
+  return transport_.HasNetworkInterface();
+}
+
+bool MediaChannelUtil::DscpEnabled() const {
+  return transport_.DscpEnabled();
+}
+
+void MediaChannelUtil::SetPreferredDscp(rtc::DiffServCodePoint new_dscp) {
+  transport_.SetPreferredDscp(new_dscp);
+}
+
+MediaSenderInfo::MediaSenderInfo() = default;
+MediaSenderInfo::~MediaSenderInfo() = default;
+
+MediaReceiverInfo::MediaReceiverInfo() = default;
+MediaReceiverInfo::~MediaReceiverInfo() = default;
+
+VoiceSenderInfo::VoiceSenderInfo() = default;
+VoiceSenderInfo::~VoiceSenderInfo() = default;
+
+VoiceReceiverInfo::VoiceReceiverInfo() = default;
+VoiceReceiverInfo::~VoiceReceiverInfo() = default;
+
+VideoSenderInfo::VideoSenderInfo() = default;
+VideoSenderInfo::~VideoSenderInfo() = default;
+
+VideoReceiverInfo::VideoReceiverInfo() = default;
+VideoReceiverInfo::~VideoReceiverInfo() = default;
+
+VoiceMediaInfo::VoiceMediaInfo() = default;
+VoiceMediaInfo::~VoiceMediaInfo() = default;
+
+VideoMediaInfo::VideoMediaInfo() = default;
+VideoMediaInfo::~VideoMediaInfo() = default;
+
+VideoMediaSendInfo::VideoMediaSendInfo() = default;
+VideoMediaSendInfo::~VideoMediaSendInfo() = default;
+
+VoiceMediaSendInfo::VoiceMediaSendInfo() = default;
+VoiceMediaSendInfo::~VoiceMediaSendInfo() = default;
+
+VideoMediaReceiveInfo::VideoMediaReceiveInfo() = default;
+VideoMediaReceiveInfo::~VideoMediaReceiveInfo() = default;
+
+VoiceMediaReceiveInfo::VoiceMediaReceiveInfo() = default;
+VoiceMediaReceiveInfo::~VoiceMediaReceiveInfo() = default;
+
+AudioSenderParameter::AudioSenderParameter() = default;
+AudioSenderParameter::~AudioSenderParameter() = default;
+
+std::map<std::string, std::string> AudioSenderParameter::ToStringMap() const {
+  auto params = SenderParameters::ToStringMap();
+  params["options"] = options.ToString();
+  return params;
+}
+
+VideoSenderParameters::VideoSenderParameters() = default;
+VideoSenderParameters::~VideoSenderParameters() = default;
+
+std::map<std::string, std::string> VideoSenderParameters::ToStringMap() const {
+  auto params = SenderParameters::ToStringMap();
+  params["conference_mode"] = (conference_mode ? "yes" : "no");
+  return params;
+}
+
+// --------------------- MediaChannelUtil::TransportForMediaChannels -----
+
+MediaChannelUtil::TransportForMediaChannels::TransportForMediaChannels(
+    webrtc::TaskQueueBase* network_thread,
+    bool enable_dscp)
+    : network_safety_(webrtc::PendingTaskSafetyFlag::CreateDetachedInactive()),
+      network_thread_(network_thread),
+
+      enable_dscp_(enable_dscp) {}
+
+MediaChannelUtil::TransportForMediaChannels::~TransportForMediaChannels() {
+  RTC_DCHECK(!network_interface_);
+}
+
+bool MediaChannelUtil::TransportForMediaChannels::SendRtcp(
+    rtc::ArrayView<const uint8_t> packet) {
+  auto send = [this, packet = rtc::CopyOnWriteBuffer(
+                         packet, kMaxRtpPacketLen)]() mutable {
+    rtc::PacketOptions rtc_options;
+    if (DscpEnabled()) {
+      rtc_options.dscp = PreferredDscp();
+    }
+    DoSendPacket(&packet, true, rtc_options);
+  };
+
+  if (network_thread_->IsCurrent()) {
+    send();
+  } else {
+    network_thread_->PostTask(SafeTask(network_safety_, std::move(send)));
+  }
+  return true;
+}
+
+bool MediaChannelUtil::TransportForMediaChannels::SendRtp(
+    rtc::ArrayView<const uint8_t> packet,
+    const webrtc::PacketOptions& options) {
+  auto send =
+      [this, packet_id = options.packet_id,
+       included_in_feedback = options.included_in_feedback,
+       included_in_allocation = options.included_in_allocation,
+       batchable = options.batchable,
+       last_packet_in_batch = options.last_packet_in_batch,
+       packet = rtc::CopyOnWriteBuffer(packet, kMaxRtpPacketLen)]() mutable {
+        rtc::PacketOptions rtc_options;
+        rtc_options.packet_id = packet_id;
+        if (DscpEnabled()) {
+          rtc_options.dscp = PreferredDscp();
+        }
+        rtc_options.info_signaled_after_sent.included_in_feedback =
+            included_in_feedback;
+        rtc_options.info_signaled_after_sent.included_in_allocation =
+            included_in_allocation;
+        rtc_options.batchable = batchable;
+        rtc_options.last_packet_in_batch = last_packet_in_batch;
+        DoSendPacket(&packet, false, rtc_options);
+      };
+
+  // TODO(bugs.webrtc.org/11993): ModuleRtpRtcpImpl2 and related classes (e.g.
+  // RTCPSender) aren't aware of the network thread and may trigger calls to
+  // this function from different threads. Update those classes to keep
+  // network traffic on the network thread.
+  if (network_thread_->IsCurrent()) {
+    send();
+  } else {
+    network_thread_->PostTask(SafeTask(network_safety_, std::move(send)));
+  }
+  return true;
+}
+
+void MediaChannelUtil::TransportForMediaChannels::SetInterface(
+    MediaChannelNetworkInterface* iface) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  iface ? network_safety_->SetAlive() : network_safety_->SetNotAlive();
+  network_interface_ = iface;
+  UpdateDscp();
+}
+
+void MediaChannelUtil::TransportForMediaChannels::UpdateDscp() {
+  rtc::DiffServCodePoint value =
+      enable_dscp_ ? preferred_dscp_ : rtc::DSCP_DEFAULT;
+  int ret = SetOptionLocked(MediaChannelNetworkInterface::ST_RTP,
+                            rtc::Socket::OPT_DSCP, value);
+  if (ret == 0)
+    SetOptionLocked(MediaChannelNetworkInterface::ST_RTCP,
+                    rtc::Socket::OPT_DSCP, value);
+}
+
+bool MediaChannelUtil::TransportForMediaChannels::DoSendPacket(
+    rtc::CopyOnWriteBuffer* packet,
+    bool rtcp,
+    const rtc::PacketOptions& options) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (!network_interface_)
+    return false;
+
+  return (!rtcp) ? network_interface_->SendPacket(packet, options)
+                 : network_interface_->SendRtcp(packet, options);
+}
+
+int MediaChannelUtil::TransportForMediaChannels::SetOption(
+    MediaChannelNetworkInterface::SocketType type,
+    rtc::Socket::Option opt,
+    int option) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  return SetOptionLocked(type, opt, option);
+}
+
+int MediaChannelUtil::TransportForMediaChannels::SetOptionLocked(
+    MediaChannelNetworkInterface::SocketType type,
+    rtc::Socket::Option opt,
+    int option) {
+  if (!network_interface_)
+    return -1;
+  return network_interface_->SetOption(type, opt, option);
+}
+
+void MediaChannelUtil::TransportForMediaChannels::SetPreferredDscp(
+    rtc::DiffServCodePoint new_dscp) {
+  if (!network_thread_->IsCurrent()) {
+    // This is currently the common path as the derived channel classes
+    // get called on the worker thread. There are still some tests though
+    // that call directly on the network thread.
+    network_thread_->PostTask(SafeTask(
+        network_safety_, [this, new_dscp]() { SetPreferredDscp(new_dscp); }));
+    return;
+  }
+
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (new_dscp == preferred_dscp_)
+    return;
+
+  preferred_dscp_ = new_dscp;
+  UpdateDscp();
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/media_channel_impl.h

@@ -0,0 +1,181 @@
+/*
+ *  Copyright 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 MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_
+#define MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <functional>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_options.h"
+#include "api/call/audio_sink.h"
+#include "api/call/transport.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/media_types.h"
+#include "api/rtc_error.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_sender_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "api/video/recordable_encoded_frame.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/stream_params.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/dscp.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/socket.h"
+#include "rtc_base/thread_annotations.h"
+// This file contains the base classes for classes that implement
+// the channel interfaces.
+// These implementation classes used to be the exposed interface names,
+// but this is in the process of being changed.
+
+namespace cricket {
+
+// The `MediaChannelUtil` class provides functionality that is used by
+// multiple MediaChannel-like objects, of both sending and receiving
+// types.
+class MediaChannelUtil {
+ public:
+  MediaChannelUtil(webrtc::TaskQueueBase* network_thread,
+                   bool enable_dscp = false);
+  virtual ~MediaChannelUtil();
+  // Returns the absolute sendtime extension id value from media channel.
+  virtual int GetRtpSendTimeExtnId() const;
+
+  webrtc::Transport* transport() { return &transport_; }
+
+  // Base methods to send packet using MediaChannelNetworkInterface.
+  // These methods are used by some tests only.
+  bool SendPacket(rtc::CopyOnWriteBuffer* packet,
+                  const rtc::PacketOptions& options);
+
+  bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
+                const rtc::PacketOptions& options);
+
+  int SetOption(MediaChannelNetworkInterface::SocketType type,
+                rtc::Socket::Option opt,
+                int option);
+
+  // Functions that form part of one or more interface classes.
+  // Not marked override, since this class does not inherit from the
+  // interfaces.
+
+  // 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. The setter and getter must only be called from
+  // worker_thread.
+  void SetExtmapAllowMixed(bool extmap_allow_mixed);
+  bool ExtmapAllowMixed() const;
+
+  void SetInterface(MediaChannelNetworkInterface* iface);
+  // Returns `true` if a non-null MediaChannelNetworkInterface pointer is held.
+  // Must be called on the network thread.
+  bool HasNetworkInterface() const;
+
+ protected:
+  bool DscpEnabled() const;
+
+  void SetPreferredDscp(rtc::DiffServCodePoint new_dscp);
+
+ private:
+  // Implementation of the webrtc::Transport interface required
+  // by Call().
+  class TransportForMediaChannels : public webrtc::Transport {
+   public:
+    TransportForMediaChannels(webrtc::TaskQueueBase* network_thread,
+                              bool enable_dscp);
+
+    virtual ~TransportForMediaChannels();
+
+    // Implementation of webrtc::Transport
+    bool SendRtp(rtc::ArrayView<const uint8_t> packet,
+                 const webrtc::PacketOptions& options) override;
+    bool SendRtcp(rtc::ArrayView<const uint8_t> packet) override;
+
+    // Not implementation of webrtc::Transport
+    void SetInterface(MediaChannelNetworkInterface* iface);
+
+    int SetOption(MediaChannelNetworkInterface::SocketType type,
+                  rtc::Socket::Option opt,
+                  int option);
+
+    bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
+                      bool rtcp,
+                      const rtc::PacketOptions& options);
+
+    bool HasNetworkInterface() const {
+      RTC_DCHECK_RUN_ON(network_thread_);
+      return network_interface_ != nullptr;
+    }
+    bool DscpEnabled() const { return enable_dscp_; }
+
+    void SetPreferredDscp(rtc::DiffServCodePoint new_dscp);
+
+   private:
+    // This is the DSCP value used for both RTP and RTCP channels if DSCP is
+    // enabled. It can be changed at any time via `SetPreferredDscp`.
+    rtc::DiffServCodePoint PreferredDscp() const {
+      RTC_DCHECK_RUN_ON(network_thread_);
+      return preferred_dscp_;
+    }
+
+    // Apply the preferred DSCP setting to the underlying network interface RTP
+    // and RTCP channels. If DSCP is disabled, then apply the default DSCP
+    // value.
+    void UpdateDscp() RTC_RUN_ON(network_thread_);
+
+    int SetOptionLocked(MediaChannelNetworkInterface::SocketType type,
+                        rtc::Socket::Option opt,
+                        int option) RTC_RUN_ON(network_thread_);
+
+    const rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety_
+        RTC_PT_GUARDED_BY(network_thread_);
+    webrtc::TaskQueueBase* const network_thread_;
+    const bool enable_dscp_;
+    MediaChannelNetworkInterface* network_interface_
+        RTC_GUARDED_BY(network_thread_) = nullptr;
+    rtc::DiffServCodePoint preferred_dscp_ RTC_GUARDED_BY(network_thread_) =
+        rtc::DSCP_DEFAULT;
+  };
+
+  bool extmap_allow_mixed_ = false;
+  TransportForMediaChannels transport_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_

TMessagesProj/jni/voip/webrtc/media/base/media_config.h

@@ -0,0 +1,98 @@
+/*
+ *  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 MEDIA_BASE_MEDIA_CONFIG_H_
+#define MEDIA_BASE_MEDIA_CONFIG_H_
+
+namespace cricket {
+
+// Construction-time settings, passed on when creating
+// MediaChannels.
+struct MediaConfig {
+  // Set DSCP value on packets. This flag comes from the
+  // PeerConnection constraint 'googDscp'.
+  // TODO(https://crbug.com/1315574): Remove the ability to set it in Chromium
+  // and delete this flag.
+  bool enable_dscp = true;
+
+  // Video-specific config.
+  struct Video {
+    // Enable WebRTC CPU Overuse Detection. This flag comes from the
+    // PeerConnection constraint 'googCpuOveruseDetection'.
+    // TODO(https://crbug.com/1315569): Remove the ability to set it in Chromium
+    // and delete this flag.
+    bool enable_cpu_adaptation = true;
+
+    // Enable WebRTC suspension of video. No video frames will be sent
+    // when the bitrate is below the configured minimum bitrate. This
+    // flag comes from the PeerConnection constraint
+    // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
+    // to VideoSendStream::Config::suspend_below_min_bitrate.
+    // TODO(https://crbug.com/1315564): Remove the ability to set it in Chromium
+    // and delete this flag.
+    bool suspend_below_min_bitrate = false;
+
+    // Enable buffering and playout timing smoothing of decoded frames.
+    // If set to true, then WebRTC will buffer and potentially drop decoded
+    // frames in order to keep a smooth rendering.
+    // If set to false, then WebRTC will hand over the frame from the decoder
+    // to the renderer as soon as possible, meaning that the renderer is
+    // responsible for smooth rendering.
+    // Note that even if this flag is set to false, dropping of frames can
+    // still happen pre-decode, e.g., dropping of higher temporal layers.
+    // This flag comes from the PeerConnection RtcConfiguration.
+    bool enable_prerenderer_smoothing = true;
+
+    // Enables periodic bandwidth probing in application-limited region.
+    bool periodic_alr_bandwidth_probing = false;
+
+    // Enables the new method to estimate the cpu load from encoding, used for
+    // cpu adaptation. This flag is intended to be controlled primarily by a
+    // Chrome origin-trial.
+    // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
+    // together with the old method of estimation.
+    bool experiment_cpu_load_estimator = false;
+
+    // Time interval between RTCP report for video
+    int rtcp_report_interval_ms = 1000;
+
+    // Enables send packet batching from the egress RTP sender.
+    bool enable_send_packet_batching = false;
+  } video;
+
+  // Audio-specific config.
+  struct Audio {
+    // Time interval between RTCP report for audio
+    int rtcp_report_interval_ms = 5000;
+  } audio;
+
+  bool operator==(const MediaConfig& o) const {
+    return enable_dscp == o.enable_dscp &&
+           video.enable_cpu_adaptation == o.video.enable_cpu_adaptation &&
+           video.suspend_below_min_bitrate ==
+               o.video.suspend_below_min_bitrate &&
+           video.enable_prerenderer_smoothing ==
+               o.video.enable_prerenderer_smoothing &&
+           video.periodic_alr_bandwidth_probing ==
+               o.video.periodic_alr_bandwidth_probing &&
+           video.experiment_cpu_load_estimator ==
+               o.video.experiment_cpu_load_estimator &&
+           video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms &&
+           video.enable_send_packet_batching ==
+               o.video.enable_send_packet_batching &&
+           audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms;
+  }
+
+  bool operator!=(const MediaConfig& o) const { return !(*this == o); }
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_MEDIA_CONFIG_H_

TMessagesProj/jni/voip/webrtc/media/base/media_constants.cc

@@ -0,0 +1,149 @@
+/*
+ *  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 "media/base/media_constants.h"
+
+namespace cricket {
+
+const int kVideoCodecClockrate = 90000;
+
+const int kVideoMtu = 1200;
+const int kVideoRtpSendBufferSize = 262144;
+const int kVideoRtpRecvBufferSize = 262144;
+
+const float kHighSystemCpuThreshold = 0.85f;
+const float kLowSystemCpuThreshold = 0.65f;
+const float kProcessCpuThreshold = 0.10f;
+
+const char kRedCodecName[] = "red";
+const char kUlpfecCodecName[] = "ulpfec";
+const char kMultiplexCodecName[] = "multiplex";
+
+// TODO(brandtr): Change this to 'flexfec' when we are confident that the
+// header format is not changing anymore.
+const char kFlexfecCodecName[] = "flexfec-03";
+
+// draft-ietf-payload-flexible-fec-scheme-02.txt
+const char kFlexfecFmtpRepairWindow[] = "repair-window";
+
+// RFC 4588 RTP Retransmission Payload Format
+const char kRtxCodecName[] = "rtx";
+const char kCodecParamRtxTime[] = "rtx-time";
+const char kCodecParamAssociatedPayloadType[] = "apt";
+
+const char kCodecParamAssociatedCodecName[] = "acn";
+// Parameters that do not follow the key-value convention
+// are treated as having the empty string as key.
+const char kCodecParamNotInNameValueFormat[] = "";
+
+const char kOpusCodecName[] = "opus";
+const char kL16CodecName[] = "L16";
+const char kG722CodecName[] = "G722";
+const char kIlbcCodecName[] = "ILBC";
+const char kPcmuCodecName[] = "PCMU";
+const char kPcmaCodecName[] = "PCMA";
+const char kCnCodecName[] = "CN";
+const char kDtmfCodecName[] = "telephone-event";
+
+// draft-spittka-payload-rtp-opus-03.txt
+const char kCodecParamPTime[] = "ptime";
+const char kCodecParamMaxPTime[] = "maxptime";
+const char kCodecParamMinPTime[] = "minptime";
+const char kCodecParamSPropStereo[] = "sprop-stereo";
+const char kCodecParamStereo[] = "stereo";
+const char kCodecParamUseInbandFec[] = "useinbandfec";
+const char kCodecParamUseDtx[] = "usedtx";
+const char kCodecParamMaxAverageBitrate[] = "maxaveragebitrate";
+const char kCodecParamMaxPlaybackRate[] = "maxplaybackrate";
+
+const char kParamValueTrue[] = "1";
+const char kParamValueEmpty[] = "";
+
+const int kOpusDefaultMaxPTime = 120;
+const int kOpusDefaultPTime = 20;
+const int kOpusDefaultMinPTime = 3;
+const int kOpusDefaultSPropStereo = 0;
+const int kOpusDefaultStereo = 0;
+const int kOpusDefaultUseInbandFec = 0;
+const int kOpusDefaultUseDtx = 0;
+const int kOpusDefaultMaxPlaybackRate = 48000;
+
+const int kPreferredMaxPTime = 120;
+const int kPreferredMinPTime = 10;
+const int kPreferredSPropStereo = 0;
+const int kPreferredStereo = 0;
+const int kPreferredUseInbandFec = 0;
+
+const char kPacketizationParamRaw[] = "raw";
+
+const char kRtcpFbParamLntf[] = "goog-lntf";
+const char kRtcpFbParamNack[] = "nack";
+const char kRtcpFbNackParamPli[] = "pli";
+const char kRtcpFbParamRemb[] = "goog-remb";
+const char kRtcpFbParamTransportCc[] = "transport-cc";
+
+const char kRtcpFbParamCcm[] = "ccm";
+const char kRtcpFbCcmParamFir[] = "fir";
+const char kRtcpFbParamRrtr[] = "rrtr";
+const char kCodecParamMaxBitrate[] = "x-google-max-bitrate";
+const char kCodecParamMinBitrate[] = "x-google-min-bitrate";
+const char kCodecParamStartBitrate[] = "x-google-start-bitrate";
+const char kCodecParamMaxQuantization[] = "x-google-max-quantization";
+
+const char kComfortNoiseCodecName[] = "CN";
+
+const char kVp8CodecName[] = "VP8";
+const char kVp9CodecName[] = "VP9";
+const char kAv1CodecName[] = "AV1";
+const char kH264CodecName[] = "H264";
+const char kH265CodecName[] = "H265";
+
+// RFC 6184 RTP Payload Format for H.264 video
+const char kH264FmtpProfileLevelId[] = "profile-level-id";
+const char kH264FmtpLevelAsymmetryAllowed[] = "level-asymmetry-allowed";
+const char kH264FmtpPacketizationMode[] = "packetization-mode";
+const char kH264FmtpSpropParameterSets[] = "sprop-parameter-sets";
+const char kH264FmtpSpsPpsIdrInKeyframe[] = "sps-pps-idr-in-keyframe";
+const char kH264ProfileLevelConstrainedBaseline[] = "42e01f";
+const char kH264ProfileLevelConstrainedHigh[] = "640c1f";
+
+// RFC 7798 RTP Payload Format for H.265 video
+const char kH265FmtpProfileSpace[] = "profile-space";
+const char kH265FmtpTierFlag[] = "tier-flag";
+const char kH265FmtpProfileId[] = "profile-id";
+const char kH265FmtpLevelId[] = "level-id";
+const char kH265FmtpProfileCompatibilityIndicator[] =
+    "profile-compatibility-indicator";
+const char kH265FmtpInteropConstraints[] = "interop-constraints";
+const char kH265FmtpTxMode[] = "tx-mode";
+
+// draft-ietf-payload-vp9
+const char kVP9ProfileId[] = "profile-id";
+
+// https://aomediacodec.github.io/av1-rtp-spec/
+const char kAv1FmtpProfile[] = "profile";
+const char kAv1FmtpLevelIdx[] = "level-idx";
+const char kAv1FmtpTier[] = "tier";
+
+const int kDefaultVideoMaxFramerate = 60;
+// Max encode quantizer for VP8/9 and AV1 encoders assuming libvpx/libaom API
+// range [0, 63]
+const int kDefaultVideoMaxQpVpx = 56;
+// Max encode quantizer for H264/5 assuming the bitstream range [0, 51].
+const int kDefaultVideoMaxQpH26x = 51;
+
+const size_t kConferenceMaxNumSpatialLayers = 3;
+const size_t kConferenceMaxNumTemporalLayers = 3;
+const size_t kConferenceDefaultNumTemporalLayers = 3;
+
+// RFC 3556 and RFC 3890
+const char kApplicationSpecificBandwidth[] = "AS";
+const char kTransportSpecificBandwidth[] = "TIAS";
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/media_constants.h

@@ -0,0 +1,170 @@
+/*
+ *  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 MEDIA_BASE_MEDIA_CONSTANTS_H_
+#define MEDIA_BASE_MEDIA_CONSTANTS_H_
+
+#include <stddef.h>
+
+#include "rtc_base/system/rtc_export.h"
+
+// This file contains constants related to media.
+
+namespace cricket {
+
+extern const int kVideoCodecClockrate;
+
+extern const int kVideoMtu;
+extern const int kVideoRtpSendBufferSize;
+extern const int kVideoRtpRecvBufferSize;
+
+// Default CPU thresholds.
+extern const float kHighSystemCpuThreshold;
+extern const float kLowSystemCpuThreshold;
+extern const float kProcessCpuThreshold;
+
+extern const char kRedCodecName[];
+extern const char kUlpfecCodecName[];
+extern const char kFlexfecCodecName[];
+extern const char kMultiplexCodecName[];
+
+extern const char kFlexfecFmtpRepairWindow[];
+
+extern const char kRtxCodecName[];
+extern const char kCodecParamRtxTime[];
+extern const char kCodecParamAssociatedPayloadType[];
+
+extern const char kCodecParamAssociatedCodecName[];
+extern const char kCodecParamNotInNameValueFormat[];
+
+extern const char kOpusCodecName[];
+extern const char kL16CodecName[];
+extern const char kG722CodecName[];
+extern const char kIlbcCodecName[];
+extern const char kPcmuCodecName[];
+extern const char kPcmaCodecName[];
+extern const char kCnCodecName[];
+extern const char kDtmfCodecName[];
+
+// Attribute parameters
+extern const char kCodecParamPTime[];
+extern const char kCodecParamMaxPTime[];
+// fmtp parameters
+extern const char kCodecParamMinPTime[];
+extern const char kCodecParamSPropStereo[];
+extern const char kCodecParamStereo[];
+extern const char kCodecParamUseInbandFec[];
+extern const char kCodecParamUseDtx[];
+extern const char kCodecParamMaxAverageBitrate[];
+extern const char kCodecParamMaxPlaybackRate[];
+
+extern const char kParamValueTrue[];
+// Parameters are stored as parameter/value pairs. For parameters who do not
+// have a value, `kParamValueEmpty` should be used as value.
+extern const char kParamValueEmpty[];
+
+// opus parameters.
+// Default value for maxptime according to
+// http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
+extern const int kOpusDefaultMaxPTime;
+extern const int kOpusDefaultPTime;
+extern const int kOpusDefaultMinPTime;
+extern const int kOpusDefaultSPropStereo;
+extern const int kOpusDefaultStereo;
+extern const int kOpusDefaultUseInbandFec;
+extern const int kOpusDefaultUseDtx;
+extern const int kOpusDefaultMaxPlaybackRate;
+
+// Prefered values in this code base. Note that they may differ from the default
+// values in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
+// Only frames larger or equal to 10 ms are currently supported in this code
+// base.
+extern const int kPreferredMaxPTime;
+extern const int kPreferredMinPTime;
+extern const int kPreferredSPropStereo;
+extern const int kPreferredStereo;
+extern const int kPreferredUseInbandFec;
+
+extern const char kPacketizationParamRaw[];
+
+// rtcp-fb message in its first experimental stages. Documentation pending.
+extern const char kRtcpFbParamLntf[];
+// rtcp-fb messages according to RFC 4585
+extern const char kRtcpFbParamNack[];
+extern const char kRtcpFbNackParamPli[];
+// rtcp-fb messages according to
+// http://tools.ietf.org/html/draft-alvestrand-rmcat-remb-00
+extern const char kRtcpFbParamRemb[];
+// rtcp-fb messages according to
+// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
+extern const char kRtcpFbParamTransportCc[];
+// ccm submessages according to RFC 5104
+extern const char kRtcpFbParamCcm[];
+extern const char kRtcpFbCcmParamFir[];
+// Receiver reference time report
+// https://tools.ietf.org/html/rfc3611 section 4.4
+extern const char kRtcpFbParamRrtr[];
+// Google specific parameters
+extern const char kCodecParamMaxBitrate[];
+extern const char kCodecParamMinBitrate[];
+extern const char kCodecParamStartBitrate[];
+extern const char kCodecParamMaxQuantization[];
+
+extern const char kComfortNoiseCodecName[];
+
+RTC_EXPORT extern const char kVp8CodecName[];
+RTC_EXPORT extern const char kVp9CodecName[];
+RTC_EXPORT extern const char kAv1CodecName[];
+RTC_EXPORT extern const char kH264CodecName[];
+RTC_EXPORT extern const char kH265CodecName[];
+
+// RFC 6184 RTP Payload Format for H.264 video
+RTC_EXPORT extern const char kH264FmtpProfileLevelId[];
+RTC_EXPORT extern const char kH264FmtpLevelAsymmetryAllowed[];
+RTC_EXPORT extern const char kH264FmtpPacketizationMode[];
+extern const char kH264FmtpSpropParameterSets[];
+extern const char kH264FmtpSpsPpsIdrInKeyframe[];
+extern const char kH264ProfileLevelConstrainedBaseline[];
+extern const char kH264ProfileLevelConstrainedHigh[];
+
+// RFC 7798 RTP Payload Format for H.265 video.
+// According to RFC 7742, the sprop parameters MUST NOT be included
+// in SDP generated by WebRTC, so for H.265 we don't handle them, though
+// current H.264 implementation honors them when receiving
+// sprop-parameter-sets in SDP.
+RTC_EXPORT extern const char kH265FmtpProfileSpace[];
+RTC_EXPORT extern const char kH265FmtpTierFlag[];
+RTC_EXPORT extern const char kH265FmtpProfileId[];
+RTC_EXPORT extern const char kH265FmtpLevelId[];
+RTC_EXPORT extern const char kH265FmtpProfileCompatibilityIndicator[];
+RTC_EXPORT extern const char kH265FmtpInteropConstraints[];
+RTC_EXPORT extern const char kH265FmtpTxMode[];
+
+// draft-ietf-payload-vp9
+extern const char kVP9ProfileId[];
+
+// https://aomediacodec.github.io/av1-rtp-spec/
+extern const char kAv1FmtpProfile[];
+extern const char kAv1FmtpLevelIdx[];
+extern const char kAv1FmtpTier[];
+
+extern const int kDefaultVideoMaxFramerate;
+extern const int kDefaultVideoMaxQpVpx;
+extern const int kDefaultVideoMaxQpH26x;
+
+extern const size_t kConferenceMaxNumSpatialLayers;
+extern const size_t kConferenceMaxNumTemporalLayers;
+extern const size_t kConferenceDefaultNumTemporalLayers;
+
+extern const char kApplicationSpecificBandwidth[];
+extern const char kTransportSpecificBandwidth[];
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_MEDIA_CONSTANTS_H_

TMessagesProj/jni/voip/webrtc/media/base/media_engine.cc

@@ -0,0 +1,291 @@
+/*
+ *  Copyright (c) 2004 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 "media/base/media_engine.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_encode.h"
+
+namespace cricket {
+
+RtpCapabilities::RtpCapabilities() = default;
+RtpCapabilities::~RtpCapabilities() = default;
+
+webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
+  webrtc::RtpParameters parameters;
+  webrtc::RtpEncodingParameters encoding;
+  parameters.encodings.push_back(encoding);
+  return parameters;
+}
+
+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
+  std::vector<uint32_t> primary_ssrcs;
+  sp.GetPrimarySsrcs(&primary_ssrcs);
+  size_t encoding_count = primary_ssrcs.size();
+
+  std::vector<webrtc::RtpEncodingParameters> encodings(encoding_count);
+  for (size_t i = 0; i < encodings.size(); ++i) {
+    encodings[i].ssrc = primary_ssrcs[i];
+  }
+
+  const std::vector<RidDescription>& rids = sp.rids();
+  RTC_DCHECK(rids.size() == 0 || rids.size() == encoding_count);
+  for (size_t i = 0; i < rids.size(); ++i) {
+    encodings[i].rid = rids[i].rid;
+  }
+
+  webrtc::RtpParameters parameters;
+  parameters.encodings = encodings;
+  parameters.rtcp.cname = sp.cname;
+  return parameters;
+}
+
+std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
+    const RtpHeaderExtensionQueryInterface& query_interface) {
+  std::vector<webrtc::RtpExtension> extensions;
+  for (const auto& entry : query_interface.GetRtpHeaderExtensions()) {
+    if (entry.direction != webrtc::RtpTransceiverDirection::kStopped)
+      extensions.emplace_back(entry.uri, *entry.preferred_id);
+  }
+  return extensions;
+}
+
+webrtc::RTCError CheckScalabilityModeValues(
+    const webrtc::RtpParameters& rtp_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec) {
+  using webrtc::RTCErrorType;
+
+  if (codec_preferences.empty()) {
+    // This is an audio sender or an extra check in the stack where the codec
+    // list is not available and we can't check the scalability_mode values.
+    return webrtc::RTCError::OK();
+  }
+
+  for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
+    if (rtp_parameters.encodings[i].codec) {
+      bool codecFound = false;
+      for (const cricket::VideoCodec& codec : codec_preferences) {
+        if (codec.MatchesRtpCodec(*rtp_parameters.encodings[i].codec)) {
+          codecFound = true;
+          send_codec = codec;
+          break;
+        }
+      }
+      if (!codecFound) {
+        LOG_AND_RETURN_ERROR(
+            RTCErrorType::INVALID_MODIFICATION,
+            "Attempted to use an unsupported codec for layer " +
+                std::to_string(i));
+      }
+    }
+    if (rtp_parameters.encodings[i].scalability_mode) {
+      if (!send_codec) {
+        bool scalabilityModeFound = false;
+        for (const cricket::VideoCodec& codec : codec_preferences) {
+          for (const auto& scalability_mode : codec.scalability_modes) {
+            if (ScalabilityModeToString(scalability_mode) ==
+                *rtp_parameters.encodings[i].scalability_mode) {
+              scalabilityModeFound = true;
+              break;
+            }
+          }
+          if (scalabilityModeFound)
+            break;
+        }
+
+        if (!scalabilityModeFound) {
+          LOG_AND_RETURN_ERROR(
+              RTCErrorType::INVALID_MODIFICATION,
+              "Attempted to set RtpParameters scalabilityMode "
+              "to an unsupported value for the current codecs.");
+        }
+      } else {
+        bool scalabilityModeFound = false;
+        for (const auto& scalability_mode : send_codec->scalability_modes) {
+          if (ScalabilityModeToString(scalability_mode) ==
+              *rtp_parameters.encodings[i].scalability_mode) {
+            scalabilityModeFound = true;
+            break;
+          }
+        }
+        if (!scalabilityModeFound) {
+          LOG_AND_RETURN_ERROR(
+              RTCErrorType::INVALID_MODIFICATION,
+              "Attempted to set RtpParameters scalabilityMode "
+              "to an unsupported value for the current codecs.");
+        }
+      }
+    }
+  }
+
+  return webrtc::RTCError::OK();
+}
+
+webrtc::RTCError CheckRtpParametersValues(
+    const webrtc::RtpParameters& rtp_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec) {
+  using webrtc::RTCErrorType;
+
+  for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
+    if (rtp_parameters.encodings[i].bitrate_priority <= 0) {
+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
+                           "Attempted to set RtpParameters bitrate_priority to "
+                           "an invalid number. bitrate_priority must be > 0.");
+    }
+    if (rtp_parameters.encodings[i].scale_resolution_down_by &&
+        *rtp_parameters.encodings[i].scale_resolution_down_by < 1.0) {
+      LOG_AND_RETURN_ERROR(
+          RTCErrorType::INVALID_RANGE,
+          "Attempted to set RtpParameters scale_resolution_down_by to an "
+          "invalid value. scale_resolution_down_by must be >= 1.0");
+    }
+    if (rtp_parameters.encodings[i].max_framerate &&
+        *rtp_parameters.encodings[i].max_framerate < 0.0) {
+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
+                           "Attempted to set RtpParameters max_framerate to an "
+                           "invalid value. max_framerate must be >= 0.0");
+    }
+    if (rtp_parameters.encodings[i].min_bitrate_bps &&
+        rtp_parameters.encodings[i].max_bitrate_bps) {
+      if (*rtp_parameters.encodings[i].max_bitrate_bps <
+          *rtp_parameters.encodings[i].min_bitrate_bps) {
+        LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_RANGE,
+                             "Attempted to set RtpParameters min bitrate "
+                             "larger than max bitrate.");
+      }
+    }
+    if (rtp_parameters.encodings[i].num_temporal_layers) {
+      if (*rtp_parameters.encodings[i].num_temporal_layers < 1 ||
+          *rtp_parameters.encodings[i].num_temporal_layers >
+              webrtc::kMaxTemporalStreams) {
+        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
+                             "Attempted to set RtpParameters "
+                             "num_temporal_layers to an invalid number.");
+      }
+    }
+
+    if (rtp_parameters.encodings[i].requested_resolution &&
+        rtp_parameters.encodings[i].scale_resolution_down_by) {
+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
+                           "Attempted to set scale_resolution_down_by and "
+                           "requested_resolution simultaniously.");
+    }
+
+    if (i > 0 && rtp_parameters.encodings[i - 1].codec !=
+                     rtp_parameters.encodings[i].codec) {
+      LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
+                           "Attempted to use different codec values for "
+                           "different encodings.");
+    }
+  }
+
+  return CheckScalabilityModeValues(rtp_parameters, codec_preferences,
+                                    send_codec);
+}
+
+webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
+    const webrtc::RtpParameters& old_rtp_parameters,
+    const webrtc::RtpParameters& rtp_parameters) {
+  return CheckRtpParametersInvalidModificationAndValues(
+      old_rtp_parameters, rtp_parameters, {}, absl::nullopt);
+}
+
+webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
+    const webrtc::RtpParameters& old_rtp_parameters,
+    const webrtc::RtpParameters& rtp_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec) {
+  using webrtc::RTCErrorType;
+  if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_MODIFICATION,
+        "Attempted to set RtpParameters with different encoding count");
+  }
+  if (rtp_parameters.rtcp != old_rtp_parameters.rtcp) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_MODIFICATION,
+        "Attempted to set RtpParameters with modified RTCP parameters");
+  }
+  if (rtp_parameters.header_extensions !=
+      old_rtp_parameters.header_extensions) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_MODIFICATION,
+        "Attempted to set RtpParameters with modified header extensions");
+  }
+  if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
+                     [](const webrtc::RtpEncodingParameters& encoding1,
+                        const webrtc::RtpEncodingParameters& encoding2) {
+                       return encoding1.rid == encoding2.rid;
+                     })) {
+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
+                         "Attempted to change RID values in the encodings.");
+  }
+  if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings,
+                     [](const webrtc::RtpEncodingParameters& encoding1,
+                        const webrtc::RtpEncodingParameters& encoding2) {
+                       return encoding1.ssrc == encoding2.ssrc;
+                     })) {
+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
+                         "Attempted to set RtpParameters with modified SSRC");
+  }
+
+  return CheckRtpParametersValues(rtp_parameters, codec_preferences,
+                                  send_codec);
+}
+
+CompositeMediaEngine::CompositeMediaEngine(
+    std::unique_ptr<webrtc::FieldTrialsView> trials,
+    std::unique_ptr<VoiceEngineInterface> audio_engine,
+    std::unique_ptr<VideoEngineInterface> video_engine)
+    : trials_(std::move(trials)),
+      voice_engine_(std::move(audio_engine)),
+      video_engine_(std::move(video_engine)) {}
+
+CompositeMediaEngine::CompositeMediaEngine(
+    std::unique_ptr<VoiceEngineInterface> audio_engine,
+    std::unique_ptr<VideoEngineInterface> video_engine)
+    : CompositeMediaEngine(nullptr,
+                           std::move(audio_engine),
+                           std::move(video_engine)) {}
+
+CompositeMediaEngine::~CompositeMediaEngine() = default;
+
+bool CompositeMediaEngine::Init() {
+  voice().Init();
+  return true;
+}
+
+VoiceEngineInterface& CompositeMediaEngine::voice() {
+  return *voice_engine_.get();
+}
+
+VideoEngineInterface& CompositeMediaEngine::video() {
+  return *video_engine_.get();
+}
+
+const VoiceEngineInterface& CompositeMediaEngine::voice() const {
+  return *voice_engine_.get();
+}
+
+const VideoEngineInterface& CompositeMediaEngine::video() const {
+  return *video_engine_.get();
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/media_engine.h

@@ -0,0 +1,239 @@
+/*
+ *  Copyright (c) 2004 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 MEDIA_BASE_MEDIA_ENGINE_H_
+#define MEDIA_BASE_MEDIA_ENGINE_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/crypto/crypto_options.h"
+#include "api/field_trials_view.h"
+#include "api/rtp_parameters.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "call/audio_state.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_config.h"
+#include "media/base/video_common.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+class AudioDeviceModule;
+class AudioMixer;
+class AudioProcessing;
+class Call;
+}  // namespace webrtc
+
+namespace cricket {
+
+// Checks that the scalability_mode value of each encoding is supported by at
+// least one video codec of the list. If the list is empty, no check is done.
+webrtc::RTCError CheckScalabilityModeValues(
+    const webrtc::RtpParameters& new_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec);
+
+// Checks the parameters have valid and supported values, and checks parameters
+// with CheckScalabilityModeValues().
+webrtc::RTCError CheckRtpParametersValues(
+    const webrtc::RtpParameters& new_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec);
+
+// Checks that the immutable values have not changed in new_parameters and
+// checks all parameters with CheckRtpParametersValues().
+webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
+    const webrtc::RtpParameters& old_parameters,
+    const webrtc::RtpParameters& new_parameters,
+    rtc::ArrayView<cricket::Codec> codec_preferences,
+    absl::optional<cricket::Codec> send_codec);
+
+// Checks that the immutable values have not changed in new_parameters and
+// checks parameters (except SVC) with CheckRtpParametersValues(). It should
+// usually be paired with a call to CheckScalabilityModeValues().
+webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
+    const webrtc::RtpParameters& old_parameters,
+    const webrtc::RtpParameters& new_parameters);
+
+struct RtpCapabilities {
+  RtpCapabilities();
+  ~RtpCapabilities();
+  std::vector<webrtc::RtpExtension> header_extensions;
+};
+
+class RtpHeaderExtensionQueryInterface {
+ public:
+  virtual ~RtpHeaderExtensionQueryInterface() = default;
+
+  // Returns a vector of RtpHeaderExtensionCapability, whose direction is
+  // kStopped if the extension is stopped (not used) by default.
+  virtual std::vector<webrtc::RtpHeaderExtensionCapability>
+  GetRtpHeaderExtensions() const = 0;
+};
+
+class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
+ public:
+  VoiceEngineInterface() = default;
+  virtual ~VoiceEngineInterface() = default;
+
+  VoiceEngineInterface(const VoiceEngineInterface&) = delete;
+  VoiceEngineInterface& operator=(const VoiceEngineInterface&) = delete;
+
+  // Initialization
+  // Starts the engine.
+  virtual void Init() = 0;
+
+  // TODO(solenberg): Remove once VoE API refactoring is done.
+  virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
+
+  virtual std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) {
+    // TODO(hta): Make pure virtual when all downstream has updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<VoiceMediaReceiveChannelInterface>
+  CreateReceiveChannel(webrtc::Call* call,
+                       const MediaConfig& config,
+                       const AudioOptions& options,
+                       const webrtc::CryptoOptions& crypto_options,
+                       webrtc::AudioCodecPairId codec_pair_id) {
+    // TODO(hta): Make pure virtual when all downstream has updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual const std::vector<AudioCodec>& send_codecs() const = 0;
+  virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
+
+  // Starts AEC dump using existing file, a maximum file size in bytes can be
+  // specified. Logging is stopped just before the size limit is exceeded.
+  // If max_size_bytes is set to a value <= 0, no limit will be used.
+  virtual bool StartAecDump(webrtc::FileWrapper file,
+                            int64_t max_size_bytes) = 0;
+
+  // Stops recording AEC dump.
+  virtual void StopAecDump() = 0;
+
+  virtual absl::optional<webrtc::AudioDeviceModule::Stats>
+  GetAudioDeviceStats() = 0;
+};
+
+class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
+ public:
+  VideoEngineInterface() = default;
+  virtual ~VideoEngineInterface() = default;
+
+  VideoEngineInterface(const VideoEngineInterface&) = delete;
+  VideoEngineInterface& operator=(const VideoEngineInterface&) = delete;
+
+  virtual std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
+    // Default implementation, delete when all is updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<VideoMediaReceiveChannelInterface>
+  CreateReceiveChannel(webrtc::Call* call,
+                       const MediaConfig& config,
+                       const VideoOptions& options,
+                       const webrtc::CryptoOptions& crypto_options) {
+    // Default implementation, delete when all is updated
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  // Retrieve list of supported codecs.
+  virtual std::vector<VideoCodec> send_codecs() const = 0;
+  virtual std::vector<VideoCodec> recv_codecs() const = 0;
+  // As above, but if include_rtx is false, don't include RTX codecs.
+  // TODO(bugs.webrtc.org/13931): Remove default implementation once
+  // upstream subclasses have converted.
+  virtual std::vector<VideoCodec> send_codecs(bool include_rtx) const {
+    RTC_DCHECK(include_rtx);
+    return send_codecs();
+  }
+  virtual std::vector<VideoCodec> recv_codecs(bool include_rtx) const {
+    RTC_DCHECK(include_rtx);
+    return recv_codecs();
+  }
+};
+
+// MediaEngineInterface is an abstraction of a media engine which can be
+// subclassed to support different media componentry backends.
+// It supports voice and video operations in the same class to facilitate
+// proper synchronization between both media types.
+class MediaEngineInterface {
+ public:
+  virtual ~MediaEngineInterface() {}
+
+  // Initialization. Needs to be called on the worker thread.
+  virtual bool Init() = 0;
+
+  virtual VoiceEngineInterface& voice() = 0;
+  virtual VideoEngineInterface& video() = 0;
+  virtual const VoiceEngineInterface& voice() const = 0;
+  virtual const VideoEngineInterface& video() const = 0;
+};
+
+// CompositeMediaEngine constructs a MediaEngine from separate
+// voice and video engine classes.
+// Optionally owns a FieldTrialsView trials map.
+class CompositeMediaEngine : public MediaEngineInterface {
+ public:
+  CompositeMediaEngine(std::unique_ptr<webrtc::FieldTrialsView> trials,
+                       std::unique_ptr<VoiceEngineInterface> audio_engine,
+                       std::unique_ptr<VideoEngineInterface> video_engine);
+  CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
+                       std::unique_ptr<VideoEngineInterface> video_engine);
+  ~CompositeMediaEngine() override;
+
+  // Always succeeds.
+  bool Init() override;
+
+  VoiceEngineInterface& voice() override;
+  VideoEngineInterface& video() override;
+  const VoiceEngineInterface& voice() const override;
+  const VideoEngineInterface& video() const override;
+
+ private:
+  const std::unique_ptr<webrtc::FieldTrialsView> trials_;
+  const std::unique_ptr<VoiceEngineInterface> voice_engine_;
+  const std::unique_ptr<VideoEngineInterface> video_engine_;
+};
+
+webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
+
+// Returns a vector of RTP extensions as visible from RtpSender/Receiver
+// GetCapabilities(). The returned vector only shows what will definitely be
+// offered by default, i.e. the list of extensions returned from
+// GetRtpHeaderExtensions() that are not kStopped.
+std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
+    const RtpHeaderExtensionQueryInterface& query_interface);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_MEDIA_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/base/rid_description.cc

@@ -0,0 +1,28 @@
+/*
+ *  Copyright 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 "media/base/rid_description.h"
+
+namespace cricket {
+
+RidDescription::RidDescription() = default;
+RidDescription::RidDescription(const std::string& rid, RidDirection direction)
+    : rid{rid}, direction{direction} {}
+RidDescription::RidDescription(const RidDescription& other) = default;
+RidDescription::~RidDescription() = default;
+RidDescription& RidDescription::operator=(const RidDescription& other) =
+    default;
+bool RidDescription::operator==(const RidDescription& other) const {
+  return rid == other.rid && direction == other.direction &&
+         payload_types == other.payload_types &&
+         restrictions == other.restrictions;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/rid_description.h

@@ -0,0 +1,93 @@
+/*
+ *  Copyright 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 MEDIA_BASE_RID_DESCRIPTION_H_
+#define MEDIA_BASE_RID_DESCRIPTION_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+namespace cricket {
+
+enum class RidDirection { kSend, kReceive };
+
+// Description of a Restriction Id (RID) according to:
+// https://tools.ietf.org/html/draft-ietf-mmusic-rid-15
+// A Restriction Identifier serves two purposes:
+//   1. Uniquely identifies an RTP stream inside an RTP session.
+//      When combined with MIDs (https://tools.ietf.org/html/rfc5888),
+//      RIDs uniquely identify an RTP stream within an RTP session.
+//      The MID will identify the media section and the RID will identify
+//      the stream within the section.
+//      RID identifiers must be unique within the media section.
+//   2. Allows indicating further restrictions to the stream.
+//      These restrictions are added according to the direction specified.
+//      The direction field identifies the direction of the RTP stream packets
+//      to which the restrictions apply. The direction is independent of the
+//      transceiver direction and can be one of {send, recv}.
+//      The following are some examples of these restrictions:
+//        a. max-width, max-height, max-fps, max-br, ...
+//        b. further restricting the codec set (from what m= section specified)
+//
+// Note: Indicating dependencies between streams (using depend) will not be
+// supported, since the WG is adopting a different approach to achieve this.
+// As of 2018-12-04, the new SVC (Scalable Video Coder) approach is still not
+// mature enough to be implemented as part of this work.
+// See: https://w3c.github.io/webrtc-svc/ for more details.
+struct RidDescription final {
+  RidDescription();
+  RidDescription(const std::string& rid, RidDirection direction);
+  RidDescription(const RidDescription& other);
+  ~RidDescription();
+  RidDescription& operator=(const RidDescription& other);
+
+  // This is currently required for unit tests of StreamParams which contains
+  // RidDescription objects and checks for equality using operator==.
+  bool operator==(const RidDescription& other) const;
+  bool operator!=(const RidDescription& other) const {
+    return !(*this == other);
+  }
+
+  // The RID identifier that uniquely identifies the stream within the session.
+  std::string rid;
+
+  // Specifies the direction for which the specified restrictions hold.
+  // This direction is either send or receive and is independent of the
+  // direction of the transceiver.
+  // https://tools.ietf.org/html/draft-ietf-mmusic-rid-15#section-4 :
+  // The "direction" field identifies the direction of the RTP Stream
+  // packets to which the indicated restrictions are applied.  It may be
+  // either "send" or "recv".  Note that these restriction directions are
+  // expressed independently of any "inactive", "sendonly", "recvonly", or
+  // "sendrecv" attributes associated with the media section.  It is, for
+  // example, valid to indicate "recv" restrictions on a "sendonly"
+  // stream; those restrictions would apply if, at a future point in time,
+  // the stream were changed to "sendrecv" or "recvonly".
+  RidDirection direction;
+
+  // The list of codec payload types for this stream.
+  // It should be a subset of the payloads supported for the media section.
+  std::vector<int> payload_types;
+
+  // Contains key-value pairs for restrictions.
+  // The keys are not validated against a known set.
+  // The meaning to infer for the values depends on each key.
+  // Examples:
+  // 1. An entry for max-width will have a value that is interpreted as an int.
+  // 2. An entry for max-bpp (bits per pixel) will have a float value.
+  // Interpretation (and validation of value) is left for the implementation.
+  // I.E. the media engines should validate values for parameters they support.
+  std::map<std::string, std::string> restrictions;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_RID_DESCRIPTION_H_

TMessagesProj/jni/voip/webrtc/media/base/rtp_utils.cc

@@ -0,0 +1,401 @@
+/*
+ *  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 "media/base/rtp_utils.h"
+
+#include <string.h>
+
+#include <vector>
+
+// PacketTimeUpdateParams is defined in asyncpacketsocket.h.
+// TODO(sergeyu): Find more appropriate place for PacketTimeUpdateParams.
+#include "media/base/turn_utils.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/byte_order.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/message_digest.h"
+
+namespace cricket {
+
+static const size_t kRtcpPayloadTypeOffset = 1;
+static const size_t kRtpExtensionHeaderLen = 4;
+static const size_t kAbsSendTimeExtensionLen = 3;
+static const size_t kOneByteExtensionHeaderLen = 1;
+static const size_t kTwoByteExtensionHeaderLen = 2;
+
+namespace {
+
+// Fake auth tag written by the sender when external authentication is enabled.
+// HMAC in packet will be compared against this value before updating packet
+// with actual HMAC value.
+static const uint8_t kFakeAuthTag[10] = {0xba, 0xdd, 0xba, 0xdd, 0xba,
+                                         0xdd, 0xba, 0xdd, 0xba, 0xdd};
+
+void UpdateAbsSendTimeExtensionValue(uint8_t* extension_data,
+                                     size_t length,
+                                     uint64_t time_us) {
+  // Absolute send time in RTP streams.
+  //
+  // The absolute send time is signaled to the receiver in-band using the
+  // general mechanism for RTP header extensions [RFC5285]. The payload
+  // of this extension (the transmitted value) is a 24-bit unsigned integer
+  // containing the sender's current time in seconds as a fixed point number
+  // with 18 bits fractional part.
+  //
+  // The form of the absolute send time extension 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
+  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  //   |  ID   | len=2 |              absolute send time               |
+  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  if (length != kAbsSendTimeExtensionLen) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+
+  // Convert microseconds to a 6.18 fixed point value in seconds.
+  uint32_t send_time = ((time_us << 18) / 1000000) & 0x00FFFFFF;
+  extension_data[0] = static_cast<uint8_t>(send_time >> 16);
+  extension_data[1] = static_cast<uint8_t>(send_time >> 8);
+  extension_data[2] = static_cast<uint8_t>(send_time);
+}
+
+// Assumes `length` is actual packet length + tag length. Updates HMAC at end of
+// the RTP packet.
+void UpdateRtpAuthTag(uint8_t* rtp,
+                      size_t length,
+                      const rtc::PacketTimeUpdateParams& packet_time_params) {
+  // If there is no key, return.
+  if (packet_time_params.srtp_auth_key.empty()) {
+    return;
+  }
+
+  size_t tag_length = packet_time_params.srtp_auth_tag_len;
+
+  // ROC (rollover counter) is at the beginning of the auth tag.
+  const size_t kRocLength = 4;
+  if (tag_length < kRocLength || tag_length > length) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+
+  uint8_t* auth_tag = rtp + (length - tag_length);
+
+  // We should have a fake HMAC value @ auth_tag.
+  RTC_DCHECK_EQ(0, memcmp(auth_tag, kFakeAuthTag, tag_length));
+
+  // Copy ROC after end of rtp packet.
+  memcpy(auth_tag, &packet_time_params.srtp_packet_index, kRocLength);
+  // Authentication of a RTP packet will have RTP packet + ROC size.
+  size_t auth_required_length = length - tag_length + kRocLength;
+
+  uint8_t output[64];
+  size_t result =
+      rtc::ComputeHmac(rtc::DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0],
+                       packet_time_params.srtp_auth_key.size(), rtp,
+                       auth_required_length, output, sizeof(output));
+
+  if (result < tag_length) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+
+  // Copy HMAC from output to packet. This is required as auth tag length
+  // may not be equal to the actual HMAC length.
+  memcpy(auth_tag, output, tag_length);
+}
+
+bool GetUint8(const void* data, size_t offset, int* value) {
+  if (!data || !value) {
+    return false;
+  }
+  *value = *(static_cast<const uint8_t*>(data) + offset);
+  return true;
+}
+
+}  // namespace
+
+bool GetRtcpType(const void* data, size_t len, int* value) {
+  if (len < kMinRtcpPacketLen) {
+    return false;
+  }
+  return GetUint8(data, kRtcpPayloadTypeOffset, value);
+}
+
+// This method returns SSRC first of RTCP packet, except if packet is SDES.
+// TODO(mallinath) - Fully implement RFC 5506. This standard doesn't restrict
+// to send non-compound packets only to feedback messages.
+bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value) {
+  // Packet should be at least of 8 bytes, to get SSRC from a RTCP packet.
+  if (!data || len < kMinRtcpPacketLen + 4 || !value)
+    return false;
+  int pl_type;
+  if (!GetRtcpType(data, len, &pl_type))
+    return false;
+  // SDES packet parsing is not supported.
+  if (pl_type == kRtcpTypeSDES)
+    return false;
+  *value = rtc::GetBE32(static_cast<const uint8_t*>(data) + 4);
+  return true;
+}
+
+bool IsValidRtpPayloadType(int payload_type) {
+  return payload_type >= 0 && payload_type <= 127;
+}
+
+bool IsValidRtpPacketSize(RtpPacketType packet_type, size_t size) {
+  RTC_DCHECK_NE(RtpPacketType::kUnknown, packet_type);
+  size_t min_packet_length = packet_type == RtpPacketType::kRtcp
+                                 ? kMinRtcpPacketLen
+                                 : kMinRtpPacketLen;
+  return size >= min_packet_length && size <= kMaxRtpPacketLen;
+}
+
+absl::string_view RtpPacketTypeToString(RtpPacketType packet_type) {
+  switch (packet_type) {
+    case RtpPacketType::kRtp:
+      return "RTP";
+    case RtpPacketType::kRtcp:
+      return "RTCP";
+    case RtpPacketType::kUnknown:
+      return "Unknown";
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+RtpPacketType InferRtpPacketType(rtc::ArrayView<const char> packet) {
+  if (webrtc::IsRtcpPacket(
+          rtc::reinterpret_array_view<const uint8_t>(packet))) {
+    return RtpPacketType::kRtcp;
+  }
+  if (webrtc::IsRtpPacket(rtc::reinterpret_array_view<const uint8_t>(packet))) {
+    return RtpPacketType::kRtp;
+  }
+  return RtpPacketType::kUnknown;
+}
+
+bool ValidateRtpHeader(const uint8_t* rtp,
+                       size_t length,
+                       size_t* header_length) {
+  if (header_length) {
+    *header_length = 0;
+  }
+
+  if (length < kMinRtpPacketLen) {
+    return false;
+  }
+
+  size_t cc_count = rtp[0] & 0x0F;
+  size_t header_length_without_extension = kMinRtpPacketLen + 4 * cc_count;
+  if (header_length_without_extension > length) {
+    return false;
+  }
+
+  // If extension bit is not set, we are done with header processing, as input
+  // length is verified above.
+  if (!(rtp[0] & 0x10)) {
+    if (header_length)
+      *header_length = header_length_without_extension;
+
+    return true;
+  }
+
+  rtp += header_length_without_extension;
+
+  if (header_length_without_extension + kRtpExtensionHeaderLen > length) {
+    return false;
+  }
+
+  // Getting extension profile length.
+  // Length is in 32 bit words.
+  uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2);
+  size_t extension_length = extension_length_in_32bits * 4;
+
+  size_t rtp_header_length = extension_length +
+                             header_length_without_extension +
+                             kRtpExtensionHeaderLen;
+
+  // Verify input length against total header size.
+  if (rtp_header_length > length) {
+    return false;
+  }
+
+  if (header_length) {
+    *header_length = rtp_header_length;
+  }
+  return true;
+}
+
+// ValidateRtpHeader() must be called before this method to make sure, we have
+// a sane rtp packet.
+bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
+                                   size_t length,
+                                   int extension_id,
+                                   uint64_t time_us) {
+  //  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|X|  CC   |M|     PT      |       sequence number         |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |                           timestamp                           |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |           synchronization source (SSRC) identifier            |
+  // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+  // |            contributing source (CSRC) identifiers             |
+  // |                             ....                              |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+  // Return if extension bit is not set.
+  if (!(rtp[0] & 0x10)) {
+    return true;
+  }
+
+  size_t cc_count = rtp[0] & 0x0F;
+  size_t header_length_without_extension = kMinRtpPacketLen + 4 * cc_count;
+
+  rtp += header_length_without_extension;
+
+  // Getting extension profile ID and length.
+  uint16_t profile_id = rtc::GetBE16(rtp);
+  // Length is in 32 bit words.
+  uint16_t extension_length_in_32bits = rtc::GetBE16(rtp + 2);
+  size_t extension_length = extension_length_in_32bits * 4;
+
+  rtp += kRtpExtensionHeaderLen;  // Moving past extension header.
+
+  constexpr uint16_t kOneByteExtensionProfileId = 0xBEDE;
+  constexpr uint16_t kTwoByteExtensionProfileId = 0x1000;
+
+  bool found = false;
+  if (profile_id == kOneByteExtensionProfileId ||
+      profile_id == kTwoByteExtensionProfileId) {
+    // OneByte extension header
+    //  0
+    //  0 1 2 3 4 5 6 7
+    // +-+-+-+-+-+-+-+-+
+    // |  ID   |length |
+    // +-+-+-+-+-+-+-+-+
+
+    //  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
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |       0xBE    |    0xDE       |           length=3            |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |  ID   | L=0   |     data      |  ID   |  L=1  |   data...
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    //       ...data   |    0 (pad)    |    0 (pad)    |  ID   | L=3   |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |                          data                                 |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+    // TwoByte extension header
+    //  0
+    //  0 1 2 3 4 5 6 7
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |      ID       |    length     |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+    //  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
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |     0x10      |     0x00      |           length=3            |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |      ID       |      L=1      |     data      |      ID       |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |      L=2      |             data              |    0 (pad)    |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |      ID       |      L=2      |             data              |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+    size_t extension_header_length = profile_id == kOneByteExtensionProfileId
+                                         ? kOneByteExtensionHeaderLen
+                                         : kTwoByteExtensionHeaderLen;
+
+    const uint8_t* extension_start = rtp;
+    const uint8_t* extension_end = extension_start + extension_length;
+
+    // rtp + 1 since the minimum size per header extension is two bytes for both
+    // one- and two-byte header extensions.
+    while (rtp + 1 < extension_end) {
+      // See RFC8285 Section 4.2-4.3 for more information about one- and
+      // two-byte header extensions.
+      const int id =
+          profile_id == kOneByteExtensionProfileId ? (*rtp & 0xF0) >> 4 : *rtp;
+      const size_t length = profile_id == kOneByteExtensionProfileId
+                                ? (*rtp & 0x0F) + 1
+                                : *(rtp + 1);
+      if (rtp + extension_header_length + length > extension_end) {
+        return false;
+      }
+      if (id == extension_id) {
+        UpdateAbsSendTimeExtensionValue(rtp + extension_header_length, length,
+                                        time_us);
+        found = true;
+        break;
+      }
+      rtp += extension_header_length + length;
+      // Counting padding bytes.
+      while ((rtp < extension_end) && (*rtp == 0)) {
+        ++rtp;
+      }
+    }
+  }
+  return found;
+}
+
+bool ApplyPacketOptions(uint8_t* data,
+                        size_t length,
+                        const rtc::PacketTimeUpdateParams& packet_time_params,
+                        uint64_t time_us) {
+  RTC_DCHECK(data);
+  RTC_DCHECK(length);
+
+  // if there is no valid `rtp_sendtime_extension_id` and `srtp_auth_key` in
+  // PacketOptions, nothing to be updated in this packet.
+  if (packet_time_params.rtp_sendtime_extension_id == -1 &&
+      packet_time_params.srtp_auth_key.empty()) {
+    return true;
+  }
+
+  // If there is a srtp auth key present then the packet must be an RTP packet.
+  // RTP packet may have been wrapped in a TURN Channel Data or TURN send
+  // indication.
+  size_t rtp_start_pos;
+  size_t rtp_length;
+  if (!UnwrapTurnPacket(data, length, &rtp_start_pos, &rtp_length)) {
+    RTC_DCHECK_NOTREACHED();
+    return false;
+  }
+
+  // Making sure we have a valid RTP packet at the end.
+  auto packet = rtc::MakeArrayView(data + rtp_start_pos, rtp_length);
+  if (!webrtc::IsRtpPacket(packet) ||
+      !ValidateRtpHeader(data + rtp_start_pos, rtp_length, nullptr)) {
+    RTC_DCHECK_NOTREACHED();
+    return false;
+  }
+
+  uint8_t* start = data + rtp_start_pos;
+  // If packet option has non default value (-1) for sendtime extension id,
+  // then we should parse the rtp packet to update the timestamp. Otherwise
+  // just calculate HMAC and update packet with it.
+  if (packet_time_params.rtp_sendtime_extension_id != -1) {
+    UpdateRtpAbsSendTimeExtension(start, rtp_length,
+                                  packet_time_params.rtp_sendtime_extension_id,
+                                  time_us);
+  }
+
+  UpdateRtpAuthTag(start, rtp_length, packet_time_params);
+  return true;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/rtp_utils.h

@@ -0,0 +1,80 @@
+/*
+ *  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 MEDIA_BASE_RTP_UTILS_H_
+#define MEDIA_BASE_RTP_UTILS_H_
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "rtc_base/byte_order.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+struct PacketTimeUpdateParams;
+}  // namespace rtc
+
+namespace cricket {
+
+const size_t kMinRtpPacketLen = 12;
+const size_t kMaxRtpPacketLen = 2048;
+const size_t kMinRtcpPacketLen = 4;
+
+enum RtcpTypes {
+  kRtcpTypeSR = 200,     // Sender report payload type.
+  kRtcpTypeRR = 201,     // Receiver report payload type.
+  kRtcpTypeSDES = 202,   // SDES payload type.
+  kRtcpTypeBye = 203,    // BYE payload type.
+  kRtcpTypeApp = 204,    // APP payload type.
+  kRtcpTypeRTPFB = 205,  // Transport layer Feedback message payload type.
+  kRtcpTypePSFB = 206,   // Payload-specific Feedback message payload type.
+};
+
+enum class RtpPacketType {
+  kRtp,
+  kRtcp,
+  kUnknown,
+};
+
+bool GetRtcpType(const void* data, size_t len, int* value);
+bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value);
+
+// Checks the packet header to determine if it can be an RTP or RTCP packet.
+RtpPacketType InferRtpPacketType(rtc::ArrayView<const char> packet);
+// True if |payload type| is 0-127.
+bool IsValidRtpPayloadType(int payload_type);
+
+// True if `size` is appropriate for the indicated packet type.
+bool IsValidRtpPacketSize(RtpPacketType packet_type, size_t size);
+
+// Returns "RTCP", "RTP" or "Unknown" according to `packet_type`.
+absl::string_view RtpPacketTypeToString(RtpPacketType packet_type);
+
+// Verifies that a packet has a valid RTP header.
+bool RTC_EXPORT ValidateRtpHeader(const uint8_t* rtp,
+                                  size_t length,
+                                  size_t* header_length);
+
+// Helper method which updates the absolute send time extension if present.
+bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
+                                   size_t length,
+                                   int extension_id,
+                                   uint64_t time_us);
+
+// Applies specified `options` to the packet. It updates the absolute send time
+// extension header if it is present present then updates HMAC.
+bool RTC_EXPORT
+ApplyPacketOptions(uint8_t* data,
+                   size_t length,
+                   const rtc::PacketTimeUpdateParams& packet_time_params,
+                   uint64_t time_us);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_RTP_UTILS_H_

TMessagesProj/jni/voip/webrtc/media/base/sdp_video_format_utils.cc

@@ -0,0 +1,180 @@
+/*
+ *  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 "media/base/sdp_video_format_utils.h"
+
+#include <cstring>
+#include <map>
+#include <utility>
+
+#include "api/video_codecs/h264_profile_level_id.h"
+#ifdef RTC_ENABLE_H265
+#include "api/video_codecs/h265_profile_tier_level.h"
+#endif
+#include "rtc_base/checks.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+namespace {
+const char kProfileLevelId[] = "profile-level-id";
+const char kH264LevelAsymmetryAllowed[] = "level-asymmetry-allowed";
+// Max frame rate for VP8 and VP9 video.
+const char kVPxFmtpMaxFrameRate[] = "max-fr";
+// Max frame size for VP8 and VP9 video.
+const char kVPxFmtpMaxFrameSize[] = "max-fs";
+const int kVPxFmtpFrameSizeSubBlockPixels = 256;
+#ifdef RTC_ENABLE_H265
+constexpr char kH265ProfileId[] = "profile-id";
+constexpr char kH265TierFlag[] = "tier-flag";
+constexpr char kH265LevelId[] = "level-id";
+#endif
+
+bool IsH264LevelAsymmetryAllowed(const CodecParameterMap& params) {
+  const auto it = params.find(kH264LevelAsymmetryAllowed);
+  return it != params.end() && strcmp(it->second.c_str(), "1") == 0;
+}
+
+// Compare H264 levels and handle the level 1b case.
+bool H264LevelIsLess(H264Level a, H264Level b) {
+  if (a == H264Level::kLevel1_b)
+    return b != H264Level::kLevel1 && b != H264Level::kLevel1_b;
+  if (b == H264Level::kLevel1_b)
+    return a == H264Level::kLevel1;
+  return a < b;
+}
+
+H264Level H264LevelMin(H264Level a, H264Level b) {
+  return H264LevelIsLess(a, b) ? a : b;
+}
+
+absl::optional<int> ParsePositiveNumberFromParams(
+    const CodecParameterMap& params,
+    const char* parameter_name) {
+  const auto max_frame_rate_it = params.find(parameter_name);
+  if (max_frame_rate_it == params.end())
+    return absl::nullopt;
+
+  const absl::optional<int> i =
+      rtc::StringToNumber<int>(max_frame_rate_it->second);
+  if (!i.has_value() || i.value() <= 0)
+    return absl::nullopt;
+  return i;
+}
+
+#ifdef RTC_ENABLE_H265
+// Compares two H265Level and return the smaller.
+H265Level H265LevelMin(H265Level a, H265Level b) {
+  return a <= b ? a : b;
+}
+
+// Returns true if none of profile-id/tier-flag/level-id is specified
+// explicitly in the param.
+bool IsDefaultH265PTL(const CodecParameterMap& params) {
+  return !params.count(kH265ProfileId) && !params.count(kH265TierFlag) &&
+         !params.count(kH265LevelId);
+}
+#endif
+
+}  // namespace
+
+#ifdef RTC_ENABLE_H265
+// Set level according to https://tools.ietf.org/html/rfc7798#section-7.1
+void H265GenerateProfileTierLevelForAnswer(
+    const CodecParameterMap& local_supported_params,
+    const CodecParameterMap& remote_offered_params,
+    CodecParameterMap* answer_params) {
+  // If local and remote haven't set profile-id/tier-flag/level-id, they
+  // are both using the default PTL In this case, don't set PTL in answer
+  // either.
+  if (IsDefaultH265PTL(local_supported_params) &&
+      IsDefaultH265PTL(remote_offered_params)) {
+    return;
+  }
+
+  // Parse profile-tier-level.
+  const absl::optional<H265ProfileTierLevel> local_profile_tier_level =
+      ParseSdpForH265ProfileTierLevel(local_supported_params);
+  const absl::optional<H265ProfileTierLevel> remote_profile_tier_level =
+      ParseSdpForH265ProfileTierLevel(remote_offered_params);
+  // Profile and tier for local and remote codec must be valid and equal.
+  RTC_DCHECK(local_profile_tier_level);
+  RTC_DCHECK(remote_profile_tier_level);
+  RTC_DCHECK_EQ(local_profile_tier_level->profile,
+                remote_profile_tier_level->profile);
+  RTC_DCHECK_EQ(local_profile_tier_level->tier,
+                remote_profile_tier_level->tier);
+
+  const H265Level answer_level = H265LevelMin(local_profile_tier_level->level,
+                                              remote_profile_tier_level->level);
+
+  // Level-id in answer is changable as long as the highest level indicated by
+  // the answer is not higher than that indicated by the offer. See
+  // https://tools.ietf.org/html/rfc7798#section-7.2.2, sub-clause 2.
+  (*answer_params)[kH265LevelId] = H265LevelToString(answer_level);
+}
+#endif
+
+// Set level according to https://tools.ietf.org/html/rfc6184#section-8.2.2.
+void H264GenerateProfileLevelIdForAnswer(
+    const CodecParameterMap& local_supported_params,
+    const CodecParameterMap& remote_offered_params,
+    CodecParameterMap* answer_params) {
+  // If both local and remote haven't set profile-level-id, they are both using
+  // the default profile. In this case, don't set profile-level-id in answer
+  // either.
+  if (!local_supported_params.count(kProfileLevelId) &&
+      !remote_offered_params.count(kProfileLevelId)) {
+    return;
+  }
+
+  // Parse profile-level-ids.
+  const absl::optional<H264ProfileLevelId> local_profile_level_id =
+      ParseSdpForH264ProfileLevelId(local_supported_params);
+  const absl::optional<H264ProfileLevelId> remote_profile_level_id =
+      ParseSdpForH264ProfileLevelId(remote_offered_params);
+  // The local and remote codec must have valid and equal H264 Profiles.
+  RTC_DCHECK(local_profile_level_id);
+  RTC_DCHECK(remote_profile_level_id);
+  RTC_DCHECK_EQ(local_profile_level_id->profile,
+                remote_profile_level_id->profile);
+
+  // Parse level information.
+  const bool level_asymmetry_allowed =
+      IsH264LevelAsymmetryAllowed(local_supported_params) &&
+      IsH264LevelAsymmetryAllowed(remote_offered_params);
+  const H264Level local_level = local_profile_level_id->level;
+  const H264Level remote_level = remote_profile_level_id->level;
+  const H264Level min_level = H264LevelMin(local_level, remote_level);
+
+  // Determine answer level. When level asymmetry is not allowed, level upgrade
+  // is not allowed, i.e., the level in the answer must be equal to or lower
+  // than the level in the offer.
+  const H264Level answer_level =
+      level_asymmetry_allowed ? local_level : min_level;
+
+  // Set the resulting profile-level-id in the answer parameters.
+  (*answer_params)[kProfileLevelId] = *H264ProfileLevelIdToString(
+      H264ProfileLevelId(local_profile_level_id->profile, answer_level));
+}
+
+absl::optional<int> ParseSdpForVPxMaxFrameRate(
+    const CodecParameterMap& params) {
+  return ParsePositiveNumberFromParams(params, kVPxFmtpMaxFrameRate);
+}
+
+absl::optional<int> ParseSdpForVPxMaxFrameSize(
+    const CodecParameterMap& params) {
+  const absl::optional<int> i =
+      ParsePositiveNumberFromParams(params, kVPxFmtpMaxFrameSize);
+  return i ? absl::make_optional(i.value() * kVPxFmtpFrameSizeSubBlockPixels)
+           : absl::nullopt;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/base/sdp_video_format_utils.h

@@ -0,0 +1,62 @@
+/*
+ *  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 MEDIA_BASE_SDP_VIDEO_FORMAT_UTILS_H_
+#define MEDIA_BASE_SDP_VIDEO_FORMAT_UTILS_H_
+
+#include "absl/types/optional.h"
+#include "api/video_codecs/sdp_video_format.h"
+
+namespace webrtc {
+// Generate codec parameters that will be used as answer in an SDP negotiation
+// based on local supported parameters and remote offered parameters. Both
+// `local_supported_params`, `remote_offered_params`, and `answer_params`
+// represent sendrecv media descriptions, i.e they are a mix of both encode and
+// decode capabilities. In theory, when the profile in `local_supported_params`
+// represent a strict superset of the profile in `remote_offered_params`, we
+// could limit the profile in `answer_params` to the profile in
+// `remote_offered_params`. However, to simplify the code, each supported H264
+// profile should be listed explicitly in the list of local supported codecs,
+// even if they are redundant. Then each local codec in the list should be
+// tested one at a time against the remote codec, and only when the profiles are
+// equal should this function be called. Therefore, this function does not need
+// to handle profile intersection, and the profile of `local_supported_params`
+// and `remote_offered_params` must be equal before calling this function. The
+// parameters that are used when negotiating are the level part of
+// profile-level-id and level-asymmetry-allowed.
+void H264GenerateProfileLevelIdForAnswer(
+    const CodecParameterMap& local_supported_params,
+    const CodecParameterMap& remote_offered_params,
+    CodecParameterMap* answer_params);
+
+#ifdef RTC_ENABLE_H265
+// Works similarly as H264GenerateProfileLevelIdForAnswer, but generates codec
+// parameters that will be used as answer for H.265.
+// Media configuration parameters, except level-id, must be used symmetrically.
+// For level-id, the highest level indicated by the answer must not be higher
+// than that indicated by the offer.
+void H265GenerateProfileTierLevelForAnswer(
+    const CodecParameterMap& local_supported_params,
+    const CodecParameterMap& remote_offered_params,
+    CodecParameterMap* answer_params);
+#endif
+
+// Parse max frame rate from SDP FMTP line. absl::nullopt is returned if the
+// field is missing or not a number.
+absl::optional<int> ParseSdpForVPxMaxFrameRate(const CodecParameterMap& params);
+
+// Parse max frame size from SDP FMTP line. absl::nullopt is returned if the
+// field is missing or not a number. Please note that the value is stored in sub
+// blocks but the returned value is in total number of pixels.
+absl::optional<int> ParseSdpForVPxMaxFrameSize(const CodecParameterMap& params);
+
+}  // namespace webrtc
+
+#endif  // MEDIA_BASE_SDP_VIDEO_FORMAT_UTILS_H_

TMessagesProj/jni/voip/webrtc/media/base/stream_params.cc

@@ -0,0 +1,240 @@
+/*
+ *  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 "media/base/stream_params.h"
+
+#include <stdint.h>
+
+#include <list>
+
+#include "absl/algorithm/container.h"
+#include "api/array_view.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace cricket {
+namespace {
+
+void AppendSsrcs(rtc::ArrayView<const uint32_t> ssrcs,
+                 rtc::SimpleStringBuilder* sb) {
+  *sb << "ssrcs:[";
+  const char* delimiter = "";
+  for (uint32_t ssrc : ssrcs) {
+    *sb << delimiter << ssrc;
+    delimiter = ",";
+  }
+  *sb << "]";
+}
+
+void AppendSsrcGroups(rtc::ArrayView<const SsrcGroup> ssrc_groups,
+                      rtc::SimpleStringBuilder* sb) {
+  *sb << "ssrc_groups:";
+  const char* delimiter = "";
+  for (const SsrcGroup& ssrc_group : ssrc_groups) {
+    *sb << delimiter << ssrc_group.ToString();
+    delimiter = ",";
+  }
+}
+
+void AppendStreamIds(rtc::ArrayView<const std::string> stream_ids,
+                     rtc::SimpleStringBuilder* sb) {
+  *sb << "stream_ids:";
+  const char* delimiter = "";
+  for (const std::string& stream_id : stream_ids) {
+    *sb << delimiter << stream_id;
+    delimiter = ",";
+  }
+}
+
+void AppendRids(rtc::ArrayView<const RidDescription> rids,
+                rtc::SimpleStringBuilder* sb) {
+  *sb << "rids:[";
+  const char* delimiter = "";
+  for (const RidDescription& rid : rids) {
+    *sb << delimiter << rid.rid;
+    delimiter = ",";
+  }
+  *sb << "]";
+}
+
+}  // namespace
+
+const char kFecSsrcGroupSemantics[] = "FEC";
+const char kFecFrSsrcGroupSemantics[] = "FEC-FR";
+const char kFidSsrcGroupSemantics[] = "FID";
+const char kSimSsrcGroupSemantics[] = "SIM";
+
+bool GetStream(const StreamParamsVec& streams,
+               const StreamSelector& selector,
+               StreamParams* stream_out) {
+  const StreamParams* found = GetStream(streams, selector);
+  if (found && stream_out)
+    *stream_out = *found;
+  return found != nullptr;
+}
+
+SsrcGroup::SsrcGroup(const std::string& usage,
+                     const std::vector<uint32_t>& ssrcs)
+    : semantics(usage), ssrcs(ssrcs) {}
+SsrcGroup::SsrcGroup(const SsrcGroup&) = default;
+SsrcGroup::SsrcGroup(SsrcGroup&&) = default;
+SsrcGroup::~SsrcGroup() = default;
+
+SsrcGroup& SsrcGroup::operator=(const SsrcGroup&) = default;
+SsrcGroup& SsrcGroup::operator=(SsrcGroup&&) = default;
+
+bool SsrcGroup::has_semantics(const std::string& semantics_in) const {
+  return (semantics == semantics_in && ssrcs.size() > 0);
+}
+
+std::string SsrcGroup::ToString() const {
+  char buf[1024];
+  rtc::SimpleStringBuilder sb(buf);
+  sb << "{";
+  sb << "semantics:" << semantics << ";";
+  AppendSsrcs(ssrcs, &sb);
+  sb << "}";
+  return sb.str();
+}
+
+StreamParams::StreamParams() = default;
+StreamParams::StreamParams(const StreamParams&) = default;
+StreamParams::StreamParams(StreamParams&&) = default;
+StreamParams::~StreamParams() = default;
+StreamParams& StreamParams::operator=(const StreamParams&) = default;
+StreamParams& StreamParams::operator=(StreamParams&&) = default;
+
+bool StreamParams::operator==(const StreamParams& other) const {
+  return (id == other.id && ssrcs == other.ssrcs &&
+          ssrc_groups == other.ssrc_groups && cname == other.cname &&
+          stream_ids_ == other.stream_ids_ &&
+          // RIDs are not required to be in the same order for equality.
+          absl::c_is_permutation(rids_, other.rids_));
+}
+
+std::string StreamParams::ToString() const {
+  char buf[2 * 1024];
+  rtc::SimpleStringBuilder sb(buf);
+  sb << "{";
+  if (!id.empty()) {
+    sb << "id:" << id << ";";
+  }
+  AppendSsrcs(ssrcs, &sb);
+  sb << ";";
+  AppendSsrcGroups(ssrc_groups, &sb);
+  sb << ";";
+  if (!cname.empty()) {
+    sb << "cname:" << cname << ";";
+  }
+  AppendStreamIds(stream_ids_, &sb);
+  sb << ";";
+  if (!rids_.empty()) {
+    AppendRids(rids_, &sb);
+    sb << ";";
+  }
+  sb << "}";
+  return sb.str();
+}
+
+void StreamParams::GenerateSsrcs(int num_layers,
+                                 bool generate_fid,
+                                 bool generate_fec_fr,
+                                 rtc::UniqueRandomIdGenerator* ssrc_generator) {
+  RTC_DCHECK_GE(num_layers, 0);
+  RTC_DCHECK(ssrc_generator);
+  std::vector<uint32_t> primary_ssrcs;
+  for (int i = 0; i < num_layers; ++i) {
+    uint32_t ssrc = ssrc_generator->GenerateId();
+    primary_ssrcs.push_back(ssrc);
+    add_ssrc(ssrc);
+  }
+
+  if (num_layers > 1) {
+    SsrcGroup simulcast(kSimSsrcGroupSemantics, primary_ssrcs);
+    ssrc_groups.push_back(simulcast);
+  }
+
+  if (generate_fid) {
+    for (uint32_t ssrc : primary_ssrcs) {
+      AddFidSsrc(ssrc, ssrc_generator->GenerateId());
+    }
+  }
+
+  if (generate_fec_fr) {
+    for (uint32_t ssrc : primary_ssrcs) {
+      AddFecFrSsrc(ssrc, ssrc_generator->GenerateId());
+    }
+  }
+}
+
+void StreamParams::GetPrimarySsrcs(std::vector<uint32_t>* ssrcs) const {
+  const SsrcGroup* sim_group = get_ssrc_group(kSimSsrcGroupSemantics);
+  if (sim_group == NULL) {
+    ssrcs->push_back(first_ssrc());
+  } else {
+    ssrcs->insert(ssrcs->end(), sim_group->ssrcs.begin(),
+                  sim_group->ssrcs.end());
+  }
+}
+
+void StreamParams::GetSecondarySsrcs(
+    const std::string& semantics,
+    const std::vector<uint32_t>& primary_ssrcs,
+    std::vector<uint32_t>* secondary_ssrcs) const {
+  for (uint32_t primary_ssrc : primary_ssrcs) {
+    uint32_t secondary_ssrc;
+    if (GetSecondarySsrc(semantics, primary_ssrc, &secondary_ssrc)) {
+      secondary_ssrcs->push_back(secondary_ssrc);
+    }
+  }
+}
+
+void StreamParams::GetFidSsrcs(const std::vector<uint32_t>& primary_ssrcs,
+                               std::vector<uint32_t>* fid_ssrcs) const {
+  return GetSecondarySsrcs(kFidSsrcGroupSemantics, primary_ssrcs, fid_ssrcs);
+}
+
+bool StreamParams::AddSecondarySsrc(const std::string& semantics,
+                                    uint32_t primary_ssrc,
+                                    uint32_t secondary_ssrc) {
+  if (!has_ssrc(primary_ssrc)) {
+    return false;
+  }
+
+  ssrcs.push_back(secondary_ssrc);
+  ssrc_groups.push_back(SsrcGroup(semantics, {primary_ssrc, secondary_ssrc}));
+  return true;
+}
+
+bool StreamParams::GetSecondarySsrc(const std::string& semantics,
+                                    uint32_t primary_ssrc,
+                                    uint32_t* secondary_ssrc) const {
+  for (const SsrcGroup& ssrc_group : ssrc_groups) {
+    if (ssrc_group.has_semantics(semantics) && ssrc_group.ssrcs.size() >= 2 &&
+        ssrc_group.ssrcs[0] == primary_ssrc) {
+      *secondary_ssrc = ssrc_group.ssrcs[1];
+      return true;
+    }
+  }
+  return false;
+}
+
+std::vector<std::string> StreamParams::stream_ids() const {
+  return stream_ids_;
+}
+
+void StreamParams::set_stream_ids(const std::vector<std::string>& stream_ids) {
+  stream_ids_ = stream_ids;
+}
+
+std::string StreamParams::first_stream_id() const {
+  return stream_ids_.empty() ? "" : stream_ids_[0];
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/stream_params.h

@@ -0,0 +1,321 @@
+/*
+ *  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.
+ */
+
+// This file contains structures for describing SSRCs from a media source such
+// as a MediaStreamTrack when it is sent across an RTP session. Multiple media
+// sources may be sent across the same RTP session, each of them will be
+// described by one StreamParams object
+// SsrcGroup is used to describe the relationship between the SSRCs that
+// are used for this media source.
+// E.x: Consider a source that is sent as 3 simulcast streams
+// Let the simulcast elements have SSRC 10, 20, 30.
+// Let each simulcast element use FEC and let the protection packets have
+// SSRC 11,21,31.
+// To describe this 4 SsrcGroups are needed,
+// StreamParams would then contain ssrc = {10,11,20,21,30,31} and
+// ssrc_groups = {{SIM,{10,20,30}, {FEC,{10,11}, {FEC, {20,21}, {FEC {30,31}}}
+// Please see RFC 5576.
+// A spec-compliant way to achieve this is to use RIDs and Simulcast attribute
+// instead of the ssrc-group. In this method, the StreamParam object will
+// have multiple RidDescriptions, each corresponding to a simulcast layer
+// and the media section will have a simulcast attribute that indicates
+// that these layers are for the same source. This also removes the extra
+// lines for redundancy streams, as the same RIDs appear in the redundancy
+// packets.
+// Note: in the spec compliant simulcast scenario, some of the RIDs might be
+// alternatives for one another (such as different encodings for same data).
+// In the context of the StreamParams class, the notion of alternatives does
+// not exist and all the RIDs will describe different layers of the same source.
+// When the StreamParams class is used to configure the media engine, simulcast
+// considerations will be used to remove the alternative layers outside of this
+// class.
+// As an example, let the simulcast layers have RID 10, 20, 30.
+// StreamParams would contain rid = { 10, 20, 30 }.
+// MediaSection would contain SimulcastDescription specifying these rids.
+// a=simulcast:send 10;20;30 (or a=simulcast:send 10,20;30 or similar).
+// See https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-13
+// and https://tools.ietf.org/html/draft-ietf-mmusic-rid-15.
+
+#ifndef MEDIA_BASE_STREAM_PARAMS_H_
+#define MEDIA_BASE_STREAM_PARAMS_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "media/base/rid_description.h"
+#include "rtc_base/unique_id_generator.h"
+
+namespace cricket {
+
+extern const char kFecSsrcGroupSemantics[];
+extern const char kFecFrSsrcGroupSemantics[];
+extern const char kFidSsrcGroupSemantics[];
+extern const char kSimSsrcGroupSemantics[];
+
+struct SsrcGroup {
+  SsrcGroup(const std::string& usage, const std::vector<uint32_t>& ssrcs);
+  SsrcGroup(const SsrcGroup&);
+  SsrcGroup(SsrcGroup&&);
+  ~SsrcGroup();
+  SsrcGroup& operator=(const SsrcGroup&);
+  SsrcGroup& operator=(SsrcGroup&&);
+
+  bool operator==(const SsrcGroup& other) const {
+    return (semantics == other.semantics && ssrcs == other.ssrcs);
+  }
+  bool operator!=(const SsrcGroup& other) const { return !(*this == other); }
+
+  bool has_semantics(const std::string& semantics) const;
+
+  std::string ToString() const;
+
+  std::string semantics;        // e.g FIX, FEC, SIM.
+  std::vector<uint32_t> ssrcs;  // SSRCs of this type.
+};
+
+// StreamParams is used to represent a sender/track in a SessionDescription.
+// In Plan B, this means that multiple StreamParams can exist within one
+// MediaContentDescription, while in UnifiedPlan this means that there is one
+// StreamParams per MediaContentDescription.
+struct StreamParams {
+  StreamParams();
+  StreamParams(const StreamParams&);
+  StreamParams(StreamParams&&);
+  ~StreamParams();
+  StreamParams& operator=(const StreamParams&);
+  StreamParams& operator=(StreamParams&&);
+
+  static StreamParams CreateLegacy(uint32_t ssrc) {
+    StreamParams stream;
+    stream.ssrcs.push_back(ssrc);
+    return stream;
+  }
+
+  bool operator==(const StreamParams& other) const;
+  bool operator!=(const StreamParams& other) const { return !(*this == other); }
+
+  uint32_t first_ssrc() const {
+    if (ssrcs.empty()) {
+      return 0;
+    }
+
+    return ssrcs[0];
+  }
+  bool has_ssrcs() const { return !ssrcs.empty(); }
+  bool has_ssrc(uint32_t ssrc) const {
+    return absl::c_linear_search(ssrcs, ssrc);
+  }
+  void add_ssrc(uint32_t ssrc) { ssrcs.push_back(ssrc); }
+  bool has_ssrc_groups() const { return !ssrc_groups.empty(); }
+  bool has_ssrc_group(const std::string& semantics) const {
+    return (get_ssrc_group(semantics) != NULL);
+  }
+  const SsrcGroup* get_ssrc_group(const std::string& semantics) const {
+    for (const SsrcGroup& ssrc_group : ssrc_groups) {
+      if (ssrc_group.has_semantics(semantics)) {
+        return &ssrc_group;
+      }
+    }
+    return NULL;
+  }
+
+  // Convenience function to add an FID ssrc for a primary_ssrc
+  // that's already been added.
+  bool AddFidSsrc(uint32_t primary_ssrc, uint32_t fid_ssrc) {
+    return AddSecondarySsrc(kFidSsrcGroupSemantics, primary_ssrc, fid_ssrc);
+  }
+
+  // Convenience function to lookup the FID ssrc for a primary_ssrc.
+  // Returns false if primary_ssrc not found or FID not defined for it.
+  bool GetFidSsrc(uint32_t primary_ssrc, uint32_t* fid_ssrc) const {
+    return GetSecondarySsrc(kFidSsrcGroupSemantics, primary_ssrc, fid_ssrc);
+  }
+
+  // Convenience function to add an FEC-FR ssrc for a primary_ssrc
+  // that's already been added.
+  bool AddFecFrSsrc(uint32_t primary_ssrc, uint32_t fecfr_ssrc) {
+    return AddSecondarySsrc(kFecFrSsrcGroupSemantics, primary_ssrc, fecfr_ssrc);
+  }
+
+  // Convenience function to lookup the FEC-FR ssrc for a primary_ssrc.
+  // Returns false if primary_ssrc not found or FEC-FR not defined for it.
+  bool GetFecFrSsrc(uint32_t primary_ssrc, uint32_t* fecfr_ssrc) const {
+    return GetSecondarySsrc(kFecFrSsrcGroupSemantics, primary_ssrc, fecfr_ssrc);
+  }
+
+  // Convenience function to populate the StreamParams with the requested number
+  // of SSRCs along with accompanying FID and FEC-FR ssrcs if requested.
+  // SSRCs are generated using the given generator.
+  void GenerateSsrcs(int num_layers,
+                     bool generate_fid,
+                     bool generate_fec_fr,
+                     rtc::UniqueRandomIdGenerator* ssrc_generator);
+
+  // Convenience to get all the SIM SSRCs if there are SIM ssrcs, or
+  // the first SSRC otherwise.
+  void GetPrimarySsrcs(std::vector<uint32_t>* ssrcs) const;
+
+  // Convenience to get all the secondary SSRCs for the given primary ssrcs
+  // of a particular semantic.
+  // If a given primary SSRC does not have a secondary SSRC, the list of
+  // secondary SSRCS will be smaller than the list of primary SSRCs.
+  void GetSecondarySsrcs(const std::string& semantic,
+                         const std::vector<uint32_t>& primary_ssrcs,
+                         std::vector<uint32_t>* fid_ssrcs) const;
+
+  // Convenience to get all the FID SSRCs for the given primary ssrcs.
+  // If a given primary SSRC does not have a FID SSRC, the list of FID
+  // SSRCS will be smaller than the list of primary SSRCs.
+  void GetFidSsrcs(const std::vector<uint32_t>& primary_ssrcs,
+                   std::vector<uint32_t>* fid_ssrcs) const;
+
+  // Stream ids serialized to SDP.
+  std::vector<std::string> stream_ids() const;
+  void set_stream_ids(const std::vector<std::string>& stream_ids);
+
+  // Returns the first stream id or "" if none exist. This method exists only
+  // as temporary backwards compatibility with the old sync_label.
+  std::string first_stream_id() const;
+
+  std::string ToString() const;
+
+  // A unique identifier of the StreamParams object. When the SDP is created,
+  // this comes from the track ID of the sender that the StreamParams object
+  // is associated with.
+  std::string id;
+  // There may be no SSRCs stored in unsignaled case when stream_ids are
+  // signaled with a=msid lines.
+  std::vector<uint32_t> ssrcs;         // All SSRCs for this source
+  std::vector<SsrcGroup> ssrc_groups;  // e.g. FID, FEC, SIM
+  std::string cname;                   // RTCP CNAME
+
+  // RID functionality according to
+  // https://tools.ietf.org/html/draft-ietf-mmusic-rid-15
+  // Each layer can be represented by a RID identifier and can also have
+  // restrictions (such as max-width, max-height, etc.)
+  // If the track has multiple layers (ex. Simulcast), each layer will be
+  // represented by a RID.
+  bool has_rids() const { return !rids_.empty(); }
+  const std::vector<RidDescription>& rids() const { return rids_; }
+  void set_rids(const std::vector<RidDescription>& rids) { rids_ = rids; }
+
+ private:
+  bool AddSecondarySsrc(const std::string& semantics,
+                        uint32_t primary_ssrc,
+                        uint32_t secondary_ssrc);
+  bool GetSecondarySsrc(const std::string& semantics,
+                        uint32_t primary_ssrc,
+                        uint32_t* secondary_ssrc) const;
+
+  // The stream IDs of the sender that the StreamParams object is associated
+  // with. In Plan B this should always be size of 1, while in Unified Plan this
+  // could be none or multiple stream IDs.
+  std::vector<std::string> stream_ids_;
+
+  std::vector<RidDescription> rids_;
+};
+
+// A Stream can be selected by either id or ssrc.
+struct StreamSelector {
+  explicit StreamSelector(uint32_t ssrc) : ssrc(ssrc) {}
+
+  explicit StreamSelector(const std::string& streamid)
+      : ssrc(0), streamid(streamid) {}
+
+  bool Matches(const StreamParams& stream) const {
+    if (ssrc == 0) {
+      return stream.id == streamid;
+    } else {
+      return stream.has_ssrc(ssrc);
+    }
+  }
+
+  uint32_t ssrc;
+  std::string streamid;
+};
+
+typedef std::vector<StreamParams> StreamParamsVec;
+
+template <class Condition>
+const StreamParams* GetStream(const StreamParamsVec& streams,
+                              Condition condition) {
+  auto found = absl::c_find_if(streams, condition);
+  return found == streams.end() ? nullptr : &(*found);
+}
+
+template <class Condition>
+StreamParams* GetStream(StreamParamsVec& streams, Condition condition) {
+  auto found = absl::c_find_if(streams, condition);
+  return found == streams.end() ? nullptr : &(*found);
+}
+
+inline bool HasStreamWithNoSsrcs(const StreamParamsVec& streams) {
+  return GetStream(streams,
+                   [](const StreamParams& sp) { return !sp.has_ssrcs(); });
+}
+
+inline const StreamParams* GetStreamBySsrc(const StreamParamsVec& streams,
+                                           uint32_t ssrc) {
+  return GetStream(
+      streams, [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); });
+}
+
+inline const StreamParams* GetStreamByIds(const StreamParamsVec& streams,
+                                          const std::string& id) {
+  return GetStream(streams,
+                   [&id](const StreamParams& sp) { return sp.id == id; });
+}
+
+inline StreamParams* GetStreamByIds(StreamParamsVec& streams,
+                                    const std::string& id) {
+  return GetStream(streams,
+                   [&id](const StreamParams& sp) { return sp.id == id; });
+}
+
+inline const StreamParams* GetStream(const StreamParamsVec& streams,
+                                     const StreamSelector& selector) {
+  return GetStream(streams, [&selector](const StreamParams& sp) {
+    return selector.Matches(sp);
+  });
+}
+
+template <class Condition>
+bool RemoveStream(StreamParamsVec* streams, Condition condition) {
+  auto iter(std::remove_if(streams->begin(), streams->end(), condition));
+  if (iter == streams->end())
+    return false;
+  streams->erase(iter, streams->end());
+  return true;
+}
+
+// Removes the stream from streams. Returns true if a stream is
+// found and removed.
+inline bool RemoveStream(StreamParamsVec* streams,
+                         const StreamSelector& selector) {
+  return RemoveStream(streams, [&selector](const StreamParams& sp) {
+    return selector.Matches(sp);
+  });
+}
+inline bool RemoveStreamBySsrc(StreamParamsVec* streams, uint32_t ssrc) {
+  return RemoveStream(
+      streams, [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); });
+}
+inline bool RemoveStreamByIds(StreamParamsVec* streams, const std::string& id) {
+  return RemoveStream(streams,
+                      [&id](const StreamParams& sp) { return sp.id == id; });
+}
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_STREAM_PARAMS_H_

TMessagesProj/jni/voip/webrtc/media/base/test_utils.cc

@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2004 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 "media/base/test_utils.h"
+
+#include <cstdint>
+
+#include "api/video/video_frame.h"
+#include "api/video/video_source_interface.h"
+
+namespace cricket {
+
+cricket::StreamParams CreateSimStreamParams(
+    const std::string& cname,
+    const std::vector<uint32_t>& ssrcs) {
+  cricket::StreamParams sp;
+  cricket::SsrcGroup sg(cricket::kSimSsrcGroupSemantics, ssrcs);
+  sp.ssrcs = ssrcs;
+  sp.ssrc_groups.push_back(sg);
+  sp.cname = cname;
+  return sp;
+}
+
+// There should be an rtx_ssrc per ssrc.
+cricket::StreamParams CreateSimWithRtxStreamParams(
+    const std::string& cname,
+    const std::vector<uint32_t>& ssrcs,
+    const std::vector<uint32_t>& rtx_ssrcs) {
+  cricket::StreamParams sp = CreateSimStreamParams(cname, ssrcs);
+  for (size_t i = 0; i < ssrcs.size(); ++i) {
+    sp.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]);
+  }
+  return sp;
+}
+
+// There should be one fec ssrc per ssrc.
+cricket::StreamParams CreatePrimaryWithFecFrStreamParams(
+    const std::string& cname,
+    uint32_t primary_ssrc,
+    uint32_t flexfec_ssrc) {
+  cricket::StreamParams sp;
+  sp.ssrcs = {primary_ssrc};
+  sp.cname = cname;
+  sp.AddFecFrSsrc(primary_ssrc, flexfec_ssrc);
+  return sp;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/test_utils.h

@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2004 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 MEDIA_BASE_TEST_UTILS_H_
+#define MEDIA_BASE_TEST_UTILS_H_
+
+#include <string>
+#include <vector>
+
+#include "media/base/media_channel.h"
+#include "media/base/video_common.h"
+#include "rtc_base/arraysize.h"
+
+namespace webrtc {
+class VideoFrame;
+}
+
+namespace cricket {
+
+// Returns size of 420 image with rounding on chroma for odd sizes.
+#define I420_SIZE(w, h) (w * h + (((w + 1) / 2) * ((h + 1) / 2)) * 2)
+// Returns size of ARGB image.
+#define ARGB_SIZE(w, h) (w * h * 4)
+
+template <class T>
+inline std::vector<T> MakeVector(const T a[], size_t s) {
+  return std::vector<T>(a, a + s);
+}
+#define MAKE_VECTOR(a) cricket::MakeVector(a, arraysize(a))
+
+// Create Simulcast StreamParams with given `ssrcs` and `cname`.
+cricket::StreamParams CreateSimStreamParams(const std::string& cname,
+                                            const std::vector<uint32_t>& ssrcs);
+// Create Simulcast stream with given `ssrcs` and `rtx_ssrcs`.
+// The number of `rtx_ssrcs` must match number of `ssrcs`.
+cricket::StreamParams CreateSimWithRtxStreamParams(
+    const std::string& cname,
+    const std::vector<uint32_t>& ssrcs,
+    const std::vector<uint32_t>& rtx_ssrcs);
+
+// Create StreamParams with single primary SSRC and corresponding FlexFEC SSRC.
+cricket::StreamParams CreatePrimaryWithFecFrStreamParams(
+    const std::string& cname,
+    uint32_t primary_ssrc,
+    uint32_t flexfec_ssrc);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_TEST_UTILS_H_

TMessagesProj/jni/voip/webrtc/media/base/turn_utils.cc

@@ -0,0 +1,126 @@
+/*
+ *  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 "media/base/turn_utils.h"
+
+#include "api/transport/stun.h"
+#include "rtc_base/byte_order.h"
+
+namespace cricket {
+
+namespace {
+
+const size_t kTurnChannelHeaderLength = 4;
+
+bool IsTurnChannelData(const uint8_t* data, size_t length) {
+  return length >= kTurnChannelHeaderLength && ((*data & 0xC0) == 0x40);
+}
+
+bool IsTurnSendIndicationPacket(const uint8_t* data, size_t length) {
+  if (length < kStunHeaderSize) {
+    return false;
+  }
+
+  uint16_t type = rtc::GetBE16(data);
+  return (type == TURN_SEND_INDICATION);
+}
+
+}  // namespace
+
+bool UnwrapTurnPacket(const uint8_t* packet,
+                      size_t packet_size,
+                      size_t* content_position,
+                      size_t* content_size) {
+  if (IsTurnChannelData(packet, packet_size)) {
+    // Turn Channel Message 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
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |         Channel Number        |            Length             |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |                                                               |
+    // /                       Application Data                        /
+    // /                                                               /
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    size_t length = rtc::GetBE16(&packet[2]);
+    if (length + kTurnChannelHeaderLength > packet_size) {
+      return false;
+    }
+
+    *content_position = kTurnChannelHeaderLength;
+    *content_size = length;
+    return true;
+  }
+
+  if (IsTurnSendIndicationPacket(packet, packet_size)) {
+    // Validate STUN message length.
+    const size_t stun_message_length = rtc::GetBE16(&packet[2]);
+    if (stun_message_length + kStunHeaderSize != packet_size) {
+      return false;
+    }
+
+    // First skip mandatory stun header which is of 20 bytes.
+    size_t pos = kStunHeaderSize;
+    // Loop through STUN attributes until we find STUN DATA attribute.
+    while (pos < packet_size) {
+      // Keep reading STUN attributes until we hit DATA attribute.
+      // Attribute will be a TLV structure.
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      // |         Type                  |            Length             |
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      // |                         Value (variable)                ....
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      // The value in the length field MUST contain the length of the Value
+      // part of the attribute, prior to padding, measured in bytes.  Since
+      // STUN aligns attributes on 32-bit boundaries, attributes whose content
+      // is not a multiple of 4 bytes are padded with 1, 2, or 3 bytes of
+      // padding so that its value contains a multiple of 4 bytes.  The
+      // padding bits are ignored, and may be any value.
+      uint16_t attr_type, attr_length;
+      const int kAttrHeaderLength = sizeof(attr_type) + sizeof(attr_length);
+
+      if (packet_size < pos + kAttrHeaderLength) {
+        return false;
+      }
+
+      // Getting attribute type and length.
+      attr_type = rtc::GetBE16(&packet[pos]);
+      attr_length = rtc::GetBE16(&packet[pos + sizeof(attr_type)]);
+
+      pos += kAttrHeaderLength;  // Skip STUN_DATA_ATTR header.
+
+      // Checking for bogus attribute length.
+      if (pos + attr_length > packet_size) {
+        return false;
+      }
+
+      if (attr_type == STUN_ATTR_DATA) {
+        *content_position = pos;
+        *content_size = attr_length;
+        return true;
+      }
+
+      pos += attr_length;
+      if ((attr_length % 4) != 0) {
+        pos += (4 - (attr_length % 4));
+      }
+    }
+
+    // There is no data attribute present in the message.
+    return false;
+  }
+
+  // This is not a TURN packet.
+  *content_position = 0;
+  *content_size = packet_size;
+  return true;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/turn_utils.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 MEDIA_BASE_TURN_UTILS_H_
+#define MEDIA_BASE_TURN_UTILS_H_
+
+#include <cstddef>
+#include <cstdint>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace cricket {
+
+// Finds data location within a TURN Channel Message or TURN Send Indication
+// message.
+bool RTC_EXPORT UnwrapTurnPacket(const uint8_t* packet,
+                                 size_t packet_size,
+                                 size_t* content_position,
+                                 size_t* content_size);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_TURN_UTILS_H_

TMessagesProj/jni/voip/webrtc/media/base/video_adapter.cc

@@ -0,0 +1,470 @@
+/*
+ *  Copyright (c) 2010 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 "media/base/video_adapter.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstdint>
+#include <cstdlib>
+#include <limits>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "media/base/video_common.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace {
+
+struct Fraction {
+  int numerator;
+  int denominator;
+
+  void DivideByGcd() {
+    int g = cricket::GreatestCommonDivisor(numerator, denominator);
+    numerator /= g;
+    denominator /= g;
+  }
+
+  // Determines number of output pixels if both width and height of an input of
+  // `input_pixels` pixels is scaled with the fraction numerator / denominator.
+  int scale_pixel_count(int input_pixels) {
+    return (numerator * numerator * static_cast<int64_t>(input_pixels)) /
+           (denominator * denominator);
+  }
+};
+
+// Round `value_to_round` to a multiple of `multiple`. Prefer rounding upwards,
+// but never more than `max_value`.
+int roundUp(int value_to_round, int multiple, int max_value) {
+  const int rounded_value =
+      (value_to_round + multiple - 1) / multiple * multiple;
+  return rounded_value <= max_value ? rounded_value
+                                    : (max_value / multiple * multiple);
+}
+
+// Generates a scale factor that makes `input_pixels` close to `target_pixels`,
+// but no higher than `max_pixels`.
+Fraction FindScale(int input_width,
+                   int input_height,
+                   int target_pixels,
+                   int max_pixels,
+                   bool variable_start_scale_factor) {
+  // This function only makes sense for a positive target.
+  RTC_DCHECK_GT(target_pixels, 0);
+  RTC_DCHECK_GT(max_pixels, 0);
+  RTC_DCHECK_GE(max_pixels, target_pixels);
+
+  const int input_pixels = input_width * input_height;
+
+  // Don't scale up original.
+  if (target_pixels >= input_pixels)
+    return Fraction{1, 1};
+
+  Fraction current_scale = Fraction{1, 1};
+  Fraction best_scale = Fraction{1, 1};
+
+  if (variable_start_scale_factor) {
+    // Start scaling down by 2/3 depending on `input_width` and `input_height`.
+    if (input_width % 3 == 0 && input_height % 3 == 0) {
+      // 2/3 (then alternates 3/4, 2/3, 3/4,...).
+      current_scale = Fraction{6, 6};
+    }
+    if (input_width % 9 == 0 && input_height % 9 == 0) {
+      // 2/3, 2/3 (then alternates 3/4, 2/3, 3/4,...).
+      current_scale = Fraction{36, 36};
+    }
+  }
+
+  // The minimum (absolute) difference between the number of output pixels and
+  // the target pixel count.
+  int min_pixel_diff = std::numeric_limits<int>::max();
+  if (input_pixels <= max_pixels) {
+    // Start condition for 1/1 case, if it is less than max.
+    min_pixel_diff = std::abs(input_pixels - target_pixels);
+  }
+
+  // Alternately scale down by 3/4 and 2/3. This results in fractions which are
+  // effectively scalable. For instance, starting at 1280x720 will result in
+  // the series (3/4) => 960x540, (1/2) => 640x360, (3/8) => 480x270,
+  // (1/4) => 320x180, (3/16) => 240x125, (1/8) => 160x90.
+  while (current_scale.scale_pixel_count(input_pixels) > target_pixels) {
+    if (current_scale.numerator % 3 == 0 &&
+        current_scale.denominator % 2 == 0) {
+      // Multiply by 2/3.
+      current_scale.numerator /= 3;
+      current_scale.denominator /= 2;
+    } else {
+      // Multiply by 3/4.
+      current_scale.numerator *= 3;
+      current_scale.denominator *= 4;
+    }
+
+    int output_pixels = current_scale.scale_pixel_count(input_pixels);
+    if (output_pixels <= max_pixels) {
+      int diff = std::abs(target_pixels - output_pixels);
+      if (diff < min_pixel_diff) {
+        min_pixel_diff = diff;
+        best_scale = current_scale;
+      }
+    }
+  }
+  best_scale.DivideByGcd();
+
+  return best_scale;
+}
+
+absl::optional<std::pair<int, int>> Swap(
+    const absl::optional<std::pair<int, int>>& in) {
+  if (!in) {
+    return absl::nullopt;
+  }
+  return std::make_pair(in->second, in->first);
+}
+
+}  // namespace
+
+namespace cricket {
+
+VideoAdapter::VideoAdapter(int source_resolution_alignment)
+    : frames_in_(0),
+      frames_out_(0),
+      frames_scaled_(0),
+      adaption_changes_(0),
+      previous_width_(0),
+      previous_height_(0),
+      variable_start_scale_factor_(!webrtc::field_trial::IsDisabled(
+          "WebRTC-Video-VariableStartScaleFactor")),
+      source_resolution_alignment_(source_resolution_alignment),
+      resolution_alignment_(source_resolution_alignment),
+      resolution_request_target_pixel_count_(std::numeric_limits<int>::max()),
+      resolution_request_max_pixel_count_(std::numeric_limits<int>::max()),
+      max_framerate_request_(std::numeric_limits<int>::max()) {}
+
+VideoAdapter::VideoAdapter() : VideoAdapter(1) {}
+
+VideoAdapter::~VideoAdapter() {}
+
+bool VideoAdapter::DropFrame(int64_t in_timestamp_ns) {
+  int max_fps = max_framerate_request_;
+  if (output_format_request_.max_fps)
+    max_fps = std::min(max_fps, *output_format_request_.max_fps);
+
+  framerate_controller_.SetMaxFramerate(max_fps);
+  return framerate_controller_.ShouldDropFrame(in_timestamp_ns);
+}
+
+bool VideoAdapter::AdaptFrameResolution(int in_width,
+                                        int in_height,
+                                        int64_t in_timestamp_ns,
+                                        int* cropped_width,
+                                        int* cropped_height,
+                                        int* out_width,
+                                        int* out_height) {
+  webrtc::MutexLock lock(&mutex_);
+  ++frames_in_;
+
+  // The max output pixel count is the minimum of the requests from
+  // OnOutputFormatRequest and OnResolutionFramerateRequest.
+  int max_pixel_count = resolution_request_max_pixel_count_;
+
+  // Select target aspect ratio and max pixel count depending on input frame
+  // orientation.
+  absl::optional<std::pair<int, int>> target_aspect_ratio;
+  if (in_width > in_height) {
+    target_aspect_ratio = output_format_request_.target_landscape_aspect_ratio;
+    if (output_format_request_.max_landscape_pixel_count)
+      max_pixel_count = std::min(
+          max_pixel_count, *output_format_request_.max_landscape_pixel_count);
+  } else {
+    target_aspect_ratio = output_format_request_.target_portrait_aspect_ratio;
+    if (output_format_request_.max_portrait_pixel_count)
+      max_pixel_count = std::min(
+          max_pixel_count, *output_format_request_.max_portrait_pixel_count);
+  }
+
+  int target_pixel_count =
+      std::min(resolution_request_target_pixel_count_, max_pixel_count);
+
+  // Drop the input frame if necessary.
+  if (max_pixel_count <= 0 || DropFrame(in_timestamp_ns)) {
+    // Show VAdapt log every 90 frames dropped. (3 seconds)
+    if ((frames_in_ - frames_out_) % 90 == 0) {
+      // TODO(fbarchard): Reduce to LS_VERBOSE when adapter info is not needed
+      // in default calls.
+      RTC_LOG(LS_INFO) << "VAdapt Drop Frame: scaled " << frames_scaled_
+                       << " / out " << frames_out_ << " / in " << frames_in_
+                       << " Changes: " << adaption_changes_
+                       << " Input: " << in_width << "x" << in_height
+                       << " timestamp: " << in_timestamp_ns
+                       << " Output fps: " << max_framerate_request_ << "/"
+                       << output_format_request_.max_fps.value_or(-1)
+                       << " alignment: " << resolution_alignment_;
+    }
+
+    // Drop frame.
+    return false;
+  }
+
+  // Calculate how the input should be cropped.
+  if (!target_aspect_ratio || target_aspect_ratio->first <= 0 ||
+      target_aspect_ratio->second <= 0) {
+    *cropped_width = in_width;
+    *cropped_height = in_height;
+  } else {
+    const float requested_aspect =
+        target_aspect_ratio->first /
+        static_cast<float>(target_aspect_ratio->second);
+    *cropped_width =
+        std::min(in_width, static_cast<int>(in_height * requested_aspect));
+    *cropped_height =
+        std::min(in_height, static_cast<int>(in_width / requested_aspect));
+  }
+  const Fraction scale =
+      FindScale(*cropped_width, *cropped_height, target_pixel_count,
+                max_pixel_count, variable_start_scale_factor_);
+  // Adjust cropping slightly to get correctly aligned output size and a perfect
+  // scale factor.
+  *cropped_width = roundUp(*cropped_width,
+                           scale.denominator * resolution_alignment_, in_width);
+  *cropped_height = roundUp(
+      *cropped_height, scale.denominator * resolution_alignment_, in_height);
+  RTC_DCHECK_EQ(0, *cropped_width % scale.denominator);
+  RTC_DCHECK_EQ(0, *cropped_height % scale.denominator);
+
+  // Calculate final output size.
+  *out_width = *cropped_width / scale.denominator * scale.numerator;
+  *out_height = *cropped_height / scale.denominator * scale.numerator;
+  RTC_DCHECK_EQ(0, *out_width % resolution_alignment_);
+  RTC_DCHECK_EQ(0, *out_height % resolution_alignment_);
+
+  ++frames_out_;
+  if (scale.numerator != scale.denominator)
+    ++frames_scaled_;
+
+  if (previous_width_ &&
+      (previous_width_ != *out_width || previous_height_ != *out_height)) {
+    ++adaption_changes_;
+    RTC_LOG(LS_INFO) << "Frame size changed: scaled " << frames_scaled_
+                     << " / out " << frames_out_ << " / in " << frames_in_
+                     << " Changes: " << adaption_changes_
+                     << " Input: " << in_width << "x" << in_height
+                     << " Scale: " << scale.numerator << "/"
+                     << scale.denominator << " Output: " << *out_width << "x"
+                     << *out_height << " fps: " << max_framerate_request_ << "/"
+                     << output_format_request_.max_fps.value_or(-1)
+                     << " alignment: " << resolution_alignment_;
+  }
+
+  previous_width_ = *out_width;
+  previous_height_ = *out_height;
+
+  return true;
+}
+
+void VideoAdapter::OnOutputFormatRequest(
+    const absl::optional<VideoFormat>& format) {
+  absl::optional<std::pair<int, int>> target_aspect_ratio;
+  absl::optional<int> max_pixel_count;
+  absl::optional<int> max_fps;
+  if (format) {
+    target_aspect_ratio = std::make_pair(format->width, format->height);
+    max_pixel_count = format->width * format->height;
+    if (format->interval > 0)
+      max_fps = rtc::kNumNanosecsPerSec / format->interval;
+  }
+  OnOutputFormatRequest(target_aspect_ratio, max_pixel_count, max_fps);
+}
+
+void VideoAdapter::OnOutputFormatRequest(
+    const absl::optional<std::pair<int, int>>& target_aspect_ratio,
+    const absl::optional<int>& max_pixel_count,
+    const absl::optional<int>& max_fps) {
+  absl::optional<std::pair<int, int>> target_landscape_aspect_ratio;
+  absl::optional<std::pair<int, int>> target_portrait_aspect_ratio;
+  if (target_aspect_ratio && target_aspect_ratio->first > 0 &&
+      target_aspect_ratio->second > 0) {
+    // Maintain input orientation.
+    const int max_side =
+        std::max(target_aspect_ratio->first, target_aspect_ratio->second);
+    const int min_side =
+        std::min(target_aspect_ratio->first, target_aspect_ratio->second);
+    target_landscape_aspect_ratio = std::make_pair(max_side, min_side);
+    target_portrait_aspect_ratio = std::make_pair(min_side, max_side);
+  }
+  OnOutputFormatRequest(target_landscape_aspect_ratio, max_pixel_count,
+                        target_portrait_aspect_ratio, max_pixel_count, max_fps);
+}
+
+void VideoAdapter::OnOutputFormatRequest(
+    const absl::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
+    const absl::optional<int>& max_landscape_pixel_count,
+    const absl::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
+    const absl::optional<int>& max_portrait_pixel_count,
+    const absl::optional<int>& max_fps) {
+  webrtc::MutexLock lock(&mutex_);
+
+  OutputFormatRequest request = {
+      .target_landscape_aspect_ratio = target_landscape_aspect_ratio,
+      .max_landscape_pixel_count = max_landscape_pixel_count,
+      .target_portrait_aspect_ratio = target_portrait_aspect_ratio,
+      .max_portrait_pixel_count = max_portrait_pixel_count,
+      .max_fps = max_fps};
+
+  if (stashed_output_format_request_) {
+    // Save the output format request for later use in case the encoder making
+    // this call would become active, because currently all active encoders use
+    // requested_resolution instead.
+    stashed_output_format_request_ = request;
+    RTC_LOG(LS_INFO) << "Stashing OnOutputFormatRequest: "
+                     << stashed_output_format_request_->ToString();
+  } else {
+    output_format_request_ = request;
+    RTC_LOG(LS_INFO) << "Setting output_format_request_: "
+                     << output_format_request_.ToString();
+  }
+
+  framerate_controller_.Reset();
+}
+
+void VideoAdapter::OnSinkWants(const rtc::VideoSinkWants& sink_wants) {
+  webrtc::MutexLock lock(&mutex_);
+  resolution_request_max_pixel_count_ = sink_wants.max_pixel_count;
+  resolution_request_target_pixel_count_ =
+      sink_wants.target_pixel_count.value_or(
+          resolution_request_max_pixel_count_);
+  max_framerate_request_ = sink_wants.max_framerate_fps;
+  resolution_alignment_ = cricket::LeastCommonMultiple(
+      source_resolution_alignment_, sink_wants.resolution_alignment);
+
+  if (!sink_wants.aggregates) {
+    RTC_LOG(LS_WARNING)
+        << "These should always be created by VideoBroadcaster!";
+    return;
+  }
+
+  // If requested_resolution is used, and there are no active encoders
+  // that are NOT using requested_resolution (aka newapi), then override
+  // calls to OnOutputFormatRequest and use values from requested_resolution
+  // instead (combined with qualityscaling based on pixel counts above).
+  if (webrtc::field_trial::IsDisabled(
+          "WebRTC-Video-RequestedResolutionOverrideOutputFormatRequest")) {
+    // kill-switch...
+    return;
+  }
+
+  if (!sink_wants.requested_resolution) {
+    if (stashed_output_format_request_) {
+      // because current active_output_format_request is based on
+      // requested_resolution logic, while current encoder(s) doesn't want that,
+      // we have to restore the stashed request.
+      RTC_LOG(LS_INFO) << "Unstashing OnOutputFormatRequest: "
+                       << stashed_output_format_request_->ToString();
+      output_format_request_ = *stashed_output_format_request_;
+      stashed_output_format_request_.reset();
+    }
+    return;
+  }
+
+  if (sink_wants.aggregates->any_active_without_requested_resolution) {
+    return;
+  }
+
+  if (!stashed_output_format_request_) {
+    // The active output format request is about to be rewritten by
+    // request_resolution. We need to save it for later use in case the encoder
+    // which doesn't use request_resolution logic become active in the future.
+    stashed_output_format_request_ = output_format_request_;
+    RTC_LOG(LS_INFO) << "Stashing OnOutputFormatRequest: "
+                     << stashed_output_format_request_->ToString();
+  }
+
+  auto res = *sink_wants.requested_resolution;
+  auto pixel_count = res.width * res.height;
+  output_format_request_.target_landscape_aspect_ratio =
+      std::make_pair(res.width, res.height);
+  output_format_request_.max_landscape_pixel_count = pixel_count;
+  output_format_request_.target_portrait_aspect_ratio =
+      std::make_pair(res.height, res.width);
+  output_format_request_.max_portrait_pixel_count = pixel_count;
+  output_format_request_.max_fps = max_framerate_request_;
+  RTC_LOG(LS_INFO) << "Setting output_format_request_ based on sink_wants: "
+                   << output_format_request_.ToString();
+}
+
+int VideoAdapter::GetTargetPixels() const {
+  webrtc::MutexLock lock(&mutex_);
+  return resolution_request_target_pixel_count_;
+}
+
+float VideoAdapter::GetMaxFramerate() const {
+  webrtc::MutexLock lock(&mutex_);
+  // Minimum of `output_format_request_.max_fps` and `max_framerate_request_` is
+  // used to throttle frame-rate.
+  int framerate =
+      std::min(max_framerate_request_,
+               output_format_request_.max_fps.value_or(max_framerate_request_));
+  if (framerate == std::numeric_limits<int>::max()) {
+    return std::numeric_limits<float>::infinity();
+  } else {
+    return max_framerate_request_;
+  }
+}
+
+std::string VideoAdapter::OutputFormatRequest::ToString() const {
+  rtc::StringBuilder oss;
+  oss << "[ ";
+  if (target_landscape_aspect_ratio == Swap(target_portrait_aspect_ratio) &&
+      max_landscape_pixel_count == max_portrait_pixel_count) {
+    if (target_landscape_aspect_ratio) {
+      oss << target_landscape_aspect_ratio->first << "x"
+          << target_landscape_aspect_ratio->second;
+    } else {
+      oss << "unset-resolution";
+    }
+    if (max_landscape_pixel_count) {
+      oss << " max_pixel_count: " << *max_landscape_pixel_count;
+    }
+  } else {
+    oss << "[ landscape: ";
+    if (target_landscape_aspect_ratio) {
+      oss << target_landscape_aspect_ratio->first << "x"
+          << target_landscape_aspect_ratio->second;
+    } else {
+      oss << "unset";
+    }
+    if (max_landscape_pixel_count) {
+      oss << " max_pixel_count: " << *max_landscape_pixel_count;
+    }
+    oss << " ] [ portrait: ";
+    if (target_portrait_aspect_ratio) {
+      oss << target_portrait_aspect_ratio->first << "x"
+          << target_portrait_aspect_ratio->second;
+    }
+    if (max_portrait_pixel_count) {
+      oss << " max_pixel_count: " << *max_portrait_pixel_count;
+    }
+    oss << " ]";
+  }
+  oss << " max_fps: ";
+  if (max_fps) {
+    oss << *max_fps;
+  } else {
+    oss << "unset";
+  }
+  oss << " ]";
+  return oss.Release();
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/video_adapter.h

@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2010 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 MEDIA_BASE_VIDEO_ADAPTER_H_
+#define MEDIA_BASE_VIDEO_ADAPTER_H_
+
+#include <stdint.h>
+
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/video/video_source_interface.h"
+#include "common_video/framerate_controller.h"
+#include "media/base/video_common.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace cricket {
+
+// VideoAdapter adapts an input video frame to an output frame based on the
+// specified input and output formats. The adaptation includes dropping frames
+// to reduce frame rate and scaling frames.
+// VideoAdapter is thread safe.
+class RTC_EXPORT VideoAdapter {
+ public:
+  VideoAdapter();
+  // The source requests output frames whose width and height are divisible
+  // by `source_resolution_alignment`.
+  explicit VideoAdapter(int source_resolution_alignment);
+  virtual ~VideoAdapter();
+
+  VideoAdapter(const VideoAdapter&) = delete;
+  VideoAdapter& operator=(const VideoAdapter&) = delete;
+
+  // Return the adapted resolution and cropping parameters given the
+  // input resolution. The input frame should first be cropped, then
+  // scaled to the final output resolution. Returns true if the frame
+  // should be adapted, and false if it should be dropped.
+  bool AdaptFrameResolution(int in_width,
+                            int in_height,
+                            int64_t in_timestamp_ns,
+                            int* cropped_width,
+                            int* cropped_height,
+                            int* out_width,
+                            int* out_height) RTC_LOCKS_EXCLUDED(mutex_);
+
+  // DEPRECATED. Please use OnOutputFormatRequest below.
+  // TODO(asapersson): Remove this once it is no longer used.
+  // Requests the output frame size and frame interval from
+  // `AdaptFrameResolution` to not be larger than `format`. Also, the input
+  // frame size will be cropped to match the requested aspect ratio. The
+  // requested aspect ratio is orientation agnostic and will be adjusted to
+  // maintain the input orientation, so it doesn't matter if e.g. 1280x720 or
+  // 720x1280 is requested.
+  // Note: Should be called from the source only.
+  void OnOutputFormatRequest(const absl::optional<VideoFormat>& format)
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Requests output frame size and frame interval from `AdaptFrameResolution`.
+  // `target_aspect_ratio`: The input frame size will be cropped to match the
+  // requested aspect ratio. The aspect ratio is orientation agnostic and will
+  // be adjusted to maintain the input orientation (i.e. it doesn't matter if
+  // e.g. <1280,720> or <720,1280> is requested).
+  // `max_pixel_count`: The maximum output frame size.
+  // `max_fps`: The maximum output framerate.
+  // Note: Should be called from the source only.
+  void OnOutputFormatRequest(
+      const absl::optional<std::pair<int, int>>& target_aspect_ratio,
+      const absl::optional<int>& max_pixel_count,
+      const absl::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Same as above, but allows setting two different target aspect ratios
+  // depending on incoming frame orientation. This gives more fine-grained
+  // control and can e.g. be used to force landscape video to be cropped to
+  // portrait video.
+  void OnOutputFormatRequest(
+      const absl::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
+      const absl::optional<int>& max_landscape_pixel_count,
+      const absl::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
+      const absl::optional<int>& max_portrait_pixel_count,
+      const absl::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Requests the output frame size from `AdaptFrameResolution` to have as close
+  // as possible to `sink_wants.target_pixel_count` pixels (if set)
+  // but no more than `sink_wants.max_pixel_count`.
+  // `sink_wants.max_framerate_fps` is essentially analogous to
+  // `sink_wants.max_pixel_count`, but for framerate rather than resolution.
+  // Set `sink_wants.max_pixel_count` and/or `sink_wants.max_framerate_fps` to
+  // std::numeric_limit<int>::max() if no upper limit is desired.
+  // The sink resolution alignment requirement is given by
+  // `sink_wants.resolution_alignment`.
+  // Note: Should be called from the sink only.
+  void OnSinkWants(const rtc::VideoSinkWants& sink_wants)
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Returns maximum image area, which shouldn't impose any adaptations.
+  // Can return `numeric_limits<int>::max()` if no limit is set.
+  int GetTargetPixels() const;
+
+  // Returns current frame-rate limit.
+  // Can return `numeric_limits<float>::infinity()` if no limit is set.
+  float GetMaxFramerate() const;
+
+ private:
+  // Determine if frame should be dropped based on input fps and requested fps.
+  bool DropFrame(int64_t in_timestamp_ns) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  int frames_in_ RTC_GUARDED_BY(mutex_);      // Number of input frames.
+  int frames_out_ RTC_GUARDED_BY(mutex_);     // Number of output frames.
+  int frames_scaled_ RTC_GUARDED_BY(mutex_);  // Number of frames scaled.
+  int adaption_changes_
+      RTC_GUARDED_BY(mutex_);  // Number of changes in scale factor.
+  int previous_width_ RTC_GUARDED_BY(mutex_);  // Previous adapter output width.
+  int previous_height_
+      RTC_GUARDED_BY(mutex_);  // Previous adapter output height.
+  const bool variable_start_scale_factor_;
+
+  // The fixed source resolution alignment requirement.
+  const int source_resolution_alignment_;
+  // The currently applied resolution alignment, as given by the requirements:
+  //  - the fixed `source_resolution_alignment_`; and
+  //  - the latest `sink_wants.resolution_alignment`.
+  int resolution_alignment_ RTC_GUARDED_BY(mutex_);
+
+  // Max number of pixels/fps requested via calls to OnOutputFormatRequest,
+  // OnResolutionFramerateRequest respectively.
+  // The adapted output format is the minimum of these.
+  struct OutputFormatRequest {
+    absl::optional<std::pair<int, int>> target_landscape_aspect_ratio;
+    absl::optional<int> max_landscape_pixel_count;
+    absl::optional<std::pair<int, int>> target_portrait_aspect_ratio;
+    absl::optional<int> max_portrait_pixel_count;
+    absl::optional<int> max_fps;
+
+    // For logging.
+    std::string ToString() const;
+  };
+
+  OutputFormatRequest output_format_request_ RTC_GUARDED_BY(mutex_);
+  int resolution_request_target_pixel_count_ RTC_GUARDED_BY(mutex_);
+  int resolution_request_max_pixel_count_ RTC_GUARDED_BY(mutex_);
+  int max_framerate_request_ RTC_GUARDED_BY(mutex_);
+
+  // Stashed OutputFormatRequest that is used to save value of
+  // OnOutputFormatRequest in case all active encoders are using
+  // requested_resolution. I.e when all active encoders are using
+  // requested_resolution, the call to OnOutputFormatRequest is ignored
+  // and the value from requested_resolution is used instead (to scale/crop
+  // frame). This allows for an application to only use
+  // RtpEncodingParameters::request_resolution and get the same behavior as if
+  // it had used VideoAdapter::OnOutputFormatRequest.
+  absl::optional<OutputFormatRequest> stashed_output_format_request_
+      RTC_GUARDED_BY(mutex_);
+
+  webrtc::FramerateController framerate_controller_ RTC_GUARDED_BY(mutex_);
+
+  // The critical section to protect the above variables.
+  mutable webrtc::Mutex mutex_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_VIDEO_ADAPTER_H_

TMessagesProj/jni/voip/webrtc/media/base/video_broadcaster.cc

@@ -0,0 +1,214 @@
+/*
+ *  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 "media/base/video_broadcaster.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_rotation.h"
+#include "media/base/video_common.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace rtc {
+
+VideoBroadcaster::VideoBroadcaster() = default;
+VideoBroadcaster::~VideoBroadcaster() = default;
+
+void VideoBroadcaster::AddOrUpdateSink(
+    VideoSinkInterface<webrtc::VideoFrame>* sink,
+    const VideoSinkWants& wants) {
+  RTC_DCHECK(sink != nullptr);
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  if (!FindSinkPair(sink)) {
+    // `Sink` is a new sink, which didn't receive previous frame.
+    previous_frame_sent_to_all_sinks_ = false;
+
+    if (last_constraints_.has_value()) {
+      RTC_LOG(LS_INFO) << __func__ << " forwarding stored constraints min_fps "
+                       << last_constraints_->min_fps.value_or(-1) << " max_fps "
+                       << last_constraints_->max_fps.value_or(-1);
+      sink->OnConstraintsChanged(*last_constraints_);
+    }
+  }
+  VideoSourceBase::AddOrUpdateSink(sink, wants);
+  UpdateWants();
+}
+
+void VideoBroadcaster::RemoveSink(
+    VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  RTC_DCHECK(sink != nullptr);
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  VideoSourceBase::RemoveSink(sink);
+  UpdateWants();
+}
+
+bool VideoBroadcaster::frame_wanted() const {
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  return !sink_pairs().empty();
+}
+
+VideoSinkWants VideoBroadcaster::wants() const {
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  return current_wants_;
+}
+
+void VideoBroadcaster::OnFrame(const webrtc::VideoFrame& frame) {
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  bool current_frame_was_discarded = false;
+  for (auto& sink_pair : sink_pairs()) {
+    if (sink_pair.wants.rotation_applied &&
+        frame.rotation() != webrtc::kVideoRotation_0) {
+      // Calls to OnFrame are not synchronized with changes to the sink wants.
+      // When rotation_applied is set to true, one or a few frames may get here
+      // with rotation still pending. Protect sinks that don't expect any
+      // pending rotation.
+      RTC_LOG(LS_VERBOSE) << "Discarding frame with unexpected rotation.";
+      sink_pair.sink->OnDiscardedFrame();
+      current_frame_was_discarded = true;
+      continue;
+    }
+    if (sink_pair.wants.black_frames) {
+      webrtc::VideoFrame black_frame =
+          webrtc::VideoFrame::Builder()
+              .set_video_frame_buffer(
+                  GetBlackFrameBuffer(frame.width(), frame.height()))
+              .set_rotation(frame.rotation())
+              .set_timestamp_us(frame.timestamp_us())
+              .set_id(frame.id())
+              .build();
+      sink_pair.sink->OnFrame(black_frame);
+    } else if (!previous_frame_sent_to_all_sinks_ && frame.has_update_rect()) {
+      // Since last frame was not sent to some sinks, no reliable update
+      // information is available, so we need to clear the update rect.
+      webrtc::VideoFrame copy = frame;
+      copy.clear_update_rect();
+      sink_pair.sink->OnFrame(copy);
+    } else {
+      sink_pair.sink->OnFrame(frame);
+    }
+  }
+  previous_frame_sent_to_all_sinks_ = !current_frame_was_discarded;
+}
+
+void VideoBroadcaster::OnDiscardedFrame() {
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  for (auto& sink_pair : sink_pairs()) {
+    sink_pair.sink->OnDiscardedFrame();
+  }
+}
+
+void VideoBroadcaster::ProcessConstraints(
+    const webrtc::VideoTrackSourceConstraints& constraints) {
+  webrtc::MutexLock lock(&sinks_and_wants_lock_);
+  RTC_LOG(LS_INFO) << __func__ << " min_fps "
+                   << constraints.min_fps.value_or(-1) << " max_fps "
+                   << constraints.max_fps.value_or(-1) << " broadcasting to "
+                   << sink_pairs().size() << " sinks.";
+  last_constraints_ = constraints;
+  for (auto& sink_pair : sink_pairs())
+    sink_pair.sink->OnConstraintsChanged(constraints);
+}
+
+void VideoBroadcaster::UpdateWants() {
+  VideoSinkWants wants;
+  wants.rotation_applied = false;
+  wants.resolution_alignment = 1;
+  wants.aggregates.emplace(VideoSinkWants::Aggregates());
+  wants.is_active = false;
+
+  // TODO(webrtc:14451) : I think it makes sense to always
+  // "ignore" encoders that are not active. But that would
+  // probably require a controlled roll out with a field trials?
+  // To play it safe, only ignore inactive encoders is there is an
+  // active encoder using the new api (requested_resolution),
+  // this means that there is only a behavioural change when using new
+  // api.
+  bool ignore_inactive_encoders_old_api = false;
+  for (auto& sink : sink_pairs()) {
+    if (sink.wants.is_active && sink.wants.requested_resolution.has_value()) {
+      ignore_inactive_encoders_old_api = true;
+      break;
+    }
+  }
+
+  for (auto& sink : sink_pairs()) {
+    if (!sink.wants.is_active &&
+        (sink.wants.requested_resolution || ignore_inactive_encoders_old_api)) {
+      continue;
+    }
+    // wants.rotation_applied == ANY(sink.wants.rotation_applied)
+    if (sink.wants.rotation_applied) {
+      wants.rotation_applied = true;
+    }
+    // wants.max_pixel_count == MIN(sink.wants.max_pixel_count)
+    if (sink.wants.max_pixel_count < wants.max_pixel_count) {
+      wants.max_pixel_count = sink.wants.max_pixel_count;
+    }
+    // Select the minimum requested target_pixel_count, if any, of all sinks so
+    // that we don't over utilize the resources for any one.
+    // TODO(sprang): Consider using the median instead, since the limit can be
+    // expressed by max_pixel_count.
+    if (sink.wants.target_pixel_count &&
+        (!wants.target_pixel_count ||
+         (*sink.wants.target_pixel_count < *wants.target_pixel_count))) {
+      wants.target_pixel_count = sink.wants.target_pixel_count;
+    }
+    // Select the minimum for the requested max framerates.
+    if (sink.wants.max_framerate_fps < wants.max_framerate_fps) {
+      wants.max_framerate_fps = sink.wants.max_framerate_fps;
+    }
+    wants.resolution_alignment = cricket::LeastCommonMultiple(
+        wants.resolution_alignment, sink.wants.resolution_alignment);
+
+    // Pick MAX(requested_resolution) since the actual can be downscaled
+    // in encoder instead.
+    if (sink.wants.requested_resolution) {
+      if (!wants.requested_resolution) {
+        wants.requested_resolution = sink.wants.requested_resolution;
+      } else {
+        wants.requested_resolution->width =
+            std::max(wants.requested_resolution->width,
+                     sink.wants.requested_resolution->width);
+        wants.requested_resolution->height =
+            std::max(wants.requested_resolution->height,
+                     sink.wants.requested_resolution->height);
+      }
+    } else if (sink.wants.is_active) {
+      wants.aggregates->any_active_without_requested_resolution = true;
+    }
+
+    wants.is_active |= sink.wants.is_active;
+  }
+
+  if (wants.target_pixel_count &&
+      *wants.target_pixel_count >= wants.max_pixel_count) {
+    wants.target_pixel_count.emplace(wants.max_pixel_count);
+  }
+  current_wants_ = wants;
+}
+
+const rtc::scoped_refptr<webrtc::VideoFrameBuffer>&
+VideoBroadcaster::GetBlackFrameBuffer(int width, int height) {
+  if (!black_frame_buffer_ || black_frame_buffer_->width() != width ||
+      black_frame_buffer_->height() != height) {
+    rtc::scoped_refptr<webrtc::I420Buffer> buffer =
+        webrtc::I420Buffer::Create(width, height);
+    webrtc::I420Buffer::SetBlack(buffer.get());
+    black_frame_buffer_ = buffer;
+  }
+
+  return black_frame_buffer_;
+}
+
+}  // namespace rtc

TMessagesProj/jni/voip/webrtc/media/base/video_broadcaster.h

@@ -0,0 +1,82 @@
+/*
+ *  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 MEDIA_BASE_VIDEO_BROADCASTER_H_
+#define MEDIA_BASE_VIDEO_BROADCASTER_H_
+
+#include "api/media_stream_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_source_interface.h"
+#include "media/base/video_source_base.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace rtc {
+
+// VideoBroadcaster broadcast video frames to sinks and combines VideoSinkWants
+// from its sinks. It does that by implementing rtc::VideoSourceInterface and
+// rtc::VideoSinkInterface. The class is threadsafe; methods may be called on
+// any thread. This is needed because VideoStreamEncoder calls AddOrUpdateSink
+// both on the worker thread and on the encoder task queue.
+class VideoBroadcaster : public VideoSourceBase,
+                         public VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  VideoBroadcaster();
+  ~VideoBroadcaster() override;
+
+  // Adds a new, or updates an already existing sink. If the sink is new and
+  // ProcessConstraints has been called previously, the new sink's
+  // OnConstraintsCalled method will be invoked with the most recent
+  // constraints.
+  void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
+                       const VideoSinkWants& wants) override;
+  void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
+  // Returns true if the next frame will be delivered to at least one sink.
+  bool frame_wanted() const;
+
+  // Returns VideoSinkWants a source is requested to fulfill. They are
+  // aggregated by all VideoSinkWants from all sinks.
+  VideoSinkWants wants() const;
+
+  // This method ensures that if a sink sets rotation_applied == true,
+  // it will never receive a frame with pending rotation. Our caller
+  // may pass in frames without precise synchronization with changes
+  // to the VideoSinkWants.
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+  void OnDiscardedFrame() override;
+
+  // Called on the network thread when constraints change. Forwards the
+  // constraints to sinks added with AddOrUpdateSink via OnConstraintsChanged.
+  void ProcessConstraints(
+      const webrtc::VideoTrackSourceConstraints& constraints);
+
+ protected:
+  void UpdateWants() RTC_EXCLUSIVE_LOCKS_REQUIRED(sinks_and_wants_lock_);
+  const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& GetBlackFrameBuffer(
+      int width,
+      int height) RTC_EXCLUSIVE_LOCKS_REQUIRED(sinks_and_wants_lock_);
+
+  mutable webrtc::Mutex sinks_and_wants_lock_;
+
+  VideoSinkWants current_wants_ RTC_GUARDED_BY(sinks_and_wants_lock_);
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> black_frame_buffer_;
+  bool previous_frame_sent_to_all_sinks_ RTC_GUARDED_BY(sinks_and_wants_lock_) =
+      true;
+  absl::optional<webrtc::VideoTrackSourceConstraints> last_constraints_
+      RTC_GUARDED_BY(sinks_and_wants_lock_);
+};
+
+}  // namespace rtc
+
+#endif  // MEDIA_BASE_VIDEO_BROADCASTER_H_

TMessagesProj/jni/voip/webrtc/media/base/video_common.cc

@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2010 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 "media/base/video_common.h"
+
+#include "api/array_view.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace cricket {
+
+struct FourCCAliasEntry {
+  uint32_t alias;
+  uint32_t canonical;
+};
+
+static const FourCCAliasEntry kFourCCAliases[] = {
+    {FOURCC_IYUV, FOURCC_I420},
+    {FOURCC_YU16, FOURCC_I422},
+    {FOURCC_YU24, FOURCC_I444},
+    {FOURCC_YUYV, FOURCC_YUY2},
+    {FOURCC_YUVS, FOURCC_YUY2},
+    {FOURCC_HDYC, FOURCC_UYVY},
+    {FOURCC_2VUY, FOURCC_UYVY},
+    {FOURCC_JPEG, FOURCC_MJPG},  // Note: JPEG has DHT while MJPG does not.
+    {FOURCC_DMB1, FOURCC_MJPG},
+    {FOURCC_BA81, FOURCC_BGGR},
+    {FOURCC_RGB3, FOURCC_RAW},
+    {FOURCC_BGR3, FOURCC_24BG},
+    {FOURCC_CM32, FOURCC_BGRA},
+    {FOURCC_CM24, FOURCC_RAW},
+};
+
+uint32_t CanonicalFourCC(uint32_t fourcc) {
+  for (uint32_t i = 0; i < arraysize(kFourCCAliases); ++i) {
+    if (kFourCCAliases[i].alias == fourcc) {
+      return kFourCCAliases[i].canonical;
+    }
+  }
+  // Not an alias, so return it as-is.
+  return fourcc;
+}
+
+// The C++ standard requires a namespace-scope definition of static const
+// integral types even when they are initialized in the declaration (see
+// [class.static.data]/4), but MSVC with /Ze is non-conforming and treats that
+// as a multiply defined symbol error. See Also:
+// http://msdn.microsoft.com/en-us/library/34h23df8.aspx
+#ifndef _MSC_EXTENSIONS
+const int64_t VideoFormat::kMinimumInterval;  // Initialized in header.
+#endif
+
+std::string VideoFormat::ToString() const {
+  std::string fourcc_name = GetFourccName(fourcc) + " ";
+  for (std::string::const_iterator i = fourcc_name.begin();
+       i < fourcc_name.end(); ++i) {
+    // Test character is printable; Avoid isprint() which asserts on negatives.
+    if (*i < 32 || *i >= 127) {
+      fourcc_name = "";
+      break;
+    }
+  }
+
+  char buf[256];
+  rtc::SimpleStringBuilder sb(buf);
+  sb << fourcc_name << width << "x" << height << "x"
+     << IntervalToFpsFloat(interval);
+  return sb.str();
+}
+
+int GreatestCommonDivisor(int a, int b) {
+  RTC_DCHECK_GE(a, 0);
+  RTC_DCHECK_GT(b, 0);
+  int c = a % b;
+  while (c != 0) {
+    a = b;
+    b = c;
+    c = a % b;
+  }
+  return b;
+}
+
+int LeastCommonMultiple(int a, int b) {
+  RTC_DCHECK_GT(a, 0);
+  RTC_DCHECK_GT(b, 0);
+  return a * (b / GreatestCommonDivisor(a, b));
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/base/video_common.h

@@ -0,0 +1,224 @@
+/*
+ *  Copyright (c) 2004 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.
+ */
+
+// Common definition for video, including fourcc and VideoFormat.
+
+#ifndef MEDIA_BASE_VIDEO_COMMON_H_
+#define MEDIA_BASE_VIDEO_COMMON_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/time_utils.h"
+
+namespace cricket {
+
+//////////////////////////////////////////////////////////////////////////////
+// Definition of FourCC codes
+//////////////////////////////////////////////////////////////////////////////
+// Convert four characters to a FourCC code.
+// Needs to be a macro otherwise the OS X compiler complains when the kFormat*
+// constants are used in a switch.
+#define CRICKET_FOURCC(a, b, c, d)                                \
+  ((static_cast<uint32_t>(a)) | (static_cast<uint32_t>(b) << 8) | \
+   (static_cast<uint32_t>(c) << 16) | (static_cast<uint32_t>(d) << 24))
+// Some pages discussing FourCC codes:
+//   http://www.fourcc.org/yuv.php
+//   http://v4l2spec.bytesex.org/spec/book1.htm
+//   http://developer.apple.com/quicktime/icefloe/dispatch020.html
+//   http://msdn.microsoft.com/library/windows/desktop/dd206750.aspx#nv12
+//   http://people.xiph.org/~xiphmont/containers/nut/nut4cc.txt
+
+// FourCC codes grouped according to implementation efficiency.
+// Primary formats should convert in 1 efficient step.
+// Secondary formats are converted in 2 steps.
+// Auxilliary formats call primary converters.
+enum FourCC {
+  // 9 Primary YUV formats: 5 planar, 2 biplanar, 2 packed.
+  FOURCC_I420 = CRICKET_FOURCC('I', '4', '2', '0'),
+  FOURCC_I422 = CRICKET_FOURCC('I', '4', '2', '2'),
+  FOURCC_I444 = CRICKET_FOURCC('I', '4', '4', '4'),
+  FOURCC_I411 = CRICKET_FOURCC('I', '4', '1', '1'),
+  FOURCC_I400 = CRICKET_FOURCC('I', '4', '0', '0'),
+  FOURCC_NV21 = CRICKET_FOURCC('N', 'V', '2', '1'),
+  FOURCC_NV12 = CRICKET_FOURCC('N', 'V', '1', '2'),
+  FOURCC_YUY2 = CRICKET_FOURCC('Y', 'U', 'Y', '2'),
+  FOURCC_UYVY = CRICKET_FOURCC('U', 'Y', 'V', 'Y'),
+
+  // 2 Secondary YUV formats: row biplanar.
+  FOURCC_M420 = CRICKET_FOURCC('M', '4', '2', '0'),
+
+  // 9 Primary RGB formats: 4 32 bpp, 2 24 bpp, 3 16 bpp.
+  FOURCC_ARGB = CRICKET_FOURCC('A', 'R', 'G', 'B'),
+  FOURCC_BGRA = CRICKET_FOURCC('B', 'G', 'R', 'A'),
+  FOURCC_ABGR = CRICKET_FOURCC('A', 'B', 'G', 'R'),
+  FOURCC_24BG = CRICKET_FOURCC('2', '4', 'B', 'G'),
+  FOURCC_RAW = CRICKET_FOURCC('r', 'a', 'w', ' '),
+  FOURCC_RGBA = CRICKET_FOURCC('R', 'G', 'B', 'A'),
+  FOURCC_RGBP = CRICKET_FOURCC('R', 'G', 'B', 'P'),  // bgr565.
+  FOURCC_RGBO = CRICKET_FOURCC('R', 'G', 'B', 'O'),  // abgr1555.
+  FOURCC_R444 = CRICKET_FOURCC('R', '4', '4', '4'),  // argb4444.
+
+  // 4 Secondary RGB formats: 4 Bayer Patterns.
+  FOURCC_RGGB = CRICKET_FOURCC('R', 'G', 'G', 'B'),
+  FOURCC_BGGR = CRICKET_FOURCC('B', 'G', 'G', 'R'),
+  FOURCC_GRBG = CRICKET_FOURCC('G', 'R', 'B', 'G'),
+  FOURCC_GBRG = CRICKET_FOURCC('G', 'B', 'R', 'G'),
+
+  // 1 Primary Compressed YUV format.
+  FOURCC_MJPG = CRICKET_FOURCC('M', 'J', 'P', 'G'),
+
+  // 5 Auxiliary YUV variations: 3 with U and V planes are swapped, 1 Alias.
+  FOURCC_YV12 = CRICKET_FOURCC('Y', 'V', '1', '2'),
+  FOURCC_YV16 = CRICKET_FOURCC('Y', 'V', '1', '6'),
+  FOURCC_YV24 = CRICKET_FOURCC('Y', 'V', '2', '4'),
+  FOURCC_YU12 = CRICKET_FOURCC('Y', 'U', '1', '2'),  // Linux version of I420.
+  FOURCC_J420 = CRICKET_FOURCC('J', '4', '2', '0'),
+  FOURCC_J400 = CRICKET_FOURCC('J', '4', '0', '0'),
+
+  // 14 Auxiliary aliases.  CanonicalFourCC() maps these to canonical FOURCC.
+  FOURCC_IYUV = CRICKET_FOURCC('I', 'Y', 'U', 'V'),  // Alias for I420.
+  FOURCC_YU16 = CRICKET_FOURCC('Y', 'U', '1', '6'),  // Alias for I422.
+  FOURCC_YU24 = CRICKET_FOURCC('Y', 'U', '2', '4'),  // Alias for I444.
+  FOURCC_YUYV = CRICKET_FOURCC('Y', 'U', 'Y', 'V'),  // Alias for YUY2.
+  FOURCC_YUVS = CRICKET_FOURCC('y', 'u', 'v', 's'),  // Alias for YUY2 on Mac.
+  FOURCC_HDYC = CRICKET_FOURCC('H', 'D', 'Y', 'C'),  // Alias for UYVY.
+  FOURCC_2VUY = CRICKET_FOURCC('2', 'v', 'u', 'y'),  // Alias for UYVY on Mac.
+  FOURCC_JPEG = CRICKET_FOURCC('J', 'P', 'E', 'G'),  // Alias for MJPG.
+  FOURCC_DMB1 = CRICKET_FOURCC('d', 'm', 'b', '1'),  // Alias for MJPG on Mac.
+  FOURCC_BA81 = CRICKET_FOURCC('B', 'A', '8', '1'),  // Alias for BGGR.
+  FOURCC_RGB3 = CRICKET_FOURCC('R', 'G', 'B', '3'),  // Alias for RAW.
+  FOURCC_BGR3 = CRICKET_FOURCC('B', 'G', 'R', '3'),  // Alias for 24BG.
+  FOURCC_CM32 = CRICKET_FOURCC(0, 0, 0, 32),  // BGRA kCMPixelFormat_32ARGB
+  FOURCC_CM24 = CRICKET_FOURCC(0, 0, 0, 24),  // RAW kCMPixelFormat_24RGB
+
+  // 1 Auxiliary compressed YUV format set aside for capturer.
+  FOURCC_H264 = CRICKET_FOURCC('H', '2', '6', '4'),
+};
+
+#undef CRICKET_FOURCC
+
+// Match any fourcc.
+
+// We move this out of the enum because using it in many places caused
+// the compiler to get grumpy, presumably since the above enum is
+// backed by an int.
+static const uint32_t FOURCC_ANY = 0xFFFFFFFF;
+
+// Converts fourcc aliases into canonical ones.
+uint32_t CanonicalFourCC(uint32_t fourcc);
+
+// Get FourCC code as a string.
+inline std::string GetFourccName(uint32_t fourcc) {
+  std::string name;
+  name.push_back(static_cast<char>(fourcc & 0xFF));
+  name.push_back(static_cast<char>((fourcc >> 8) & 0xFF));
+  name.push_back(static_cast<char>((fourcc >> 16) & 0xFF));
+  name.push_back(static_cast<char>((fourcc >> 24) & 0xFF));
+  return name;
+}
+
+//////////////////////////////////////////////////////////////////////////////
+// Definition of VideoFormat.
+//////////////////////////////////////////////////////////////////////////////
+
+// VideoFormat with Plain Old Data for global variables.
+struct VideoFormatPod {
+  int width;         // Number of pixels.
+  int height;        // Number of pixels.
+  int64_t interval;  // Nanoseconds.
+  uint32_t fourcc;  // Color space. FOURCC_ANY means that any color space is OK.
+};
+
+struct RTC_EXPORT VideoFormat : VideoFormatPod {
+  static const int64_t kMinimumInterval =
+      rtc::kNumNanosecsPerSec / 10000;  // 10k fps.
+
+  VideoFormat() { Construct(0, 0, 0, 0); }
+
+  VideoFormat(int w, int h, int64_t interval_ns, uint32_t cc) {
+    Construct(w, h, interval_ns, cc);
+  }
+
+  explicit VideoFormat(const VideoFormatPod& format) {
+    Construct(format.width, format.height, format.interval, format.fourcc);
+  }
+
+  void Construct(int w, int h, int64_t interval_ns, uint32_t cc) {
+    width = w;
+    height = h;
+    interval = interval_ns;
+    fourcc = cc;
+  }
+
+  static int64_t FpsToInterval(int fps) {
+    return fps ? rtc::kNumNanosecsPerSec / fps : kMinimumInterval;
+  }
+
+  static int IntervalToFps(int64_t interval) {
+    if (!interval) {
+      return 0;
+    }
+    return static_cast<int>(rtc::kNumNanosecsPerSec / interval);
+  }
+
+  static float IntervalToFpsFloat(int64_t interval) {
+    if (!interval) {
+      return 0.f;
+    }
+    return static_cast<float>(rtc::kNumNanosecsPerSec) /
+           static_cast<float>(interval);
+  }
+
+  bool operator==(const VideoFormat& format) const {
+    return width == format.width && height == format.height &&
+           interval == format.interval && fourcc == format.fourcc;
+  }
+
+  bool operator!=(const VideoFormat& format) const {
+    return !(*this == format);
+  }
+
+  bool operator<(const VideoFormat& format) const {
+    return (fourcc < format.fourcc) ||
+           (fourcc == format.fourcc && width < format.width) ||
+           (fourcc == format.fourcc && width == format.width &&
+            height < format.height) ||
+           (fourcc == format.fourcc && width == format.width &&
+            height == format.height && interval > format.interval);
+  }
+
+  int framerate() const { return IntervalToFps(interval); }
+
+  // Check if both width and height are 0.
+  bool IsSize0x0() const { return 0 == width && 0 == height; }
+
+  // Check if this format is less than another one by comparing the resolution
+  // and frame rate.
+  bool IsPixelRateLess(const VideoFormat& format) const {
+    return width * height * framerate() <
+           format.width * format.height * format.framerate();
+  }
+
+  // Get a string presentation in the form of "fourcc width x height x fps"
+  std::string ToString() const;
+};
+
+// Returns the largest positive integer that divides both `a` and `b`.
+int GreatestCommonDivisor(int a, int b);
+
+// Returns the smallest positive integer that is divisible by both `a` and `b`.
+int LeastCommonMultiple(int a, int b);
+
+}  // namespace cricket
+
+#endif  // MEDIA_BASE_VIDEO_COMMON_H_

TMessagesProj/jni/voip/webrtc/media/base/video_source_base.cc

@@ -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.
+ */
+
+#include "media/base/video_source_base.h"
+
+#include <algorithm>
+
+#include "absl/algorithm/container.h"
+#include "rtc_base/checks.h"
+
+namespace rtc {
+
+VideoSourceBase::VideoSourceBase() = default;
+VideoSourceBase::~VideoSourceBase() = default;
+
+void VideoSourceBase::AddOrUpdateSink(
+    VideoSinkInterface<webrtc::VideoFrame>* sink,
+    const VideoSinkWants& wants) {
+  RTC_DCHECK(sink != nullptr);
+
+  SinkPair* sink_pair = FindSinkPair(sink);
+  if (!sink_pair) {
+    sinks_.push_back(SinkPair(sink, wants));
+  } else {
+    sink_pair->wants = wants;
+  }
+}
+
+void VideoSourceBase::RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  RTC_DCHECK(sink != nullptr);
+  RTC_DCHECK(FindSinkPair(sink));
+  sinks_.erase(std::remove_if(sinks_.begin(), sinks_.end(),
+                              [sink](const SinkPair& sink_pair) {
+                                return sink_pair.sink == sink;
+                              }),
+               sinks_.end());
+}
+
+VideoSourceBase::SinkPair* VideoSourceBase::FindSinkPair(
+    const VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  auto sink_pair_it = absl::c_find_if(
+      sinks_,
+      [sink](const SinkPair& sink_pair) { return sink_pair.sink == sink; });
+  if (sink_pair_it != sinks_.end()) {
+    return &*sink_pair_it;
+  }
+  return nullptr;
+}
+
+VideoSourceBaseGuarded::VideoSourceBaseGuarded() = default;
+VideoSourceBaseGuarded::~VideoSourceBaseGuarded() = default;
+
+void VideoSourceBaseGuarded::AddOrUpdateSink(
+    VideoSinkInterface<webrtc::VideoFrame>* sink,
+    const VideoSinkWants& wants) {
+  RTC_DCHECK_RUN_ON(&source_sequence_);
+  RTC_DCHECK(sink != nullptr);
+
+  SinkPair* sink_pair = FindSinkPair(sink);
+  if (!sink_pair) {
+    sinks_.push_back(SinkPair(sink, wants));
+  } else {
+    sink_pair->wants = wants;
+  }
+}
+
+void VideoSourceBaseGuarded::RemoveSink(
+    VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  RTC_DCHECK_RUN_ON(&source_sequence_);
+  RTC_DCHECK(sink != nullptr);
+  RTC_DCHECK(FindSinkPair(sink));
+  sinks_.erase(std::remove_if(sinks_.begin(), sinks_.end(),
+                              [sink](const SinkPair& sink_pair) {
+                                return sink_pair.sink == sink;
+                              }),
+               sinks_.end());
+}
+
+VideoSourceBaseGuarded::SinkPair* VideoSourceBaseGuarded::FindSinkPair(
+    const VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  RTC_DCHECK_RUN_ON(&source_sequence_);
+  auto sink_pair_it = absl::c_find_if(
+      sinks_,
+      [sink](const SinkPair& sink_pair) { return sink_pair.sink == sink; });
+  if (sink_pair_it != sinks_.end()) {
+    return &*sink_pair_it;
+  }
+  return nullptr;
+}
+
+const std::vector<VideoSourceBaseGuarded::SinkPair>&
+VideoSourceBaseGuarded::sink_pairs() const {
+  RTC_DCHECK_RUN_ON(&source_sequence_);
+  return sinks_;
+}
+
+}  // namespace rtc

TMessagesProj/jni/voip/webrtc/media/base/video_source_base.h

@@ -0,0 +1,83 @@
+/*
+ *  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 MEDIA_BASE_VIDEO_SOURCE_BASE_H_
+#define MEDIA_BASE_VIDEO_SOURCE_BASE_H_
+
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace rtc {
+
+// VideoSourceBase is not thread safe. Before using this class, consider using
+// VideoSourceBaseGuarded below instead, which is an identical implementation
+// but applies a sequence checker to help protect internal state.
+// TODO(bugs.webrtc.org/12780): Delete this class.
+class VideoSourceBase : public VideoSourceInterface<webrtc::VideoFrame> {
+ public:
+  VideoSourceBase();
+  ~VideoSourceBase() override;
+  void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
+                       const VideoSinkWants& wants) override;
+  void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
+ protected:
+  struct SinkPair {
+    SinkPair(VideoSinkInterface<webrtc::VideoFrame>* sink, VideoSinkWants wants)
+        : sink(sink), wants(wants) {}
+    VideoSinkInterface<webrtc::VideoFrame>* sink;
+    VideoSinkWants wants;
+  };
+  SinkPair* FindSinkPair(const VideoSinkInterface<webrtc::VideoFrame>* sink);
+
+  const std::vector<SinkPair>& sink_pairs() const { return sinks_; }
+
+ private:
+  std::vector<SinkPair> sinks_;
+};
+
+// VideoSourceBaseGuarded assumes that operations related to sinks, occur on the
+// same TQ/thread that the object was constructed on.
+class VideoSourceBaseGuarded : public VideoSourceInterface<webrtc::VideoFrame> {
+ public:
+  VideoSourceBaseGuarded();
+  ~VideoSourceBaseGuarded() override;
+
+  void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
+                       const VideoSinkWants& wants) override;
+  void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
+ protected:
+  struct SinkPair {
+    SinkPair(VideoSinkInterface<webrtc::VideoFrame>* sink, VideoSinkWants wants)
+        : sink(sink), wants(wants) {}
+    VideoSinkInterface<webrtc::VideoFrame>* sink;
+    VideoSinkWants wants;
+  };
+
+  SinkPair* FindSinkPair(const VideoSinkInterface<webrtc::VideoFrame>* sink);
+  const std::vector<SinkPair>& sink_pairs() const;
+
+  // Keep the `source_sequence_` checker protected to allow sub classes the
+  // ability to call Detach() if/when appropriate.
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker source_sequence_;
+
+ private:
+  std::vector<SinkPair> sinks_ RTC_GUARDED_BY(&source_sequence_);
+};
+
+}  // namespace rtc
+
+#endif  // MEDIA_BASE_VIDEO_SOURCE_BASE_H_

TMessagesProj/jni/voip/webrtc/media/engine/adm_helpers.cc

@@ -0,0 +1,82 @@
+/*
+ *  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 "media/engine/adm_helpers.h"
+
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace adm_helpers {
+
+// On Windows Vista and newer, Microsoft introduced the concept of "Default
+// Communications Device". This means that there are two types of default
+// devices (old Wave Audio style default and Default Communications Device).
+//
+// On Windows systems which only support Wave Audio style default, uses either
+// -1 or 0 to select the default device.
+//
+// Using a #define for AUDIO_DEVICE since we will call *different* versions of
+// the ADM functions, depending on the ID type.
+#if defined(WEBRTC_WIN)
+#define AUDIO_DEVICE_ID \
+  (AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice)
+#else
+#define AUDIO_DEVICE_ID (0u)
+#endif  // defined(WEBRTC_WIN)
+
+void Init(AudioDeviceModule* adm) {
+  RTC_DCHECK(adm);
+
+  RTC_CHECK_EQ(0, adm->Init()) << "Failed to initialize the ADM.";
+
+  // Playout device.
+  {
+    if (adm->SetPlayoutDevice(AUDIO_DEVICE_ID) != 0) {
+      RTC_LOG(LS_ERROR) << "Unable to set playout device.";
+      return;
+    }
+    if (adm->InitSpeaker() != 0) {
+      RTC_LOG(LS_ERROR) << "Unable to access speaker.";
+    }
+
+    // Set number of channels
+    bool available = false;
+    if (adm->StereoPlayoutIsAvailable(&available) != 0) {
+      RTC_LOG(LS_ERROR) << "Failed to query stereo playout.";
+    }
+    if (adm->SetStereoPlayout(available) != 0) {
+      RTC_LOG(LS_ERROR) << "Failed to set stereo playout mode.";
+    }
+  }
+
+  // Recording device.
+  {
+    if (adm->SetRecordingDevice(AUDIO_DEVICE_ID) != 0) {
+      RTC_LOG(LS_ERROR) << "Unable to set recording device.";
+      return;
+    }
+    if (adm->InitMicrophone() != 0) {
+      RTC_LOG(LS_ERROR) << "Unable to access microphone.";
+    }
+
+    // Set number of channels
+    bool available = false;
+    if (adm->StereoRecordingIsAvailable(&available) != 0) {
+      RTC_LOG(LS_ERROR) << "Failed to query stereo recording.";
+    }
+    if (adm->SetStereoRecording(available) != 0) {
+      RTC_LOG(LS_ERROR) << "Failed to set stereo recording mode.";
+    }
+  }
+}
+}  // namespace adm_helpers
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/engine/adm_helpers.h

@@ -0,0 +1,25 @@
+/*
+ *  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 MEDIA_ENGINE_ADM_HELPERS_H_
+#define MEDIA_ENGINE_ADM_HELPERS_H_
+
+namespace webrtc {
+
+class AudioDeviceModule;
+
+namespace adm_helpers {
+
+void Init(AudioDeviceModule* adm);
+
+}  // namespace adm_helpers
+}  // namespace webrtc
+
+#endif  // MEDIA_ENGINE_ADM_HELPERS_H_

TMessagesProj/jni/voip/webrtc/media/engine/fake_video_codec_factory.cc

@@ -0,0 +1,69 @@
+/*
+ *  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 "media/engine/fake_video_codec_factory.h"
+
+#include <memory>
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/fake_decoder.h"
+#include "test/fake_encoder.h"
+
+namespace {
+
+static const char kFakeCodecFactoryCodecName[] = "FakeCodec";
+
+}  // anonymous namespace
+
+namespace webrtc {
+
+FakeVideoEncoderFactory::FakeVideoEncoderFactory() = default;
+
+// static
+std::unique_ptr<VideoEncoder> FakeVideoEncoderFactory::CreateVideoEncoder() {
+  return std::make_unique<test::FakeEncoder>(Clock::GetRealTimeClock());
+}
+
+std::vector<SdpVideoFormat> FakeVideoEncoderFactory::GetSupportedFormats()
+    const {
+  return std::vector<SdpVideoFormat>(
+      1, SdpVideoFormat(kFakeCodecFactoryCodecName));
+}
+
+std::unique_ptr<VideoEncoder> FakeVideoEncoderFactory::CreateVideoEncoder(
+    const SdpVideoFormat& format) {
+  return std::make_unique<test::FakeEncoder>(Clock::GetRealTimeClock());
+}
+
+FakeVideoDecoderFactory::FakeVideoDecoderFactory() = default;
+
+// static
+std::unique_ptr<VideoDecoder> FakeVideoDecoderFactory::CreateVideoDecoder() {
+  return std::make_unique<test::FakeDecoder>();
+}
+
+std::vector<SdpVideoFormat> FakeVideoDecoderFactory::GetSupportedFormats()
+    const {
+  return std::vector<SdpVideoFormat>(
+      1, SdpVideoFormat(kFakeCodecFactoryCodecName));
+}
+
+std::unique_ptr<VideoDecoder> FakeVideoDecoderFactory::CreateVideoDecoder(
+    const SdpVideoFormat& format) {
+  return std::make_unique<test::FakeDecoder>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/engine/fake_video_codec_factory.h

@@ -0,0 +1,53 @@
+/*
+ *  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 MEDIA_ENGINE_FAKE_VIDEO_CODEC_FACTORY_H_
+#define MEDIA_ENGINE_FAKE_VIDEO_CODEC_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Provides a fake video encoder instance that produces frames large enough for
+// the given bitrate constraints.
+class RTC_EXPORT FakeVideoEncoderFactory : public VideoEncoderFactory {
+ public:
+  FakeVideoEncoderFactory();
+
+  static std::unique_ptr<VideoEncoder> CreateVideoEncoder();
+
+  // VideoEncoderFactory implementation
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override;
+};
+
+// Provides a fake video decoder instance that ignores the given bitstream and
+// produces frames.
+class RTC_EXPORT FakeVideoDecoderFactory : public VideoDecoderFactory {
+ public:
+  FakeVideoDecoderFactory();
+
+  static std::unique_ptr<VideoDecoder> CreateVideoDecoder();
+
+  // VideoDecoderFactory implementation
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      const SdpVideoFormat& format) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MEDIA_ENGINE_FAKE_VIDEO_CODEC_FACTORY_H_

TMessagesProj/jni/voip/webrtc/media/engine/fake_webrtc_call.cc

@@ -0,0 +1,763 @@
+/*
+ *  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 "media/engine/fake_webrtc_call.h"
+
+#include <cstdint>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "api/call/audio_sink.h"
+#include "api/units/timestamp.h"
+#include "call/packet_receiver.h"
+#include "media/base/media_channel.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/thread.h"
+#include "video/config/encoder_stream_factory.h"
+
+namespace cricket {
+
+using ::webrtc::ParseRtpSsrc;
+
+FakeAudioSendStream::FakeAudioSendStream(
+    int id,
+    const webrtc::AudioSendStream::Config& config)
+    : id_(id), config_(config) {}
+
+void FakeAudioSendStream::Reconfigure(
+    const webrtc::AudioSendStream::Config& config,
+    webrtc::SetParametersCallback callback) {
+  config_ = config;
+  webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK());
+}
+
+const webrtc::AudioSendStream::Config& FakeAudioSendStream::GetConfig() const {
+  return config_;
+}
+
+void FakeAudioSendStream::SetStats(
+    const webrtc::AudioSendStream::Stats& stats) {
+  stats_ = stats;
+}
+
+FakeAudioSendStream::TelephoneEvent
+FakeAudioSendStream::GetLatestTelephoneEvent() const {
+  return latest_telephone_event_;
+}
+
+bool FakeAudioSendStream::SendTelephoneEvent(int payload_type,
+                                             int payload_frequency,
+                                             int event,
+                                             int duration_ms) {
+  latest_telephone_event_.payload_type = payload_type;
+  latest_telephone_event_.payload_frequency = payload_frequency;
+  latest_telephone_event_.event_code = event;
+  latest_telephone_event_.duration_ms = duration_ms;
+  return true;
+}
+
+void FakeAudioSendStream::SetMuted(bool muted) {
+  muted_ = muted;
+}
+
+webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
+  return stats_;
+}
+
+webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats(
+    bool /*has_remote_tracks*/) const {
+  return stats_;
+}
+
+FakeAudioReceiveStream::FakeAudioReceiveStream(
+    int id,
+    const webrtc::AudioReceiveStreamInterface::Config& config)
+    : id_(id), config_(config) {}
+
+const webrtc::AudioReceiveStreamInterface::Config&
+FakeAudioReceiveStream::GetConfig() const {
+  return config_;
+}
+
+void FakeAudioReceiveStream::SetStats(
+    const webrtc::AudioReceiveStreamInterface::Stats& stats) {
+  stats_ = stats;
+}
+
+bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
+                                              size_t length) const {
+  return last_packet_ == rtc::Buffer(data, length);
+}
+
+bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                        size_t length,
+                                        int64_t /* packet_time_us */) {
+  ++received_packets_;
+  last_packet_.SetData(packet, length);
+  return true;
+}
+
+void FakeAudioReceiveStream::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  config_.frame_transformer = std::move(frame_transformer);
+}
+
+void FakeAudioReceiveStream::SetDecoderMap(
+    std::map<int, webrtc::SdpAudioFormat> decoder_map) {
+  config_.decoder_map = std::move(decoder_map);
+}
+
+void FakeAudioReceiveStream::SetNackHistory(int history_ms) {
+  config_.rtp.nack.rtp_history_ms = history_ms;
+}
+
+void FakeAudioReceiveStream::SetNonSenderRttMeasurement(bool enabled) {
+  config_.enable_non_sender_rtt = enabled;
+}
+
+void FakeAudioReceiveStream::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  config_.frame_decryptor = std::move(frame_decryptor);
+}
+
+webrtc::AudioReceiveStreamInterface::Stats FakeAudioReceiveStream::GetStats(
+    bool get_and_clear_legacy_stats) const {
+  return stats_;
+}
+
+void FakeAudioReceiveStream::SetSink(webrtc::AudioSinkInterface* sink) {
+  sink_ = sink;
+}
+
+void FakeAudioReceiveStream::SetGain(float gain) {
+  gain_ = gain;
+}
+
+FakeVideoSendStream::FakeVideoSendStream(
+    webrtc::VideoSendStream::Config config,
+    webrtc::VideoEncoderConfig encoder_config)
+    : sending_(false),
+      config_(std::move(config)),
+      codec_settings_set_(false),
+      resolution_scaling_enabled_(false),
+      framerate_scaling_enabled_(false),
+      source_(nullptr),
+      num_swapped_frames_(0) {
+  RTC_DCHECK(config.encoder_settings.encoder_factory != nullptr);
+  RTC_DCHECK(config.encoder_settings.bitrate_allocator_factory != nullptr);
+  ReconfigureVideoEncoder(std::move(encoder_config));
+}
+
+FakeVideoSendStream::~FakeVideoSendStream() {
+  if (source_)
+    source_->RemoveSink(this);
+}
+
+const webrtc::VideoSendStream::Config& FakeVideoSendStream::GetConfig() const {
+  return config_;
+}
+
+const webrtc::VideoEncoderConfig& FakeVideoSendStream::GetEncoderConfig()
+    const {
+  return encoder_config_;
+}
+
+const std::vector<webrtc::VideoStream>& FakeVideoSendStream::GetVideoStreams()
+    const {
+  return video_streams_;
+}
+
+bool FakeVideoSendStream::IsSending() const {
+  return sending_;
+}
+
+bool FakeVideoSendStream::GetVp8Settings(
+    webrtc::VideoCodecVP8* settings) const {
+  if (!codec_settings_set_) {
+    return false;
+  }
+
+  *settings = codec_specific_settings_.vp8;
+  return true;
+}
+
+bool FakeVideoSendStream::GetVp9Settings(
+    webrtc::VideoCodecVP9* settings) const {
+  if (!codec_settings_set_) {
+    return false;
+  }
+
+  *settings = codec_specific_settings_.vp9;
+  return true;
+}
+
+bool FakeVideoSendStream::GetH264Settings(
+    webrtc::VideoCodecH264* settings) const {
+  if (!codec_settings_set_) {
+    return false;
+  }
+
+  *settings = codec_specific_settings_.h264;
+  return true;
+}
+
+bool FakeVideoSendStream::GetAv1Settings(
+    webrtc::VideoCodecAV1* settings) const {
+  if (!codec_settings_set_) {
+    return false;
+  }
+
+  *settings = codec_specific_settings_.av1;
+  return true;
+}
+
+int FakeVideoSendStream::GetNumberOfSwappedFrames() const {
+  return num_swapped_frames_;
+}
+
+int FakeVideoSendStream::GetLastWidth() const {
+  return last_frame_->width();
+}
+
+int FakeVideoSendStream::GetLastHeight() const {
+  return last_frame_->height();
+}
+
+int64_t FakeVideoSendStream::GetLastTimestamp() const {
+  RTC_DCHECK(last_frame_->ntp_time_ms() == 0);
+  return last_frame_->render_time_ms();
+}
+
+void FakeVideoSendStream::OnFrame(const webrtc::VideoFrame& frame) {
+  ++num_swapped_frames_;
+  if (!last_frame_ || frame.width() != last_frame_->width() ||
+      frame.height() != last_frame_->height() ||
+      frame.rotation() != last_frame_->rotation()) {
+    if (encoder_config_.video_stream_factory) {
+      // Note: only tests set their own EncoderStreamFactory...
+      video_streams_ =
+          encoder_config_.video_stream_factory->CreateEncoderStreams(
+              frame.width(), frame.height(), encoder_config_);
+    } else {
+      webrtc::VideoEncoder::EncoderInfo encoder_info;
+      rtc::scoped_refptr<
+          webrtc::VideoEncoderConfig::VideoStreamFactoryInterface>
+          factory = rtc::make_ref_counted<cricket::EncoderStreamFactory>(
+              encoder_config_.video_format.name, encoder_config_.max_qp,
+              encoder_config_.content_type ==
+                  webrtc::VideoEncoderConfig::ContentType::kScreen,
+              encoder_config_.legacy_conference_mode, encoder_info);
+
+      video_streams_ = factory->CreateEncoderStreams(
+          frame.width(), frame.height(), encoder_config_);
+    }
+  }
+  last_frame_ = frame;
+}
+
+void FakeVideoSendStream::SetStats(
+    const webrtc::VideoSendStream::Stats& stats) {
+  stats_ = stats;
+}
+
+webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
+  return stats_;
+}
+
+void FakeVideoSendStream::ReconfigureVideoEncoder(
+    webrtc::VideoEncoderConfig config) {
+  ReconfigureVideoEncoder(std::move(config), nullptr);
+}
+
+void FakeVideoSendStream::ReconfigureVideoEncoder(
+    webrtc::VideoEncoderConfig config,
+    webrtc::SetParametersCallback callback) {
+  int width, height;
+  if (last_frame_) {
+    width = last_frame_->width();
+    height = last_frame_->height();
+  } else {
+    width = height = 0;
+  }
+  if (config.video_stream_factory) {
+    // Note: only tests set their own EncoderStreamFactory...
+    video_streams_ = config.video_stream_factory->CreateEncoderStreams(
+        width, height, config);
+  } else {
+    webrtc::VideoEncoder::EncoderInfo encoder_info;
+    rtc::scoped_refptr<webrtc::VideoEncoderConfig::VideoStreamFactoryInterface>
+        factory = rtc::make_ref_counted<cricket::EncoderStreamFactory>(
+            config.video_format.name, config.max_qp,
+            config.content_type ==
+                webrtc::VideoEncoderConfig::ContentType::kScreen,
+            config.legacy_conference_mode, encoder_info);
+
+    video_streams_ = factory->CreateEncoderStreams(width, height, config);
+  }
+
+  if (config.encoder_specific_settings != nullptr) {
+    const unsigned char num_temporal_layers = static_cast<unsigned char>(
+        video_streams_.back().num_temporal_layers.value_or(1));
+    if (config_.rtp.payload_name == "VP8") {
+      config.encoder_specific_settings->FillVideoCodecVp8(
+          &codec_specific_settings_.vp8);
+      if (!video_streams_.empty()) {
+        codec_specific_settings_.vp8.numberOfTemporalLayers =
+            num_temporal_layers;
+      }
+    } else if (config_.rtp.payload_name == "VP9") {
+      config.encoder_specific_settings->FillVideoCodecVp9(
+          &codec_specific_settings_.vp9);
+      if (!video_streams_.empty()) {
+        codec_specific_settings_.vp9.numberOfTemporalLayers =
+            num_temporal_layers;
+      }
+    } else if (config_.rtp.payload_name == "H264") {
+      codec_specific_settings_.h264.numberOfTemporalLayers =
+          num_temporal_layers;
+    } else if (config_.rtp.payload_name == "AV1") {
+      config.encoder_specific_settings->FillVideoCodecAv1(
+          &codec_specific_settings_.av1);
+    } else {
+      ADD_FAILURE() << "Unsupported encoder payload: "
+                    << config_.rtp.payload_name;
+    }
+  }
+  codec_settings_set_ = config.encoder_specific_settings != nullptr;
+  encoder_config_ = std::move(config);
+  ++num_encoder_reconfigurations_;
+  webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK());
+}
+
+void FakeVideoSendStream::Start() {
+  sending_ = true;
+}
+
+void FakeVideoSendStream::Stop() {
+  sending_ = false;
+}
+
+void FakeVideoSendStream::AddAdaptationResource(
+    rtc::scoped_refptr<webrtc::Resource> resource) {}
+
+std::vector<rtc::scoped_refptr<webrtc::Resource>>
+FakeVideoSendStream::GetAdaptationResources() {
+  return {};
+}
+
+void FakeVideoSendStream::SetSource(
+    rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+    const webrtc::DegradationPreference& degradation_preference) {
+  if (source_)
+    source_->RemoveSink(this);
+  source_ = source;
+  switch (degradation_preference) {
+    case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
+      resolution_scaling_enabled_ = true;
+      framerate_scaling_enabled_ = false;
+      break;
+    case webrtc::DegradationPreference::MAINTAIN_RESOLUTION:
+      resolution_scaling_enabled_ = false;
+      framerate_scaling_enabled_ = true;
+      break;
+    case webrtc::DegradationPreference::BALANCED:
+      resolution_scaling_enabled_ = true;
+      framerate_scaling_enabled_ = true;
+      break;
+    case webrtc::DegradationPreference::DISABLED:
+      resolution_scaling_enabled_ = false;
+      framerate_scaling_enabled_ = false;
+      break;
+  }
+  if (source)
+    source->AddOrUpdateSink(this, resolution_scaling_enabled_
+                                      ? sink_wants_
+                                      : rtc::VideoSinkWants());
+}
+
+void FakeVideoSendStream::GenerateKeyFrame(
+    const std::vector<std::string>& rids) {
+  keyframes_requested_by_rid_ = rids;
+}
+
+void FakeVideoSendStream::InjectVideoSinkWants(
+    const rtc::VideoSinkWants& wants) {
+  sink_wants_ = wants;
+  source_->AddOrUpdateSink(this, wants);
+}
+
+FakeVideoReceiveStream::FakeVideoReceiveStream(
+    webrtc::VideoReceiveStreamInterface::Config config)
+    : config_(std::move(config)), receiving_(false) {}
+
+const webrtc::VideoReceiveStreamInterface::Config&
+FakeVideoReceiveStream::GetConfig() const {
+  return config_;
+}
+
+bool FakeVideoReceiveStream::IsReceiving() const {
+  return receiving_;
+}
+
+void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame) {
+  config_.renderer->OnFrame(frame);
+}
+
+webrtc::VideoReceiveStreamInterface::Stats FakeVideoReceiveStream::GetStats()
+    const {
+  return stats_;
+}
+
+void FakeVideoReceiveStream::Start() {
+  receiving_ = true;
+}
+
+void FakeVideoReceiveStream::Stop() {
+  receiving_ = false;
+}
+
+void FakeVideoReceiveStream::SetStats(
+    const webrtc::VideoReceiveStreamInterface::Stats& stats) {
+  stats_ = stats;
+}
+
+FakeFlexfecReceiveStream::FakeFlexfecReceiveStream(
+    const webrtc::FlexfecReceiveStream::Config config)
+    : config_(std::move(config)) {}
+
+const webrtc::FlexfecReceiveStream::Config&
+FakeFlexfecReceiveStream::GetConfig() const {
+  return config_;
+}
+
+void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented.";
+}
+
+FakeCall::FakeCall(webrtc::test::ScopedKeyValueConfig* field_trials)
+    : FakeCall(rtc::Thread::Current(), rtc::Thread::Current(), field_trials) {}
+
+FakeCall::FakeCall(webrtc::TaskQueueBase* worker_thread,
+                   webrtc::TaskQueueBase* network_thread,
+                   webrtc::test::ScopedKeyValueConfig* field_trials)
+    : network_thread_(network_thread),
+      worker_thread_(worker_thread),
+      audio_network_state_(webrtc::kNetworkUp),
+      video_network_state_(webrtc::kNetworkUp),
+      num_created_send_streams_(0),
+      num_created_receive_streams_(0),
+      trials_(field_trials ? field_trials : &fallback_trials_) {}
+
+FakeCall::~FakeCall() {
+  EXPECT_EQ(0u, video_send_streams_.size());
+  EXPECT_EQ(0u, audio_send_streams_.size());
+  EXPECT_EQ(0u, video_receive_streams_.size());
+  EXPECT_EQ(0u, audio_receive_streams_.size());
+}
+
+const std::vector<FakeVideoSendStream*>& FakeCall::GetVideoSendStreams() {
+  return video_send_streams_;
+}
+
+const std::vector<FakeVideoReceiveStream*>& FakeCall::GetVideoReceiveStreams() {
+  return video_receive_streams_;
+}
+
+const FakeVideoReceiveStream* FakeCall::GetVideoReceiveStream(uint32_t ssrc) {
+  for (const auto* p : GetVideoReceiveStreams()) {
+    if (p->GetConfig().rtp.remote_ssrc == ssrc) {
+      return p;
+    }
+  }
+  return nullptr;
+}
+
+const std::vector<FakeAudioSendStream*>& FakeCall::GetAudioSendStreams() {
+  return audio_send_streams_;
+}
+
+const FakeAudioSendStream* FakeCall::GetAudioSendStream(uint32_t ssrc) {
+  for (const auto* p : GetAudioSendStreams()) {
+    if (p->GetConfig().rtp.ssrc == ssrc) {
+      return p;
+    }
+  }
+  return nullptr;
+}
+
+const std::vector<FakeAudioReceiveStream*>& FakeCall::GetAudioReceiveStreams() {
+  return audio_receive_streams_;
+}
+
+const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
+  for (const auto* p : GetAudioReceiveStreams()) {
+    if (p->GetConfig().rtp.remote_ssrc == ssrc) {
+      return p;
+    }
+  }
+  return nullptr;
+}
+
+const std::vector<FakeFlexfecReceiveStream*>&
+FakeCall::GetFlexfecReceiveStreams() {
+  return flexfec_receive_streams_;
+}
+
+webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
+  switch (media) {
+    case webrtc::MediaType::AUDIO:
+      return audio_network_state_;
+    case webrtc::MediaType::VIDEO:
+      return video_network_state_;
+    case webrtc::MediaType::DATA:
+    case webrtc::MediaType::ANY:
+      ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
+      return webrtc::kNetworkDown;
+  }
+  // Even though all the values for the enum class are listed above,the compiler
+  // will emit a warning as the method may be called with a value outside of the
+  // valid enum range, unless this case is also handled.
+  ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
+  return webrtc::kNetworkDown;
+}
+
+webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  FakeAudioSendStream* fake_stream =
+      new FakeAudioSendStream(next_stream_id_++, config);
+  audio_send_streams_.push_back(fake_stream);
+  ++num_created_send_streams_;
+  return fake_stream;
+}
+
+void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+  auto it = absl::c_find(audio_send_streams_,
+                         static_cast<FakeAudioSendStream*>(send_stream));
+  if (it == audio_send_streams_.end()) {
+    ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
+  } else {
+    delete *it;
+    audio_send_streams_.erase(it);
+  }
+}
+
+webrtc::AudioReceiveStreamInterface* FakeCall::CreateAudioReceiveStream(
+    const webrtc::AudioReceiveStreamInterface::Config& config) {
+  audio_receive_streams_.push_back(
+      new FakeAudioReceiveStream(next_stream_id_++, config));
+  ++num_created_receive_streams_;
+  return audio_receive_streams_.back();
+}
+
+void FakeCall::DestroyAudioReceiveStream(
+    webrtc::AudioReceiveStreamInterface* receive_stream) {
+  auto it = absl::c_find(audio_receive_streams_,
+                         static_cast<FakeAudioReceiveStream*>(receive_stream));
+  if (it == audio_receive_streams_.end()) {
+    ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
+  } else {
+    delete *it;
+    audio_receive_streams_.erase(it);
+  }
+}
+
+webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
+    webrtc::VideoSendStream::Config config,
+    webrtc::VideoEncoderConfig encoder_config) {
+  FakeVideoSendStream* fake_stream =
+      new FakeVideoSendStream(std::move(config), std::move(encoder_config));
+  video_send_streams_.push_back(fake_stream);
+  ++num_created_send_streams_;
+  return fake_stream;
+}
+
+void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
+  auto it = absl::c_find(video_send_streams_,
+                         static_cast<FakeVideoSendStream*>(send_stream));
+  if (it == video_send_streams_.end()) {
+    ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
+  } else {
+    delete *it;
+    video_send_streams_.erase(it);
+  }
+}
+
+webrtc::VideoReceiveStreamInterface* FakeCall::CreateVideoReceiveStream(
+    webrtc::VideoReceiveStreamInterface::Config config) {
+  video_receive_streams_.push_back(
+      new FakeVideoReceiveStream(std::move(config)));
+  ++num_created_receive_streams_;
+  return video_receive_streams_.back();
+}
+
+void FakeCall::DestroyVideoReceiveStream(
+    webrtc::VideoReceiveStreamInterface* receive_stream) {
+  auto it = absl::c_find(video_receive_streams_,
+                         static_cast<FakeVideoReceiveStream*>(receive_stream));
+  if (it == video_receive_streams_.end()) {
+    ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter.";
+  } else {
+    delete *it;
+    video_receive_streams_.erase(it);
+  }
+}
+
+webrtc::FlexfecReceiveStream* FakeCall::CreateFlexfecReceiveStream(
+    const webrtc::FlexfecReceiveStream::Config config) {
+  FakeFlexfecReceiveStream* fake_stream =
+      new FakeFlexfecReceiveStream(std::move(config));
+  flexfec_receive_streams_.push_back(fake_stream);
+  ++num_created_receive_streams_;
+  return fake_stream;
+}
+
+void FakeCall::DestroyFlexfecReceiveStream(
+    webrtc::FlexfecReceiveStream* receive_stream) {
+  auto it =
+      absl::c_find(flexfec_receive_streams_,
+                   static_cast<FakeFlexfecReceiveStream*>(receive_stream));
+  if (it == flexfec_receive_streams_.end()) {
+    ADD_FAILURE()
+        << "DestroyFlexfecReceiveStream called with unknown parameter.";
+  } else {
+    delete *it;
+    flexfec_receive_streams_.erase(it);
+  }
+}
+
+void FakeCall::AddAdaptationResource(
+    rtc::scoped_refptr<webrtc::Resource> resource) {}
+
+webrtc::PacketReceiver* FakeCall::Receiver() {
+  return this;
+}
+
+void FakeCall::DeliverRtpPacket(
+    webrtc::MediaType media_type,
+    webrtc::RtpPacketReceived packet,
+    OnUndemuxablePacketHandler undemuxable_packet_handler) {
+  if (!DeliverPacketInternal(media_type, packet.Ssrc(), packet.Buffer(),
+                             packet.arrival_time())) {
+    if (undemuxable_packet_handler(packet)) {
+      DeliverPacketInternal(media_type, packet.Ssrc(), packet.Buffer(),
+                            packet.arrival_time());
+    }
+  }
+  last_received_rtp_packet_ = packet;
+}
+
+bool FakeCall::DeliverPacketInternal(webrtc::MediaType media_type,
+                                     uint32_t ssrc,
+                                     const rtc::CopyOnWriteBuffer& packet,
+                                     webrtc::Timestamp arrival_time) {
+  EXPECT_GE(packet.size(), 12u);
+  RTC_DCHECK(arrival_time.IsFinite());
+  RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
+             media_type == webrtc::MediaType::VIDEO);
+
+  if (media_type == webrtc::MediaType::VIDEO) {
+    for (auto receiver : video_receive_streams_) {
+      if (receiver->GetConfig().rtp.remote_ssrc == ssrc ||
+          receiver->GetConfig().rtp.rtx_ssrc == ssrc) {
+        ++delivered_packets_by_ssrc_[ssrc];
+        return true;
+      }
+    }
+  }
+  if (media_type == webrtc::MediaType::AUDIO) {
+    for (auto receiver : audio_receive_streams_) {
+      if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
+        receiver->DeliverRtp(packet.cdata(), packet.size(), arrival_time.us());
+        ++delivered_packets_by_ssrc_[ssrc];
+        return true;
+      }
+    }
+  }
+  return false;
+}
+
+void FakeCall::SetStats(const webrtc::Call::Stats& stats) {
+  stats_ = stats;
+}
+
+int FakeCall::GetNumCreatedSendStreams() const {
+  return num_created_send_streams_;
+}
+
+int FakeCall::GetNumCreatedReceiveStreams() const {
+  return num_created_receive_streams_;
+}
+
+webrtc::Call::Stats FakeCall::GetStats() const {
+  return stats_;
+}
+
+webrtc::TaskQueueBase* FakeCall::network_thread() const {
+  return network_thread_;
+}
+
+webrtc::TaskQueueBase* FakeCall::worker_thread() const {
+  return worker_thread_;
+}
+
+void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
+                                         webrtc::NetworkState state) {
+  switch (media) {
+    case webrtc::MediaType::AUDIO:
+      audio_network_state_ = state;
+      break;
+    case webrtc::MediaType::VIDEO:
+      video_network_state_ = state;
+      break;
+    case webrtc::MediaType::DATA:
+    case webrtc::MediaType::ANY:
+      ADD_FAILURE()
+          << "SignalChannelNetworkState called with unknown parameter.";
+  }
+}
+
+void FakeCall::OnAudioTransportOverheadChanged(
+    int transport_overhead_per_packet) {}
+
+void FakeCall::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
+                                  uint32_t local_ssrc) {
+  auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
+  fake_stream.SetLocalSsrc(local_ssrc);
+}
+
+void FakeCall::OnLocalSsrcUpdated(webrtc::VideoReceiveStreamInterface& stream,
+                                  uint32_t local_ssrc) {
+  auto& fake_stream = static_cast<FakeVideoReceiveStream&>(stream);
+  fake_stream.SetLocalSsrc(local_ssrc);
+}
+
+void FakeCall::OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream,
+                                  uint32_t local_ssrc) {
+  auto& fake_stream = static_cast<FakeFlexfecReceiveStream&>(stream);
+  fake_stream.SetLocalSsrc(local_ssrc);
+}
+
+void FakeCall::OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
+                                 absl::string_view sync_group) {
+  auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
+  fake_stream.SetSyncGroup(sync_group);
+}
+
+void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  last_sent_packet_ = sent_packet;
+  if (sent_packet.packet_id >= 0) {
+    last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
+  }
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/engine/fake_webrtc_call.h

@@ -0,0 +1,518 @@
+/*
+ *  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.
+ */
+
+// This file contains fake implementations, for use in unit tests, of the
+// following classes:
+//
+//   webrtc::Call
+//   webrtc::AudioSendStream
+//   webrtc::AudioReceiveStreamInterface
+//   webrtc::VideoSendStream
+//   webrtc::VideoReceiveStreamInterface
+
+#ifndef MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
+#define MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video/video_frame.h"
+#include "call/audio_receive_stream.h"
+#include "call/audio_send_stream.h"
+#include "call/call.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/test/mock_rtp_transport_controller_send.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/buffer.h"
+#include "test/scoped_key_value_config.h"
+
+namespace cricket {
+class FakeAudioSendStream final : public webrtc::AudioSendStream {
+ public:
+  struct TelephoneEvent {
+    int payload_type = -1;
+    int payload_frequency = -1;
+    int event_code = 0;
+    int duration_ms = 0;
+  };
+
+  explicit FakeAudioSendStream(int id,
+                               const webrtc::AudioSendStream::Config& config);
+
+  int id() const { return id_; }
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
+  void SetStats(const webrtc::AudioSendStream::Stats& stats);
+  TelephoneEvent GetLatestTelephoneEvent() const;
+  bool IsSending() const { return sending_; }
+  bool muted() const { return muted_; }
+
+ private:
+  // webrtc::AudioSendStream implementation.
+  void Reconfigure(const webrtc::AudioSendStream::Config& config,
+                   webrtc::SetParametersCallback callback) override;
+  void Start() override { sending_ = true; }
+  void Stop() override { sending_ = false; }
+  void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
+  }
+  bool SendTelephoneEvent(int payload_type,
+                          int payload_frequency,
+                          int event,
+                          int duration_ms) override;
+  void SetMuted(bool muted) override;
+  webrtc::AudioSendStream::Stats GetStats() const override;
+  webrtc::AudioSendStream::Stats GetStats(
+      bool has_remote_tracks) const override;
+
+  int id_ = -1;
+  TelephoneEvent latest_telephone_event_;
+  webrtc::AudioSendStream::Config config_;
+  webrtc::AudioSendStream::Stats stats_;
+  bool sending_ = false;
+  bool muted_ = false;
+};
+
+class FakeAudioReceiveStream final
+    : public webrtc::AudioReceiveStreamInterface {
+ public:
+  explicit FakeAudioReceiveStream(
+      int id,
+      const webrtc::AudioReceiveStreamInterface::Config& config);
+
+  int id() const { return id_; }
+  const webrtc::AudioReceiveStreamInterface::Config& GetConfig() const;
+  void SetStats(const webrtc::AudioReceiveStreamInterface::Stats& stats);
+  int received_packets() const { return received_packets_; }
+  bool VerifyLastPacket(const uint8_t* data, size_t length) const;
+  const webrtc::AudioSinkInterface* sink() const { return sink_; }
+  float gain() const { return gain_; }
+  bool DeliverRtp(const uint8_t* packet, size_t length, int64_t packet_time_us);
+  bool started() const { return started_; }
+  int base_mininum_playout_delay_ms() const {
+    return base_mininum_playout_delay_ms_;
+  }
+
+  void SetLocalSsrc(uint32_t local_ssrc) {
+    config_.rtp.local_ssrc = local_ssrc;
+  }
+
+  void SetSyncGroup(absl::string_view sync_group) {
+    config_.sync_group = std::string(sync_group);
+  }
+
+  uint32_t remote_ssrc() const override { return config_.rtp.remote_ssrc; }
+  void Start() override { started_ = true; }
+  void Stop() override { started_ = false; }
+  bool IsRunning() const override { return started_; }
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  void SetDecoderMap(
+      std::map<int, webrtc::SdpAudioFormat> decoder_map) override;
+  void SetNackHistory(int history_ms) override;
+  void SetNonSenderRttMeasurement(bool enabled) override;
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+
+  webrtc::AudioReceiveStreamInterface::Stats GetStats(
+      bool get_and_clear_legacy_stats) const override;
+  void SetSink(webrtc::AudioSinkInterface* sink) override;
+  void SetGain(float gain) override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
+    base_mininum_playout_delay_ms_ = delay_ms;
+    return true;
+  }
+  int GetBaseMinimumPlayoutDelayMs() const override {
+    return base_mininum_playout_delay_ms_;
+  }
+  std::vector<webrtc::RtpSource> GetSources() const override {
+    return std::vector<webrtc::RtpSource>();
+  }
+
+ private:
+  int id_ = -1;
+  webrtc::AudioReceiveStreamInterface::Config config_;
+  webrtc::AudioReceiveStreamInterface::Stats stats_;
+  int received_packets_ = 0;
+  webrtc::AudioSinkInterface* sink_ = nullptr;
+  float gain_ = 1.0f;
+  rtc::Buffer last_packet_;
+  bool started_ = false;
+  int base_mininum_playout_delay_ms_ = 0;
+};
+
+class FakeVideoSendStream final
+    : public webrtc::VideoSendStream,
+      public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  FakeVideoSendStream(webrtc::VideoSendStream::Config config,
+                      webrtc::VideoEncoderConfig encoder_config);
+  ~FakeVideoSendStream() override;
+  const webrtc::VideoSendStream::Config& GetConfig() const;
+  const webrtc::VideoEncoderConfig& GetEncoderConfig() const;
+  const std::vector<webrtc::VideoStream>& GetVideoStreams() const;
+
+  bool IsSending() const;
+  bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
+  bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
+  bool GetH264Settings(webrtc::VideoCodecH264* settings) const;
+  bool GetAv1Settings(webrtc::VideoCodecAV1* settings) const;
+
+  int GetNumberOfSwappedFrames() const;
+  int GetLastWidth() const;
+  int GetLastHeight() const;
+  int64_t GetLastTimestamp() const;
+  void SetStats(const webrtc::VideoSendStream::Stats& stats);
+  int num_encoder_reconfigurations() const {
+    return num_encoder_reconfigurations_;
+  }
+
+  bool resolution_scaling_enabled() const {
+    return resolution_scaling_enabled_;
+  }
+  bool framerate_scaling_enabled() const { return framerate_scaling_enabled_; }
+  void InjectVideoSinkWants(const rtc::VideoSinkWants& wants);
+
+  rtc::VideoSourceInterface<webrtc::VideoFrame>* source() const {
+    return source_;
+  }
+  void GenerateKeyFrame(const std::vector<std::string>& rids);
+  const std::vector<std::string>& GetKeyFramesRequested() const {
+    return keyframes_requested_by_rid_;
+  }
+
+ private:
+  // rtc::VideoSinkInterface<VideoFrame> implementation.
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+  // webrtc::VideoSendStream implementation.
+  void Start() override;
+  void Stop() override;
+  bool started() override { return IsSending(); }
+  void AddAdaptationResource(
+      rtc::scoped_refptr<webrtc::Resource> resource) override;
+  std::vector<rtc::scoped_refptr<webrtc::Resource>> GetAdaptationResources()
+      override;
+  void SetSource(
+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+      const webrtc::DegradationPreference& degradation_preference) override;
+  webrtc::VideoSendStream::Stats GetStats() override;
+
+  void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override;
+  void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config,
+                               webrtc::SetParametersCallback callback) override;
+
+  bool sending_;
+  webrtc::VideoSendStream::Config config_;
+  webrtc::VideoEncoderConfig encoder_config_;
+  std::vector<webrtc::VideoStream> video_streams_;
+  rtc::VideoSinkWants sink_wants_;
+
+  bool codec_settings_set_;
+  union CodecSpecificSettings {
+    webrtc::VideoCodecVP8 vp8;
+    webrtc::VideoCodecVP9 vp9;
+    webrtc::VideoCodecH264 h264;
+    webrtc::VideoCodecAV1 av1;
+  } codec_specific_settings_;
+  bool resolution_scaling_enabled_;
+  bool framerate_scaling_enabled_;
+  rtc::VideoSourceInterface<webrtc::VideoFrame>* source_;
+  int num_swapped_frames_;
+  absl::optional<webrtc::VideoFrame> last_frame_;
+  webrtc::VideoSendStream::Stats stats_;
+  int num_encoder_reconfigurations_ = 0;
+  std::vector<std::string> keyframes_requested_by_rid_;
+};
+
+class FakeVideoReceiveStream final
+    : public webrtc::VideoReceiveStreamInterface {
+ public:
+  explicit FakeVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface::Config config);
+
+  const webrtc::VideoReceiveStreamInterface::Config& GetConfig() const;
+
+  bool IsReceiving() const;
+
+  void InjectFrame(const webrtc::VideoFrame& frame);
+
+  void SetStats(const webrtc::VideoReceiveStreamInterface::Stats& stats);
+
+  std::vector<webrtc::RtpSource> GetSources() const override {
+    return std::vector<webrtc::RtpSource>();
+  }
+
+  int base_mininum_playout_delay_ms() const {
+    return base_mininum_playout_delay_ms_;
+  }
+
+  void SetLocalSsrc(uint32_t local_ssrc) {
+    config_.rtp.local_ssrc = local_ssrc;
+  }
+
+  void UpdateRtxSsrc(uint32_t ssrc) { config_.rtp.rtx_ssrc = ssrc; }
+
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override {}
+
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override {}
+
+  RecordingState SetAndGetRecordingState(RecordingState state,
+                                         bool generate_key_frame) override {
+    return RecordingState();
+  }
+  void GenerateKeyFrame() override {}
+
+  void SetRtcpMode(webrtc::RtcpMode mode) override {
+    config_.rtp.rtcp_mode = mode;
+  }
+
+  void SetFlexFecProtection(webrtc::RtpPacketSinkInterface* sink) override {
+    config_.rtp.packet_sink_ = sink;
+    config_.rtp.protected_by_flexfec = (sink != nullptr);
+  }
+
+  void SetLossNotificationEnabled(bool enabled) override {
+    config_.rtp.lntf.enabled = enabled;
+  }
+
+  void SetNackHistory(webrtc::TimeDelta history) override {
+    config_.rtp.nack.rtp_history_ms = history.ms();
+  }
+
+  void SetProtectionPayloadTypes(int red_payload_type,
+                                 int ulpfec_payload_type) override {
+    config_.rtp.red_payload_type = red_payload_type;
+    config_.rtp.ulpfec_payload_type = ulpfec_payload_type;
+  }
+
+  void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) override {
+    config_.rtp.rtcp_xr = rtcp_xr;
+  }
+
+  void SetAssociatedPayloadTypes(std::map<int, int> associated_payload_types) {
+    config_.rtp.rtx_associated_payload_types =
+        std::move(associated_payload_types);
+  }
+
+  void Start() override;
+  void Stop() override;
+
+  webrtc::VideoReceiveStreamInterface::Stats GetStats() const override;
+
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
+    base_mininum_playout_delay_ms_ = delay_ms;
+    return true;
+  }
+
+  int GetBaseMinimumPlayoutDelayMs() const override {
+    return base_mininum_playout_delay_ms_;
+  }
+
+ private:
+  webrtc::VideoReceiveStreamInterface::Config config_;
+  bool receiving_;
+  webrtc::VideoReceiveStreamInterface::Stats stats_;
+
+  int base_mininum_playout_delay_ms_ = 0;
+};
+
+class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
+ public:
+  explicit FakeFlexfecReceiveStream(
+      const webrtc::FlexfecReceiveStream::Config config);
+
+  void SetLocalSsrc(uint32_t local_ssrc) {
+    config_.rtp.local_ssrc = local_ssrc;
+  }
+
+  void SetRtcpMode(webrtc::RtcpMode mode) override { config_.rtcp_mode = mode; }
+
+  int payload_type() const override { return config_.payload_type; }
+  void SetPayloadType(int payload_type) override {
+    config_.payload_type = payload_type;
+  }
+
+  const webrtc::FlexfecReceiveStream::Config& GetConfig() const;
+
+  uint32_t remote_ssrc() const { return config_.rtp.remote_ssrc; }
+
+  const webrtc::ReceiveStatistics* GetStats() const override { return nullptr; }
+
+ private:
+  void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
+
+  webrtc::FlexfecReceiveStream::Config config_;
+};
+
+class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
+ public:
+  explicit FakeCall(webrtc::test::ScopedKeyValueConfig* field_trials = nullptr);
+  FakeCall(webrtc::TaskQueueBase* worker_thread,
+           webrtc::TaskQueueBase* network_thread,
+           webrtc::test::ScopedKeyValueConfig* field_trials = nullptr);
+  ~FakeCall() override;
+
+  webrtc::MockRtpTransportControllerSend* GetMockTransportControllerSend() {
+    return &transport_controller_send_;
+  }
+
+  const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
+  const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
+
+  const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
+  const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
+  const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
+  const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
+  const FakeVideoReceiveStream* GetVideoReceiveStream(uint32_t ssrc);
+
+  const std::vector<FakeFlexfecReceiveStream*>& GetFlexfecReceiveStreams();
+
+  rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
+  const webrtc::RtpPacketReceived& last_received_rtp_packet() const {
+    return last_received_rtp_packet_;
+  }
+  size_t GetDeliveredPacketsForSsrc(uint32_t ssrc) const {
+    auto it = delivered_packets_by_ssrc_.find(ssrc);
+    return it != delivered_packets_by_ssrc_.end() ? it->second : 0u;
+  }
+
+  // This is useful if we care about the last media packet (with id populated)
+  // but not the last ICE packet (with -1 ID).
+  int last_sent_nonnegative_packet_id() const {
+    return last_sent_nonnegative_packet_id_;
+  }
+
+  webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const;
+  int GetNumCreatedSendStreams() const;
+  int GetNumCreatedReceiveStreams() const;
+  void SetStats(const webrtc::Call::Stats& stats);
+
+  void SetClientBitratePreferences(
+      const webrtc::BitrateSettings& preferences) override {}
+
+  void SetFieldTrial(const std::string& field_trial_string) {
+    trials_overrides_ = std::make_unique<webrtc::test::ScopedKeyValueConfig>(
+        *trials_, field_trial_string);
+  }
+
+  const webrtc::FieldTrialsView& trials() const override { return *trials_; }
+
+ private:
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
+  webrtc::AudioReceiveStreamInterface* CreateAudioReceiveStream(
+      const webrtc::AudioReceiveStreamInterface::Config& config) override;
+  void DestroyAudioReceiveStream(
+      webrtc::AudioReceiveStreamInterface* receive_stream) override;
+
+  webrtc::VideoSendStream* CreateVideoSendStream(
+      webrtc::VideoSendStream::Config config,
+      webrtc::VideoEncoderConfig encoder_config) override;
+  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
+
+  webrtc::VideoReceiveStreamInterface* CreateVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface::Config config) override;
+  void DestroyVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface* receive_stream) override;
+
+  webrtc::FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      const webrtc::FlexfecReceiveStream::Config config) override;
+  void DestroyFlexfecReceiveStream(
+      webrtc::FlexfecReceiveStream* receive_stream) override;
+
+  void AddAdaptationResource(
+      rtc::scoped_refptr<webrtc::Resource> resource) override;
+
+  webrtc::PacketReceiver* Receiver() override;
+
+  void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override {}
+
+  void DeliverRtpPacket(
+      webrtc::MediaType media_type,
+      webrtc::RtpPacketReceived packet,
+      OnUndemuxablePacketHandler un_demuxable_packet_handler) override;
+
+  bool DeliverPacketInternal(webrtc::MediaType media_type,
+                             uint32_t ssrc,
+                             const rtc::CopyOnWriteBuffer& packet,
+                             webrtc::Timestamp arrival_time);
+
+  webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend()
+      override {
+    return &transport_controller_send_;
+  }
+
+  webrtc::Call::Stats GetStats() const override;
+
+  webrtc::TaskQueueBase* network_thread() const override;
+  webrtc::TaskQueueBase* worker_thread() const override;
+
+  void SignalChannelNetworkState(webrtc::MediaType media,
+                                 webrtc::NetworkState state) override;
+  void OnAudioTransportOverheadChanged(
+      int transport_overhead_per_packet) override;
+  void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
+                          uint32_t local_ssrc) override;
+  void OnLocalSsrcUpdated(webrtc::VideoReceiveStreamInterface& stream,
+                          uint32_t local_ssrc) override;
+  void OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream,
+                          uint32_t local_ssrc) override;
+  void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
+                         absl::string_view sync_group) override;
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+
+  webrtc::TaskQueueBase* const network_thread_;
+  webrtc::TaskQueueBase* const worker_thread_;
+
+  ::testing::NiceMock<webrtc::MockRtpTransportControllerSend>
+      transport_controller_send_;
+
+  webrtc::NetworkState audio_network_state_;
+  webrtc::NetworkState video_network_state_;
+  rtc::SentPacket last_sent_packet_;
+  webrtc::RtpPacketReceived last_received_rtp_packet_;
+  int last_sent_nonnegative_packet_id_ = -1;
+  int next_stream_id_ = 665;
+  webrtc::Call::Stats stats_;
+  std::vector<FakeVideoSendStream*> video_send_streams_;
+  std::vector<FakeAudioSendStream*> audio_send_streams_;
+  std::vector<FakeVideoReceiveStream*> video_receive_streams_;
+  std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
+  std::vector<FakeFlexfecReceiveStream*> flexfec_receive_streams_;
+  std::map<uint32_t, size_t> delivered_packets_by_ssrc_;
+
+  int num_created_send_streams_;
+  int num_created_receive_streams_;
+
+  // The field trials that are in use, either supplied by caller
+  // or pointer to &fallback_trials_.
+  webrtc::test::ScopedKeyValueConfig* trials_;
+
+  // fallback_trials_ is used if caller does not provide any field trials.
+  webrtc::test::ScopedKeyValueConfig fallback_trials_;
+
+  // An extra field trial that can be set using SetFieldTrial.
+  std::unique_ptr<webrtc::test::ScopedKeyValueConfig> trials_overrides_;
+};
+
+}  // namespace cricket
+#endif  // MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_

TMessagesProj/jni/voip/webrtc/media/engine/fake_webrtc_video_engine.cc

@@ -0,0 +1,304 @@
+/*
+ *  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 "media/engine/fake_webrtc_video_engine.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "media/base/codec.h"
+#include "media/base/media_constants.h"
+#include "media/engine/simulcast_encoder_adapter.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/time_utils.h"
+
+namespace cricket {
+
+namespace {
+
+static constexpr webrtc::TimeDelta kEventTimeout =
+    webrtc::TimeDelta::Seconds(10);
+
+bool IsScalabilityModeSupported(
+    const std::vector<webrtc::SdpVideoFormat>& formats,
+    absl::optional<std::string> scalability_mode) {
+  if (!scalability_mode.has_value()) {
+    return true;
+  }
+  for (const auto& format : formats) {
+    for (const auto& mode : format.scalability_modes) {
+      if (ScalabilityModeToString(mode) == scalability_mode)
+        return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace
+
+// Decoder.
+FakeWebRtcVideoDecoder::FakeWebRtcVideoDecoder(
+    FakeWebRtcVideoDecoderFactory* factory)
+    : num_frames_received_(0), factory_(factory) {}
+
+FakeWebRtcVideoDecoder::~FakeWebRtcVideoDecoder() {
+  if (factory_) {
+    factory_->DecoderDestroyed(this);
+  }
+}
+
+bool FakeWebRtcVideoDecoder::Configure(const Settings& settings) {
+  return true;
+}
+
+int32_t FakeWebRtcVideoDecoder::Decode(const webrtc::EncodedImage&,
+                                       int64_t) {
+  num_frames_received_++;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeWebRtcVideoDecoder::RegisterDecodeCompleteCallback(
+    webrtc::DecodedImageCallback*) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeWebRtcVideoDecoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int FakeWebRtcVideoDecoder::GetNumFramesReceived() const {
+  return num_frames_received_;
+}
+
+// Decoder factory.
+FakeWebRtcVideoDecoderFactory::FakeWebRtcVideoDecoderFactory()
+    : num_created_decoders_(0) {}
+
+std::vector<webrtc::SdpVideoFormat>
+FakeWebRtcVideoDecoderFactory::GetSupportedFormats() const {
+  std::vector<webrtc::SdpVideoFormat> formats;
+
+  for (const webrtc::SdpVideoFormat& format : supported_codec_formats_) {
+    // Don't add same codec twice.
+    if (!format.IsCodecInList(formats))
+      formats.push_back(format);
+  }
+
+  return formats;
+}
+
+std::unique_ptr<webrtc::VideoDecoder>
+FakeWebRtcVideoDecoderFactory::CreateVideoDecoder(
+    const webrtc::SdpVideoFormat& format) {
+  if (format.IsCodecInList(supported_codec_formats_)) {
+    num_created_decoders_++;
+    std::unique_ptr<FakeWebRtcVideoDecoder> decoder =
+        std::make_unique<FakeWebRtcVideoDecoder>(this);
+    decoders_.push_back(decoder.get());
+    return decoder;
+  }
+
+  return nullptr;
+}
+
+void FakeWebRtcVideoDecoderFactory::DecoderDestroyed(
+    FakeWebRtcVideoDecoder* decoder) {
+  decoders_.erase(std::remove(decoders_.begin(), decoders_.end(), decoder),
+                  decoders_.end());
+}
+
+void FakeWebRtcVideoDecoderFactory::AddSupportedVideoCodecType(
+    const std::string& name) {
+  // This is to match the default H264 params of cricket::VideoCodec.
+  cricket::VideoCodec video_codec = cricket::CreateVideoCodec(name);
+  supported_codec_formats_.push_back(
+      webrtc::SdpVideoFormat(video_codec.name, video_codec.params));
+}
+
+int FakeWebRtcVideoDecoderFactory::GetNumCreatedDecoders() {
+  return num_created_decoders_;
+}
+
+const std::vector<FakeWebRtcVideoDecoder*>&
+FakeWebRtcVideoDecoderFactory::decoders() {
+  return decoders_;
+}
+
+// Encoder.
+FakeWebRtcVideoEncoder::FakeWebRtcVideoEncoder(
+    FakeWebRtcVideoEncoderFactory* factory)
+    : num_frames_encoded_(0), factory_(factory) {}
+
+FakeWebRtcVideoEncoder::~FakeWebRtcVideoEncoder() {
+  if (factory_) {
+    factory_->EncoderDestroyed(this);
+  }
+}
+
+void FakeWebRtcVideoEncoder::SetFecControllerOverride(
+    webrtc::FecControllerOverride* fec_controller_override) {
+  // Ignored.
+}
+
+int32_t FakeWebRtcVideoEncoder::InitEncode(
+    const webrtc::VideoCodec* codecSettings,
+    const VideoEncoder::Settings& settings) {
+  webrtc::MutexLock lock(&mutex_);
+  codec_settings_ = *codecSettings;
+  init_encode_event_.Set();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeWebRtcVideoEncoder::Encode(
+    const webrtc::VideoFrame& inputImage,
+    const std::vector<webrtc::VideoFrameType>* frame_types) {
+  webrtc::MutexLock lock(&mutex_);
+  ++num_frames_encoded_;
+  init_encode_event_.Set();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeWebRtcVideoEncoder::RegisterEncodeCompleteCallback(
+    webrtc::EncodedImageCallback* callback) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeWebRtcVideoEncoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void FakeWebRtcVideoEncoder::SetRates(const RateControlParameters& parameters) {
+}
+
+webrtc::VideoEncoder::EncoderInfo FakeWebRtcVideoEncoder::GetEncoderInfo()
+    const {
+  EncoderInfo info;
+  info.is_hardware_accelerated = true;
+  return info;
+}
+
+bool FakeWebRtcVideoEncoder::WaitForInitEncode() {
+  return init_encode_event_.Wait(kEventTimeout);
+}
+
+webrtc::VideoCodec FakeWebRtcVideoEncoder::GetCodecSettings() {
+  webrtc::MutexLock lock(&mutex_);
+  return codec_settings_;
+}
+
+int FakeWebRtcVideoEncoder::GetNumEncodedFrames() {
+  webrtc::MutexLock lock(&mutex_);
+  return num_frames_encoded_;
+}
+
+// Video encoder factory.
+FakeWebRtcVideoEncoderFactory::FakeWebRtcVideoEncoderFactory()
+    : num_created_encoders_(0), vp8_factory_mode_(false) {}
+
+std::vector<webrtc::SdpVideoFormat>
+FakeWebRtcVideoEncoderFactory::GetSupportedFormats() const {
+  std::vector<webrtc::SdpVideoFormat> formats;
+
+  for (const webrtc::SdpVideoFormat& format : formats_) {
+    // Don't add same codec twice.
+    if (!format.IsCodecInList(formats))
+      formats.push_back(format);
+  }
+
+  return formats;
+}
+
+webrtc::VideoEncoderFactory::CodecSupport
+FakeWebRtcVideoEncoderFactory::QueryCodecSupport(
+    const webrtc::SdpVideoFormat& format,
+    absl::optional<std::string> scalability_mode) const {
+  std::vector<webrtc::SdpVideoFormat> supported_formats;
+  for (const auto& f : formats_) {
+    if (format.IsSameCodec(f))
+      supported_formats.push_back(f);
+  }
+  if (format.IsCodecInList(formats_)) {
+    return {.is_supported = IsScalabilityModeSupported(supported_formats,
+                                                       scalability_mode)};
+  }
+  return {.is_supported = false};
+}
+
+std::unique_ptr<webrtc::VideoEncoder>
+FakeWebRtcVideoEncoderFactory::CreateVideoEncoder(
+    const webrtc::SdpVideoFormat& format) {
+  webrtc::MutexLock lock(&mutex_);
+  std::unique_ptr<webrtc::VideoEncoder> encoder;
+  if (format.IsCodecInList(formats_)) {
+    if (absl::EqualsIgnoreCase(format.name, kVp8CodecName) &&
+        !vp8_factory_mode_) {
+      // The simulcast adapter will ask this factory for multiple VP8
+      // encoders. Enter vp8_factory_mode so that we now create these encoders
+      // instead of more adapters.
+      vp8_factory_mode_ = true;
+      encoder = std::make_unique<webrtc::SimulcastEncoderAdapter>(this, format);
+    } else {
+      num_created_encoders_++;
+      created_video_encoder_event_.Set();
+      encoder = std::make_unique<FakeWebRtcVideoEncoder>(this);
+      encoders_.push_back(static_cast<FakeWebRtcVideoEncoder*>(encoder.get()));
+    }
+  }
+  return encoder;
+}
+
+bool FakeWebRtcVideoEncoderFactory::WaitForCreatedVideoEncoders(
+    int num_encoders) {
+  int64_t start_offset_ms = rtc::TimeMillis();
+  int64_t wait_time = kEventTimeout.ms();
+  do {
+    if (GetNumCreatedEncoders() >= num_encoders)
+      return true;
+    wait_time = kEventTimeout.ms() - (rtc::TimeMillis() - start_offset_ms);
+  } while (wait_time > 0 && created_video_encoder_event_.Wait(
+                                webrtc::TimeDelta::Millis(wait_time)));
+  return false;
+}
+
+void FakeWebRtcVideoEncoderFactory::EncoderDestroyed(
+    FakeWebRtcVideoEncoder* encoder) {
+  webrtc::MutexLock lock(&mutex_);
+  encoders_.erase(std::remove(encoders_.begin(), encoders_.end(), encoder),
+                  encoders_.end());
+}
+
+void FakeWebRtcVideoEncoderFactory::AddSupportedVideoCodec(
+    const webrtc::SdpVideoFormat& format) {
+  formats_.push_back(format);
+}
+
+void FakeWebRtcVideoEncoderFactory::AddSupportedVideoCodecType(
+    const std::string& name,
+    const std::vector<webrtc::ScalabilityMode>& scalability_modes) {
+  // This is to match the default H264 params of cricket::VideoCodec.
+  cricket::VideoCodec video_codec = cricket::CreateVideoCodec(name);
+  formats_.push_back(webrtc::SdpVideoFormat(
+      video_codec.name, video_codec.params,
+      {scalability_modes.begin(), scalability_modes.end()}));
+}
+
+int FakeWebRtcVideoEncoderFactory::GetNumCreatedEncoders() {
+  webrtc::MutexLock lock(&mutex_);
+  return num_created_encoders_;
+}
+
+const std::vector<FakeWebRtcVideoEncoder*>
+FakeWebRtcVideoEncoderFactory::encoders() {
+  webrtc::MutexLock lock(&mutex_);
+  return encoders_;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/engine/fake_webrtc_video_engine.h

@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2010 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 MEDIA_ENGINE_FAKE_WEBRTC_VIDEO_ENGINE_H_
+#define MEDIA_ENGINE_FAKE_WEBRTC_VIDEO_ENGINE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/fec_controller_override.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace cricket {
+
+class FakeWebRtcVideoDecoderFactory;
+class FakeWebRtcVideoEncoderFactory;
+
+// Fake class for mocking out webrtc::VideoDecoder
+class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder {
+ public:
+  explicit FakeWebRtcVideoDecoder(FakeWebRtcVideoDecoderFactory* factory);
+  ~FakeWebRtcVideoDecoder();
+
+  bool Configure(const Settings& settings) override;
+  int32_t Decode(const webrtc::EncodedImage&, int64_t) override;
+  int32_t RegisterDecodeCompleteCallback(
+      webrtc::DecodedImageCallback*) override;
+  int32_t Release() override;
+
+  int GetNumFramesReceived() const;
+
+ private:
+  int num_frames_received_;
+  FakeWebRtcVideoDecoderFactory* factory_;
+};
+
+// Fake class for mocking out webrtc::VideoDecoderFactory.
+class FakeWebRtcVideoDecoderFactory : public webrtc::VideoDecoderFactory {
+ public:
+  FakeWebRtcVideoDecoderFactory();
+
+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
+      const webrtc::SdpVideoFormat& format) override;
+
+  void DecoderDestroyed(FakeWebRtcVideoDecoder* decoder);
+  void AddSupportedVideoCodecType(const std::string& name);
+  int GetNumCreatedDecoders();
+  const std::vector<FakeWebRtcVideoDecoder*>& decoders();
+
+ private:
+  std::vector<webrtc::SdpVideoFormat> supported_codec_formats_;
+  std::vector<FakeWebRtcVideoDecoder*> decoders_;
+  int num_created_decoders_;
+};
+
+// Fake class for mocking out webrtc::VideoEnoder
+class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
+ public:
+  explicit FakeWebRtcVideoEncoder(FakeWebRtcVideoEncoderFactory* factory);
+  ~FakeWebRtcVideoEncoder();
+
+  void SetFecControllerOverride(
+      webrtc::FecControllerOverride* fec_controller_override) override;
+  int32_t InitEncode(const webrtc::VideoCodec* codecSettings,
+                     const VideoEncoder::Settings& settings) override;
+  int32_t Encode(
+      const webrtc::VideoFrame& inputImage,
+      const std::vector<webrtc::VideoFrameType>* frame_types) override;
+  int32_t RegisterEncodeCompleteCallback(
+      webrtc::EncodedImageCallback* callback) override;
+  int32_t Release() override;
+  void SetRates(const RateControlParameters& parameters) override;
+  webrtc::VideoEncoder::EncoderInfo GetEncoderInfo() const override;
+
+  bool WaitForInitEncode();
+  webrtc::VideoCodec GetCodecSettings();
+  int GetNumEncodedFrames();
+
+ private:
+  webrtc::Mutex mutex_;
+  rtc::Event init_encode_event_;
+  int num_frames_encoded_ RTC_GUARDED_BY(mutex_);
+  webrtc::VideoCodec codec_settings_ RTC_GUARDED_BY(mutex_);
+  FakeWebRtcVideoEncoderFactory* factory_;
+};
+
+// Fake class for mocking out webrtc::VideoEncoderFactory.
+class FakeWebRtcVideoEncoderFactory : public webrtc::VideoEncoderFactory {
+ public:
+  FakeWebRtcVideoEncoderFactory();
+
+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
+  webrtc::VideoEncoderFactory::CodecSupport QueryCodecSupport(
+      const webrtc::SdpVideoFormat& format,
+      absl::optional<std::string> scalability_mode) const override;
+  std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(
+      const webrtc::SdpVideoFormat& format) override;
+
+  bool WaitForCreatedVideoEncoders(int num_encoders);
+  void EncoderDestroyed(FakeWebRtcVideoEncoder* encoder);
+  void set_encoders_have_internal_sources(bool internal_source);
+  void AddSupportedVideoCodec(const webrtc::SdpVideoFormat& format);
+  void AddSupportedVideoCodecType(
+      const std::string& name,
+      const std::vector<webrtc::ScalabilityMode>& scalability_modes = {});
+  int GetNumCreatedEncoders();
+  const std::vector<FakeWebRtcVideoEncoder*> encoders();
+
+ private:
+  webrtc::Mutex mutex_;
+  rtc::Event created_video_encoder_event_;
+  std::vector<webrtc::SdpVideoFormat> formats_;
+  std::vector<FakeWebRtcVideoEncoder*> encoders_ RTC_GUARDED_BY(mutex_);
+  int num_created_encoders_ RTC_GUARDED_BY(mutex_);
+  bool vp8_factory_mode_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_FAKE_WEBRTC_VIDEO_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/engine/internal_decoder_factory.cc

@@ -0,0 +1,105 @@
+/*
+ *  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 "media/engine/internal_decoder_factory.h"
+
+#include "absl/strings/match.h"
+#include "api/video_codecs/av1_profile.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
+#include "media/base/codec.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/h264/include/h264.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+#if defined(RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY)
+#include "modules/video_coding/codecs/av1/dav1d_decoder.h"  // nogncheck
+#endif
+
+namespace webrtc {
+namespace {
+#if defined(RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY)
+constexpr bool kDav1dIsIncluded = true;
+#else
+constexpr bool kDav1dIsIncluded = false;
+std::unique_ptr<VideoDecoder> CreateDav1dDecoder() {
+  return nullptr;
+}
+#endif
+
+}  // namespace
+
+std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
+    const {
+  std::vector<SdpVideoFormat> formats;
+  formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
+  for (const SdpVideoFormat& format : SupportedVP9DecoderCodecs())
+    formats.push_back(format);
+  for (const SdpVideoFormat& h264_format : SupportedH264DecoderCodecs())
+    formats.push_back(h264_format);
+
+  if (kDav1dIsIncluded) {
+    formats.push_back(SdpVideoFormat(cricket::kAv1CodecName));
+    formats.push_back(
+        SdpVideoFormat(cricket::kAv1CodecName,
+                       {{cricket::kAv1FmtpProfile,
+                         AV1ProfileToString(AV1Profile::kProfile1).data()}}));
+  }
+
+  return formats;
+}
+
+VideoDecoderFactory::CodecSupport InternalDecoderFactory::QueryCodecSupport(
+    const SdpVideoFormat& format,
+    bool reference_scaling) const {
+  // Query for supported formats and check if the specified format is supported.
+  // Return unsupported if an invalid combination of format and
+  // reference_scaling is specified.
+  if (reference_scaling) {
+    VideoCodecType codec = PayloadStringToCodecType(format.name);
+    if (codec != kVideoCodecVP9 && codec != kVideoCodecAV1) {
+      return {/*is_supported=*/false, /*is_power_efficient=*/false};
+    }
+  }
+
+  CodecSupport codec_support;
+  codec_support.is_supported = format.IsCodecInList(GetSupportedFormats());
+  return codec_support;
+}
+
+std::unique_ptr<VideoDecoder> InternalDecoderFactory::CreateVideoDecoder(
+    const SdpVideoFormat& format) {
+  if (!format.IsCodecInList(GetSupportedFormats())) {
+    RTC_LOG(LS_WARNING) << "Trying to create decoder for unsupported format. "
+                        << format.ToString();
+    return nullptr;
+  }
+
+  if (absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName))
+    return VP8Decoder::Create();
+  if (absl::EqualsIgnoreCase(format.name, cricket::kVp9CodecName))
+    return VP9Decoder::Create();
+  if (absl::EqualsIgnoreCase(format.name, cricket::kH264CodecName))
+    return H264Decoder::Create();
+
+  if (absl::EqualsIgnoreCase(format.name, cricket::kAv1CodecName) &&
+      kDav1dIsIncluded) {
+    return CreateDav1dDecoder();
+  }
+
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/engine/internal_decoder_factory.h

@@ -0,0 +1,35 @@
+/*
+ *  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 MEDIA_ENGINE_INTERNAL_DECODER_FACTORY_H_
+#define MEDIA_ENGINE_INTERNAL_DECODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT InternalDecoderFactory : public VideoDecoderFactory {
+ public:
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  CodecSupport QueryCodecSupport(const SdpVideoFormat& format,
+                                 bool reference_scaling) const override;
+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      const SdpVideoFormat& format) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MEDIA_ENGINE_INTERNAL_DECODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/media/engine/internal_encoder_factory.cc

@@ -0,0 +1,66 @@
+/*
+ *  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 "media/engine/internal_encoder_factory.h"
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "api/video_codecs/video_encoder_factory_template.h"
+#if defined(RTC_USE_LIBAOM_AV1_ENCODER)
+#include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h"  // nogncheck
+#endif
+#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
+#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
+#if defined(WEBRTC_USE_H264)
+#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"  // nogncheck
+#endif
+
+namespace webrtc {
+namespace {
+
+using Factory =
+    VideoEncoderFactoryTemplate<webrtc::LibvpxVp8EncoderTemplateAdapter,
+#if defined(WEBRTC_USE_H264)
+                                webrtc::OpenH264EncoderTemplateAdapter,
+#endif
+#if defined(RTC_USE_LIBAOM_AV1_ENCODER)
+                                webrtc::LibaomAv1EncoderTemplateAdapter,
+#endif
+                                webrtc::LibvpxVp9EncoderTemplateAdapter>;
+}  // namespace
+
+std::vector<SdpVideoFormat> InternalEncoderFactory::GetSupportedFormats()
+    const {
+  return Factory().GetSupportedFormats();
+}
+
+std::unique_ptr<VideoEncoder> InternalEncoderFactory::CreateVideoEncoder(
+    const SdpVideoFormat& format) {
+  auto original_format =
+      FuzzyMatchSdpVideoFormat(Factory().GetSupportedFormats(), format);
+  return original_format ? Factory().CreateVideoEncoder(*original_format)
+                         : nullptr;
+}
+
+VideoEncoderFactory::CodecSupport InternalEncoderFactory::QueryCodecSupport(
+    const SdpVideoFormat& format,
+    absl::optional<std::string> scalability_mode) const {
+  auto original_format =
+      FuzzyMatchSdpVideoFormat(Factory().GetSupportedFormats(), format);
+  return original_format
+             ? Factory().QueryCodecSupport(*original_format, scalability_mode)
+             : VideoEncoderFactory::CodecSupport{.is_supported = false};
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/engine/internal_encoder_factory.h

@@ -0,0 +1,34 @@
+/*
+ *  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 MEDIA_ENGINE_INTERNAL_ENCODER_FACTORY_H_
+#define MEDIA_ENGINE_INTERNAL_ENCODER_FACTORY_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/video_codecs/video_encoder_factory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+class RTC_EXPORT InternalEncoderFactory : public VideoEncoderFactory {
+ public:
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  CodecSupport QueryCodecSupport(
+      const SdpVideoFormat& format,
+      absl::optional<std::string> scalability_mode) const override;
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MEDIA_ENGINE_INTERNAL_ENCODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/media/engine/multiplex_codec_factory.cc

@@ -0,0 +1,117 @@
+/*
+ *  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 "media/engine/multiplex_codec_factory.h"
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/environment/environment.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "media/base/codec.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
+#include "rtc_base/logging.h"
+
+namespace {
+
+bool IsMultiplexCodec(const cricket::VideoCodec& codec) {
+  return absl::EqualsIgnoreCase(codec.name.c_str(),
+                                cricket::kMultiplexCodecName);
+}
+
+}  // anonymous namespace
+
+namespace webrtc {
+
+constexpr const char* kMultiplexAssociatedCodecName = cricket::kVp9CodecName;
+
+MultiplexEncoderFactory::MultiplexEncoderFactory(
+    std::unique_ptr<VideoEncoderFactory> factory,
+    bool supports_augmenting_data)
+    : factory_(std::move(factory)),
+      supports_augmenting_data_(supports_augmenting_data) {}
+
+std::vector<SdpVideoFormat> MultiplexEncoderFactory::GetSupportedFormats()
+    const {
+  std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
+  for (const auto& format : formats) {
+    if (absl::EqualsIgnoreCase(format.name, kMultiplexAssociatedCodecName)) {
+      SdpVideoFormat multiplex_format = format;
+      multiplex_format.parameters[cricket::kCodecParamAssociatedCodecName] =
+          format.name;
+      multiplex_format.name = cricket::kMultiplexCodecName;
+      formats.push_back(multiplex_format);
+      break;
+    }
+  }
+  return formats;
+}
+
+std::unique_ptr<VideoEncoder> MultiplexEncoderFactory::CreateVideoEncoder(
+    const SdpVideoFormat& format) {
+  if (!IsMultiplexCodec(cricket::CreateVideoCodec(format)))
+    return factory_->CreateVideoEncoder(format);
+  const auto& it =
+      format.parameters.find(cricket::kCodecParamAssociatedCodecName);
+  if (it == format.parameters.end()) {
+    RTC_LOG(LS_ERROR) << "No assicated codec for multiplex.";
+    return nullptr;
+  }
+  SdpVideoFormat associated_format = format;
+  associated_format.name = it->second;
+  return std::unique_ptr<VideoEncoder>(new MultiplexEncoderAdapter(
+      factory_.get(), associated_format, supports_augmenting_data_));
+}
+
+MultiplexDecoderFactory::MultiplexDecoderFactory(
+    std::unique_ptr<VideoDecoderFactory> factory,
+    bool supports_augmenting_data)
+    : factory_(std::move(factory)),
+      supports_augmenting_data_(supports_augmenting_data) {}
+
+std::vector<SdpVideoFormat> MultiplexDecoderFactory::GetSupportedFormats()
+    const {
+  std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
+  std::vector<SdpVideoFormat> augmented_formats = formats;
+  for (const auto& format : formats) {
+    if (absl::EqualsIgnoreCase(format.name, kMultiplexAssociatedCodecName)) {
+      SdpVideoFormat multiplex_format = format;
+      multiplex_format.parameters[cricket::kCodecParamAssociatedCodecName] =
+          format.name;
+      multiplex_format.name = cricket::kMultiplexCodecName;
+      augmented_formats.push_back(multiplex_format);
+    }
+  }
+  return augmented_formats;
+}
+
+std::unique_ptr<VideoDecoder> MultiplexDecoderFactory::Create(
+    const Environment& env,
+    const SdpVideoFormat& format) {
+  if (!IsMultiplexCodec(cricket::CreateVideoCodec(format))) {
+    return factory_->Create(env, format);
+  }
+  auto it = format.parameters.find(cricket::kCodecParamAssociatedCodecName);
+  if (it == format.parameters.end()) {
+    RTC_LOG(LS_ERROR) << "No assicated codec for multiplex.";
+    return nullptr;
+  }
+  SdpVideoFormat associated_format = format;
+  associated_format.name = it->second;
+  return std::make_unique<MultiplexDecoderAdapter>(
+      env, factory_.get(), associated_format, supports_augmenting_data_);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/engine/multiplex_codec_factory.h

@@ -0,0 +1,80 @@
+/*
+ *  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 MEDIA_ENGINE_MULTIPLEX_CODEC_FACTORY_H_
+#define MEDIA_ENGINE_MULTIPLEX_CODEC_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/environment/environment.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+// Multiplex codec is a completely modular/optional codec that allows users to
+// send more than a frame's opaque content(RGB/YUV) over video channels.
+// - Allows sending Alpha channel over the wire iff input is
+// I420ABufferInterface. Users can expect to receive I420ABufferInterface as the
+// decoded video frame buffer. I420A data is split into YUV/AXX portions,
+// encoded/decoded seperately and bitstreams are concatanated.
+// - Allows sending augmenting data over the wire attached to the frame. This
+// attached data portion is not encoded in any way and sent as it is. Users can
+// input AugmentedVideoFrameBuffer and can expect the same interface as the
+// decoded video frame buffer.
+// - Showcases an example of how to add a custom codec in webrtc video channel.
+// How to use it end-to-end:
+// - Wrap your existing VideoEncoderFactory implemention with
+// MultiplexEncoderFactory and VideoDecoderFactory implemention with
+// MultiplexDecoderFactory below. For actual coding, multiplex creates encoder
+// and decoder instance(s) using these factories.
+// - Use Multiplex*coderFactory classes in CreatePeerConnectionFactory() calls.
+// - Select "multiplex" codec in SDP negotiation.
+class RTC_EXPORT MultiplexEncoderFactory : public VideoEncoderFactory {
+ public:
+  // `supports_augmenting_data` defines if the encoder would support augmenting
+  // data. If set, the encoder expects to receive video frame buffers of type
+  // AugmentedVideoFrameBuffer.
+  MultiplexEncoderFactory(std::unique_ptr<VideoEncoderFactory> factory,
+                          bool supports_augmenting_data = false);
+
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override;
+
+ private:
+  std::unique_ptr<VideoEncoderFactory> factory_;
+  const bool supports_augmenting_data_;
+};
+
+class RTC_EXPORT MultiplexDecoderFactory : public VideoDecoderFactory {
+ public:
+  // `supports_augmenting_data` defines if the decoder would support augmenting
+  // data. If set, the decoder is expected to output video frame buffers of type
+  // AugmentedVideoFrameBuffer.
+  MultiplexDecoderFactory(std::unique_ptr<VideoDecoderFactory> factory,
+                          bool supports_augmenting_data = false);
+
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<VideoDecoder> Create(const Environment& env,
+                                       const SdpVideoFormat& format) override;
+
+ private:
+  std::unique_ptr<VideoDecoderFactory> factory_;
+  const bool supports_augmenting_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MEDIA_ENGINE_MULTIPLEX_CODEC_FACTORY_H_

TMessagesProj/jni/voip/webrtc/media/engine/null_webrtc_video_engine.h

@@ -0,0 +1,54 @@
+/*
+ *  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 MEDIA_ENGINE_NULL_WEBRTC_VIDEO_ENGINE_H_
+#define MEDIA_ENGINE_NULL_WEBRTC_VIDEO_ENGINE_H_
+
+#include <vector>
+
+#include "media/base/media_channel.h"
+#include "media/base/media_engine.h"
+
+namespace webrtc {
+
+class Call;
+
+}  // namespace webrtc
+
+namespace cricket {
+
+// Video engine implementation that does nothing and can be used in
+// CompositeMediaEngine.
+class NullWebRtcVideoEngine : public VideoEngineInterface {
+ public:
+  std::vector<VideoCodec> send_codecs(bool) const override {
+    return std::vector<VideoCodec>();
+  }
+
+  std::vector<VideoCodec> recv_codecs(bool) const override {
+    return std::vector<VideoCodec>();
+  }
+  std::vector<VideoCodec> send_codecs() const override {
+    return std::vector<VideoCodec>();
+  }
+
+  std::vector<VideoCodec> recv_codecs() const override {
+    return std::vector<VideoCodec>();
+  }
+
+  std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
+      const override {
+    return {};
+  }
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_NULL_WEBRTC_VIDEO_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/engine/payload_type_mapper.cc

@@ -0,0 +1,160 @@
+/*
+ *  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 "media/engine/payload_type_mapper.h"
+
+#include <utility>
+
+#include "absl/strings/ascii.h"
+#include "api/audio_codecs/audio_format.h"
+#include "media/base/codec.h"
+#include "media/base/media_constants.h"
+
+namespace cricket {
+
+webrtc::SdpAudioFormat AudioCodecToSdpAudioFormat(const AudioCodec& ac) {
+  return webrtc::SdpAudioFormat(ac.name, ac.clockrate, ac.channels, ac.params);
+}
+
+PayloadTypeMapper::PayloadTypeMapper()
+    // RFC 3551 reserves payload type numbers in the range 96-127 exclusively
+    // for dynamic assignment. Once those are used up, it is recommended that
+    // payload types unassigned by the RFC are used for dynamic payload type
+    // mapping, before any static payload ids. At this point, we only support
+    // mapping within the exclusive range.
+    : next_unused_payload_type_(96),
+      max_payload_type_(127),
+      mappings_(
+          {// Static payload type assignments according to RFC 3551.
+           {{kPcmuCodecName, 8000, 1}, 0},
+           {{"GSM", 8000, 1}, 3},
+           {{"G723", 8000, 1}, 4},
+           {{"DVI4", 8000, 1}, 5},
+           {{"DVI4", 16000, 1}, 6},
+           {{"LPC", 8000, 1}, 7},
+           {{kPcmaCodecName, 8000, 1}, 8},
+           {{kG722CodecName, 8000, 1}, 9},
+           {{kL16CodecName, 44100, 2}, 10},
+           {{kL16CodecName, 44100, 1}, 11},
+           {{"QCELP", 8000, 1}, 12},
+           {{kCnCodecName, 8000, 1}, 13},
+           // RFC 4566 is a bit ambiguous on the contents of the "encoding
+           // parameters" field, which, for audio, encodes the number of
+           // channels. It is "optional and may be omitted if the number of
+           // channels is one". Does that necessarily imply that an omitted
+           // encoding parameter means one channel?  Since RFC 3551 doesn't
+           // specify a value for this parameter for MPA, I've included both 0
+           // and 1 here, to increase the chances it will be correctly used if
+           // someone implements an MPEG audio encoder/decoder.
+           {{"MPA", 90000, 0}, 14},
+           {{"MPA", 90000, 1}, 14},
+           {{"G728", 8000, 1}, 15},
+           {{"DVI4", 11025, 1}, 16},
+           {{"DVI4", 22050, 1}, 17},
+           {{"G729", 8000, 1}, 18},
+
+           // Payload type assignments currently used by WebRTC.
+           // Includes data to reduce collisions (and thus reassignments)
+           {{kIlbcCodecName, 8000, 1}, 102},
+           {{kCnCodecName, 16000, 1}, 105},
+           {{kCnCodecName, 32000, 1}, 106},
+           {{kOpusCodecName,
+             48000,
+             2,
+             {{kCodecParamMinPTime, "10"},
+              {kCodecParamUseInbandFec, kParamValueTrue}}},
+            111},
+           // RED for opus is assigned in the lower range, starting at the top.
+           // Note that the FMTP refers to the opus payload type.
+           {{kRedCodecName,
+             48000,
+             2,
+             {{kCodecParamNotInNameValueFormat, "111/111"}}},
+            63},
+           // TODO(solenberg): Remove the hard coded 16k,32k,48k DTMF once we
+           // assign payload types dynamically for send side as well.
+           {{kDtmfCodecName, 48000, 1}, 110},
+           {{kDtmfCodecName, 32000, 1}, 112},
+           {{kDtmfCodecName, 16000, 1}, 113},
+           {{kDtmfCodecName, 8000, 1}, 126}}) {
+  // TODO(ossu): Try to keep this as change-proof as possible until we're able
+  // to remove the payload type constants from everywhere in the code.
+  for (const auto& mapping : mappings_) {
+    used_payload_types_.insert(mapping.second);
+  }
+}
+
+PayloadTypeMapper::~PayloadTypeMapper() = default;
+
+absl::optional<int> PayloadTypeMapper::GetMappingFor(
+    const webrtc::SdpAudioFormat& format) {
+  auto iter = mappings_.find(format);
+  if (iter != mappings_.end())
+    return iter->second;
+
+  for (; next_unused_payload_type_ <= max_payload_type_;
+       ++next_unused_payload_type_) {
+    int payload_type = next_unused_payload_type_;
+    if (used_payload_types_.find(payload_type) == used_payload_types_.end()) {
+      used_payload_types_.insert(payload_type);
+      mappings_[format] = payload_type;
+      ++next_unused_payload_type_;
+      return payload_type;
+    }
+  }
+
+  return absl::nullopt;
+}
+
+absl::optional<int> PayloadTypeMapper::FindMappingFor(
+    const webrtc::SdpAudioFormat& format) const {
+  auto iter = mappings_.find(format);
+  if (iter != mappings_.end())
+    return iter->second;
+
+  return absl::nullopt;
+}
+
+absl::optional<AudioCodec> PayloadTypeMapper::ToAudioCodec(
+    const webrtc::SdpAudioFormat& format) {
+  // TODO(ossu): We can safely set bitrate to zero here, since that field is
+  // not presented in the SDP. It is used to ferry around some target bitrate
+  // values for certain codecs (ISAC and Opus) and in ways it really
+  // shouldn't. It should be removed once we no longer use CodecInsts in the
+  // ACM or NetEq.
+  auto opt_payload_type = GetMappingFor(format);
+  if (opt_payload_type) {
+    AudioCodec codec =
+        cricket::CreateAudioCodec(*opt_payload_type, format.name,
+                                  format.clockrate_hz, format.num_channels);
+    codec.params = format.parameters;
+    return std::move(codec);
+  }
+
+  return absl::nullopt;
+}
+
+bool PayloadTypeMapper::SdpAudioFormatOrdering::operator()(
+    const webrtc::SdpAudioFormat& a,
+    const webrtc::SdpAudioFormat& b) const {
+  if (a.clockrate_hz == b.clockrate_hz) {
+    if (a.num_channels == b.num_channels) {
+      int name_cmp =
+          absl::AsciiStrToLower(a.name).compare(absl::AsciiStrToLower(b.name));
+      if (name_cmp == 0)
+        return a.parameters < b.parameters;
+      return name_cmp < 0;
+    }
+    return a.num_channels < b.num_channels;
+  }
+  return a.clockrate_hz < b.clockrate_hz;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/engine/payload_type_mapper.h

@@ -0,0 +1,57 @@
+/*
+ *  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 MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_
+#define MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_
+
+#include <map>
+#include <set>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_format.h"
+#include "media/base/codec.h"
+
+namespace cricket {
+
+webrtc::SdpAudioFormat AudioCodecToSdpAudioFormat(const AudioCodec& ac);
+
+class PayloadTypeMapper {
+ public:
+  PayloadTypeMapper();
+  ~PayloadTypeMapper();
+
+  // Finds the current payload type for `format` or assigns a new one, if no
+  // current mapping exists. Will return an empty value if it was unable to
+  // create a mapping, i.e. if all dynamic payload type ids have been used up.
+  absl::optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format);
+
+  // Finds the current payload type for `format`, if any. Returns an empty value
+  // if no payload type mapping exists for the format.
+  absl::optional<int> FindMappingFor(
+      const webrtc::SdpAudioFormat& format) const;
+
+  // Like GetMappingFor, but fills in an AudioCodec structure with the necessary
+  // information instead.
+  absl::optional<AudioCodec> ToAudioCodec(const webrtc::SdpAudioFormat& format);
+
+ private:
+  struct SdpAudioFormatOrdering {
+    bool operator()(const webrtc::SdpAudioFormat& a,
+                    const webrtc::SdpAudioFormat& b) const;
+  };
+
+  int next_unused_payload_type_;
+  int max_payload_type_;
+  std::map<webrtc::SdpAudioFormat, int, SdpAudioFormatOrdering> mappings_;
+  std::set<int> used_payload_types_;
+};
+
+}  // namespace cricket
+#endif  // MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_

TMessagesProj/jni/voip/webrtc/media/engine/simulcast_encoder_adapter.cc

@@ -0,0 +1,995 @@
+/*
+ *  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 "media/engine/simulcast_encoder_adapter.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "api/field_trials_view.h"
+#include "api/scoped_refptr.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
+#include "media/base/media_constants.h"
+#include "media/base/video_common.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/include/video_error_codes_utils.h"
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "rtc_base/logging.h"
+
+namespace {
+
+// Max qp for lowest spatial resolution when doing simulcast.
+const unsigned int kLowestResMaxQp = 45;
+
+absl::optional<unsigned int> GetScreenshareBoostedQpValue(
+    const webrtc::FieldTrialsView& field_trials) {
+  std::string experiment_group =
+      field_trials.Lookup("WebRTC-BoostedScreenshareQp");
+  unsigned int qp;
+  if (sscanf(experiment_group.c_str(), "%u", &qp) != 1)
+    return absl::nullopt;
+  qp = std::min(qp, 63u);
+  qp = std::max(qp, 1u);
+  return qp;
+}
+
+uint32_t SumStreamMaxBitrate(int streams, const webrtc::VideoCodec& codec) {
+  uint32_t bitrate_sum = 0;
+  for (int i = 0; i < streams; ++i) {
+    bitrate_sum += codec.simulcastStream[i].maxBitrate;
+  }
+  return bitrate_sum;
+}
+
+int CountAllStreams(const webrtc::VideoCodec& codec) {
+  int total_streams_count =
+      codec.numberOfSimulcastStreams < 1 ? 1 : codec.numberOfSimulcastStreams;
+  uint32_t simulcast_max_bitrate =
+      SumStreamMaxBitrate(total_streams_count, codec);
+  if (simulcast_max_bitrate == 0) {
+    total_streams_count = 1;
+  }
+  return total_streams_count;
+}
+
+int CountActiveStreams(const webrtc::VideoCodec& codec) {
+  if (codec.numberOfSimulcastStreams < 1) {
+    return 1;
+  }
+  int total_streams_count = CountAllStreams(codec);
+  int active_streams_count = 0;
+  for (int i = 0; i < total_streams_count; ++i) {
+    if (codec.simulcastStream[i].active) {
+      ++active_streams_count;
+    }
+  }
+  return active_streams_count;
+}
+
+int VerifyCodec(const webrtc::VideoCodec* codec_settings) {
+  if (codec_settings == nullptr) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings->maxFramerate < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  // allow zero to represent an unspecified maxBitRate
+  if (codec_settings->maxBitrate > 0 &&
+      codec_settings->startBitrate > codec_settings->maxBitrate) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings->width <= 1 || codec_settings->height <= 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings->codecType == webrtc::kVideoCodecVP8 &&
+      codec_settings->VP8().automaticResizeOn &&
+      CountActiveStreams(*codec_settings) > 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool StreamQualityCompare(const webrtc::SimulcastStream& a,
+                          const webrtc::SimulcastStream& b) {
+  return std::tie(a.height, a.width, a.maxBitrate, a.maxFramerate) <
+         std::tie(b.height, b.width, b.maxBitrate, b.maxFramerate);
+}
+
+void GetLowestAndHighestQualityStreamIndixes(
+    rtc::ArrayView<webrtc::SimulcastStream> streams,
+    int* lowest_quality_stream_idx,
+    int* highest_quality_stream_idx) {
+  const auto lowest_highest_quality_streams =
+      absl::c_minmax_element(streams, StreamQualityCompare);
+  *lowest_quality_stream_idx =
+      std::distance(streams.begin(), lowest_highest_quality_streams.first);
+  *highest_quality_stream_idx =
+      std::distance(streams.begin(), lowest_highest_quality_streams.second);
+}
+
+std::vector<uint32_t> GetStreamStartBitratesKbps(
+    const webrtc::VideoCodec& codec) {
+  std::vector<uint32_t> start_bitrates;
+  std::unique_ptr<webrtc::VideoBitrateAllocator> rate_allocator =
+      std::make_unique<webrtc::SimulcastRateAllocator>(codec);
+  webrtc::VideoBitrateAllocation allocation =
+      rate_allocator->Allocate(webrtc::VideoBitrateAllocationParameters(
+          codec.startBitrate * 1000, codec.maxFramerate));
+
+  int total_streams_count = CountAllStreams(codec);
+  for (int i = 0; i < total_streams_count; ++i) {
+    uint32_t stream_bitrate = allocation.GetSpatialLayerSum(i) / 1000;
+    start_bitrates.push_back(stream_bitrate);
+  }
+  return start_bitrates;
+}
+
+}  // namespace
+
+namespace webrtc {
+
+SimulcastEncoderAdapter::EncoderContext::EncoderContext(
+    std::unique_ptr<VideoEncoder> encoder,
+    bool prefer_temporal_support,
+    VideoEncoder::EncoderInfo primary_info,
+    VideoEncoder::EncoderInfo fallback_info)
+    : encoder_(std::move(encoder)),
+      prefer_temporal_support_(prefer_temporal_support),
+      primary_info_(std::move(primary_info)),
+      fallback_info_(std::move(fallback_info)) {}
+
+void SimulcastEncoderAdapter::EncoderContext::Release() {
+  if (encoder_) {
+    encoder_->Release();
+    encoder_->RegisterEncodeCompleteCallback(nullptr);
+  }
+}
+
+SimulcastEncoderAdapter::StreamContext::StreamContext(
+    SimulcastEncoderAdapter* parent,
+    std::unique_ptr<EncoderContext> encoder_context,
+    std::unique_ptr<FramerateController> framerate_controller,
+    int stream_idx,
+    uint16_t width,
+    uint16_t height,
+    bool is_paused)
+    : parent_(parent),
+      encoder_context_(std::move(encoder_context)),
+      framerate_controller_(std::move(framerate_controller)),
+      stream_idx_(stream_idx),
+      width_(width),
+      height_(height),
+      is_keyframe_needed_(false),
+      is_paused_(is_paused) {
+  if (parent_) {
+    encoder_context_->encoder().RegisterEncodeCompleteCallback(this);
+  }
+}
+
+SimulcastEncoderAdapter::StreamContext::StreamContext(StreamContext&& rhs)
+    : parent_(rhs.parent_),
+      encoder_context_(std::move(rhs.encoder_context_)),
+      framerate_controller_(std::move(rhs.framerate_controller_)),
+      stream_idx_(rhs.stream_idx_),
+      width_(rhs.width_),
+      height_(rhs.height_),
+      is_keyframe_needed_(rhs.is_keyframe_needed_),
+      is_paused_(rhs.is_paused_) {
+  if (parent_) {
+    encoder_context_->encoder().RegisterEncodeCompleteCallback(this);
+  }
+}
+
+SimulcastEncoderAdapter::StreamContext::~StreamContext() {
+  if (encoder_context_) {
+    encoder_context_->Release();
+  }
+}
+
+std::unique_ptr<SimulcastEncoderAdapter::EncoderContext>
+SimulcastEncoderAdapter::StreamContext::ReleaseEncoderContext() && {
+  encoder_context_->Release();
+  return std::move(encoder_context_);
+}
+
+void SimulcastEncoderAdapter::StreamContext::OnKeyframe(Timestamp timestamp) {
+  is_keyframe_needed_ = false;
+  if (framerate_controller_) {
+    framerate_controller_->KeepFrame(timestamp.us() * 1000);
+  }
+}
+
+bool SimulcastEncoderAdapter::StreamContext::ShouldDropFrame(
+    Timestamp timestamp) {
+  if (!framerate_controller_) {
+    return false;
+  }
+  return framerate_controller_->ShouldDropFrame(timestamp.us() * 1000);
+}
+
+EncodedImageCallback::Result
+SimulcastEncoderAdapter::StreamContext::OnEncodedImage(
+    const EncodedImage& encoded_image,
+    const CodecSpecificInfo* codec_specific_info) {
+  RTC_CHECK(parent_);  // If null, this method should never be called.
+  return parent_->OnEncodedImage(stream_idx_, encoded_image,
+                                 codec_specific_info);
+}
+
+void SimulcastEncoderAdapter::StreamContext::OnDroppedFrame(
+    DropReason /*reason*/) {
+  RTC_CHECK(parent_);  // If null, this method should never be called.
+  parent_->OnDroppedFrame(stream_idx_);
+}
+
+SimulcastEncoderAdapter::SimulcastEncoderAdapter(VideoEncoderFactory* factory,
+                                                 const SdpVideoFormat& format)
+    : SimulcastEncoderAdapter(factory,
+                              nullptr,
+                              format,
+                              FieldTrialBasedConfig()) {}
+
+SimulcastEncoderAdapter::SimulcastEncoderAdapter(
+    VideoEncoderFactory* primary_factory,
+    VideoEncoderFactory* fallback_factory,
+    const SdpVideoFormat& format,
+    const FieldTrialsView& field_trials)
+    : inited_(0),
+      primary_encoder_factory_(primary_factory),
+      fallback_encoder_factory_(fallback_factory),
+      video_format_(format),
+      total_streams_count_(0),
+      bypass_mode_(false),
+      encoded_complete_callback_(nullptr),
+      experimental_boosted_screenshare_qp_(
+          GetScreenshareBoostedQpValue(field_trials)),
+      boost_base_layer_quality_(
+          RateControlSettings::ParseFromKeyValueConfig(&field_trials)
+              .Vp8BoostBaseLayerQuality()),
+      prefer_temporal_support_on_base_layer_(field_trials.IsEnabled(
+          "WebRTC-Video-PreferTemporalSupportOnBaseLayer")) {
+  RTC_DCHECK(primary_factory);
+
+  // The adapter is typically created on the worker thread, but operated on
+  // the encoder task queue.
+  encoder_queue_.Detach();
+}
+
+SimulcastEncoderAdapter::~SimulcastEncoderAdapter() {
+  RTC_DCHECK(!Initialized());
+  DestroyStoredEncoders();
+}
+
+void SimulcastEncoderAdapter::SetFecControllerOverride(
+    FecControllerOverride* /*fec_controller_override*/) {
+  // Ignored.
+}
+
+int SimulcastEncoderAdapter::Release() {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+
+  while (!stream_contexts_.empty()) {
+    // Move the encoder instances and put it on the `cached_encoder_contexts_`
+    // where it may possibly be reused from (ordering does not matter).
+    cached_encoder_contexts_.push_front(
+        std::move(stream_contexts_.back()).ReleaseEncoderContext());
+    stream_contexts_.pop_back();
+  }
+
+  bypass_mode_ = false;
+
+  // It's legal to move the encoder to another queue now.
+  encoder_queue_.Detach();
+
+  inited_.store(0);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::InitEncode(
+    const VideoCodec* codec_settings,
+    const VideoEncoder::Settings& settings) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+
+  if (settings.number_of_cores < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  int ret = VerifyCodec(codec_settings);
+  if (ret < 0) {
+    return ret;
+  }
+
+  Release();
+
+  codec_ = *codec_settings;
+  total_streams_count_ = CountAllStreams(*codec_settings);
+
+  bool is_legacy_singlecast = codec_.numberOfSimulcastStreams == 0;
+  int lowest_quality_stream_idx = 0;
+  int highest_quality_stream_idx = 0;
+  if (!is_legacy_singlecast) {
+    GetLowestAndHighestQualityStreamIndixes(
+        rtc::ArrayView<SimulcastStream>(codec_.simulcastStream,
+                                        total_streams_count_),
+        &lowest_quality_stream_idx, &highest_quality_stream_idx);
+  }
+
+  std::unique_ptr<EncoderContext> encoder_context = FetchOrCreateEncoderContext(
+      /*is_lowest_quality_stream=*/(
+          is_legacy_singlecast ||
+          codec_.simulcastStream[lowest_quality_stream_idx].active));
+  if (encoder_context == nullptr) {
+    return WEBRTC_VIDEO_CODEC_MEMORY;
+  }
+
+  // Two distinct scenarios:
+  // * Singlecast (total_streams_count == 1) or simulcast with simulcast-capable
+  //   underlaying encoder implementation if active_streams_count > 1. SEA
+  //   operates in bypass mode: original settings are passed to the underlaying
+  //   encoder, frame encode complete callback is not intercepted.
+  // * Multi-encoder simulcast or singlecast if layers are deactivated
+  //   (active_streams_count >= 1). SEA creates N=active_streams_count encoders
+  //   and configures each to produce a single stream.
+
+  int active_streams_count = CountActiveStreams(*codec_settings);
+  // If we only have a single active layer it is better to create an encoder
+  // with only one configured layer than creating it with all-but-one disabled
+  // layers because that way we control scaling.
+  bool separate_encoders_needed =
+      !encoder_context->encoder().GetEncoderInfo().supports_simulcast ||
+      active_streams_count == 1;
+  RTC_LOG(LS_INFO) << "[SEA] InitEncode: total_streams_count: "
+                   << total_streams_count_
+                   << ", active_streams_count: " << active_streams_count
+                   << ", separate_encoders_needed: "
+                   << (separate_encoders_needed ? "true" : "false");
+  // Singlecast or simulcast with simulcast-capable underlaying encoder.
+  if (total_streams_count_ == 1 || !separate_encoders_needed) {
+    RTC_LOG(LS_INFO) << "[SEA] InitEncode: Single-encoder mode";
+    int ret = encoder_context->encoder().InitEncode(&codec_, settings);
+    if (ret >= 0) {
+      stream_contexts_.emplace_back(
+          /*parent=*/nullptr, std::move(encoder_context),
+          /*framerate_controller=*/nullptr, /*stream_idx=*/0, codec_.width,
+          codec_.height, /*is_paused=*/active_streams_count == 0);
+      bypass_mode_ = true;
+
+      DestroyStoredEncoders();
+      inited_.store(1);
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    encoder_context->Release();
+    if (total_streams_count_ == 1) {
+      RTC_LOG(LS_ERROR) << "[SEA] InitEncode: failed with error code: "
+                        << WebRtcVideoCodecErrorToString(ret);
+      return ret;
+    }
+  }
+
+  // Multi-encoder simulcast or singlecast (deactivated layers).
+  std::vector<uint32_t> stream_start_bitrate_kbps =
+      GetStreamStartBitratesKbps(codec_);
+
+  for (int stream_idx = 0; stream_idx < total_streams_count_; ++stream_idx) {
+    if (!is_legacy_singlecast && !codec_.simulcastStream[stream_idx].active) {
+      continue;
+    }
+
+    if (encoder_context == nullptr) {
+      encoder_context = FetchOrCreateEncoderContext(
+          /*is_lowest_quality_stream=*/stream_idx == lowest_quality_stream_idx);
+    }
+    if (encoder_context == nullptr) {
+      Release();
+      return WEBRTC_VIDEO_CODEC_MEMORY;
+    }
+
+    VideoCodec stream_codec = MakeStreamCodec(
+        codec_, stream_idx, stream_start_bitrate_kbps[stream_idx],
+        /*is_lowest_quality_stream=*/stream_idx == lowest_quality_stream_idx,
+        /*is_highest_quality_stream=*/stream_idx == highest_quality_stream_idx);
+
+    RTC_LOG(LS_INFO) << "[SEA] Multi-encoder mode: initializing stream: "
+                     << stream_idx << ", active: "
+                     << (codec_.simulcastStream[stream_idx].active ? "true"
+                                                                   : "false");
+    int ret = encoder_context->encoder().InitEncode(&stream_codec, settings);
+    if (ret < 0) {
+      encoder_context.reset();
+      Release();
+      RTC_LOG(LS_ERROR) << "[SEA] InitEncode: failed with error code: "
+                        << WebRtcVideoCodecErrorToString(ret);
+      return ret;
+    }
+
+    // Intercept frame encode complete callback only for upper streams, where
+    // we need to set a correct stream index. Set `parent` to nullptr for the
+    // lowest stream to bypass the callback.
+    SimulcastEncoderAdapter* parent = stream_idx > 0 ? this : nullptr;
+
+    bool is_paused = stream_start_bitrate_kbps[stream_idx] == 0;
+    stream_contexts_.emplace_back(
+        parent, std::move(encoder_context),
+        std::make_unique<FramerateController>(stream_codec.maxFramerate),
+        stream_idx, stream_codec.width, stream_codec.height, is_paused);
+  }
+
+  // To save memory, don't store encoders that we don't use.
+  DestroyStoredEncoders();
+
+  inited_.store(1);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::Encode(
+    const VideoFrame& input_image,
+    const std::vector<VideoFrameType>* frame_types) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+
+  if (!Initialized()) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (encoded_complete_callback_ == nullptr) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  if (encoder_info_override_.requested_resolution_alignment()) {
+    const int alignment =
+        *encoder_info_override_.requested_resolution_alignment();
+    if (input_image.width() % alignment != 0 ||
+        input_image.height() % alignment != 0) {
+      RTC_LOG(LS_WARNING) << "Frame " << input_image.width() << "x"
+                          << input_image.height() << " not divisible by "
+                          << alignment;
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    if (encoder_info_override_.apply_alignment_to_all_simulcast_layers()) {
+      for (const auto& layer : stream_contexts_) {
+        if (layer.width() % alignment != 0 || layer.height() % alignment != 0) {
+          RTC_LOG(LS_WARNING)
+              << "Codec " << layer.width() << "x" << layer.height()
+              << " not divisible by " << alignment;
+          return WEBRTC_VIDEO_CODEC_ERROR;
+        }
+      }
+    }
+  }
+
+  bool is_keyframe_needed = false;
+  for (const auto& layer : stream_contexts_) {
+    if (layer.is_keyframe_needed()) {
+      // This is legacy behavior, generating a keyframe on all layers
+      // when generating one for a layer that became active for the first time
+      // or after being disabled.
+      is_keyframe_needed = true;
+      break;
+    }
+  }
+
+  // Temporary thay may hold the result of texture to i420 buffer conversion.
+  rtc::scoped_refptr<VideoFrameBuffer> src_buffer;
+  int src_width = input_image.width();
+  int src_height = input_image.height();
+
+  for (auto& layer : stream_contexts_) {
+    // Don't encode frames in resolutions that we don't intend to send.
+    if (layer.is_paused()) {
+      continue;
+    }
+
+    // Convert timestamp from RTP 90kHz clock.
+    const Timestamp frame_timestamp =
+        Timestamp::Micros((1000 * input_image.timestamp()) / 90);
+
+    // If adapter is passed through and only one sw encoder does simulcast,
+    // frame types for all streams should be passed to the encoder unchanged.
+    // Otherwise a single per-encoder frame type is passed.
+    std::vector<VideoFrameType> stream_frame_types(
+        bypass_mode_
+            ? std::max<unsigned char>(codec_.numberOfSimulcastStreams, 1)
+            : 1,
+        VideoFrameType::kVideoFrameDelta);
+
+    bool keyframe_requested = false;
+    if (is_keyframe_needed) {
+      std::fill(stream_frame_types.begin(), stream_frame_types.end(),
+                VideoFrameType::kVideoFrameKey);
+      keyframe_requested = true;
+    } else if (frame_types) {
+      if (bypass_mode_) {
+        // In bypass mode, we effectively pass on frame_types.
+        RTC_DCHECK_EQ(frame_types->size(), stream_frame_types.size());
+        stream_frame_types = *frame_types;
+        keyframe_requested =
+            absl::c_any_of(*frame_types, [](const VideoFrameType frame_type) {
+              return frame_type == VideoFrameType::kVideoFrameKey;
+            });
+      } else {
+        size_t stream_idx = static_cast<size_t>(layer.stream_idx());
+        if (frame_types->size() >= stream_idx &&
+            (*frame_types)[stream_idx] == VideoFrameType::kVideoFrameKey) {
+          stream_frame_types[0] = VideoFrameType::kVideoFrameKey;
+          keyframe_requested = true;
+        }
+      }
+    }
+    if (keyframe_requested) {
+      layer.OnKeyframe(frame_timestamp);
+    } else if (layer.ShouldDropFrame(frame_timestamp)) {
+      continue;
+    }
+
+    // If scaling isn't required, because the input resolution
+    // matches the destination or the input image is empty (e.g.
+    // a keyframe request for encoders with internal camera
+    // sources) or the source image has a native handle, pass the image on
+    // directly. Otherwise, we'll scale it to match what the encoder expects
+    // (below).
+    // For texture frames, the underlying encoder is expected to be able to
+    // correctly sample/scale the source texture.
+    // TODO(perkj): ensure that works going forward, and figure out how this
+    // affects webrtc:5683.
+    if ((layer.width() == src_width && layer.height() == src_height) ||
+        (input_image.video_frame_buffer()->type() ==
+             VideoFrameBuffer::Type::kNative &&
+         layer.encoder().GetEncoderInfo().supports_native_handle)) {
+      int ret = layer.encoder().Encode(input_image, &stream_frame_types);
+      if (ret != WEBRTC_VIDEO_CODEC_OK) {
+        return ret;
+      }
+    } else {
+      if (src_buffer == nullptr) {
+        src_buffer = input_image.video_frame_buffer();
+      }
+      rtc::scoped_refptr<VideoFrameBuffer> dst_buffer =
+          src_buffer->Scale(layer.width(), layer.height());
+      if (!dst_buffer) {
+        RTC_LOG(LS_ERROR) << "Failed to scale video frame";
+        return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+      }
+
+      // UpdateRect is not propagated to lower simulcast layers currently.
+      // TODO(ilnik): Consider scaling UpdateRect together with the buffer.
+      VideoFrame frame(input_image);
+      frame.set_video_frame_buffer(dst_buffer);
+      frame.set_rotation(webrtc::kVideoRotation_0);
+      frame.set_update_rect(
+          VideoFrame::UpdateRect{0, 0, frame.width(), frame.height()});
+      int ret = layer.encoder().Encode(frame, &stream_frame_types);
+      if (ret != WEBRTC_VIDEO_CODEC_OK) {
+        return ret;
+      }
+    }
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int SimulcastEncoderAdapter::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+  encoded_complete_callback_ = callback;
+  if (!stream_contexts_.empty() && stream_contexts_.front().stream_idx() == 0) {
+    // Bypass frame encode complete callback for the lowest layer since there is
+    // no need to override frame's spatial index.
+    stream_contexts_.front().encoder().RegisterEncodeCompleteCallback(callback);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void SimulcastEncoderAdapter::SetRates(
+    const RateControlParameters& parameters) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+
+  if (!Initialized()) {
+    RTC_LOG(LS_WARNING) << "SetRates while not initialized";
+    return;
+  }
+
+  if (parameters.framerate_fps < 1.0) {
+    RTC_LOG(LS_WARNING) << "Invalid framerate: " << parameters.framerate_fps;
+    return;
+  }
+
+  codec_.maxFramerate = static_cast<uint32_t>(parameters.framerate_fps + 0.5);
+
+  if (bypass_mode_) {
+    stream_contexts_.front().encoder().SetRates(parameters);
+    return;
+  }
+
+  for (StreamContext& layer_context : stream_contexts_) {
+    int stream_idx = layer_context.stream_idx();
+    uint32_t stream_bitrate_kbps =
+        parameters.bitrate.GetSpatialLayerSum(stream_idx) / 1000;
+
+    // Need a key frame if we have not sent this stream before.
+    if (stream_bitrate_kbps > 0 && layer_context.is_paused()) {
+      layer_context.set_is_keyframe_needed();
+    }
+    layer_context.set_is_paused(stream_bitrate_kbps == 0);
+
+    // Slice the temporal layers out of the full allocation and pass it on to
+    // the encoder handling the current simulcast stream.
+    RateControlParameters stream_parameters = parameters;
+    stream_parameters.bitrate = VideoBitrateAllocation();
+    for (int i = 0; i < kMaxTemporalStreams; ++i) {
+      if (parameters.bitrate.HasBitrate(stream_idx, i)) {
+        stream_parameters.bitrate.SetBitrate(
+            0, i, parameters.bitrate.GetBitrate(stream_idx, i));
+      }
+    }
+
+    // Assign link allocation proportionally to spatial layer allocation.
+    if (!parameters.bandwidth_allocation.IsZero() &&
+        parameters.bitrate.get_sum_bps() > 0) {
+      stream_parameters.bandwidth_allocation =
+          DataRate::BitsPerSec((parameters.bandwidth_allocation.bps() *
+                                stream_parameters.bitrate.get_sum_bps()) /
+                               parameters.bitrate.get_sum_bps());
+      // Make sure we don't allocate bandwidth lower than target bitrate.
+      if (stream_parameters.bandwidth_allocation.bps() <
+          stream_parameters.bitrate.get_sum_bps()) {
+        stream_parameters.bandwidth_allocation =
+            DataRate::BitsPerSec(stream_parameters.bitrate.get_sum_bps());
+      }
+    }
+
+    stream_parameters.framerate_fps = std::min<double>(
+        parameters.framerate_fps,
+        layer_context.target_fps().value_or(parameters.framerate_fps));
+
+    layer_context.encoder().SetRates(stream_parameters);
+  }
+}
+
+void SimulcastEncoderAdapter::OnPacketLossRateUpdate(float packet_loss_rate) {
+  for (auto& c : stream_contexts_) {
+    c.encoder().OnPacketLossRateUpdate(packet_loss_rate);
+  }
+}
+
+void SimulcastEncoderAdapter::OnRttUpdate(int64_t rtt_ms) {
+  for (auto& c : stream_contexts_) {
+    c.encoder().OnRttUpdate(rtt_ms);
+  }
+}
+
+void SimulcastEncoderAdapter::OnLossNotification(
+    const LossNotification& loss_notification) {
+  for (auto& c : stream_contexts_) {
+    c.encoder().OnLossNotification(loss_notification);
+  }
+}
+
+// TODO(brandtr): Add task checker to this member function, when all encoder
+// callbacks are coming in on the encoder queue.
+EncodedImageCallback::Result SimulcastEncoderAdapter::OnEncodedImage(
+    size_t stream_idx,
+    const EncodedImage& encodedImage,
+    const CodecSpecificInfo* codecSpecificInfo) {
+  EncodedImage stream_image(encodedImage);
+  CodecSpecificInfo stream_codec_specific = *codecSpecificInfo;
+
+  stream_image.SetSimulcastIndex(stream_idx);
+
+  return encoded_complete_callback_->OnEncodedImage(stream_image,
+                                                    &stream_codec_specific);
+}
+
+void SimulcastEncoderAdapter::OnDroppedFrame(size_t stream_idx) {
+  // Not yet implemented.
+}
+
+bool SimulcastEncoderAdapter::Initialized() const {
+  return inited_.load() == 1;
+}
+
+void SimulcastEncoderAdapter::DestroyStoredEncoders() {
+  while (!cached_encoder_contexts_.empty()) {
+    cached_encoder_contexts_.pop_back();
+  }
+}
+
+std::unique_ptr<SimulcastEncoderAdapter::EncoderContext>
+SimulcastEncoderAdapter::FetchOrCreateEncoderContext(
+    bool is_lowest_quality_stream) const {
+  bool prefer_temporal_support = fallback_encoder_factory_ != nullptr &&
+                                 is_lowest_quality_stream &&
+                                 prefer_temporal_support_on_base_layer_;
+
+  // Toggling of `prefer_temporal_support` requires encoder recreation. Find
+  // and reuse encoder with desired `prefer_temporal_support`. Otherwise, if
+  // there is no such encoder in the cache, create a new instance.
+  auto encoder_context_iter =
+      std::find_if(cached_encoder_contexts_.begin(),
+                   cached_encoder_contexts_.end(), [&](auto& encoder_context) {
+                     return encoder_context->prefer_temporal_support() ==
+                            prefer_temporal_support;
+                   });
+
+  std::unique_ptr<SimulcastEncoderAdapter::EncoderContext> encoder_context;
+  if (encoder_context_iter != cached_encoder_contexts_.end()) {
+    encoder_context = std::move(*encoder_context_iter);
+    cached_encoder_contexts_.erase(encoder_context_iter);
+  } else {
+    std::unique_ptr<VideoEncoder> primary_encoder =
+        primary_encoder_factory_->CreateVideoEncoder(video_format_);
+
+    std::unique_ptr<VideoEncoder> fallback_encoder;
+    if (fallback_encoder_factory_ != nullptr) {
+      fallback_encoder =
+          fallback_encoder_factory_->CreateVideoEncoder(video_format_);
+    }
+
+    std::unique_ptr<VideoEncoder> encoder;
+    VideoEncoder::EncoderInfo primary_info;
+    VideoEncoder::EncoderInfo fallback_info;
+
+    if (primary_encoder != nullptr) {
+      primary_info = primary_encoder->GetEncoderInfo();
+      fallback_info = primary_info;
+
+      if (fallback_encoder == nullptr) {
+        encoder = std::move(primary_encoder);
+      } else {
+        encoder = CreateVideoEncoderSoftwareFallbackWrapper(
+            std::move(fallback_encoder), std::move(primary_encoder),
+            prefer_temporal_support);
+      }
+    } else if (fallback_encoder != nullptr) {
+      RTC_LOG(LS_WARNING) << "Failed to create primary " << video_format_.name
+                          << " encoder. Use fallback encoder.";
+      fallback_info = fallback_encoder->GetEncoderInfo();
+      primary_info = fallback_info;
+      encoder = std::move(fallback_encoder);
+    } else {
+      RTC_LOG(LS_ERROR) << "Failed to create primary and fallback "
+                        << video_format_.name << " encoders.";
+      return nullptr;
+    }
+
+    encoder_context = std::make_unique<SimulcastEncoderAdapter::EncoderContext>(
+        std::move(encoder), prefer_temporal_support, primary_info,
+        fallback_info);
+  }
+
+  encoder_context->encoder().RegisterEncodeCompleteCallback(
+      encoded_complete_callback_);
+  return encoder_context;
+}
+
+webrtc::VideoCodec SimulcastEncoderAdapter::MakeStreamCodec(
+    const webrtc::VideoCodec& codec,
+    int stream_idx,
+    uint32_t start_bitrate_kbps,
+    bool is_lowest_quality_stream,
+    bool is_highest_quality_stream) {
+  webrtc::VideoCodec codec_params = codec;
+  const SimulcastStream& stream_params = codec.simulcastStream[stream_idx];
+
+  codec_params.numberOfSimulcastStreams = 0;
+  codec_params.width = stream_params.width;
+  codec_params.height = stream_params.height;
+  codec_params.maxBitrate = stream_params.maxBitrate;
+  codec_params.minBitrate = stream_params.minBitrate;
+  codec_params.maxFramerate = stream_params.maxFramerate;
+  codec_params.qpMax = stream_params.qpMax;
+  codec_params.active = stream_params.active;
+  // By default, `scalability_mode` comes from SimulcastStream when
+  // SimulcastEncoderAdapter is used. This allows multiple encodings of L1Tx,
+  // but SimulcastStream currently does not support multiple spatial layers.
+  ScalabilityMode scalability_mode = stream_params.GetScalabilityMode();
+  // To support the full set of scalability modes in the event that this is the
+  // only active encoding, prefer VideoCodec::GetScalabilityMode() if all other
+  // encodings are inactive.
+  if (codec.GetScalabilityMode().has_value()) {
+    bool only_active_stream = true;
+    for (int i = 0; i < codec.numberOfSimulcastStreams; ++i) {
+      if (i != stream_idx && codec.simulcastStream[i].active) {
+        only_active_stream = false;
+        break;
+      }
+    }
+    if (only_active_stream) {
+      scalability_mode = codec.GetScalabilityMode().value();
+    }
+  }
+  codec_params.SetScalabilityMode(scalability_mode);
+  // Settings that are based on stream/resolution.
+  if (is_lowest_quality_stream) {
+    // Settings for lowest spatial resolutions.
+    if (codec.mode == VideoCodecMode::kScreensharing) {
+      if (experimental_boosted_screenshare_qp_) {
+        codec_params.qpMax = *experimental_boosted_screenshare_qp_;
+      }
+    } else if (boost_base_layer_quality_) {
+      codec_params.qpMax = kLowestResMaxQp;
+    }
+  }
+  if (codec.codecType == webrtc::kVideoCodecVP8) {
+    codec_params.VP8()->numberOfTemporalLayers =
+        stream_params.numberOfTemporalLayers;
+    if (!is_highest_quality_stream) {
+      // For resolutions below CIF, set the codec `complexity` parameter to
+      // kComplexityHigher, which maps to cpu_used = -4.
+      int pixels_per_frame = codec_params.width * codec_params.height;
+      if (pixels_per_frame < 352 * 288) {
+        codec_params.SetVideoEncoderComplexity(
+            webrtc::VideoCodecComplexity::kComplexityHigher);
+      }
+      // Turn off denoising for all streams but the highest resolution.
+      codec_params.VP8()->denoisingOn = false;
+    }
+  } else if (codec.codecType == webrtc::kVideoCodecH264) {
+    codec_params.H264()->numberOfTemporalLayers =
+        stream_params.numberOfTemporalLayers;
+  }
+
+  // Cap start bitrate to the min bitrate in order to avoid strange codec
+  // behavior.
+  codec_params.startBitrate =
+      std::max(stream_params.minBitrate, start_bitrate_kbps);
+
+  // Legacy screenshare mode is only enabled for the first simulcast layer
+  codec_params.legacy_conference_mode =
+      codec.legacy_conference_mode && stream_idx == 0;
+
+  return codec_params;
+}
+
+void SimulcastEncoderAdapter::OverrideFromFieldTrial(
+    VideoEncoder::EncoderInfo* info) const {
+  if (encoder_info_override_.requested_resolution_alignment()) {
+    info->requested_resolution_alignment = cricket::LeastCommonMultiple(
+        info->requested_resolution_alignment,
+        *encoder_info_override_.requested_resolution_alignment());
+    info->apply_alignment_to_all_simulcast_layers =
+        info->apply_alignment_to_all_simulcast_layers ||
+        encoder_info_override_.apply_alignment_to_all_simulcast_layers();
+  }
+  // Override resolution bitrate limits unless they're set already.
+  if (info->resolution_bitrate_limits.empty() &&
+      !encoder_info_override_.resolution_bitrate_limits().empty()) {
+    info->resolution_bitrate_limits =
+        encoder_info_override_.resolution_bitrate_limits();
+  }
+}
+
+VideoEncoder::EncoderInfo SimulcastEncoderAdapter::GetEncoderInfo() const {
+  if (stream_contexts_.size() == 1) {
+    // Not using simulcast adapting functionality, just pass through.
+    VideoEncoder::EncoderInfo info =
+        stream_contexts_.front().encoder().GetEncoderInfo();
+    OverrideFromFieldTrial(&info);
+    return info;
+  }
+
+  VideoEncoder::EncoderInfo encoder_info;
+  encoder_info.implementation_name = "SimulcastEncoderAdapter";
+  encoder_info.requested_resolution_alignment = 1;
+  encoder_info.apply_alignment_to_all_simulcast_layers = false;
+  encoder_info.supports_native_handle = true;
+  encoder_info.scaling_settings.thresholds = absl::nullopt;
+
+  if (stream_contexts_.empty()) {
+    // GetEncoderInfo queried before InitEncode. Only alignment info is needed
+    // to be filled.
+    // Create one encoder and query it.
+
+    std::unique_ptr<SimulcastEncoderAdapter::EncoderContext> encoder_context =
+        FetchOrCreateEncoderContext(/*is_lowest_quality_stream=*/true);
+    if (encoder_context == nullptr) {
+      return encoder_info;
+    }
+
+    const VideoEncoder::EncoderInfo& primary_info =
+        encoder_context->PrimaryInfo();
+    const VideoEncoder::EncoderInfo& fallback_info =
+        encoder_context->FallbackInfo();
+
+    encoder_info.requested_resolution_alignment = cricket::LeastCommonMultiple(
+        primary_info.requested_resolution_alignment,
+        fallback_info.requested_resolution_alignment);
+
+    encoder_info.apply_alignment_to_all_simulcast_layers =
+        primary_info.apply_alignment_to_all_simulcast_layers ||
+        fallback_info.apply_alignment_to_all_simulcast_layers;
+
+    if (!primary_info.supports_simulcast || !fallback_info.supports_simulcast) {
+      encoder_info.apply_alignment_to_all_simulcast_layers = true;
+    }
+
+    cached_encoder_contexts_.emplace_back(std::move(encoder_context));
+
+    OverrideFromFieldTrial(&encoder_info);
+    return encoder_info;
+  }
+
+  encoder_info.scaling_settings = VideoEncoder::ScalingSettings::kOff;
+
+  for (size_t i = 0; i < stream_contexts_.size(); ++i) {
+    VideoEncoder::EncoderInfo encoder_impl_info =
+        stream_contexts_[i].encoder().GetEncoderInfo();
+    if (i == 0) {
+      // Encoder name indicates names of all sub-encoders.
+      encoder_info.implementation_name += " (";
+      encoder_info.implementation_name += encoder_impl_info.implementation_name;
+
+      encoder_info.supports_native_handle =
+          encoder_impl_info.supports_native_handle;
+      encoder_info.has_trusted_rate_controller =
+          encoder_impl_info.has_trusted_rate_controller;
+      encoder_info.is_hardware_accelerated =
+          encoder_impl_info.is_hardware_accelerated;
+      encoder_info.is_qp_trusted = encoder_impl_info.is_qp_trusted;
+    } else {
+      encoder_info.implementation_name += ", ";
+      encoder_info.implementation_name += encoder_impl_info.implementation_name;
+
+      // Native handle supported if any encoder supports it.
+      encoder_info.supports_native_handle |=
+          encoder_impl_info.supports_native_handle;
+
+      // Trusted rate controller only if all encoders have it.
+      encoder_info.has_trusted_rate_controller &=
+          encoder_impl_info.has_trusted_rate_controller;
+
+      // Uses hardware support if any of the encoders uses it.
+      // For example, if we are having issues with down-scaling due to
+      // pipelining delay in HW encoders we need higher encoder usage
+      // thresholds in CPU adaptation.
+      encoder_info.is_hardware_accelerated |=
+          encoder_impl_info.is_hardware_accelerated;
+
+      // Treat QP from frame/slice/tile header as average QP only if all
+      // encoders report it as average QP.
+      encoder_info.is_qp_trusted =
+          encoder_info.is_qp_trusted.value_or(true) &&
+          encoder_impl_info.is_qp_trusted.value_or(true);
+    }
+    encoder_info.fps_allocation[i] = encoder_impl_info.fps_allocation[0];
+    encoder_info.requested_resolution_alignment = cricket::LeastCommonMultiple(
+        encoder_info.requested_resolution_alignment,
+        encoder_impl_info.requested_resolution_alignment);
+    // request alignment on all layers if any of the encoders may need it, or
+    // if any non-top layer encoder requests a non-trivial alignment.
+    if (encoder_impl_info.apply_alignment_to_all_simulcast_layers ||
+        (encoder_impl_info.requested_resolution_alignment > 1 &&
+         (codec_.simulcastStream[i].height < codec_.height ||
+          codec_.simulcastStream[i].width < codec_.width))) {
+      encoder_info.apply_alignment_to_all_simulcast_layers = true;
+    }
+  }
+  encoder_info.implementation_name += ")";
+
+  OverrideFromFieldTrial(&encoder_info);
+
+  return encoder_info;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/engine/simulcast_encoder_adapter.h

@@ -0,0 +1,200 @@
+/*
+ *  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 MEDIA_ENGINE_SIMULCAST_ENCODER_ADAPTER_H_
+#define MEDIA_ENGINE_SIMULCAST_ENCODER_ADAPTER_H_
+
+#include <atomic>
+#include <list>
+#include <memory>
+#include <stack>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/fec_controller_override.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "common_video/framerate_controller.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/experiments/encoder_info_settings.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// SimulcastEncoderAdapter implements simulcast support by creating multiple
+// webrtc::VideoEncoder instances with the given VideoEncoderFactory.
+// The object is created and destroyed on the worker thread, but all public
+// interfaces should be called from the encoder task queue.
+class RTC_EXPORT SimulcastEncoderAdapter : public VideoEncoder {
+ public:
+  // TODO(bugs.webrtc.org/11000): Remove when downstream usage is gone.
+  SimulcastEncoderAdapter(VideoEncoderFactory* primarty_factory,
+                          const SdpVideoFormat& format);
+  // `primary_factory` produces the first-choice encoders to use.
+  // `fallback_factory`, if non-null, is used to create fallback encoder that
+  // will be used if InitEncode() fails for the primary encoder.
+  SimulcastEncoderAdapter(VideoEncoderFactory* primary_factory,
+                          VideoEncoderFactory* fallback_factory,
+                          const SdpVideoFormat& format,
+                          const FieldTrialsView& field_trials);
+  ~SimulcastEncoderAdapter() override;
+
+  // Implements VideoEncoder.
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+  int Release() override;
+  int InitEncode(const VideoCodec* codec_settings,
+                 const VideoEncoder::Settings& settings) override;
+  int Encode(const VideoFrame& input_image,
+             const std::vector<VideoFrameType>* frame_types) override;
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
+  void SetRates(const RateControlParameters& parameters) override;
+  void OnPacketLossRateUpdate(float packet_loss_rate) override;
+  void OnRttUpdate(int64_t rtt_ms) override;
+  void OnLossNotification(const LossNotification& loss_notification) override;
+
+  EncoderInfo GetEncoderInfo() const override;
+
+ private:
+  class EncoderContext {
+   public:
+    EncoderContext(std::unique_ptr<VideoEncoder> encoder,
+                   bool prefer_temporal_support,
+                   VideoEncoder::EncoderInfo primary_info,
+                   VideoEncoder::EncoderInfo fallback_info);
+    EncoderContext& operator=(EncoderContext&&) = delete;
+
+    VideoEncoder& encoder() { return *encoder_; }
+    bool prefer_temporal_support() { return prefer_temporal_support_; }
+    void Release();
+
+    const VideoEncoder::EncoderInfo& PrimaryInfo() { return primary_info_; }
+
+    const VideoEncoder::EncoderInfo& FallbackInfo() { return fallback_info_; }
+
+   private:
+    std::unique_ptr<VideoEncoder> encoder_;
+    bool prefer_temporal_support_;
+    const VideoEncoder::EncoderInfo primary_info_;
+    const VideoEncoder::EncoderInfo fallback_info_;
+  };
+
+  class StreamContext : public EncodedImageCallback {
+   public:
+    StreamContext(SimulcastEncoderAdapter* parent,
+                  std::unique_ptr<EncoderContext> encoder_context,
+                  std::unique_ptr<FramerateController> framerate_controller,
+                  int stream_idx,
+                  uint16_t width,
+                  uint16_t height,
+                  bool send_stream);
+    StreamContext(StreamContext&& rhs);
+    StreamContext& operator=(StreamContext&&) = delete;
+    ~StreamContext() override;
+
+    Result OnEncodedImage(
+        const EncodedImage& encoded_image,
+        const CodecSpecificInfo* codec_specific_info) override;
+    void OnDroppedFrame(DropReason reason) override;
+
+    VideoEncoder& encoder() { return encoder_context_->encoder(); }
+    const VideoEncoder& encoder() const { return encoder_context_->encoder(); }
+    int stream_idx() const { return stream_idx_; }
+    uint16_t width() const { return width_; }
+    uint16_t height() const { return height_; }
+    bool is_keyframe_needed() const {
+      return !is_paused_ && is_keyframe_needed_;
+    }
+    void set_is_keyframe_needed() { is_keyframe_needed_ = true; }
+    bool is_paused() const { return is_paused_; }
+    void set_is_paused(bool is_paused) { is_paused_ = is_paused; }
+    absl::optional<double> target_fps() const {
+      return framerate_controller_ == nullptr
+                 ? absl::nullopt
+                 : absl::optional<double>(
+                       framerate_controller_->GetMaxFramerate());
+    }
+
+    std::unique_ptr<EncoderContext> ReleaseEncoderContext() &&;
+    void OnKeyframe(Timestamp timestamp);
+    bool ShouldDropFrame(Timestamp timestamp);
+
+   private:
+    SimulcastEncoderAdapter* const parent_;
+    std::unique_ptr<EncoderContext> encoder_context_;
+    std::unique_ptr<FramerateController> framerate_controller_;
+    const int stream_idx_;
+    const uint16_t width_;
+    const uint16_t height_;
+    bool is_keyframe_needed_;
+    bool is_paused_;
+  };
+
+  bool Initialized() const;
+
+  void DestroyStoredEncoders();
+
+  // This method creates encoder. May reuse previously created encoders from
+  // `cached_encoder_contexts_`. It's const because it's used from
+  // const GetEncoderInfo().
+  std::unique_ptr<EncoderContext> FetchOrCreateEncoderContext(
+      bool is_lowest_quality_stream) const;
+
+  webrtc::VideoCodec MakeStreamCodec(const webrtc::VideoCodec& codec,
+                                     int stream_idx,
+                                     uint32_t start_bitrate_kbps,
+                                     bool is_lowest_quality_stream,
+                                     bool is_highest_quality_stream);
+
+  EncodedImageCallback::Result OnEncodedImage(
+      size_t stream_idx,
+      const EncodedImage& encoded_image,
+      const CodecSpecificInfo* codec_specific_info);
+
+  void OnDroppedFrame(size_t stream_idx);
+
+  void OverrideFromFieldTrial(VideoEncoder::EncoderInfo* info) const;
+
+  std::atomic<int> inited_;
+  VideoEncoderFactory* const primary_encoder_factory_;
+  VideoEncoderFactory* const fallback_encoder_factory_;
+  const SdpVideoFormat video_format_;
+  VideoCodec codec_;
+  int total_streams_count_;
+  bool bypass_mode_;
+  std::vector<StreamContext> stream_contexts_;
+  EncodedImageCallback* encoded_complete_callback_;
+
+  // Used for checking the single-threaded access of the encoder interface.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_;
+
+  // Store previously created and released encoders , so they don't have to be
+  // recreated. Remaining encoders are destroyed by the destructor.
+  // Marked as `mutable` becuase we may need to temporarily create encoder in
+  // GetEncoderInfo(), which is const.
+  mutable std::list<std::unique_ptr<EncoderContext>> cached_encoder_contexts_;
+
+  const absl::optional<unsigned int> experimental_boosted_screenshare_qp_;
+  const bool boost_base_layer_quality_;
+  const bool prefer_temporal_support_on_base_layer_;
+
+  const SimulcastEncoderAdapterEncoderInfoSettings encoder_info_override_;
+};
+
+}  // namespace webrtc
+
+#endif  // MEDIA_ENGINE_SIMULCAST_ENCODER_ADAPTER_H_

TMessagesProj/jni/voip/webrtc/media/engine/webrtc_media_engine.cc

@@ -0,0 +1,185 @@
+/*
+ *  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 "media/engine/webrtc_media_engine.h"
+
+#include <algorithm>
+#include <map>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/match.h"
+#include "media/base/media_constants.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace cricket {
+namespace {
+// Remove mutually exclusive extensions with lower priority.
+void DiscardRedundantExtensions(
+    std::vector<webrtc::RtpExtension>* extensions,
+    rtc::ArrayView<const char* const> extensions_decreasing_prio) {
+  RTC_DCHECK(extensions);
+  bool found = false;
+  for (const char* uri : extensions_decreasing_prio) {
+    auto it = absl::c_find_if(
+        *extensions,
+        [uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; });
+    if (it != extensions->end()) {
+      if (found) {
+        extensions->erase(it);
+      }
+      found = true;
+    }
+  }
+}
+}  // namespace
+
+bool ValidateRtpExtensions(
+    rtc::ArrayView<const webrtc::RtpExtension> extensions,
+    rtc::ArrayView<const webrtc::RtpExtension> old_extensions) {
+  bool id_used[1 + webrtc::RtpExtension::kMaxId] = {false};
+  for (const auto& extension : extensions) {
+    if (extension.id < webrtc::RtpExtension::kMinId ||
+        extension.id > webrtc::RtpExtension::kMaxId) {
+      RTC_LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString();
+      return false;
+    }
+    if (id_used[extension.id]) {
+      RTC_LOG(LS_ERROR) << "Duplicate RTP extension ID: "
+                        << extension.ToString();
+      return false;
+    }
+    id_used[extension.id] = true;
+  }
+  // Validate the extension list against the already negotiated extensions.
+  // Re-registering is OK, re-mapping (either same URL at new ID or same
+  // ID used with new URL) is an illegal remap.
+
+  // This is required in order to avoid a crash when registering an
+  // extension. A better structure would use the registered extensions
+  // in the RTPSender. This requires spinning through:
+  //
+  // WebRtcVoiceMediaChannel::::WebRtcAudioSendStream::stream_ (pointer)
+  // AudioSendStream::rtp_rtcp_module_ (pointer)
+  // ModuleRtpRtcpImpl2::rtp_sender_ (pointer)
+  // RtpSenderContext::packet_generator (struct member)
+  // RTPSender::rtp_header_extension_map_ (class member)
+  //
+  // Getting at this seems like a hard slog.
+  if (!old_extensions.empty()) {
+    absl::string_view urimap[1 + webrtc::RtpExtension::kMaxId];
+    std::map<absl::string_view, int> idmap;
+    for (const auto& old_extension : old_extensions) {
+      urimap[old_extension.id] = old_extension.uri;
+      idmap[old_extension.uri] = old_extension.id;
+    }
+    for (const auto& extension : extensions) {
+      if (!urimap[extension.id].empty() &&
+          urimap[extension.id] != extension.uri) {
+        RTC_LOG(LS_ERROR) << "Extension negotiation failure: " << extension.id
+                          << " was mapped to " << urimap[extension.id]
+                          << " but is proposed changed to " << extension.uri;
+        return false;
+      }
+      const auto& it = idmap.find(extension.uri);
+      if (it != idmap.end() && it->second != extension.id) {
+        RTC_LOG(LS_ERROR) << "Extension negotation failure: " << extension.uri
+                          << " was identified by " << it->second
+                          << " but is proposed changed to " << extension.id;
+        return false;
+      }
+    }
+  }
+  return true;
+}
+
+std::vector<webrtc::RtpExtension> FilterRtpExtensions(
+    const std::vector<webrtc::RtpExtension>& extensions,
+    bool (*supported)(absl::string_view),
+    bool filter_redundant_extensions,
+    const webrtc::FieldTrialsView& trials) {
+  // Don't check against old parameters; this should have been done earlier.
+  RTC_DCHECK(ValidateRtpExtensions(extensions, {}));
+  RTC_DCHECK(supported);
+  std::vector<webrtc::RtpExtension> result;
+
+  // Ignore any extensions that we don't recognize.
+  for (const auto& extension : extensions) {
+    if (supported(extension.uri)) {
+      result.push_back(extension);
+    } else {
+      RTC_LOG(LS_WARNING) << "Unsupported RTP extension: "
+                          << extension.ToString();
+    }
+  }
+
+  // Sort by name, ascending (prioritise encryption), so that we don't reset
+  // extensions if they were specified in a different order (also allows us
+  // to use std::unique below).
+  absl::c_sort(result, [](const webrtc::RtpExtension& rhs,
+                          const webrtc::RtpExtension& lhs) {
+    return rhs.encrypt == lhs.encrypt ? rhs.uri < lhs.uri
+                                      : rhs.encrypt > lhs.encrypt;
+  });
+
+  // Remove unnecessary extensions (used on send side).
+  if (filter_redundant_extensions) {
+    auto it = std::unique(
+        result.begin(), result.end(),
+        [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
+          return rhs.uri == lhs.uri && rhs.encrypt == lhs.encrypt;
+        });
+    result.erase(it, result.end());
+
+    // Keep just the highest priority extension of any in the following lists.
+    if (absl::StartsWith(trials.Lookup("WebRTC-FilterAbsSendTimeExtension"),
+                         "Enabled")) {
+      static const char* const kBweExtensionPriorities[] = {
+          webrtc::RtpExtension::kTransportSequenceNumberUri,
+          webrtc::RtpExtension::kAbsSendTimeUri,
+          webrtc::RtpExtension::kTimestampOffsetUri};
+      DiscardRedundantExtensions(&result, kBweExtensionPriorities);
+    } else {
+      static const char* const kBweExtensionPriorities[] = {
+          webrtc::RtpExtension::kAbsSendTimeUri,
+          webrtc::RtpExtension::kTimestampOffsetUri};
+      DiscardRedundantExtensions(&result, kBweExtensionPriorities);
+    }
+  }
+  return result;
+}
+
+webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec) {
+  webrtc::BitrateConstraints config;
+  int bitrate_kbps = 0;
+  if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
+      bitrate_kbps > 0) {
+    config.min_bitrate_bps = bitrate_kbps * 1000;
+  } else {
+    config.min_bitrate_bps = 0;
+  }
+  if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
+      bitrate_kbps > 0) {
+    config.start_bitrate_bps = bitrate_kbps * 1000;
+  } else {
+    // Do not reconfigure start bitrate unless it's specified and positive.
+    config.start_bitrate_bps = -1;
+  }
+  if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
+      bitrate_kbps > 0) {
+    config.max_bitrate_bps = bitrate_kbps * 1000;
+  } else {
+    config.max_bitrate_bps = -1;
+  }
+  return config;
+}
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/engine/webrtc_media_engine.h

@@ -0,0 +1,45 @@
+/*
+ *  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 MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_
+#define MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/rtp_parameters.h"
+#include "api/transport/bitrate_settings.h"
+#include "media/base/codec.h"
+
+namespace cricket {
+
+// Verify that extension IDs are within 1-byte extension range and are not
+// overlapping, and that they form a legal change from previously registerd
+// extensions (if any).
+bool ValidateRtpExtensions(
+    rtc::ArrayView<const webrtc::RtpExtension> extennsions,
+    rtc::ArrayView<const webrtc::RtpExtension> old_extensions);
+
+// Discard any extensions not validated by the 'supported' predicate. Duplicate
+// extensions are removed if 'filter_redundant_extensions' is set, and also any
+// mutually exclusive extensions (see implementation for details) are removed.
+std::vector<webrtc::RtpExtension> FilterRtpExtensions(
+    const std::vector<webrtc::RtpExtension>& extensions,
+    bool (*supported)(absl::string_view),
+    bool filter_redundant_extensions,
+    const webrtc::FieldTrialsView& trials);
+
+webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec);
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/engine/webrtc_video_engine.h

@@ -0,0 +1,901 @@
+/*
+ *  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 MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
+#define MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <functional>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/transport.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/rtc_error.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_sender_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "api/video/recordable_encoded_frame.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "api/video/video_stream_encoder_settings.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "call/call.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/rtp_config.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_config.h"
+#include "media/base/media_engine.h"
+#include "media/base/stream_params.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread_annotations.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+class VideoDecoderFactory;
+class VideoEncoderFactory;
+}  // namespace webrtc
+
+namespace cricket {
+
+// Public for testing.
+// Inputs StreamStats for all types of substreams (kMedia, kRtx, kFlexfec) and
+// merges any non-kMedia substream stats object into its referenced kMedia-type
+// substream. The resulting substreams are all kMedia. This means, for example,
+// that packet and byte counters of RTX and FlexFEC streams are accounted for in
+// the relevant RTP media stream's stats. This makes the resulting StreamStats
+// objects ready to be turned into "outbound-rtp" stats objects for GetStats()
+// which does not create separate stream stats objects for complementary
+// streams.
+std::map<uint32_t, webrtc::VideoSendStream::StreamStats>
+MergeInfoAboutOutboundRtpSubstreamsForTesting(
+    const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>& substreams);
+
+// WebRtcVideoEngine is used for the new native WebRTC Video API (webrtc:1667).
+class WebRtcVideoEngine : public VideoEngineInterface {
+ public:
+  // These video codec factories represents all video codecs, i.e. both software
+  // and external hardware codecs.
+  WebRtcVideoEngine(
+      std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
+      std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
+      const webrtc::FieldTrialsView& trials);
+
+  ~WebRtcVideoEngine() override;
+
+  std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
+      override;
+  std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options) override;
+
+  std::vector<VideoCodec> send_codecs() const override {
+    return send_codecs(true);
+  }
+  std::vector<VideoCodec> recv_codecs() const override {
+    return recv_codecs(true);
+  }
+  std::vector<VideoCodec> send_codecs(bool include_rtx) const override;
+  std::vector<VideoCodec> recv_codecs(bool include_rtx) const override;
+  std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
+      const override;
+
+ private:
+  const std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory_;
+  const std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory_;
+  const std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
+      bitrate_allocator_factory_;
+  const webrtc::FieldTrialsView& trials_;
+};
+
+struct VideoCodecSettings {
+  explicit VideoCodecSettings(const VideoCodec& codec);
+
+  // Checks if all members of |*this| are equal to the corresponding members
+  // of `other`.
+  bool operator==(const VideoCodecSettings& other) const;
+  bool operator!=(const VideoCodecSettings& other) const;
+
+  // Checks if all members of `a`, except `flexfec_payload_type`, are equal
+  // to the corresponding members of `b`.
+  static bool EqualsDisregardingFlexfec(const VideoCodecSettings& a,
+                                        const VideoCodecSettings& b);
+
+  VideoCodec codec;
+  webrtc::UlpfecConfig ulpfec;
+  int flexfec_payload_type;  // -1 if absent.
+  int rtx_payload_type;      // -1 if absent.
+  absl::optional<int> rtx_time;
+};
+
+class WebRtcVideoSendChannel : public MediaChannelUtil,
+                               public VideoMediaSendChannelInterface,
+                               public webrtc::EncoderSwitchRequestCallback {
+ public:
+  WebRtcVideoSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::VideoEncoderFactory* encoder_factory,
+      webrtc::VideoDecoderFactory* decoder_factory,
+      webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory);
+  ~WebRtcVideoSendChannel() override;
+
+  MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
+  // Type manipulations
+  VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
+  VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+  // Functions imported from MediaChannelUtil
+  bool HasNetworkInterface() const override {
+    return MediaChannelUtil::HasNetworkInterface();
+  }
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
+    MediaChannelUtil::SetExtmapAllowMixed(extmap_allow_mixed);
+  }
+  bool ExtmapAllowMixed() const override {
+    return MediaChannelUtil::ExtmapAllowMixed();
+  }
+
+  // Common functions between sender and receiver
+  void SetInterface(MediaChannelNetworkInterface* iface) override;
+  // VideoMediaSendChannelInterface implementation
+  bool SetSenderParameters(const VideoSenderParameters& params) override;
+  webrtc::RTCError SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters,
+      webrtc::SetParametersCallback callback) override;
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  absl::optional<Codec> GetSendCodec() const override;
+  bool SetSend(bool send) override;
+  bool SetVideoSend(
+      uint32_t ssrc,
+      const VideoOptions* options,
+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
+  bool AddSendStream(const StreamParams& sp) override;
+  bool RemoveSendStream(uint32_t ssrc) override;
+  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
+  bool GetStats(VideoMediaSendInfo* info) override;
+
+  void OnPacketSent(const rtc::SentPacket& sent_packet) override;
+  void OnReadyToSend(bool ready) override;
+  void OnNetworkRouteChanged(absl::string_view transport_name,
+                             const rtc::NetworkRoute& network_route) override;
+
+  // Set a frame encryptor to a particular ssrc that will intercept all
+  // outgoing video frames and attempt to encrypt them and forward the result
+  // to the packetizer.
+  void SetFrameEncryptor(uint32_t ssrc,
+                         rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
+                             frame_encryptor) override;
+
+  // note: The encoder_selector object must remain valid for the lifetime of the
+  // MediaChannel, unless replaced.
+  void SetEncoderSelector(uint32_t ssrc,
+                          webrtc::VideoEncoderFactory::EncoderSelectorInterface*
+                              encoder_selector) override;
+
+  void SetSendCodecChangedCallback(
+      absl::AnyInvocable<void()> callback) override {
+    send_codec_changed_callback_ = std::move(callback);
+  }
+
+  void SetSsrcListChangedCallback(
+      absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
+    ssrc_list_changed_callback_ = std::move(callback);
+  }
+
+  // Implemented for VideoMediaChannelTest.
+  bool sending() const {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return sending_;
+  }
+
+  // AdaptReason is used for expressing why a WebRtcVideoSendStream request
+  // a lower input frame size than the currently configured camera input frame
+  // size. There can be more than one reason OR:ed together.
+  enum AdaptReason {
+    ADAPTREASON_NONE = 0,
+    ADAPTREASON_CPU = 1,
+    ADAPTREASON_BANDWIDTH = 2,
+  };
+
+  // Implements webrtc::EncoderSwitchRequestCallback.
+  void RequestEncoderFallback() override;
+  void RequestEncoderSwitch(const webrtc::SdpVideoFormat& format,
+                            bool allow_default_fallback) override;
+
+  void GenerateSendKeyFrame(uint32_t ssrc,
+                            const std::vector<std::string>& rids) override;
+
+  void SetEncoderToPacketizerFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  // Information queries to support SetReceiverFeedbackParameters
+  webrtc::RtcpMode SendCodecRtcpMode() const override {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return send_params_.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
+                                          : webrtc::RtcpMode::kCompound;
+  }
+
+  bool SendCodecHasLntf() const override {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    if (!send_codec()) {
+      return false;
+    }
+    return HasLntf(send_codec()->codec);
+  }
+  bool SendCodecHasNack() const override {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    if (!send_codec()) {
+      return false;
+    }
+    return HasNack(send_codec()->codec);
+  }
+  absl::optional<int> SendCodecRtxTime() const override {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    if (!send_codec()) {
+      return absl::nullopt;
+    }
+    return send_codec()->rtx_time;
+  }
+
+ private:
+  struct ChangedSenderParameters {
+    // These optionals are unset if not changed.
+    absl::optional<VideoCodecSettings> send_codec;
+    absl::optional<std::vector<VideoCodecSettings>> negotiated_codecs;
+    absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
+    absl::optional<std::string> mid;
+    absl::optional<bool> extmap_allow_mixed;
+    absl::optional<int> max_bandwidth_bps;
+    absl::optional<bool> conference_mode;
+    absl::optional<webrtc::RtcpMode> rtcp_mode;
+  };
+
+  bool GetChangedSenderParameters(const VideoSenderParameters& params,
+                                  ChangedSenderParameters* changed_params) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  bool ApplyChangedParams(const ChangedSenderParameters& changed_params);
+  bool ValidateSendSsrcAvailability(const StreamParams& sp) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  // Populates `rtx_associated_payload_types`, `raw_payload_types` and
+  // `decoders` based on codec settings provided by `recv_codecs`.
+  // `recv_codecs` must be non-empty and all other parameters must be empty.
+  static void ExtractCodecInformation(
+      rtc::ArrayView<const VideoCodecSettings> recv_codecs,
+      std::map<int, int>& rtx_associated_payload_types,
+      std::set<int>& raw_payload_types,
+      std::vector<webrtc::VideoReceiveStreamInterface::Decoder>& decoders);
+
+  // Wrapper for the sender part.
+  class WebRtcVideoSendStream {
+   public:
+    WebRtcVideoSendStream(
+        webrtc::Call* call,
+        const StreamParams& sp,
+        webrtc::VideoSendStream::Config config,
+        const VideoOptions& options,
+        bool enable_cpu_overuse_detection,
+        int max_bitrate_bps,
+        const absl::optional<VideoCodecSettings>& codec_settings,
+        const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
+        const VideoSenderParameters& send_params);
+    ~WebRtcVideoSendStream();
+
+    void SetSenderParameters(const ChangedSenderParameters& send_params);
+    webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters,
+                                      webrtc::SetParametersCallback callback);
+    webrtc::RtpParameters GetRtpParameters() const;
+
+    void SetFrameEncryptor(
+        rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor);
+
+    bool SetVideoSend(const VideoOptions* options,
+                      rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
+
+    // note: The encoder_selector object must remain valid for the lifetime of
+    // the MediaChannel, unless replaced.
+    void SetEncoderSelector(
+        webrtc::VideoEncoderFactory::EncoderSelectorInterface*
+            encoder_selector);
+
+    void SetSend(bool send);
+
+    const std::vector<uint32_t>& GetSsrcs() const;
+    // Returns per ssrc VideoSenderInfos. Useful for simulcast scenario.
+    std::vector<VideoSenderInfo> GetPerLayerVideoSenderInfos(bool log_stats);
+    // Aggregates per ssrc VideoSenderInfos to single VideoSenderInfo for
+    // legacy reasons. Used in old GetStats API and track stats.
+    VideoSenderInfo GetAggregatedVideoSenderInfo(
+        const std::vector<VideoSenderInfo>& infos) const;
+    void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
+
+    void SetEncoderToPacketizerFrameTransformer(
+        rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+            frame_transformer);
+    void GenerateKeyFrame(const std::vector<std::string>& rids);
+
+   private:
+    // Parameters needed to reconstruct the underlying stream.
+    // webrtc::VideoSendStream doesn't support setting a lot of options on the
+    // fly, so when those need to be changed we tear down and reconstruct with
+    // similar parameters depending on which options changed etc.
+    struct VideoSendStreamParameters {
+      VideoSendStreamParameters(
+          webrtc::VideoSendStream::Config config,
+          const VideoOptions& options,
+          int max_bitrate_bps,
+          const absl::optional<VideoCodecSettings>& codec_settings);
+      webrtc::VideoSendStream::Config config;
+      VideoOptions options;
+      int max_bitrate_bps;
+      bool conference_mode;
+      absl::optional<VideoCodecSettings> codec_settings;
+      // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
+      // typically changes when setting a new resolution or reconfiguring
+      // bitrates.
+      webrtc::VideoEncoderConfig encoder_config;
+    };
+
+    rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
+    ConfigureVideoEncoderSettings(const VideoCodec& codec);
+    void SetCodec(const VideoCodecSettings& codec);
+    void RecreateWebRtcStream();
+    webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
+        const VideoCodec& codec) const;
+    void ReconfigureEncoder(webrtc::SetParametersCallback callback);
+
+    // Calls Start or Stop according to whether or not `sending_` is true.
+    void UpdateSendState();
+
+    webrtc::DegradationPreference GetDegradationPreference() const
+        RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
+
+    RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
+    webrtc::TaskQueueBase* const worker_thread_;
+    const std::vector<uint32_t> ssrcs_ RTC_GUARDED_BY(&thread_checker_);
+    const std::vector<SsrcGroup> ssrc_groups_ RTC_GUARDED_BY(&thread_checker_);
+    webrtc::Call* const call_;
+    const bool enable_cpu_overuse_detection_;
+    rtc::VideoSourceInterface<webrtc::VideoFrame>* source_
+        RTC_GUARDED_BY(&thread_checker_);
+
+    webrtc::VideoSendStream* stream_ RTC_GUARDED_BY(&thread_checker_);
+
+    // Contains settings that are the same for all streams in the MediaChannel,
+    // such as codecs, header extensions, and the global bitrate limit for the
+    // entire channel.
+    VideoSendStreamParameters parameters_ RTC_GUARDED_BY(&thread_checker_);
+    // Contains settings that are unique for each stream, such as max_bitrate.
+    // Does *not* contain codecs, however.
+    // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
+    // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
+    // one stream per MediaChannel.
+    webrtc::RtpParameters rtp_parameters_ RTC_GUARDED_BY(&thread_checker_);
+
+    bool sending_ RTC_GUARDED_BY(&thread_checker_);
+
+    // TODO(asapersson): investigate why setting
+    // DegrationPreferences::MAINTAIN_RESOLUTION isn't sufficient to disable
+    // downscaling everywhere in the pipeline.
+    const bool disable_automatic_resize_;
+  };
+
+  void Construct(webrtc::Call* call, WebRtcVideoEngine* engine);
+
+  // Get all codecs that are compatible with the receiver.
+  std::vector<VideoCodecSettings> SelectSendVideoCodecs(
+      const std::vector<VideoCodecSettings>& remote_mapped_codecs) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  void FillSenderStats(VideoMediaSendInfo* info, bool log_stats)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
+                                    VideoMediaInfo* info)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  void FillSendCodecStats(VideoMediaSendInfo* video_media_info)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  // Accessor function for send_codec_. Introduced in order to ensure
+  // that a receive channel does not touch the send codec directly.
+  // Can go away once these are different classes.
+  // TODO(bugs.webrtc.org/13931): Remove this function
+  absl::optional<VideoCodecSettings>& send_codec() { return send_codec_; }
+  const absl::optional<VideoCodecSettings>& send_codec() const {
+    return send_codec_;
+  }
+  webrtc::TaskQueueBase* const worker_thread_;
+  webrtc::ScopedTaskSafety task_safety_;
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_thread_checker_{
+      webrtc::SequenceChecker::kDetached};
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
+
+  uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
+  bool sending_ RTC_GUARDED_BY(thread_checker_);
+  bool receiving_ RTC_GUARDED_BY(&thread_checker_);
+  webrtc::Call* const call_;
+
+  rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_
+      RTC_GUARDED_BY(thread_checker_);
+
+  // Delay for unsignaled streams, which may be set before the stream exists.
+  int default_recv_base_minimum_delay_ms_ RTC_GUARDED_BY(thread_checker_) = 0;
+
+  const MediaConfig::Video video_config_ RTC_GUARDED_BY(thread_checker_);
+
+  // Using primary-ssrc (first ssrc) as key.
+  std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
+      RTC_GUARDED_BY(thread_checker_);
+  // When the channel and demuxer get reconfigured, there is a window of time
+  // where we have to be prepared for packets arriving based on the old demuxer
+  // criteria because the streams live on the worker thread and the demuxer
+  // lives on the network thread. Because packets are posted from the network
+  // thread to the worker thread, they can still be in-flight when streams are
+  // reconfgured. This can happen when `demuxer_criteria_id_` and
+  // `demuxer_criteria_completed_id_` don't match. During this time, we do not
+  // want to create unsignalled receive streams and should instead drop the
+  // packets. E.g:
+  // * If RemoveRecvStream(old_ssrc) was recently called, there may be packets
+  //   in-flight for that ssrc. This happens when a receiver becomes inactive.
+  // * If we go from one to many m= sections, the demuxer may change from
+  //   forwarding all packets to only forwarding the configured ssrcs, so there
+  //   is a risk of receiving ssrcs for other, recently added m= sections.
+  uint32_t demuxer_criteria_id_ RTC_GUARDED_BY(thread_checker_) = 0;
+  uint32_t demuxer_criteria_completed_id_ RTC_GUARDED_BY(thread_checker_) = 0;
+  absl::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
+      RTC_GUARDED_BY(thread_checker_);
+  std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
+  std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(thread_checker_);
+
+  absl::optional<VideoCodecSettings> send_codec_
+      RTC_GUARDED_BY(thread_checker_);
+  std::vector<VideoCodecSettings> negotiated_codecs_
+      RTC_GUARDED_BY(thread_checker_);
+
+  std::vector<webrtc::RtpExtension> send_rtp_extensions_
+      RTC_GUARDED_BY(thread_checker_);
+
+  webrtc::VideoEncoderFactory* const encoder_factory_
+      RTC_GUARDED_BY(thread_checker_);
+  webrtc::VideoDecoderFactory* const decoder_factory_
+      RTC_GUARDED_BY(thread_checker_);
+  webrtc::VideoBitrateAllocatorFactory* const bitrate_allocator_factory_
+      RTC_GUARDED_BY(thread_checker_);
+  std::vector<VideoCodecSettings> recv_codecs_ RTC_GUARDED_BY(thread_checker_);
+  webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_
+      RTC_GUARDED_BY(thread_checker_);
+  std::vector<webrtc::RtpExtension> recv_rtp_extensions_
+      RTC_GUARDED_BY(thread_checker_);
+  // See reason for keeping track of the FlexFEC payload type separately in
+  // comment in WebRtcVideoChannel::ChangedReceiverParameters.
+  int recv_flexfec_payload_type_ RTC_GUARDED_BY(thread_checker_);
+  webrtc::BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
+  // TODO(deadbeef): Don't duplicate information between
+  // send_params/recv_params, rtp_extensions, options, etc.
+  VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
+  VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
+  VideoReceiverParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
+  int64_t last_send_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
+  int64_t last_receive_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
+  const bool discard_unknown_ssrc_packets_ RTC_GUARDED_BY(thread_checker_);
+  // This is a stream param that comes from the remote description, but wasn't
+  // signaled with any a=ssrc lines. It holds information that was signaled
+  // before the unsignaled receive stream is created when the first packet is
+  // received.
+  StreamParams unsignaled_stream_params_ RTC_GUARDED_BY(thread_checker_);
+  // Per peer connection crypto options that last for the lifetime of the peer
+  // connection.
+  const webrtc::CryptoOptions crypto_options_ RTC_GUARDED_BY(thread_checker_);
+
+  // Optional frame transformer set on unsignaled streams.
+  rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+      unsignaled_frame_transformer_ RTC_GUARDED_BY(thread_checker_);
+
+  // RTP parameters that need to be set when creating a video receive stream.
+  // Only used in Receiver mode - in Both mode, it reads those things from the
+  // codec.
+  webrtc::VideoReceiveStreamInterface::Config::Rtp rtp_config_;
+
+  // Callback invoked whenever the send codec changes.
+  // TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
+  absl::AnyInvocable<void()> send_codec_changed_callback_;
+  // Callback invoked whenever the list of SSRCs changes.
+  absl::AnyInvocable<void(const std::set<uint32_t>&)>
+      ssrc_list_changed_callback_;
+};
+
+class WebRtcVideoReceiveChannel : public MediaChannelUtil,
+                                  public VideoMediaReceiveChannelInterface {
+ public:
+  WebRtcVideoReceiveChannel(webrtc::Call* call,
+                            const MediaConfig& config,
+                            const VideoOptions& options,
+                            const webrtc::CryptoOptions& crypto_options,
+                            webrtc::VideoDecoderFactory* decoder_factory);
+  ~WebRtcVideoReceiveChannel() override;
+
+ public:
+  MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
+  VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
+    return this;
+  }
+  VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  // Common functions between sender and receiver
+  void SetInterface(MediaChannelNetworkInterface* iface) override;
+  // VideoMediaReceiveChannelInterface implementation
+  bool SetReceiverParameters(const VideoReceiverParameters& params) override;
+  webrtc::RtpParameters GetRtpReceiverParameters(uint32_t ssrc) const override;
+  webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
+  void SetReceive(bool receive) override;
+  bool AddRecvStream(const StreamParams& sp) override;
+  bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
+    // Invokes private AddRecvStream variant function
+    return AddRecvStream(sp, true);
+  }
+  bool RemoveRecvStream(uint32_t ssrc) override;
+  void ResetUnsignaledRecvStream() override;
+  absl::optional<uint32_t> GetUnsignaledSsrc() const override;
+  void OnDemuxerCriteriaUpdatePending() override;
+  void OnDemuxerCriteriaUpdateComplete() override;
+  bool SetSink(uint32_t ssrc,
+               rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+  void SetDefaultSink(
+      rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+  bool GetStats(VideoMediaReceiveInfo* info) override;
+  void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override;
+  bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
+
+  absl::optional<int> GetBaseMinimumPlayoutDelayMs(
+      uint32_t ssrc) const override;
+
+  // Choose one of the available SSRCs (or default if none) as the current
+  // receiver report SSRC.
+  void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
+
+  // E2E Encrypted Video Frame API
+  // Set a frame decryptor to a particular ssrc that will intercept all
+  // incoming video frames and attempt to decrypt them before forwarding the
+  // result.
+  void SetFrameDecryptor(uint32_t ssrc,
+                         rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+  void SetRecordableEncodedFrameCallback(
+      uint32_t ssrc,
+      std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
+      override;
+  void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
+  void RequestRecvKeyFrame(uint32_t ssrc) override;
+  void SetDepacketizerToDecoderFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
+
+  void SetReceiverFeedbackParameters(bool lntf_enabled,
+                                     bool nack_enabled,
+                                     webrtc::RtcpMode rtcp_mode,
+                                     absl::optional<int> rtx_time) override;
+
+ private:
+  class WebRtcVideoReceiveStream;
+  struct ChangedReceiverParameters {
+    // These optionals are unset if not changed.
+    absl::optional<std::vector<VideoCodecSettings>> codec_settings;
+    absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
+    // Keep track of the FlexFEC payload type separately from `codec_settings`.
+    // This allows us to recreate the FlexfecReceiveStream separately from the
+    // VideoReceiveStreamInterface when the FlexFEC payload type is changed.
+    absl::optional<int> flexfec_payload_type;
+  };
+
+  // Finds VideoReceiveStreamInterface corresponding to ssrc. Aware of
+  // unsignalled ssrc handling.
+  WebRtcVideoReceiveStream* FindReceiveStream(uint32_t ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  void ProcessReceivedPacket(webrtc::RtpPacketReceived packet)
+      RTC_RUN_ON(thread_checker_);
+
+  // Expected to be invoked once per packet that belongs to this channel that
+  // can not be demuxed.
+  // Returns true if a new default stream has been created.
+  bool MaybeCreateDefaultReceiveStream(
+      const webrtc::RtpPacketReceived& parsed_packet)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  void ReCreateDefaultReceiveStream(uint32_t ssrc,
+                                    absl::optional<uint32_t> rtx_ssrc);
+  // Add a receive stream. Used for testing.
+  bool AddRecvStream(const StreamParams& sp, bool default_stream);
+
+  void ConfigureReceiverRtp(
+      webrtc::VideoReceiveStreamInterface::Config* config,
+      webrtc::FlexfecReceiveStream::Config* flexfec_config,
+      const StreamParams& sp) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  // Called when the local ssrc changes. Sets `rtcp_receiver_report_ssrc_` and
+  // updates the receive streams.
+  void SetReceiverReportSsrc(uint32_t ssrc) RTC_RUN_ON(&thread_checker_);
+
+  // Wrapper for the receiver part, contains configs etc. that are needed to
+  // reconstruct the underlying VideoReceiveStreamInterface.
+  class WebRtcVideoReceiveStream
+      : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+   public:
+    WebRtcVideoReceiveStream(
+        webrtc::Call* call,
+        const StreamParams& sp,
+        webrtc::VideoReceiveStreamInterface::Config config,
+        bool default_stream,
+        const std::vector<VideoCodecSettings>& recv_codecs,
+        const webrtc::FlexfecReceiveStream::Config& flexfec_config);
+    ~WebRtcVideoReceiveStream();
+
+    webrtc::VideoReceiveStreamInterface& stream();
+    // Return value may be nullptr.
+    webrtc::FlexfecReceiveStream* flexfec_stream();
+
+    const std::vector<uint32_t>& GetSsrcs() const;
+
+    std::vector<webrtc::RtpSource> GetSources();
+
+    // Does not return codecs, nor header extensions,  they are filled by the
+    // owning WebRtcVideoChannel.
+    webrtc::RtpParameters GetRtpParameters() const;
+
+    // TODO(deadbeef): Move these feedback parameters into the recv parameters.
+    void SetFeedbackParameters(bool lntf_enabled,
+                               bool nack_enabled,
+                               webrtc::RtcpMode rtcp_mode,
+                               absl::optional<int> rtx_time);
+    void SetReceiverParameters(const ChangedReceiverParameters& recv_params);
+
+    void OnFrame(const webrtc::VideoFrame& frame) override;
+    bool IsDefaultStream() const;
+
+    void SetFrameDecryptor(
+        rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor);
+
+    bool SetBaseMinimumPlayoutDelayMs(int delay_ms);
+
+    int GetBaseMinimumPlayoutDelayMs() const;
+
+    void SetSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
+
+    VideoReceiverInfo GetVideoReceiverInfo(bool log_stats);
+
+    void SetRecordableEncodedFrameCallback(
+        std::function<void(const webrtc::RecordableEncodedFrame&)> callback);
+    void ClearRecordableEncodedFrameCallback();
+    void GenerateKeyFrame();
+
+    void SetDepacketizerToDecoderFrameTransformer(
+        rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+            frame_transformer);
+
+    void SetLocalSsrc(uint32_t local_ssrc);
+    void UpdateRtxSsrc(uint32_t ssrc);
+    void StartReceiveStream();
+    void StopReceiveStream();
+
+   private:
+    // Attempts to reconfigure an already existing `flexfec_stream_`, create
+    // one if the configuration is now complete or remove a flexfec stream
+    // when disabled.
+    void SetFlexFecPayload(int payload_type);
+
+    void RecreateReceiveStream();
+    void CreateReceiveStream();
+
+    // Applies a new receive codecs configration to `config_`. Returns true
+    // if the internal stream needs to be reconstructed, or false if no changes
+    // were applied.
+    bool ReconfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
+
+    webrtc::Call* const call_;
+    const StreamParams stream_params_;
+
+    // Both `stream_` and `flexfec_stream_` are managed by `this`. They are
+    // destroyed by calling call_->DestroyVideoReceiveStream and
+    // call_->DestroyFlexfecReceiveStream, respectively.
+    webrtc::VideoReceiveStreamInterface* stream_;
+    const bool default_stream_;
+    webrtc::VideoReceiveStreamInterface::Config config_;
+    webrtc::FlexfecReceiveStream::Config flexfec_config_;
+    webrtc::FlexfecReceiveStream* flexfec_stream_;
+
+    webrtc::Mutex sink_lock_;
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink_
+        RTC_GUARDED_BY(sink_lock_);
+    int64_t first_frame_timestamp_ RTC_GUARDED_BY(sink_lock_);
+    // Start NTP time is estimated as current remote NTP time (estimated from
+    // RTCP) minus the elapsed time, as soon as remote NTP time is available.
+    int64_t estimated_remote_start_ntp_time_ms_ RTC_GUARDED_BY(sink_lock_);
+
+    RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
+    bool receiving_ RTC_GUARDED_BY(&thread_checker_);
+  };
+  bool GetChangedReceiverParameters(const VideoReceiverParameters& params,
+                                    ChangedReceiverParameters* changed_params)
+      const RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
+      RTC_GUARDED_BY(thread_checker_);
+  void FillReceiverStats(VideoMediaReceiveInfo* info, bool log_stats)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  void FillReceiveCodecStats(VideoMediaReceiveInfo* video_media_info)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  StreamParams unsignaled_stream_params() {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return unsignaled_stream_params_;
+  }
+  // Variables.
+  webrtc::TaskQueueBase* const worker_thread_;
+  webrtc::ScopedTaskSafety task_safety_;
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_thread_checker_{
+      webrtc::SequenceChecker::kDetached};
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker thread_checker_;
+
+  uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
+  bool receiving_ RTC_GUARDED_BY(&thread_checker_);
+  webrtc::Call* const call_;
+
+  rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_
+      RTC_GUARDED_BY(thread_checker_);
+
+  // Delay for unsignaled streams, which may be set before the stream exists.
+  int default_recv_base_minimum_delay_ms_ RTC_GUARDED_BY(thread_checker_) = 0;
+
+  const MediaConfig::Video video_config_ RTC_GUARDED_BY(thread_checker_);
+
+  // When the channel and demuxer get reconfigured, there is a window of time
+  // where we have to be prepared for packets arriving based on the old demuxer
+  // criteria because the streams live on the worker thread and the demuxer
+  // lives on the network thread. Because packets are posted from the network
+  // thread to the worker thread, they can still be in-flight when streams are
+  // reconfgured. This can happen when `demuxer_criteria_id_` and
+  // `demuxer_criteria_completed_id_` don't match. During this time, we do not
+  // want to create unsignalled receive streams and should instead drop the
+  // packets. E.g:
+  // * If RemoveRecvStream(old_ssrc) was recently called, there may be packets
+  //   in-flight for that ssrc. This happens when a receiver becomes inactive.
+  // * If we go from one to many m= sections, the demuxer may change from
+  //   forwarding all packets to only forwarding the configured ssrcs, so there
+  //   is a risk of receiving ssrcs for other, recently added m= sections.
+  uint32_t demuxer_criteria_id_ RTC_GUARDED_BY(thread_checker_) = 0;
+  uint32_t demuxer_criteria_completed_id_ RTC_GUARDED_BY(thread_checker_) = 0;
+  absl::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
+      RTC_GUARDED_BY(thread_checker_);
+  std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
+  std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(thread_checker_);
+
+  absl::optional<VideoCodecSettings> send_codec_
+      RTC_GUARDED_BY(thread_checker_);
+  std::vector<VideoCodecSettings> negotiated_codecs_
+      RTC_GUARDED_BY(thread_checker_);
+
+  std::vector<webrtc::RtpExtension> send_rtp_extensions_
+      RTC_GUARDED_BY(thread_checker_);
+
+  webrtc::VideoDecoderFactory* const decoder_factory_
+      RTC_GUARDED_BY(thread_checker_);
+  std::vector<VideoCodecSettings> recv_codecs_ RTC_GUARDED_BY(thread_checker_);
+  webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_
+      RTC_GUARDED_BY(thread_checker_);
+  std::vector<webrtc::RtpExtension> recv_rtp_extensions_
+      RTC_GUARDED_BY(thread_checker_);
+  // See reason for keeping track of the FlexFEC payload type separately in
+  // comment in WebRtcVideoChannel::ChangedReceiverParameters.
+  int recv_flexfec_payload_type_ RTC_GUARDED_BY(thread_checker_);
+  webrtc::BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
+  // TODO(deadbeef): Don't duplicate information between
+  // send_params/recv_params, rtp_extensions, options, etc.
+  VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
+  VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
+  VideoReceiverParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
+  int64_t last_receive_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
+  const bool discard_unknown_ssrc_packets_ RTC_GUARDED_BY(thread_checker_);
+  // This is a stream param that comes from the remote description, but wasn't
+  // signaled with any a=ssrc lines. It holds information that was signaled
+  // before the unsignaled receive stream is created when the first packet is
+  // received.
+  StreamParams unsignaled_stream_params_ RTC_GUARDED_BY(thread_checker_);
+  // Per peer connection crypto options that last for the lifetime of the peer
+  // connection.
+  const webrtc::CryptoOptions crypto_options_ RTC_GUARDED_BY(thread_checker_);
+
+  // Optional frame transformer set on unsignaled streams.
+  rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+      unsignaled_frame_transformer_ RTC_GUARDED_BY(thread_checker_);
+
+  // RTP parameters that need to be set when creating a video receive stream.
+  // Only used in Receiver mode - in Both mode, it reads those things from the
+  // codec.
+  webrtc::VideoReceiveStreamInterface::Config::Rtp rtp_config_;
+
+  // Callback invoked whenever the send codec changes.
+  // TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
+  absl::AnyInvocable<void()> send_codec_changed_callback_;
+  // Callback invoked whenever the list of SSRCs changes.
+  absl::AnyInvocable<void(const std::set<uint32_t>&)>
+      ssrc_list_changed_callback_;
+
+  const int receive_buffer_size_;
+};
+
+// Keeping the old name "WebRtcVideoChannel" around because some external
+// customers are using cricket::WebRtcVideoChannel::AdaptReason
+// TODO(bugs.webrtc.org/15216): Move this enum to an interface class and
+// delete this workaround.
+class WebRtcVideoChannel : public WebRtcVideoSendChannel {
+ public:
+  // Make all the values of AdaptReason available as
+  // WebRtcVideoChannel::ADAPT_xxx.
+  using WebRtcVideoSendChannel::AdaptReason;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/engine/webrtc_voice_engine.h

@@ -0,0 +1,516 @@
+/*
+ *  Copyright (c) 2004 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 MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
+#define MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio/audio_frame_processor.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_options.h"
+#include "api/call/audio_sink.h"
+#include "api/call/transport.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/rtc_error.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_sender_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "call/audio_send_stream.h"
+#include "call/audio_state.h"
+#include "call/call.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_config.h"
+#include "media/base/media_engine.h"
+#include "media/base/rtp_utils.h"
+#include "media/base/stream_params.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+class AudioFrameProcessor;
+}
+
+namespace cricket {
+
+class AudioSource;
+
+// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
+// It uses the WebRtc VoiceEngine library for audio handling.
+class WebRtcVoiceEngine final : public VoiceEngineInterface {
+  friend class WebRtcVoiceSendChannel;
+  friend class WebRtcVoiceReceiveChannel;
+
+ public:
+  WebRtcVoiceEngine(
+      webrtc::TaskQueueFactory* task_queue_factory,
+      webrtc::AudioDeviceModule* adm,
+      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
+      rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+      rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
+      std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor,
+      const webrtc::FieldTrialsView& trials);
+
+  WebRtcVoiceEngine() = delete;
+  WebRtcVoiceEngine(const WebRtcVoiceEngine&) = delete;
+  WebRtcVoiceEngine& operator=(const WebRtcVoiceEngine&) = delete;
+
+  ~WebRtcVoiceEngine() override;
+
+  // Does initialization that needs to occur on the worker thread.
+  void Init() override;
+  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
+
+  std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) override;
+
+  std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const AudioOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::AudioCodecPairId codec_pair_id) override;
+
+  const std::vector<AudioCodec>& send_codecs() const override;
+  const std::vector<AudioCodec>& recv_codecs() const override;
+  std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
+      const override;
+
+  // Starts AEC dump using an existing file. A maximum file size in bytes can be
+  // specified. When the maximum file size is reached, logging is stopped and
+  // the file is closed. If max_size_bytes is set to <= 0, no limit will be
+  // used.
+  bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
+
+  // Stops AEC dump.
+  void StopAecDump() override;
+
+  absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
+      override;
+
+ private:
+  // Every option that is "set" will be applied. Every option not "set" will be
+  // ignored. This allows us to selectively turn on and off different options
+  // easily at any time.
+  void ApplyOptions(const AudioOptions& options);
+
+  webrtc::TaskQueueFactory* const task_queue_factory_;
+  std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>
+      low_priority_worker_queue_;
+
+  webrtc::AudioDeviceModule* adm();
+  webrtc::AudioProcessing* apm() const;
+  webrtc::AudioState* audio_state();
+
+  std::vector<AudioCodec> CollectCodecs(
+      const std::vector<webrtc::AudioCodecSpec>& specs) const;
+
+  webrtc::SequenceChecker signal_thread_checker_{
+      webrtc::SequenceChecker::kDetached};
+  webrtc::SequenceChecker worker_thread_checker_{
+      webrtc::SequenceChecker::kDetached};
+
+  // The audio device module.
+  rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
+  rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
+  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
+  // The audio processing module.
+  rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
+  // Asynchronous audio processing.
+  std::unique_ptr<webrtc::AudioFrameProcessor> audio_frame_processor_;
+  // The primary instance of WebRtc VoiceEngine.
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  std::vector<AudioCodec> send_codecs_;
+  std::vector<AudioCodec> recv_codecs_;
+  bool is_dumping_aec_ = false;
+  bool initialized_ = false;
+
+  // Jitter buffer settings for new streams.
+  size_t audio_jitter_buffer_max_packets_ = 200;
+  bool audio_jitter_buffer_fast_accelerate_ = false;
+  int audio_jitter_buffer_min_delay_ms_ = 0;
+
+  const bool minimized_remsampling_on_mobile_trial_enabled_;
+};
+
+class WebRtcVoiceSendChannel final : public MediaChannelUtil,
+                                     public VoiceMediaSendChannelInterface {
+ public:
+  WebRtcVoiceSendChannel(WebRtcVoiceEngine* engine,
+                         const MediaConfig& config,
+                         const AudioOptions& options,
+                         const webrtc::CryptoOptions& crypto_options,
+                         webrtc::Call* call,
+                         webrtc::AudioCodecPairId codec_pair_id);
+
+  WebRtcVoiceSendChannel() = delete;
+  WebRtcVoiceSendChannel(const WebRtcVoiceSendChannel&) = delete;
+  WebRtcVoiceSendChannel& operator=(const WebRtcVoiceSendChannel&) = delete;
+
+  ~WebRtcVoiceSendChannel() override;
+
+  MediaType media_type() const override { return MEDIA_TYPE_AUDIO; }
+  VideoMediaSendChannelInterface* AsVideoSendChannel() override {
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+  VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
+
+  absl::optional<Codec> GetSendCodec() const override;
+
+  // Functions imported from MediaChannelUtil
+  void SetInterface(MediaChannelNetworkInterface* iface) override {
+    MediaChannelUtil::SetInterface(iface);
+  }
+
+  bool HasNetworkInterface() const override {
+    return MediaChannelUtil::HasNetworkInterface();
+  }
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
+    MediaChannelUtil::SetExtmapAllowMixed(extmap_allow_mixed);
+  }
+  bool ExtmapAllowMixed() const override {
+    return MediaChannelUtil::ExtmapAllowMixed();
+  }
+
+  const AudioOptions& options() const { return options_; }
+
+  bool SetSenderParameters(const AudioSenderParameter& params) override;
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  webrtc::RTCError SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters,
+      webrtc::SetParametersCallback callback) override;
+
+  void SetSend(bool send) override;
+  bool SetAudioSend(uint32_t ssrc,
+                    bool enable,
+                    const AudioOptions* options,
+                    AudioSource* source) override;
+  bool AddSendStream(const StreamParams& sp) override;
+  bool RemoveSendStream(uint32_t ssrc) override;
+
+  void SetSsrcListChangedCallback(
+      absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override;
+
+  // E2EE Frame API
+  // Set a frame encryptor to a particular ssrc that will intercept all
+  // outgoing audio payloads frames and attempt to encrypt them and forward the
+  // result to the packetizer.
+  void SetFrameEncryptor(uint32_t ssrc,
+                         rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
+                             frame_encryptor) override;
+
+  bool CanInsertDtmf() override;
+  bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
+
+  void OnPacketSent(const rtc::SentPacket& sent_packet) override;
+  void OnNetworkRouteChanged(absl::string_view transport_name,
+                             const rtc::NetworkRoute& network_route) override;
+  void OnReadyToSend(bool ready) override;
+  bool GetStats(VoiceMediaSendInfo* info) override;
+
+  // Sets a frame transformer between encoder and packetizer, to transform
+  // encoded frames before sending them out the network.
+  void SetEncoderToPacketizerFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+  bool SenderNackEnabled() const override {
+    if (!send_codec_spec_) {
+      return false;
+    }
+    return send_codec_spec_->nack_enabled;
+  }
+  bool SenderNonSenderRttEnabled() const override {
+    if (!send_codec_spec_) {
+      return false;
+    }
+    return send_codec_spec_->enable_non_sender_rtt;
+  }
+  bool SendCodecHasNack() const override { return SenderNackEnabled(); }
+
+  void SetSendCodecChangedCallback(
+      absl::AnyInvocable<void()> callback) override {
+    send_codec_changed_callback_ = std::move(callback);
+  }
+
+ private:
+  bool SetOptions(const AudioOptions& options);
+  bool SetSendCodecs(const std::vector<Codec>& codecs,
+                     absl::optional<Codec> preferred_codec);
+  bool SetLocalSource(uint32_t ssrc, AudioSource* source);
+  bool MuteStream(uint32_t ssrc, bool mute);
+
+  WebRtcVoiceEngine* engine() { return engine_; }
+  bool SetMaxSendBitrate(int bps);
+  void SetupRecording();
+
+  webrtc::TaskQueueBase* const worker_thread_;
+  webrtc::ScopedTaskSafety task_safety_;
+  webrtc::SequenceChecker network_thread_checker_{
+      webrtc::SequenceChecker::kDetached};
+
+  WebRtcVoiceEngine* const engine_ = nullptr;
+  std::vector<AudioCodec> send_codecs_;
+
+  int max_send_bitrate_bps_ = 0;
+  AudioOptions options_;
+  absl::optional<int> dtmf_payload_type_;
+  int dtmf_payload_freq_ = -1;
+  bool enable_non_sender_rtt_ = false;
+  bool send_ = false;
+  webrtc::Call* const call_ = nullptr;
+
+  const MediaConfig::Audio audio_config_;
+
+  class WebRtcAudioSendStream;
+
+  std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
+  std::vector<webrtc::RtpExtension> send_rtp_extensions_;
+  std::string mid_;
+
+  absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+      send_codec_spec_;
+
+  // TODO(kwiberg): Per-SSRC codec pair IDs?
+  const webrtc::AudioCodecPairId codec_pair_id_;
+
+  // Per peer connection crypto options that last for the lifetime of the peer
+  // connection.
+  const webrtc::CryptoOptions crypto_options_;
+  rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+      unsignaled_frame_transformer_;
+
+  void FillSendCodecStats(VoiceMediaSendInfo* voice_media_info);
+
+  // Callback invoked whenever the send codec changes.
+  // TODO(bugs.webrtc.org/13931): Remove again when coupling isn't needed.
+  absl::AnyInvocable<void()> send_codec_changed_callback_;
+  // Callback invoked whenever the list of SSRCs changes.
+  absl::AnyInvocable<void(const std::set<uint32_t>&)>
+      ssrc_list_changed_callback_;
+};
+
+class WebRtcVoiceReceiveChannel final
+    : public MediaChannelUtil,
+      public VoiceMediaReceiveChannelInterface {
+ public:
+  WebRtcVoiceReceiveChannel(WebRtcVoiceEngine* engine,
+                            const MediaConfig& config,
+                            const AudioOptions& options,
+                            const webrtc::CryptoOptions& crypto_options,
+                            webrtc::Call* call,
+                            webrtc::AudioCodecPairId codec_pair_id);
+
+  WebRtcVoiceReceiveChannel() = delete;
+  WebRtcVoiceReceiveChannel(const WebRtcVoiceReceiveChannel&) = delete;
+  WebRtcVoiceReceiveChannel& operator=(const WebRtcVoiceReceiveChannel&) =
+      delete;
+
+  ~WebRtcVoiceReceiveChannel() override;
+
+  MediaType media_type() const override { return MEDIA_TYPE_AUDIO; }
+
+  VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+  VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
+    return this;
+  }
+
+  const AudioOptions& options() const { return options_; }
+
+  void SetInterface(MediaChannelNetworkInterface* iface) override {
+    MediaChannelUtil::SetInterface(iface);
+  }
+  bool SetReceiverParameters(const AudioReceiverParameters& params) override;
+  webrtc::RtpParameters GetRtpReceiverParameters(uint32_t ssrc) const override;
+  webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
+
+  void SetPlayout(bool playout) override;
+  bool AddRecvStream(const StreamParams& sp) override;
+  bool RemoveRecvStream(uint32_t ssrc) override;
+  void ResetUnsignaledRecvStream() override;
+  absl::optional<uint32_t> GetUnsignaledSsrc() const override;
+
+  void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
+
+  void OnDemuxerCriteriaUpdatePending() override;
+  void OnDemuxerCriteriaUpdateComplete() override;
+
+  // E2EE Frame API
+  // Set a frame decryptor to a particular ssrc that will intercept all
+  // incoming audio payloads and attempt to decrypt them before forwarding the
+  // result.
+  void SetFrameDecryptor(uint32_t ssrc,
+                         rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+
+  bool SetOutputVolume(uint32_t ssrc, double volume) override;
+  // Applies the new volume to current and future unsignaled streams.
+  bool SetDefaultOutputVolume(double volume) override;
+
+  bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
+  absl::optional<int> GetBaseMinimumPlayoutDelayMs(
+      uint32_t ssrc) const override;
+
+  void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override;
+  bool GetStats(VoiceMediaReceiveInfo* info,
+                bool get_and_clear_legacy_stats) override;
+
+  // Set the audio sink for an existing stream.
+  void SetRawAudioSink(
+      uint32_t ssrc,
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+  // Will set the audio sink on the latest unsignaled stream, future or
+  // current. Only one stream at a time will use the sink.
+  void SetDefaultRawAudioSink(
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+
+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
+
+  void SetDepacketizerToDecoderFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+  void SetReceiveNackEnabled(bool enabled) override;
+  void SetReceiveNonSenderRttEnabled(bool enabled) override;
+
+ private:
+  bool SetOptions(const AudioOptions& options);
+  bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
+  bool SetLocalSource(uint32_t ssrc, AudioSource* source);
+  bool MuteStream(uint32_t ssrc, bool mute);
+
+  WebRtcVoiceEngine* engine() { return engine_; }
+  void SetupRecording();
+
+  // Expected to be invoked once per packet that belongs to this channel that
+  // can not be demuxed. Returns true if a default receive stream has been
+  // created.
+  bool MaybeCreateDefaultReceiveStream(const webrtc::RtpPacketReceived& packet);
+  // Check if 'ssrc' is an unsignaled stream, and if so mark it as not being
+  // unsignaled anymore (i.e. it is now removed, or signaled), and return true.
+  bool MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc);
+
+  webrtc::TaskQueueBase* const worker_thread_;
+  webrtc::ScopedTaskSafety task_safety_;
+  webrtc::SequenceChecker network_thread_checker_{
+      webrtc::SequenceChecker::kDetached};
+
+  WebRtcVoiceEngine* const engine_ = nullptr;
+
+  // TODO(kwiberg): decoder_map_ and recv_codecs_ store the exact same
+  // information, in slightly different formats. Eliminate recv_codecs_.
+  std::map<int, webrtc::SdpAudioFormat> decoder_map_;
+  std::vector<AudioCodec> recv_codecs_;
+
+  AudioOptions options_;
+  bool recv_nack_enabled_ = false;
+  bool enable_non_sender_rtt_ = false;
+  bool playout_ = false;
+  webrtc::Call* const call_ = nullptr;
+
+  const MediaConfig::Audio audio_config_;
+
+  // Queue of unsignaled SSRCs; oldest at the beginning.
+  std::vector<uint32_t> unsignaled_recv_ssrcs_;
+
+  // This is a stream param that comes from the remote description, but wasn't
+  // signaled with any a=ssrc lines. It holds the information that was signaled
+  // before the unsignaled receive stream is created when the first packet is
+  // received.
+  StreamParams unsignaled_stream_params_;
+
+  // Volume for unsignaled streams, which may be set before the stream exists.
+  double default_recv_volume_ = 1.0;
+
+  // Delay for unsignaled streams, which may be set before the stream exists.
+  int default_recv_base_minimum_delay_ms_ = 0;
+
+  // Sink for latest unsignaled stream - may be set before the stream exists.
+  std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
+  // Default SSRC to use for RTCP receiver reports in case of no signaled
+  // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
+  // and https://code.google.com/p/chromium/issues/detail?id=547661
+  uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
+
+  std::string mid_;
+
+  class WebRtcAudioReceiveStream;
+
+  std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
+  std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
+  webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_;
+
+  absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+      send_codec_spec_;
+
+  // TODO(kwiberg): Per-SSRC codec pair IDs?
+  const webrtc::AudioCodecPairId codec_pair_id_;
+
+  // Per peer connection crypto options that last for the lifetime of the peer
+  // connection.
+  const webrtc::CryptoOptions crypto_options_;
+  // Unsignaled streams have an option to have a frame decryptor set on them.
+  rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+      unsignaled_frame_decryptor_;
+  rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+      unsignaled_frame_transformer_;
+
+  void FillReceiveCodecStats(VoiceMediaReceiveInfo* voice_media_info);
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_WEBRTC_VOICE_ENGINE_H_

TMessagesProj/jni/voip/webrtc/media/sctp/OWNERS

@@ -0,0 +1,3 @@
+boivie@webrtc.org
+deadbeef@webrtc.org
+orphis@webrtc.org

TMessagesProj/jni/voip/webrtc/media/sctp/dcsctp_transport.cc

@@ -0,0 +1,668 @@
+/*
+ *  Copyright 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 "media/sctp/dcsctp_transport.h"
+
+#include <atomic>
+#include <cstdint>
+#include <limits>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "media/base/media_channel.h"
+#include "net/dcsctp/public/dcsctp_socket_factory.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/public/text_pcap_packet_observer.h"
+#include "net/dcsctp/public/types.h"
+#include "p2p/base/packet_transport_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/socket.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+using ::dcsctp::SendPacketStatus;
+
+// When there is packet loss for a long time, the SCTP retry timers will use
+// exponential backoff, which can grow to very long durations and when the
+// connection recovers, it may take a long time to reach the new backoff
+// duration. By limiting it to a reasonable limit, the time to recover reduces.
+constexpr dcsctp::DurationMs kMaxTimerBackoffDuration =
+    dcsctp::DurationMs(3000);
+
+enum class WebrtcPPID : dcsctp::PPID::UnderlyingType {
+  // https://www.rfc-editor.org/rfc/rfc8832.html#section-8.1
+  kDCEP = 50,
+  // https://www.rfc-editor.org/rfc/rfc8831.html#section-8
+  kString = 51,
+  kBinaryPartial = 52,  // Deprecated
+  kBinary = 53,
+  kStringPartial = 54,  // Deprecated
+  kStringEmpty = 56,
+  kBinaryEmpty = 57,
+};
+
+WebrtcPPID ToPPID(DataMessageType message_type, size_t size) {
+  switch (message_type) {
+    case DataMessageType::kControl:
+      return WebrtcPPID::kDCEP;
+    case DataMessageType::kText:
+      return size > 0 ? WebrtcPPID::kString : WebrtcPPID::kStringEmpty;
+    case DataMessageType::kBinary:
+      return size > 0 ? WebrtcPPID::kBinary : WebrtcPPID::kBinaryEmpty;
+  }
+}
+
+absl::optional<DataMessageType> ToDataMessageType(dcsctp::PPID ppid) {
+  switch (static_cast<WebrtcPPID>(ppid.value())) {
+    case WebrtcPPID::kDCEP:
+      return DataMessageType::kControl;
+    case WebrtcPPID::kString:
+    case WebrtcPPID::kStringPartial:
+    case WebrtcPPID::kStringEmpty:
+      return DataMessageType::kText;
+    case WebrtcPPID::kBinary:
+    case WebrtcPPID::kBinaryPartial:
+    case WebrtcPPID::kBinaryEmpty:
+      return DataMessageType::kBinary;
+  }
+  return absl::nullopt;
+}
+
+absl::optional<cricket::SctpErrorCauseCode> ToErrorCauseCode(
+    dcsctp::ErrorKind error) {
+  switch (error) {
+    case dcsctp::ErrorKind::kParseFailed:
+      return cricket::SctpErrorCauseCode::kUnrecognizedParameters;
+    case dcsctp::ErrorKind::kPeerReported:
+      return cricket::SctpErrorCauseCode::kUserInitiatedAbort;
+    case dcsctp::ErrorKind::kWrongSequence:
+    case dcsctp::ErrorKind::kProtocolViolation:
+      return cricket::SctpErrorCauseCode::kProtocolViolation;
+    case dcsctp::ErrorKind::kResourceExhaustion:
+      return cricket::SctpErrorCauseCode::kOutOfResource;
+    case dcsctp::ErrorKind::kTooManyRetries:
+    case dcsctp::ErrorKind::kUnsupportedOperation:
+    case dcsctp::ErrorKind::kNoError:
+    case dcsctp::ErrorKind::kNotConnected:
+      // No SCTP error cause code matches those
+      break;
+  }
+  return absl::nullopt;
+}
+
+bool IsEmptyPPID(dcsctp::PPID ppid) {
+  WebrtcPPID webrtc_ppid = static_cast<WebrtcPPID>(ppid.value());
+  return webrtc_ppid == WebrtcPPID::kStringEmpty ||
+         webrtc_ppid == WebrtcPPID::kBinaryEmpty;
+}
+}  // namespace
+
+DcSctpTransport::DcSctpTransport(rtc::Thread* network_thread,
+                                 rtc::PacketTransportInternal* transport,
+                                 Clock* clock)
+    : DcSctpTransport(network_thread,
+                      transport,
+                      clock,
+                      std::make_unique<dcsctp::DcSctpSocketFactory>()) {}
+
+DcSctpTransport::DcSctpTransport(
+    rtc::Thread* network_thread,
+    rtc::PacketTransportInternal* transport,
+    Clock* clock,
+    std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory)
+    : network_thread_(network_thread),
+      transport_(transport),
+      clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
+      socket_factory_(std::move(socket_factory)),
+      task_queue_timeout_factory_(
+          *network_thread,
+          [this]() { return TimeMillis(); },
+          [this](dcsctp::TimeoutID timeout_id) {
+            socket_->HandleTimeout(timeout_id);
+          }) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  static std::atomic<int> instance_count = 0;
+  rtc::StringBuilder sb;
+  sb << debug_name_ << instance_count++;
+  debug_name_ = sb.Release();
+  ConnectTransportSignals();
+}
+
+DcSctpTransport::~DcSctpTransport() {
+  if (socket_) {
+    socket_->Close();
+  }
+}
+
+void DcSctpTransport::SetOnConnectedCallback(std::function<void()> callback) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  on_connected_callback_ = std::move(callback);
+}
+
+void DcSctpTransport::SetDataChannelSink(DataChannelSink* sink) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  data_channel_sink_ = sink;
+  if (data_channel_sink_ && ready_to_send_data_) {
+    data_channel_sink_->OnReadyToSend();
+  }
+}
+
+void DcSctpTransport::SetDtlsTransport(
+    rtc::PacketTransportInternal* transport) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  DisconnectTransportSignals();
+  transport_ = transport;
+  ConnectTransportSignals();
+  MaybeConnectSocket();
+}
+
+bool DcSctpTransport::Start(int local_sctp_port,
+                            int remote_sctp_port,
+                            int max_message_size) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DCHECK(max_message_size > 0);
+  RTC_DLOG(LS_INFO) << debug_name_ << "->Start(local=" << local_sctp_port
+                    << ", remote=" << remote_sctp_port
+                    << ", max_message_size=" << max_message_size << ")";
+
+  if (!socket_) {
+    dcsctp::DcSctpOptions options;
+    options.local_port = local_sctp_port;
+    options.remote_port = remote_sctp_port;
+    options.max_message_size = max_message_size;
+    options.max_timer_backoff_duration = kMaxTimerBackoffDuration;
+    // Don't close the connection automatically on too many retransmissions.
+    options.max_retransmissions = absl::nullopt;
+    options.max_init_retransmits = absl::nullopt;
+
+    std::unique_ptr<dcsctp::PacketObserver> packet_observer;
+    if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE)) {
+      packet_observer =
+          std::make_unique<dcsctp::TextPcapPacketObserver>(debug_name_);
+    }
+
+    socket_ = socket_factory_->Create(debug_name_, *this,
+                                      std::move(packet_observer), options);
+  } else {
+    if (local_sctp_port != socket_->options().local_port ||
+        remote_sctp_port != socket_->options().remote_port) {
+      RTC_LOG(LS_ERROR)
+          << debug_name_ << "->Start(local=" << local_sctp_port
+          << ", remote=" << remote_sctp_port
+          << "): Can't change ports on already started transport.";
+      return false;
+    }
+    socket_->SetMaxMessageSize(max_message_size);
+  }
+
+  MaybeConnectSocket();
+
+  return true;
+}
+
+bool DcSctpTransport::OpenStream(int sid) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_INFO) << debug_name_ << "->OpenStream(" << sid << ").";
+
+  StreamState stream_state;
+  stream_states_.insert_or_assign(dcsctp::StreamID(static_cast<uint16_t>(sid)),
+                                  stream_state);
+  return true;
+}
+
+bool DcSctpTransport::ResetStream(int sid) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_INFO) << debug_name_ << "->ResetStream(" << sid << ").";
+  if (!socket_) {
+    RTC_LOG(LS_ERROR) << debug_name_ << "->ResetStream(sid=" << sid
+                      << "): Transport is not started.";
+    return false;
+  }
+
+  dcsctp::StreamID streams[1] = {dcsctp::StreamID(static_cast<uint16_t>(sid))};
+
+  auto it = stream_states_.find(streams[0]);
+  if (it == stream_states_.end()) {
+    RTC_LOG(LS_ERROR) << debug_name_ << "->ResetStream(sid=" << sid
+                      << "): Stream is not open.";
+    return false;
+  }
+
+  StreamState& stream_state = it->second;
+  if (stream_state.closure_initiated || stream_state.incoming_reset_done ||
+      stream_state.outgoing_reset_done) {
+    // The closing procedure was already initiated by the remote, don't do
+    // anything.
+    return false;
+  }
+  stream_state.closure_initiated = true;
+  socket_->ResetStreams(streams);
+  return true;
+}
+
+RTCError DcSctpTransport::SendData(int sid,
+                                   const SendDataParams& params,
+                                   const rtc::CopyOnWriteBuffer& payload) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->SendData(sid=" << sid
+                       << ", type=" << static_cast<int>(params.type)
+                       << ", length=" << payload.size() << ").";
+
+  if (!socket_) {
+    RTC_LOG(LS_ERROR) << debug_name_
+                      << "->SendData(...): Transport is not started.";
+    return RTCError(RTCErrorType::INVALID_STATE);
+  }
+
+  // It is possible for a message to be sent from the signaling thread at the
+  // same time a data-channel is closing, but before the signaling thread is
+  // aware of it. So we need to keep track of currently active data channels and
+  // skip sending messages for the ones that are not open or closing.
+  // The sending errors are not impacting the data channel API contract as
+  // it is allowed to discard queued messages when the channel is closing.
+  auto stream_state =
+      stream_states_.find(dcsctp::StreamID(static_cast<uint16_t>(sid)));
+  if (stream_state == stream_states_.end()) {
+    RTC_LOG(LS_VERBOSE) << "Skipping message on non-open stream with sid: "
+                        << sid;
+    return RTCError(RTCErrorType::INVALID_STATE);
+  }
+
+  if (stream_state->second.closure_initiated ||
+      stream_state->second.incoming_reset_done ||
+      stream_state->second.outgoing_reset_done) {
+    RTC_LOG(LS_VERBOSE) << "Skipping message on closing stream with sid: "
+                        << sid;
+    return RTCError(RTCErrorType::INVALID_STATE);
+  }
+
+  auto max_message_size = socket_->options().max_message_size;
+  if (max_message_size > 0 && payload.size() > max_message_size) {
+    RTC_LOG(LS_WARNING) << debug_name_
+                        << "->SendData(...): "
+                           "Trying to send packet bigger "
+                           "than the max message size: "
+                        << payload.size() << " vs max of " << max_message_size;
+    return RTCError(RTCErrorType::INVALID_RANGE);
+  }
+
+  std::vector<uint8_t> message_payload(payload.cdata(),
+                                       payload.cdata() + payload.size());
+  if (message_payload.empty()) {
+    // https://www.rfc-editor.org/rfc/rfc8831.html#section-6.6
+    // SCTP does not support the sending of empty user messages. Therefore, if
+    // an empty message has to be sent, the appropriate PPID (WebRTC String
+    // Empty or WebRTC Binary Empty) is used, and the SCTP user message of one
+    // zero byte is sent.
+    message_payload.push_back('\0');
+  }
+
+  dcsctp::DcSctpMessage message(
+      dcsctp::StreamID(static_cast<uint16_t>(sid)),
+      dcsctp::PPID(static_cast<uint16_t>(ToPPID(params.type, payload.size()))),
+      std::move(message_payload));
+
+  dcsctp::SendOptions send_options;
+  send_options.unordered = dcsctp::IsUnordered(!params.ordered);
+  if (params.max_rtx_ms.has_value()) {
+    RTC_DCHECK(*params.max_rtx_ms >= 0 &&
+               *params.max_rtx_ms <= std::numeric_limits<uint16_t>::max());
+    send_options.lifetime = dcsctp::DurationMs(*params.max_rtx_ms);
+  }
+  if (params.max_rtx_count.has_value()) {
+    RTC_DCHECK(*params.max_rtx_count >= 0 &&
+               *params.max_rtx_count <= std::numeric_limits<uint16_t>::max());
+    send_options.max_retransmissions = *params.max_rtx_count;
+  }
+
+  dcsctp::SendStatus error = socket_->Send(std::move(message), send_options);
+  switch (error) {
+    case dcsctp::SendStatus::kSuccess:
+      return RTCError::OK();
+    case dcsctp::SendStatus::kErrorResourceExhaustion:
+      ready_to_send_data_ = false;
+      return RTCError(RTCErrorType::RESOURCE_EXHAUSTED);
+    default:
+      absl::string_view message = dcsctp::ToString(error);
+      RTC_LOG(LS_ERROR) << debug_name_
+                        << "->SendData(...): send() failed with error "
+                        << message << ".";
+      return RTCError(RTCErrorType::NETWORK_ERROR, message);
+  }
+}
+
+bool DcSctpTransport::ReadyToSendData() {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  return ready_to_send_data_;
+}
+
+int DcSctpTransport::max_message_size() const {
+  if (!socket_) {
+    RTC_LOG(LS_ERROR) << debug_name_
+                      << "->max_message_size(...): Transport is not started.";
+    return 0;
+  }
+  return socket_->options().max_message_size;
+}
+
+absl::optional<int> DcSctpTransport::max_outbound_streams() const {
+  if (!socket_)
+    return absl::nullopt;
+  return socket_->options().announced_maximum_outgoing_streams;
+}
+
+absl::optional<int> DcSctpTransport::max_inbound_streams() const {
+  if (!socket_)
+    return absl::nullopt;
+  return socket_->options().announced_maximum_incoming_streams;
+}
+
+void DcSctpTransport::set_debug_name_for_testing(const char* debug_name) {
+  debug_name_ = debug_name;
+}
+
+SendPacketStatus DcSctpTransport::SendPacketWithStatus(
+    rtc::ArrayView<const uint8_t> data) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DCHECK(socket_);
+
+  if (data.size() > (socket_->options().mtu)) {
+    RTC_LOG(LS_ERROR) << debug_name_
+                      << "->SendPacket(...): "
+                         "SCTP seems to have made a packet that is bigger "
+                         "than its official MTU: "
+                      << data.size() << " vs max of " << socket_->options().mtu;
+    return SendPacketStatus::kError;
+  }
+  TRACE_EVENT0("webrtc", "DcSctpTransport::SendPacket");
+
+  if (!transport_ || !transport_->writable())
+    return SendPacketStatus::kError;
+
+  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->SendPacket(length=" << data.size()
+                       << ")";
+
+  auto result =
+      transport_->SendPacket(reinterpret_cast<const char*>(data.data()),
+                             data.size(), rtc::PacketOptions(), 0);
+
+  if (result < 0) {
+    RTC_LOG(LS_WARNING) << debug_name_ << "->SendPacket(length=" << data.size()
+                        << ") failed with error: " << transport_->GetError()
+                        << ".";
+
+    if (rtc::IsBlockingError(transport_->GetError())) {
+      return SendPacketStatus::kTemporaryFailure;
+    }
+    return SendPacketStatus::kError;
+  }
+  return SendPacketStatus::kSuccess;
+}
+
+std::unique_ptr<dcsctp::Timeout> DcSctpTransport::CreateTimeout(
+    TaskQueueBase::DelayPrecision precision) {
+  return task_queue_timeout_factory_.CreateTimeout(precision);
+}
+
+dcsctp::TimeMs DcSctpTransport::TimeMillis() {
+  return dcsctp::TimeMs(clock_->TimeInMilliseconds());
+}
+
+uint32_t DcSctpTransport::GetRandomInt(uint32_t low, uint32_t high) {
+  return random_.Rand(low, high);
+}
+
+void DcSctpTransport::OnTotalBufferedAmountLow() {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (!ready_to_send_data_) {
+    ready_to_send_data_ = true;
+    if (data_channel_sink_) {
+      data_channel_sink_->OnReadyToSend();
+    }
+  }
+}
+
+void DcSctpTransport::OnMessageReceived(dcsctp::DcSctpMessage message) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->OnMessageReceived(sid="
+                       << message.stream_id().value()
+                       << ", ppid=" << message.ppid().value()
+                       << ", length=" << message.payload().size() << ").";
+  auto type = ToDataMessageType(message.ppid());
+  if (!type.has_value()) {
+    RTC_LOG(LS_VERBOSE) << debug_name_
+                        << "->OnMessageReceived(): Received an unknown PPID "
+                        << message.ppid().value()
+                        << " on an SCTP packet. Dropping.";
+    return;
+  }
+  receive_buffer_.Clear();
+  if (!IsEmptyPPID(message.ppid()))
+    receive_buffer_.AppendData(message.payload().data(),
+                               message.payload().size());
+
+  if (data_channel_sink_) {
+    data_channel_sink_->OnDataReceived(message.stream_id().value(), *type,
+                                       receive_buffer_);
+  }
+}
+
+void DcSctpTransport::OnError(dcsctp::ErrorKind error,
+                              absl::string_view message) {
+  if (error == dcsctp::ErrorKind::kResourceExhaustion) {
+    // Indicates that a message failed to be enqueued, because the send buffer
+    // is full, which is a very common (and wanted) state for high throughput
+    // sending/benchmarks.
+    RTC_LOG(LS_VERBOSE) << debug_name_
+                        << "->OnError(error=" << dcsctp::ToString(error)
+                        << ", message=" << message << ").";
+  } else {
+    RTC_LOG(LS_ERROR) << debug_name_
+                      << "->OnError(error=" << dcsctp::ToString(error)
+                      << ", message=" << message << ").";
+  }
+}
+
+void DcSctpTransport::OnAborted(dcsctp::ErrorKind error,
+                                absl::string_view message) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_LOG(LS_ERROR) << debug_name_
+                    << "->OnAborted(error=" << dcsctp::ToString(error)
+                    << ", message=" << message << ").";
+  ready_to_send_data_ = false;
+  RTCError rtc_error(RTCErrorType::OPERATION_ERROR_WITH_DATA,
+                     std::string(message));
+  rtc_error.set_error_detail(RTCErrorDetailType::SCTP_FAILURE);
+  auto code = ToErrorCauseCode(error);
+  if (code.has_value()) {
+    rtc_error.set_sctp_cause_code(static_cast<uint16_t>(*code));
+  }
+  if (data_channel_sink_) {
+    data_channel_sink_->OnTransportClosed(rtc_error);
+  }
+}
+
+void DcSctpTransport::OnConnected() {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_INFO) << debug_name_ << "->OnConnected().";
+  ready_to_send_data_ = true;
+  if (data_channel_sink_) {
+    data_channel_sink_->OnReadyToSend();
+  }
+  if (on_connected_callback_) {
+    on_connected_callback_();
+  }
+}
+
+void DcSctpTransport::OnClosed() {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_INFO) << debug_name_ << "->OnClosed().";
+  ready_to_send_data_ = false;
+}
+
+void DcSctpTransport::OnConnectionRestarted() {
+  RTC_DLOG(LS_INFO) << debug_name_ << "->OnConnectionRestarted().";
+}
+
+void DcSctpTransport::OnStreamsResetFailed(
+    rtc::ArrayView<const dcsctp::StreamID> outgoing_streams,
+    absl::string_view reason) {
+  // TODO(orphis): Need a test to check for correct behavior
+  for (auto& stream_id : outgoing_streams) {
+    RTC_LOG(LS_WARNING)
+        << debug_name_
+        << "->OnStreamsResetFailed(...): Outgoing stream reset failed"
+        << ", sid=" << stream_id.value() << ", reason: " << reason << ".";
+  }
+}
+
+void DcSctpTransport::OnStreamsResetPerformed(
+    rtc::ArrayView<const dcsctp::StreamID> outgoing_streams) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  for (auto& stream_id : outgoing_streams) {
+    RTC_LOG(LS_INFO) << debug_name_
+                     << "->OnStreamsResetPerformed(...): Outgoing stream reset"
+                     << ", sid=" << stream_id.value();
+
+    auto it = stream_states_.find(stream_id);
+    if (it == stream_states_.end()) {
+      // Ignoring an outgoing stream reset for a closed stream
+      return;
+    }
+
+    StreamState& stream_state = it->second;
+    stream_state.outgoing_reset_done = true;
+
+    if (stream_state.incoming_reset_done) {
+      //  When the close was not initiated locally, we can signal the end of the
+      //  data channel close procedure when the remote ACKs the reset.
+      if (data_channel_sink_) {
+        data_channel_sink_->OnChannelClosed(stream_id.value());
+      }
+      stream_states_.erase(stream_id);
+    }
+  }
+}
+
+void DcSctpTransport::OnIncomingStreamsReset(
+    rtc::ArrayView<const dcsctp::StreamID> incoming_streams) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  for (auto& stream_id : incoming_streams) {
+    RTC_LOG(LS_INFO) << debug_name_
+                     << "->OnIncomingStreamsReset(...): Incoming stream reset"
+                     << ", sid=" << stream_id.value();
+
+    auto it = stream_states_.find(stream_id);
+    if (it == stream_states_.end())
+      return;
+
+    StreamState& stream_state = it->second;
+    stream_state.incoming_reset_done = true;
+
+    if (!stream_state.closure_initiated) {
+      // When receiving an incoming stream reset event for a non local close
+      // procedure, the transport needs to reset the stream in the other
+      // direction too.
+      dcsctp::StreamID streams[1] = {stream_id};
+      socket_->ResetStreams(streams);
+      if (data_channel_sink_) {
+        data_channel_sink_->OnChannelClosing(stream_id.value());
+      }
+    }
+
+    if (stream_state.outgoing_reset_done) {
+      // The close procedure that was initiated locally is complete when we
+      // receive and incoming reset event.
+      if (data_channel_sink_) {
+        data_channel_sink_->OnChannelClosed(stream_id.value());
+      }
+      stream_states_.erase(stream_id);
+    }
+  }
+}
+
+void DcSctpTransport::ConnectTransportSignals() {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (!transport_) {
+    return;
+  }
+  transport_->SignalWritableState.connect(
+      this, &DcSctpTransport::OnTransportWritableState);
+  transport_->SignalReadPacket.connect(this,
+                                       &DcSctpTransport::OnTransportReadPacket);
+  transport_->SignalClosed.connect(this, &DcSctpTransport::OnTransportClosed);
+}
+
+void DcSctpTransport::DisconnectTransportSignals() {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (!transport_) {
+    return;
+  }
+  transport_->SignalWritableState.disconnect(this);
+  transport_->SignalReadPacket.disconnect(this);
+  transport_->SignalClosed.disconnect(this);
+}
+
+void DcSctpTransport::OnTransportWritableState(
+    rtc::PacketTransportInternal* transport) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DCHECK_EQ(transport_, transport);
+  RTC_DLOG(LS_VERBOSE) << debug_name_
+                       << "->OnTransportWritableState(), writable="
+                       << transport->writable();
+  MaybeConnectSocket();
+}
+
+void DcSctpTransport::OnTransportReadPacket(
+    rtc::PacketTransportInternal* transport,
+    const char* data,
+    size_t length,
+    const int64_t& /* packet_time_us */,
+    int flags) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (flags) {
+    // We are only interested in SCTP packets.
+    return;
+  }
+
+  RTC_DLOG(LS_VERBOSE) << debug_name_
+                       << "->OnTransportReadPacket(), length=" << length;
+  if (socket_) {
+    socket_->ReceivePacket(rtc::ArrayView<const uint8_t>(
+        reinterpret_cast<const uint8_t*>(data), length));
+  }
+}
+
+void DcSctpTransport::OnTransportClosed(
+    rtc::PacketTransportInternal* transport) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->OnTransportClosed().";
+  if (data_channel_sink_) {
+    data_channel_sink_->OnTransportClosed({});
+  }
+}
+
+void DcSctpTransport::MaybeConnectSocket() {
+  if (transport_ && transport_->writable() && socket_ &&
+      socket_->state() == dcsctp::SocketState::kClosed) {
+    socket_->Connect();
+  }
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/media/sctp/dcsctp_transport.h

@@ -0,0 +1,141 @@
+/*
+ *  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 MEDIA_SCTP_DCSCTP_TRANSPORT_H_
+#define MEDIA_SCTP_DCSCTP_TRANSPORT_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "media/sctp/sctp_transport_internal.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/dcsctp_socket_factory.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/timer/task_queue_timeout.h"
+#include "p2p/base/packet_transport_internal.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/random.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class DcSctpTransport : public cricket::SctpTransportInternal,
+                        public dcsctp::DcSctpSocketCallbacks,
+                        public sigslot::has_slots<> {
+ public:
+  DcSctpTransport(rtc::Thread* network_thread,
+                  rtc::PacketTransportInternal* transport,
+                  Clock* clock);
+  DcSctpTransport(rtc::Thread* network_thread,
+                  rtc::PacketTransportInternal* transport,
+                  Clock* clock,
+                  std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory);
+  ~DcSctpTransport() override;
+
+  // cricket::SctpTransportInternal
+  void SetOnConnectedCallback(std::function<void()> callback) override;
+  void SetDataChannelSink(DataChannelSink* sink) override;
+  void SetDtlsTransport(rtc::PacketTransportInternal* transport) override;
+  bool Start(int local_sctp_port,
+             int remote_sctp_port,
+             int max_message_size) override;
+  bool OpenStream(int sid) override;
+  bool ResetStream(int sid) override;
+  RTCError SendData(int sid,
+                    const SendDataParams& params,
+                    const rtc::CopyOnWriteBuffer& payload) override;
+  bool ReadyToSendData() override;
+  int max_message_size() const override;
+  absl::optional<int> max_outbound_streams() const override;
+  absl::optional<int> max_inbound_streams() const override;
+  void set_debug_name_for_testing(const char* debug_name) override;
+
+ private:
+  // dcsctp::DcSctpSocketCallbacks
+  dcsctp::SendPacketStatus SendPacketWithStatus(
+      rtc::ArrayView<const uint8_t> data) override;
+  std::unique_ptr<dcsctp::Timeout> CreateTimeout(
+      TaskQueueBase::DelayPrecision precision) override;
+  dcsctp::TimeMs TimeMillis() override;
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override;
+  void OnTotalBufferedAmountLow() override;
+  void OnMessageReceived(dcsctp::DcSctpMessage message) override;
+  void OnError(dcsctp::ErrorKind error, absl::string_view message) override;
+  void OnAborted(dcsctp::ErrorKind error, absl::string_view message) override;
+  void OnConnected() override;
+  void OnClosed() override;
+  void OnConnectionRestarted() override;
+  void OnStreamsResetFailed(
+      rtc::ArrayView<const dcsctp::StreamID> outgoing_streams,
+      absl::string_view reason) override;
+  void OnStreamsResetPerformed(
+      rtc::ArrayView<const dcsctp::StreamID> outgoing_streams) override;
+  void OnIncomingStreamsReset(
+      rtc::ArrayView<const dcsctp::StreamID> incoming_streams) override;
+
+  // Transport callbacks
+  void ConnectTransportSignals();
+  void DisconnectTransportSignals();
+  void OnTransportWritableState(rtc::PacketTransportInternal* transport);
+  void OnTransportReadPacket(rtc::PacketTransportInternal* transport,
+                             const char* data,
+                             size_t length,
+                             const int64_t& /* packet_time_us */,
+                             int flags);
+  void OnTransportClosed(rtc::PacketTransportInternal* transport);
+
+  void MaybeConnectSocket();
+
+  rtc::Thread* network_thread_;
+  rtc::PacketTransportInternal* transport_;
+  Clock* clock_;
+  Random random_;
+
+  std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory_;
+  dcsctp::TaskQueueTimeoutFactory task_queue_timeout_factory_;
+  std::unique_ptr<dcsctp::DcSctpSocketInterface> socket_;
+  std::string debug_name_ = "DcSctpTransport";
+  rtc::CopyOnWriteBuffer receive_buffer_;
+
+  // Used to keep track of the state of data channels.
+  // Reset needs to happen both ways before signaling the transport
+  // is closed.
+  struct StreamState {
+    // True when the local connection has initiated the reset.
+    // If a connection receives a reset for a stream that isn't
+    // already being reset locally, it needs to fire the signal
+    // SignalClosingProcedureStartedRemotely.
+    bool closure_initiated = false;
+    // True when the local connection received OnIncomingStreamsReset
+    bool incoming_reset_done = false;
+    // True when the local connection received OnStreamsResetPerformed
+    bool outgoing_reset_done = false;
+  };
+
+  // Map of all currently open or closing data channels
+  flat_map<dcsctp::StreamID, StreamState> stream_states_
+      RTC_GUARDED_BY(network_thread_);
+  bool ready_to_send_data_ RTC_GUARDED_BY(network_thread_) = false;
+  std::function<void()> on_connected_callback_ RTC_GUARDED_BY(network_thread_);
+  DataChannelSink* data_channel_sink_ RTC_GUARDED_BY(network_thread_) = nullptr;
+};
+
+}  // namespace webrtc
+
+#endif  // MEDIA_SCTP_DCSCTP_TRANSPORT_H_

TMessagesProj/jni/voip/webrtc/media/sctp/sctp_transport_factory.cc

@@ -0,0 +1,38 @@
+/*
+ *  Copyright 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 "media/sctp/sctp_transport_factory.h"
+
+#include "rtc_base/system/unused.h"
+
+#ifdef WEBRTC_HAVE_DCSCTP
+#include "media/sctp/dcsctp_transport.h"    // nogncheck
+#include "system_wrappers/include/clock.h"  // nogncheck
+#endif
+
+namespace cricket {
+
+SctpTransportFactory::SctpTransportFactory(rtc::Thread* network_thread)
+    : network_thread_(network_thread) {
+  RTC_UNUSED(network_thread_);
+}
+
+std::unique_ptr<SctpTransportInternal>
+SctpTransportFactory::CreateSctpTransport(
+    rtc::PacketTransportInternal* transport) {
+  std::unique_ptr<SctpTransportInternal> result;
+#ifdef WEBRTC_HAVE_DCSCTP
+  result = std::unique_ptr<SctpTransportInternal>(new webrtc::DcSctpTransport(
+      network_thread_, transport, webrtc::Clock::GetRealTimeClock()));
+#endif
+  return result;
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/media/sctp/sctp_transport_factory.h

@@ -0,0 +1,35 @@
+/*
+ *  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 MEDIA_SCTP_SCTP_TRANSPORT_FACTORY_H_
+#define MEDIA_SCTP_SCTP_TRANSPORT_FACTORY_H_
+
+#include <memory>
+
+#include "api/transport/sctp_transport_factory_interface.h"
+#include "media/sctp/sctp_transport_internal.h"
+#include "rtc_base/thread.h"
+
+namespace cricket {
+
+class SctpTransportFactory : public webrtc::SctpTransportFactoryInterface {
+ public:
+  explicit SctpTransportFactory(rtc::Thread* network_thread);
+
+  std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
+      rtc::PacketTransportInternal* transport) override;
+
+ private:
+  rtc::Thread* network_thread_;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_SCTP_SCTP_TRANSPORT_FACTORY_H__

TMessagesProj/jni/voip/webrtc/media/sctp/sctp_transport_internal.h

@@ -0,0 +1,150 @@
+/*
+ *  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 MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_
+#define MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_
+
+// TODO(deadbeef): Move SCTP code out of media/, and make it not depend on
+// anything in media/.
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/rtc_error.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "media/base/media_channel.h"
+#include "p2p/base/packet_transport_internal.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/thread.h"
+
+namespace cricket {
+
+// Constants that are important to API users
+// The size of the SCTP association send buffer. 256kB, the usrsctp default.
+constexpr int kSctpSendBufferSize = 256 * 1024;
+
+// The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs)
+// are 0-based, the highest usable SID is 1023.
+//
+// It's recommended to use the maximum of 65535 in:
+// https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2
+// However, we use 1024 in order to save memory. usrsctp allocates 104 bytes
+// for each pair of incoming/outgoing streams (on a 64-bit system), so 65535
+// streams would waste ~6MB.
+//
+// Note: "max" and "min" here are inclusive.
+constexpr uint16_t kMaxSctpStreams = 1024;
+constexpr uint16_t kMaxSctpSid = kMaxSctpStreams - 1;
+constexpr uint16_t kMinSctpSid = 0;
+// The maximum number of streams that can be negotiated according to spec.
+constexpr uint16_t kSpecMaxSctpSid = 65535;
+
+// This is the default SCTP port to use. It is passed along the wire and the
+// connectee and connector must be using the same port. It is not related to the
+// ports at the IP level. (Corresponds to: sockaddr_conn.sconn_port in
+// usrsctp.h)
+const int kSctpDefaultPort = 5000;
+
+// Error cause codes defined at
+// https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24
+enum class SctpErrorCauseCode : uint16_t {
+  kInvalidStreamIdentifier = 1,
+  kMissingMandatoryParameter = 2,
+  kStaleCookieError = 3,
+  kOutOfResource = 4,
+  kUnresolvableAddress = 5,
+  kUnrecognizedChunkType = 6,
+  kInvalidMandatoryParameter = 7,
+  kUnrecognizedParameters = 8,
+  kNoUserData = 9,
+  kCookieReceivedWhileShuttingDown = 10,
+  kRestartWithNewAddresses = 11,
+  kUserInitiatedAbort = 12,
+  kProtocolViolation = 13,
+};
+
+// Abstract SctpTransport interface for use internally (by PeerConnection etc.).
+// Exists to allow mock/fake SctpTransports to be created.
+class SctpTransportInternal {
+ public:
+  virtual ~SctpTransportInternal() {}
+
+  virtual void SetOnConnectedCallback(std::function<void()> callback) = 0;
+  virtual void SetDataChannelSink(webrtc::DataChannelSink* sink) = 0;
+
+  // Changes what underlying DTLS transport is uses. Used when switching which
+  // bundled transport the SctpTransport uses.
+  virtual void SetDtlsTransport(rtc::PacketTransportInternal* transport) = 0;
+
+  // When Start is called, connects as soon as possible; this can be called
+  // before DTLS completes, in which case the connection will begin when DTLS
+  // completes. This method can be called multiple times, though not if either
+  // of the ports are changed.
+  //
+  // `local_sctp_port` and `remote_sctp_port` are passed along the wire and the
+  // listener and connector must be using the same port. They are not related
+  // to the ports at the IP level. If set to -1, we default to
+  // kSctpDefaultPort.
+  // `max_message_size_` sets the max message size on the connection.
+  // It must be smaller than or equal to kSctpSendBufferSize.
+  // It can be changed by a secons Start() call.
+  //
+  // TODO(deadbeef): Support calling Start with different local/remote ports
+  // and create a new association? Not clear if this is something we need to
+  // support though. See: https://github.com/w3c/webrtc-pc/issues/979
+  virtual bool Start(int local_sctp_port,
+                     int remote_sctp_port,
+                     int max_message_size) = 0;
+
+  // NOTE: Initially there was a "Stop" method here, but it was never used, so
+  // it was removed.
+
+  // Informs SctpTransport that `sid` will start being used. Returns false if
+  // it is impossible to use `sid`, or if it's already in use.
+  // Until calling this, can't send data using `sid`.
+  // TODO(deadbeef): Actually implement the "returns false if `sid` can't be
+  // used" part. See:
+  // https://bugs.chromium.org/p/chromium/issues/detail?id=619849
+  virtual bool OpenStream(int sid) = 0;
+  // The inverse of OpenStream. Begins the closing procedure, which will
+  // eventually result in SignalClosingProcedureComplete on the side that
+  // initiates it, and both SignalClosingProcedureStartedRemotely and
+  // SignalClosingProcedureComplete on the other side.
+  virtual bool ResetStream(int sid) = 0;
+  // Send data down this channel.
+  // Returns RTCError::OK() if successful an error otherwise. Notably
+  // RTCErrorType::RESOURCE_EXHAUSTED for blocked operations.
+  virtual webrtc::RTCError SendData(int sid,
+                                    const webrtc::SendDataParams& params,
+                                    const rtc::CopyOnWriteBuffer& payload) = 0;
+
+  // Indicates when the SCTP socket is created and not blocked by congestion
+  // control. This changes to false when SDR_BLOCK is returned from SendData,
+  // and
+  // changes to true when SignalReadyToSendData is fired. The underlying DTLS/
+  // ICE channels may be unwritable while ReadyToSendData is true, because data
+  // can still be queued in usrsctp.
+  virtual bool ReadyToSendData() = 0;
+  // Returns the current max message size, set with Start().
+  virtual int max_message_size() const = 0;
+  // Returns the current negotiated max # of outbound streams.
+  // Will return absl::nullopt if negotiation is incomplete.
+  virtual absl::optional<int> max_outbound_streams() const = 0;
+  // Returns the current negotiated max # of inbound streams.
+  virtual absl::optional<int> max_inbound_streams() const = 0;
+
+  // Helper for debugging.
+  virtual void set_debug_name_for_testing(const char* debug_name) = 0;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_