Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 660 insertions, 0 deletions

diff --git a/third_party/libwebrtc/media/engine/webrtc_video_engine.h b/third_party/libwebrtc/media/engine/webrtc_video_engine.h
new file mode 100644
index 0000000000..ca49f17736
--- /dev/null
+++ b/third_party/libwebrtc/media/engine/webrtc_video_engine.h
@@ -0,0 +1,660 @@
+/*
+ *  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 <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/call/transport.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/transport/field_trial_based_config.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_codecs/sdp_video_format.h"
+#include "call/call.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "media/base/media_channel_impl.h"
+#include "media/base/media_engine.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"
+
+namespace webrtc {
+class VideoDecoderFactory;
+class VideoEncoderFactory;
+}  // namespace webrtc
+
+namespace cricket {
+
+class WebRtcVideoChannel;
+
+// 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;
+
+  VideoMediaChannel* CreateMediaChannel(
+      webrtc::Call* call,
+      const MediaConfig& config,
+      const VideoOptions& options,
+      const webrtc::CryptoOptions& crypto_options,
+      webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
+      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_;
+};
+
+class WebRtcVideoChannel : public VideoMediaChannel,
+                           public webrtc::Transport,
+                           public webrtc::EncoderSwitchRequestCallback {
+ public:
+  WebRtcVideoChannel(
+      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);
+  ~WebRtcVideoChannel() override;
+
+  // VideoMediaChannel implementation
+  bool SetSendParameters(const VideoSendParameters& params) override;
+  bool SetRecvParameters(const VideoRecvParameters& params) override;
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  webrtc::RTCError SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters,
+      webrtc::SetParametersCallback callback) override;
+  webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
+  webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
+  bool GetSendCodec(VideoCodec* send_codec) 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;
+  bool AddRecvStream(const StreamParams& sp) override;
+  bool AddRecvStream(const StreamParams& sp, bool default_stream);
+  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;
+  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
+  bool GetSendStats(VideoMediaSendInfo* info) override;
+  bool GetReceiveStats(VideoMediaReceiveInfo* info) override;
+
+  void OnPacketReceived(const webrtc::RtpPacketReceived& packet) 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;
+  void SetInterface(MediaChannelNetworkInterface* iface) 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;
+  // 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 SetVideoCodecSwitchingEnabled(bool enabled) override;
+
+  bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
+
+  absl::optional<int> GetBaseMinimumPlayoutDelayMs(
+      uint32_t ssrc) const override;
+
+  // Implemented for VideoMediaChannelTest.
+  bool sending() const {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return sending_;
+  }
+
+  StreamParams unsignaled_stream_params() {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return unsignaled_stream_params_;
+  }
+
+  // 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,
+  };
+
+  static constexpr int kDefaultQpMax = 56;
+
+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
+
+  // Implements webrtc::EncoderSwitchRequestCallback.
+  void RequestEncoderFallback() override;
+  void RequestEncoderSwitch(const webrtc::SdpVideoFormat& format,
+                            bool allow_default_fallback) 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 GenerateSendKeyFrame(uint32_t ssrc,
+                            const std::vector<std::string>& rids) override;
+
+  void SetEncoderToPacketizerFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  void SetDepacketizerToDecoderFrameTransformer(
+      uint32_t ssrc,
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+ private:
+  class WebRtcVideoReceiveStream;
+
+  // Finds VideoReceiveStreamInterface corresponding to ssrc. Aware of
+  // unsignalled ssrc handling.
+  WebRtcVideoReceiveStream* FindReceiveStream(uint32_t ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  struct VideoCodecSettings {
+    VideoCodecSettings();
+
+    // 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;
+  };
+
+  struct ChangedSendParameters {
+    // 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;
+  };
+
+  struct ChangedRecvParameters {
+    // 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;
+  };
+
+  bool GetChangedSendParameters(const VideoSendParameters& params,
+                                ChangedSendParameters* changed_params) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  bool ApplyChangedParams(const ChangedSendParameters& changed_params);
+  bool GetChangedRecvParameters(const VideoRecvParameters& params,
+                                ChangedRecvParameters* changed_params) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(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 ReCreateDefaulReceiveStream(uint32_t ssrc,
+                                   absl::optional<uint32_t> rtx_ssrc);
+  void ConfigureReceiverRtp(
+      webrtc::VideoReceiveStreamInterface::Config* config,
+      webrtc::FlexfecReceiveStream::Config* flexfec_config,
+      const StreamParams& sp) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  bool ValidateSendSsrcAvailability(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_);
+
+  static std::string CodecSettingsVectorToString(
+      const std::vector<VideoCodecSettings>& codecs);
+
+  // 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);
+
+  // 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 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 VideoSendParameters& send_params);
+    ~WebRtcVideoSendStream();
+
+    void SetSendParameters(const ChangedSendParameters& 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,
+    // and whether or not the encoding in `rtp_parameters_` is active.
+    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_;
+  };
+
+  // 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, 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 SetRecvParameters(const ChangedRecvParameters& 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);
+
+   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();
+    void StartReceiveStream();
+
+    // 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_);
+  };
+
+  void Construct(webrtc::Call* call, WebRtcVideoEngine* engine);
+
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* data, size_t len) override;
+
+  // Generate the list of codec parameters to pass down based on the negotiated
+  // "codecs". Note that VideoCodecSettings correspond to concrete codecs like
+  // VP8, VP9, H264 while VideoCodecs correspond also to "virtual" codecs like
+  // RTX, ULPFEC, FLEXFEC.
+  static std::vector<VideoCodecSettings> MapCodecs(
+      const std::vector<VideoCodec>& codecs);
+  // 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_);
+
+  static bool NonFlexfecReceiveCodecsHaveChanged(
+      std::vector<VideoCodecSettings> before,
+      std::vector<VideoCodecSettings> after);
+
+  void FillSenderStats(VideoMediaSendInfo* info, bool log_stats)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+  void FillReceiverStats(VideoMediaReceiveInfo* 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_);
+  void FillReceiveCodecStats(VideoMediaReceiveInfo* video_media_info)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
+
+  webrtc::TaskQueueBase* const worker_thread_;
+  webrtc::ScopedTaskSafety task_safety_;
+  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_thread_checker_;
+  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_);
+  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_);
+  std::map<uint32_t, WebRtcVideoReceiveStream*> receive_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::ChangedRecvParameters.
+  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.
+  VideoSendParameters send_params_ RTC_GUARDED_BY(thread_checker_);
+  VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
+  VideoRecvParameters recv_params_ RTC_GUARDED_BY(thread_checker_);
+  int64_t last_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_);
+
+  // TODO(bugs.webrtc.org/11341): Remove this and relevant PC API. Presence
+  // of multiple negotiated codecs allows generic encoder fallback on failures.
+  // Presence of EncoderSelector allows switching to specific encoders.
+  bool allow_codec_switching_ = false;
+};
+
+}  // namespace cricket
+
+#endif  // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_