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+/*
+ * 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_