| /* |
| * 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 WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |
| #define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/base/asyncinvoker.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/networkroute.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/media/base/videosinkinterface.h" |
| #include "webrtc/media/base/videosourceinterface.h" |
| #include "webrtc/call.h" |
| #include "webrtc/media/base/mediaengine.h" |
| #include "webrtc/media/engine/webrtcvideochannelfactory.h" |
| #include "webrtc/media/engine/webrtcvideodecoderfactory.h" |
| #include "webrtc/media/engine/webrtcvideoencoderfactory.h" |
| #include "webrtc/transport.h" |
| #include "webrtc/video_frame.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/video_send_stream.h" |
| |
| namespace webrtc { |
| class VideoDecoder; |
| class VideoEncoder; |
| struct MediaConfig; |
| } |
| |
| namespace rtc { |
| class Thread; |
| } // namespace rtc |
| |
| namespace cricket { |
| |
| class VideoCapturer; |
| class VideoFrame; |
| class VideoProcessor; |
| class VideoRenderer; |
| class VoiceMediaChannel; |
| class WebRtcDecoderObserver; |
| class WebRtcEncoderObserver; |
| class WebRtcLocalStreamInfo; |
| class WebRtcRenderAdapter; |
| class WebRtcVideoChannelRecvInfo; |
| class WebRtcVideoChannelSendInfo; |
| class WebRtcVoiceEngine; |
| class WebRtcVoiceMediaChannel; |
| |
| struct Device; |
| |
| // Exposed here for unittests. |
| std::vector<VideoCodec> DefaultVideoCodecList(); |
| |
| class UnsignalledSsrcHandler { |
| public: |
| enum Action { |
| kDropPacket, |
| kDeliverPacket, |
| }; |
| virtual Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel, |
| uint32_t ssrc) = 0; |
| virtual ~UnsignalledSsrcHandler() = default; |
| }; |
| |
| // TODO(pbos): Remove, use external handlers only. |
| class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler { |
| public: |
| DefaultUnsignalledSsrcHandler(); |
| Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel, |
| uint32_t ssrc) override; |
| |
| rtc::VideoSinkInterface<VideoFrame>* GetDefaultSink() const; |
| void SetDefaultSink(VideoMediaChannel* channel, |
| rtc::VideoSinkInterface<VideoFrame>* sink); |
| virtual ~DefaultUnsignalledSsrcHandler() = default; |
| |
| private: |
| uint32_t default_recv_ssrc_; |
| rtc::VideoSinkInterface<VideoFrame>* default_sink_; |
| }; |
| |
| // WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667). |
| class WebRtcVideoEngine2 { |
| public: |
| WebRtcVideoEngine2(); |
| virtual ~WebRtcVideoEngine2(); |
| |
| // Basic video engine implementation. |
| void Init(); |
| |
| WebRtcVideoChannel2* CreateChannel(webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options); |
| |
| const std::vector<VideoCodec>& codecs() const; |
| RtpCapabilities GetCapabilities() const; |
| |
| // Set a WebRtcVideoDecoderFactory for external decoding. Video engine does |
| // not take the ownership of |decoder_factory|. The caller needs to make sure |
| // that |decoder_factory| outlives the video engine. |
| void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory); |
| // Set a WebRtcVideoEncoderFactory for external encoding. Video engine does |
| // not take the ownership of |encoder_factory|. The caller needs to make sure |
| // that |encoder_factory| outlives the video engine. |
| virtual void SetExternalEncoderFactory( |
| WebRtcVideoEncoderFactory* encoder_factory); |
| |
| private: |
| std::vector<VideoCodec> GetSupportedCodecs() const; |
| |
| std::vector<VideoCodec> video_codecs_; |
| |
| bool initialized_; |
| |
| WebRtcVideoDecoderFactory* external_decoder_factory_; |
| WebRtcVideoEncoderFactory* external_encoder_factory_; |
| std::unique_ptr<WebRtcVideoEncoderFactory> simulcast_encoder_factory_; |
| }; |
| |
| class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport { |
| public: |
| WebRtcVideoChannel2(webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const std::vector<VideoCodec>& recv_codecs, |
| WebRtcVideoEncoderFactory* external_encoder_factory, |
| WebRtcVideoDecoderFactory* external_decoder_factory); |
| ~WebRtcVideoChannel2() override; |
| |
| // VideoMediaChannel implementation |
| rtc::DiffServCodePoint PreferredDscp() const override; |
| |
| bool SetSendParameters(const VideoSendParameters& params) override; |
| bool SetRecvParameters(const VideoRecvParameters& params) override; |
| webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override; |
| bool SetRtpSendParameters(uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) override; |
| webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override; |
| bool SetRtpReceiveParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) override; |
| bool GetSendCodec(VideoCodec* send_codec) override; |
| bool SetSend(bool send) override; |
| bool SetVideoSend( |
| uint32_t ssrc, |
| bool enable, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<cricket::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; |
| bool SetSink(uint32_t ssrc, |
| rtc::VideoSinkInterface<VideoFrame>* sink) override; |
| bool GetStats(VideoMediaInfo* info) override; |
| |
| void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) override; |
| void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) override; |
| void OnReadyToSend(bool ready) override; |
| void OnNetworkRouteChanged(const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) override; |
| void SetInterface(NetworkInterface* iface) override; |
| |
| // Implemented for VideoMediaChannelTest. |
| bool sending() const { 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, |
| }; |
| |
| private: |
| class WebRtcVideoReceiveStream; |
| struct VideoCodecSettings { |
| VideoCodecSettings(); |
| |
| bool operator==(const VideoCodecSettings& other) const; |
| bool operator!=(const VideoCodecSettings& other) const; |
| |
| VideoCodec codec; |
| webrtc::FecConfig fec; |
| int rtx_payload_type; |
| }; |
| |
| struct ChangedSendParameters { |
| // These optionals are unset if not changed. |
| rtc::Optional<VideoCodecSettings> codec; |
| rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; |
| rtc::Optional<int> max_bandwidth_bps; |
| rtc::Optional<bool> conference_mode; |
| rtc::Optional<webrtc::RtcpMode> rtcp_mode; |
| }; |
| |
| struct ChangedRecvParameters { |
| // These optionals are unset if not changed. |
| rtc::Optional<std::vector<VideoCodecSettings>> codec_settings; |
| rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; |
| }; |
| |
| bool GetChangedSendParameters(const VideoSendParameters& params, |
| ChangedSendParameters* changed_params) const; |
| bool GetChangedRecvParameters(const VideoRecvParameters& params, |
| ChangedRecvParameters* changed_params) const; |
| |
| void SetMaxSendBandwidth(int bps); |
| |
| void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config, |
| const StreamParams& sp) const; |
| bool CodecIsExternallySupported(const std::string& name) const; |
| bool ValidateSendSsrcAvailability(const StreamParams& sp) const |
| EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
| bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const |
| EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
| void DeleteReceiveStream(WebRtcVideoReceiveStream* stream) |
| EXCLUSIVE_LOCKS_REQUIRED(stream_crit_); |
| |
| static std::string CodecSettingsVectorToString( |
| const std::vector<VideoCodecSettings>& codecs); |
| |
| // Wrapper for the sender part, this is where the source is connected and |
| // frames are then converted from cricket frames to webrtc frames. |
| class WebRtcVideoSendStream |
| : public rtc::VideoSinkInterface<cricket::VideoFrame>, |
| public rtc::VideoSourceInterface<webrtc::VideoFrame>, |
| public webrtc::LoadObserver { |
| public: |
| WebRtcVideoSendStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| WebRtcVideoEncoderFactory* external_encoder_factory, |
| bool enable_cpu_overuse_detection, |
| int max_bitrate_bps, |
| const rtc::Optional<VideoCodecSettings>& codec_settings, |
| const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions, |
| const VideoSendParameters& send_params); |
| virtual ~WebRtcVideoSendStream(); |
| |
| void SetSendParameters(const ChangedSendParameters& send_params); |
| bool SetRtpParameters(const webrtc::RtpParameters& parameters); |
| webrtc::RtpParameters GetRtpParameters() const; |
| |
| // Implements rtc::VideoSourceInterface<webrtc::VideoFrame>. |
| // WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream |
| // in |stream_|. The reason is that WebRtcVideoSendStream receives |
| // cricket::VideoFrames and forwards webrtc::VideoFrames to |source_|. |
| // TODO(perkj, nisse): Refactor WebRtcVideoSendStream to directly connect |
| // the camera input |source_| |
| void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) override; |
| void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override; |
| |
| void OnFrame(const cricket::VideoFrame& frame) override; |
| bool SetVideoSend(bool mute, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<cricket::VideoFrame>* source); |
| void DisconnectSource(); |
| |
| void SetSend(bool send); |
| |
| // Implements webrtc::LoadObserver. |
| void OnLoadUpdate(Load load) override; |
| |
| const std::vector<uint32_t>& GetSsrcs() const; |
| VideoSenderInfo GetVideoSenderInfo(bool log_stats); |
| void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info); |
| |
| 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 rtc::Optional<VideoCodecSettings>& codec_settings); |
| webrtc::VideoSendStream::Config config; |
| VideoOptions options; |
| int max_bitrate_bps; |
| bool conference_mode; |
| rtc::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; |
| }; |
| |
| struct AllocatedEncoder { |
| AllocatedEncoder(webrtc::VideoEncoder* encoder, |
| webrtc::VideoCodecType type, |
| bool external); |
| webrtc::VideoEncoder* encoder; |
| webrtc::VideoEncoder* external_encoder; |
| webrtc::VideoCodecType type; |
| bool external; |
| }; |
| |
| struct VideoFrameInfo { |
| // Initial encoder configuration (QCIF, 176x144) frame (to ensure that |
| // hardware encoders can be initialized). This gives us low memory usage |
| // but also makes it so configuration errors are discovered at the time we |
| // apply the settings rather than when we get the first frame (waiting for |
| // the first frame to know that you gave a bad codec parameter could make |
| // debugging hard). |
| // TODO(pbos): Consider setting up encoders lazily. |
| VideoFrameInfo() |
| : width(176), |
| height(144), |
| rotation(webrtc::kVideoRotation_0), |
| is_texture(false) {} |
| int width; |
| int height; |
| webrtc::VideoRotation rotation; |
| bool is_texture; |
| }; |
| |
| union VideoEncoderSettings { |
| webrtc::VideoCodecH264 h264; |
| webrtc::VideoCodecVP8 vp8; |
| webrtc::VideoCodecVP9 vp9; |
| }; |
| |
| static std::vector<webrtc::VideoStream> CreateVideoStreams( |
| const VideoCodec& codec, |
| const VideoOptions& options, |
| int max_bitrate_bps, |
| size_t num_streams); |
| static std::vector<webrtc::VideoStream> CreateSimulcastVideoStreams( |
| const VideoCodec& codec, |
| const VideoOptions& options, |
| int max_bitrate_bps, |
| size_t num_streams); |
| |
| void* ConfigureVideoEncoderSettings(const VideoCodec& codec) |
| EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| |
| AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec) |
| EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| void DestroyVideoEncoder(AllocatedEncoder* encoder) |
| EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| void SetCodec(const VideoCodecSettings& codec) |
| EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| webrtc::VideoEncoderConfig CreateVideoEncoderConfig( |
| const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| void ReconfigureEncoder() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); |
| |
| // 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() EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| |
| void UpdateHistograms() const EXCLUSIVE_LOCKS_REQUIRED(lock_); |
| |
| rtc::ThreadChecker thread_checker_; |
| rtc::AsyncInvoker invoker_; |
| rtc::Thread* worker_thread_; |
| const std::vector<uint32_t> ssrcs_; |
| const std::vector<SsrcGroup> ssrc_groups_; |
| webrtc::Call* const call_; |
| rtc::VideoSinkWants sink_wants_; |
| // Counter used for deciding if the video resolution is currently |
| // restricted by CPU usage. It is reset if |source_| is changed. |
| int cpu_restricted_counter_; |
| // Total number of times resolution as been requested to be changed due to |
| // CPU adaptation. |
| int number_of_cpu_adapt_changes_; |
| // Total number of frames sent to |stream_|. |
| int frame_count_ GUARDED_BY(lock_); |
| // Total number of cpu restricted frames sent to |stream_|. |
| int cpu_restricted_frame_count_ GUARDED_BY(lock_); |
| rtc::VideoSourceInterface<cricket::VideoFrame>* source_; |
| WebRtcVideoEncoderFactory* const external_encoder_factory_ |
| GUARDED_BY(lock_); |
| |
| rtc::CriticalSection lock_; |
| webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_ |
| GUARDED_BY(lock_); |
| // 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_ GUARDED_BY(lock_); |
| // 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_ GUARDED_BY(lock_); |
| bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); |
| VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); |
| AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); |
| VideoFrameInfo last_frame_info_ GUARDED_BY(lock_); |
| |
| bool sending_ GUARDED_BY(lock_); |
| |
| // The timestamp of the last frame received |
| // Used to generate timestamp for the black frame when source is removed |
| int64_t last_frame_timestamp_us_ GUARDED_BY(lock_); |
| }; |
| |
| // Wrapper for the receiver part, contains configs etc. that are needed to |
| // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper |
| // between rtc::VideoSinkInterface<webrtc::VideoFrame> and |
| // rtc::VideoSinkInterface<cricket::VideoFrame>. |
| class WebRtcVideoReceiveStream |
| : public rtc::VideoSinkInterface<webrtc::VideoFrame> { |
| public: |
| WebRtcVideoReceiveStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoReceiveStream::Config config, |
| WebRtcVideoDecoderFactory* external_decoder_factory, |
| bool default_stream, |
| const std::vector<VideoCodecSettings>& recv_codecs, |
| bool red_disabled_by_remote_side); |
| ~WebRtcVideoReceiveStream(); |
| |
| const std::vector<uint32_t>& GetSsrcs() const; |
| rtc::Optional<uint32_t> GetFirstPrimarySsrc() const; |
| |
| void SetLocalSsrc(uint32_t local_ssrc); |
| // TODO(deadbeef): Move these feedback parameters into the recv parameters. |
| void SetFeedbackParameters(bool nack_enabled, |
| bool remb_enabled, |
| bool transport_cc_enabled, |
| webrtc::RtcpMode rtcp_mode); |
| void SetRecvParameters(const ChangedRecvParameters& recv_params); |
| |
| void OnFrame(const webrtc::VideoFrame& frame) override; |
| bool IsDefaultStream() const; |
| |
| void SetSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink); |
| |
| VideoReceiverInfo GetVideoReceiverInfo(bool log_stats); |
| |
| // Used to disable RED/FEC when the remote description doesn't contain those |
| // codecs. This is needed to be able to work around an RTX bug which is only |
| // happening if the remote side doesn't send RED, but the local side is |
| // configured to receive RED. |
| // TODO(holmer): Remove this after a couple of Chrome versions, M53-54 |
| // time frame. |
| void SetFecDisabledRemotely(bool disable); |
| |
| private: |
| struct AllocatedDecoder { |
| AllocatedDecoder(webrtc::VideoDecoder* decoder, |
| webrtc::VideoCodecType type, |
| bool external); |
| webrtc::VideoDecoder* decoder; |
| // Decoder wrapped into a fallback decoder to permit software fallback. |
| webrtc::VideoDecoder* external_decoder; |
| webrtc::VideoCodecType type; |
| bool external; |
| }; |
| |
| void RecreateWebRtcStream(); |
| |
| void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs, |
| std::vector<AllocatedDecoder>* old_codecs); |
| AllocatedDecoder CreateOrReuseVideoDecoder( |
| std::vector<AllocatedDecoder>* old_decoder, |
| const VideoCodec& codec); |
| void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders); |
| |
| std::string GetCodecNameFromPayloadType(int payload_type); |
| |
| webrtc::Call* const call_; |
| StreamParams stream_params_; |
| |
| webrtc::VideoReceiveStream* stream_; |
| const bool default_stream_; |
| webrtc::VideoReceiveStream::Config config_; |
| bool red_disabled_by_remote_side_; |
| |
| WebRtcVideoDecoderFactory* const external_decoder_factory_; |
| std::vector<AllocatedDecoder> allocated_decoders_; |
| |
| rtc::CriticalSection sink_lock_; |
| rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ GUARDED_BY(sink_lock_); |
| // Expands remote RTP timestamps to int64_t to be able to estimate how long |
| // the stream has been running. |
| rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_ |
| GUARDED_BY(sink_lock_); |
| int64_t first_frame_timestamp_ 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_ GUARDED_BY(sink_lock_); |
| }; |
| |
| void Construct(webrtc::Call* call, WebRtcVideoEngine2* 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; |
| |
| static std::vector<VideoCodecSettings> MapCodecs( |
| const std::vector<VideoCodec>& codecs); |
| std::vector<VideoCodecSettings> FilterSupportedCodecs( |
| const std::vector<VideoCodecSettings>& mapped_codecs) const; |
| static bool ReceiveCodecsHaveChanged(std::vector<VideoCodecSettings> before, |
| std::vector<VideoCodecSettings> after); |
| |
| void FillSenderStats(VideoMediaInfo* info, bool log_stats); |
| void FillReceiverStats(VideoMediaInfo* info, bool log_stats); |
| void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats, |
| VideoMediaInfo* info); |
| |
| rtc::ThreadChecker thread_checker_; |
| |
| uint32_t rtcp_receiver_report_ssrc_; |
| bool sending_; |
| webrtc::Call* const call_; |
| |
| DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_; |
| UnsignalledSsrcHandler* const unsignalled_ssrc_handler_; |
| |
| const MediaConfig::Video video_config_; |
| |
| rtc::CriticalSection stream_crit_; |
| // Using primary-ssrc (first ssrc) as key. |
| std::map<uint32_t, WebRtcVideoSendStream*> send_streams_ |
| GUARDED_BY(stream_crit_); |
| std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_ |
| GUARDED_BY(stream_crit_); |
| std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_); |
| std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_); |
| |
| rtc::Optional<VideoCodecSettings> send_codec_; |
| rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_; |
| |
| WebRtcVideoEncoderFactory* const external_encoder_factory_; |
| WebRtcVideoDecoderFactory* const external_decoder_factory_; |
| std::vector<VideoCodecSettings> recv_codecs_; |
| std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| webrtc::Call::Config::BitrateConfig bitrate_config_; |
| // TODO(deadbeef): Don't duplicate information between |
| // send_params/recv_params, rtp_extensions, options, etc. |
| VideoSendParameters send_params_; |
| VideoOptions default_send_options_; |
| VideoRecvParameters recv_params_; |
| bool red_disabled_by_remote_side_; |
| int64_t last_stats_log_ms_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |