| /* |
| * 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/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_engine.h" |
| #include "media/engine/unhandled_packets_buffer.h" |
| #include "rtc_base/async_invoker.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "rtc_base/thread_checker.h" |
| |
| namespace webrtc { |
| class VideoDecoderFactory; |
| class VideoEncoderFactory; |
| struct MediaConfig; |
| } // namespace webrtc |
| |
| namespace rtc { |
| class Thread; |
| } // namespace rtc |
| |
| namespace cricket { |
| |
| class WebRtcVideoChannel; |
| |
| class UnsignalledSsrcHandler { |
| public: |
| enum Action { |
| kDropPacket, |
| kDeliverPacket, |
| }; |
| virtual Action OnUnsignalledSsrc(WebRtcVideoChannel* channel, |
| uint32_t ssrc) = 0; |
| virtual ~UnsignalledSsrcHandler() = default; |
| }; |
| |
| // TODO(pbos): Remove, use external handlers only. |
| class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler { |
| public: |
| DefaultUnsignalledSsrcHandler(); |
| Action OnUnsignalledSsrc(WebRtcVideoChannel* channel, uint32_t ssrc) override; |
| |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* GetDefaultSink() const; |
| void SetDefaultSink(WebRtcVideoChannel* channel, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink); |
| |
| virtual ~DefaultUnsignalledSsrcHandler() = default; |
| |
| private: |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_; |
| }; |
| |
| // 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); |
| |
| ~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> codecs() const override; |
| RtpCapabilities GetCapabilities() 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_; |
| }; |
| |
| class WebRtcVideoChannel : public VideoMediaChannel, |
| public webrtc::Transport, |
| public webrtc::EncoderFailureCallback { |
| 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) 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, |
| 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; |
| bool SetSink(uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override; |
| void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override; |
| bool GetStats(VideoMediaInfo* info) override; |
| |
| void OnPacketReceived(rtc::CopyOnWriteBuffer packet, |
| int64_t packet_time_us) override; |
| void OnRtcpReceived(rtc::CopyOnWriteBuffer packet, |
| int64_t packet_time_us) override; |
| void OnReadyToSend(bool ready) override; |
| void OnNetworkRouteChanged(const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) override; |
| void SetInterface( |
| NetworkInterface* iface, |
| const webrtc::MediaTransportConfig& media_transport_config) 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; |
| |
| 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_; |
| } |
| |
| absl::optional<uint32_t> GetDefaultReceiveStreamSsrc(); |
| |
| 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; |
| |
| // Take the buffered packets for |ssrcs| and feed them into DeliverPacket. |
| // This method does nothing unless unknown_ssrc_packet_buffer_ is configured. |
| void BackfillBufferedPackets(rtc::ArrayView<const uint32_t> ssrcs); |
| |
| // Implements webrtc::EncoderFailureCallback. |
| void OnEncoderFailure() override; |
| |
| private: |
| class WebRtcVideoReceiveStream; |
| 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. |
| }; |
| |
| 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 |
| // VideoReceiveStream 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_); |
| |
| void ConfigureReceiverRtp( |
| webrtc::VideoReceiveStream::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); |
| |
| // Wrapper for the sender part. |
| class WebRtcVideoSendStream |
| : public rtc::VideoSourceInterface<webrtc::VideoFrame> { |
| 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); |
| virtual ~WebRtcVideoSendStream(); |
| |
| void SetSendParameters(const ChangedSendParameters& send_params); |
| webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters); |
| webrtc::RtpParameters GetRtpParameters() const; |
| |
| void SetFrameEncryptor( |
| rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor); |
| |
| // Implements rtc::VideoSourceInterface<webrtc::VideoFrame>. |
| // WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream |
| // in |stream_|. This is done to proxy VideoSinkWants from the encoder to |
| // the worker thread. |
| void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) override; |
| void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override; |
| |
| bool SetVideoSend(const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source); |
| |
| void SetSend(bool send); |
| |
| const std::vector<uint32_t>& GetSsrcs() const; |
| VideoSenderInfo GetVideoSenderInfo(bool log_stats); |
| void FillBitrateInfo(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 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(); |
| |
| // 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::ThreadChecker thread_checker_; |
| rtc::Thread* 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_); |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_ |
| 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_); |
| |
| const bool use_standard_bytes_stats_; |
| |
| // In order for the |invoker_| to protect other members from being |
| // destructed as they are used in asynchronous tasks it has to be destructed |
| // first. |
| rtc::AsyncInvoker invoker_; |
| }; |
| |
| // Wrapper for the receiver part, contains configs etc. that are needed to |
| // reconstruct the underlying VideoReceiveStream. |
| class WebRtcVideoReceiveStream |
| : public rtc::VideoSinkInterface<webrtc::VideoFrame> { |
| public: |
| WebRtcVideoReceiveStream( |
| WebRtcVideoChannel* channel, |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoReceiveStream::Config config, |
| webrtc::VideoDecoderFactory* decoder_factory, |
| bool default_stream, |
| const std::vector<VideoCodecSettings>& recv_codecs, |
| const webrtc::FlexfecReceiveStream::Config& flexfec_config); |
| ~WebRtcVideoReceiveStream(); |
| |
| 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; |
| |
| void SetLocalSsrc(uint32_t local_ssrc); |
| // TODO(deadbeef): Move these feedback parameters into the recv parameters. |
| void SetFeedbackParameters(bool lntf_enabled, |
| 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 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); |
| |
| private: |
| void RecreateWebRtcVideoStream(); |
| void MaybeRecreateWebRtcFlexfecStream(); |
| |
| void MaybeAssociateFlexfecWithVideo(); |
| void MaybeDissociateFlexfecFromVideo(); |
| |
| void ConfigureCodecs(const std::vector<VideoCodecSettings>& recv_codecs); |
| void ConfigureFlexfecCodec(int flexfec_payload_type); |
| |
| std::string GetCodecNameFromPayloadType(int payload_type); |
| |
| WebRtcVideoChannel* const channel_; |
| 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::VideoReceiveStream* stream_; |
| const bool default_stream_; |
| webrtc::VideoReceiveStream::Config config_; |
| webrtc::FlexfecReceiveStream::Config flexfec_config_; |
| webrtc::FlexfecReceiveStream* flexfec_stream_; |
| |
| webrtc::VideoDecoderFactory* const decoder_factory_; |
| |
| rtc::CriticalSection sink_lock_; |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink_ |
| RTC_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_ |
| 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_); |
| |
| const bool use_standard_bytes_stats_; |
| }; |
| |
| 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(VideoMediaInfo* info, bool log_stats) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_); |
| void FillReceiverStats(VideoMediaInfo* 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 FillSendAndReceiveCodecStats(VideoMediaInfo* video_media_info) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_); |
| |
| rtc::Thread* worker_thread_; |
| rtc::ThreadChecker 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_GUARDED_BY(thread_checker_); |
| |
| DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_ |
| RTC_GUARDED_BY(thread_checker_); |
| UnsignalledSsrcHandler* const unsignalled_ssrc_handler_ |
| 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_); |
| 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_); |
| |
| absl::optional<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_); |
| 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_); |
| |
| // Buffer for unhandled packets. |
| std::unique_ptr<UnhandledPacketsBuffer> unknown_ssrc_packet_buffer_ |
| RTC_GUARDED_BY(thread_checker_); |
| |
| // In order for the |invoker_| to protect other members from being destructed |
| // as they are used in asynchronous tasks it has to be destructed first. |
| rtc::AsyncInvoker invoker_; |
| }; |
| |
| class EncoderStreamFactory |
| : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface { |
| public: |
| EncoderStreamFactory(std::string codec_name, |
| int max_qp, |
| bool is_screenshare, |
| bool conference_mode); |
| |
| private: |
| std::vector<webrtc::VideoStream> CreateEncoderStreams( |
| int width, |
| int height, |
| const webrtc::VideoEncoderConfig& encoder_config) override; |
| |
| const std::string codec_name_; |
| const int max_qp_; |
| const bool is_screenshare_; |
| // Allows a screenshare specific configuration, which enables temporal |
| // layering and various settings. |
| const bool conference_mode_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // MEDIA_ENGINE_WEBRTC_VIDEO_ENGINE_H_ |
