/* * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef VIDEO_VIDEO_RECEIVE_STREAM_H_ #define VIDEO_VIDEO_RECEIVE_STREAM_H_ #include #include #include "call/rtp_packet_sink_interface.h" #include "call/syncable.h" #include "call/video_receive_stream.h" #include "common_video/include/incoming_video_stream.h" #include "common_video/libyuv/include/webrtc_libyuv.h" #include "modules/rtp_rtcp/include/flexfec_receiver.h" #include "modules/video_coding/frame_buffer2.h" #include "modules/video_coding/video_coding_impl.h" #include "rtc_base/sequenced_task_checker.h" #include "system_wrappers/include/clock.h" #include "video/receive_statistics_proxy.h" #include "video/rtp_streams_synchronizer.h" #include "video/rtp_video_stream_receiver.h" #include "video/transport_adapter.h" #include "video/video_stream_decoder.h" namespace webrtc { class CallStats; class IvfFileWriter; class ProcessThread; class RTPFragmentationHeader; class RtpStreamReceiverInterface; class RtpStreamReceiverControllerInterface; class RtxReceiveStream; class VCMTiming; class VCMJitterEstimator; namespace internal { class VideoReceiveStream : public webrtc::VideoReceiveStream, public rtc::VideoSinkInterface, public EncodedImageCallback, public NackSender, public KeyFrameRequestSender, public video_coding::OnCompleteFrameCallback, public Syncable, public CallStatsObserver { public: VideoReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller, int num_cpu_cores, PacketRouter* packet_router, VideoReceiveStream::Config config, ProcessThread* process_thread, CallStats* call_stats); ~VideoReceiveStream() override; const Config& config() const { return config_; } void SignalNetworkState(NetworkState state); bool DeliverRtcp(const uint8_t* packet, size_t length); void SetSync(Syncable* audio_syncable); // Implements webrtc::VideoReceiveStream. void Start() override; void Stop() override; webrtc::VideoReceiveStream::Stats GetStats() const override; // Takes ownership of the file, is responsible for closing it later. // Calling this method will close and finalize any current log. // Giving rtc::kInvalidPlatformFileValue disables logging. // If a frame to be written would make the log too large the write fails and // the log is closed and finalized. A |byte_limit| of 0 means no limit. void EnableEncodedFrameRecording(rtc::PlatformFile file, size_t byte_limit) override; void AddSecondarySink(RtpPacketSinkInterface* sink) override; void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override; // Implements rtc::VideoSinkInterface. void OnFrame(const VideoFrame& video_frame) override; // Implements EncodedImageCallback. EncodedImageCallback::Result OnEncodedImage( const EncodedImage& encoded_image, const CodecSpecificInfo* codec_specific_info, const RTPFragmentationHeader* fragmentation) override; // Implements NackSender. void SendNack(const std::vector& sequence_numbers) override; // Implements KeyFrameRequestSender. void RequestKeyFrame() override; // Implements video_coding::OnCompleteFrameCallback. void OnCompleteFrame( std::unique_ptr frame) override; // Implements CallStatsObserver::OnRttUpdate void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override; // Implements Syncable. int id() const override; rtc::Optional GetInfo() const override; uint32_t GetPlayoutTimestamp() const override; void SetMinimumPlayoutDelay(int delay_ms) override; private: static void DecodeThreadFunction(void* ptr); bool Decode(); rtc::SequencedTaskChecker worker_sequence_checker_; rtc::SequencedTaskChecker module_process_sequence_checker_; TransportAdapter transport_adapter_; const VideoReceiveStream::Config config_; const int num_cpu_cores_; ProcessThread* const process_thread_; Clock* const clock_; rtc::PlatformThread decode_thread_; CallStats* const call_stats_; // Shared by media and rtx stream receivers, since the latter has no RtpRtcp // module of its own. const std::unique_ptr rtp_receive_statistics_; std::unique_ptr timing_; // Jitter buffer experiment. vcm::VideoReceiver video_receiver_; std::unique_ptr> incoming_video_stream_; ReceiveStatisticsProxy stats_proxy_; RtpVideoStreamReceiver rtp_video_stream_receiver_; std::unique_ptr video_stream_decoder_; RtpStreamsSynchronizer rtp_stream_sync_; rtc::CriticalSection ivf_writer_lock_; std::unique_ptr ivf_writer_ RTC_GUARDED_BY(ivf_writer_lock_); // Members for the new jitter buffer experiment. std::unique_ptr jitter_estimator_; std::unique_ptr frame_buffer_; std::unique_ptr media_receiver_; std::unique_ptr rtx_receive_stream_; std::unique_ptr rtx_receiver_; // Whenever we are in an undecodable state (stream has just started or due to // a decoding error) we require a keyframe to restart the stream. bool keyframe_required_ = true; // If we have successfully decoded any frame. bool frame_decoded_ = false; int64_t last_keyframe_request_ms_ = 0; }; } // namespace internal } // namespace webrtc #endif // VIDEO_VIDEO_RECEIVE_STREAM_H_