
This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameters 'audio call video': #!/bin/bash find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I02c5db956846a88a268a300ba086703a02d62e36 Reviewed-on: https://webrtc-review.googlesource.com/83722 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23628}
168 lines
6.0 KiB
C++
168 lines
6.0 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef VIDEO_VIDEO_RECEIVE_STREAM_H_
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#define VIDEO_VIDEO_RECEIVE_STREAM_H_
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#include <memory>
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#include <vector>
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#include "call/rtp_packet_sink_interface.h"
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#include "call/syncable.h"
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#include "call/video_receive_stream.h"
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#include "common_video/libyuv/include/webrtc_libyuv.h"
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#include "modules/rtp_rtcp/include/flexfec_receiver.h"
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#include "modules/video_coding/frame_buffer2.h"
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#include "modules/video_coding/video_coding_impl.h"
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#include "rtc_base/sequenced_task_checker.h"
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#include "system_wrappers/include/clock.h"
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#include "video/receive_statistics_proxy.h"
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#include "video/rtp_streams_synchronizer.h"
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#include "video/rtp_video_stream_receiver.h"
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#include "video/transport_adapter.h"
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#include "video/video_stream_decoder.h"
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namespace webrtc {
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class CallStats;
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class IvfFileWriter;
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class ProcessThread;
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class RTPFragmentationHeader;
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class RtpStreamReceiverInterface;
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class RtpStreamReceiverControllerInterface;
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class RtxReceiveStream;
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class VCMTiming;
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class VCMJitterEstimator;
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namespace internal {
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class VideoReceiveStream : public webrtc::VideoReceiveStream,
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public rtc::VideoSinkInterface<VideoFrame>,
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public EncodedImageCallback,
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public NackSender,
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public KeyFrameRequestSender,
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public video_coding::OnCompleteFrameCallback,
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public Syncable,
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public CallStatsObserver {
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public:
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VideoReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
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int num_cpu_cores,
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PacketRouter* packet_router,
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VideoReceiveStream::Config config,
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ProcessThread* process_thread,
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CallStats* call_stats);
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~VideoReceiveStream() override;
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const Config& config() const { return config_; }
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void SignalNetworkState(NetworkState state);
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bool DeliverRtcp(const uint8_t* packet, size_t length);
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void SetSync(Syncable* audio_syncable);
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// Implements webrtc::VideoReceiveStream.
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void Start() override;
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void Stop() override;
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webrtc::VideoReceiveStream::Stats GetStats() const override;
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// Takes ownership of the file, is responsible for closing it later.
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// Calling this method will close and finalize any current log.
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// Giving rtc::kInvalidPlatformFileValue disables logging.
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// If a frame to be written would make the log too large the write fails and
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// the log is closed and finalized. A |byte_limit| of 0 means no limit.
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void EnableEncodedFrameRecording(rtc::PlatformFile file,
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size_t byte_limit) override;
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void AddSecondarySink(RtpPacketSinkInterface* sink) override;
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void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;
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// Implements rtc::VideoSinkInterface<VideoFrame>.
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void OnFrame(const VideoFrame& video_frame) override;
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// Implements EncodedImageCallback.
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EncodedImageCallback::Result OnEncodedImage(
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const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info,
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const RTPFragmentationHeader* fragmentation) override;
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// Implements NackSender.
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void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
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// Implements KeyFrameRequestSender.
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void RequestKeyFrame() override;
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// Implements video_coding::OnCompleteFrameCallback.
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void OnCompleteFrame(
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std::unique_ptr<video_coding::EncodedFrame> frame) override;
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// Implements CallStatsObserver::OnRttUpdate
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void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
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// Implements Syncable.
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int id() const override;
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absl::optional<Syncable::Info> GetInfo() const override;
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uint32_t GetPlayoutTimestamp() const override;
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void SetMinimumPlayoutDelay(int delay_ms) override;
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private:
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static void DecodeThreadFunction(void* ptr);
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bool Decode();
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rtc::SequencedTaskChecker worker_sequence_checker_;
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rtc::SequencedTaskChecker module_process_sequence_checker_;
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TransportAdapter transport_adapter_;
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const VideoReceiveStream::Config config_;
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const int num_cpu_cores_;
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ProcessThread* const process_thread_;
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Clock* const clock_;
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rtc::PlatformThread decode_thread_;
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CallStats* const call_stats_;
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// Shared by media and rtx stream receivers, since the latter has no RtpRtcp
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// module of its own.
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const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
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std::unique_ptr<VCMTiming> timing_; // Jitter buffer experiment.
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vcm::VideoReceiver video_receiver_;
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std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
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ReceiveStatisticsProxy stats_proxy_;
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RtpVideoStreamReceiver rtp_video_stream_receiver_;
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std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
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RtpStreamsSynchronizer rtp_stream_sync_;
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rtc::CriticalSection ivf_writer_lock_;
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std::unique_ptr<IvfFileWriter> ivf_writer_ RTC_GUARDED_BY(ivf_writer_lock_);
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// Members for the new jitter buffer experiment.
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std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
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std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
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std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
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std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
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std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
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// Whenever we are in an undecodable state (stream has just started or due to
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// a decoding error) we require a keyframe to restart the stream.
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bool keyframe_required_ = true;
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// If we have successfully decoded any frame.
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bool frame_decoded_ = false;
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int64_t last_keyframe_request_ms_ = 0;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // VIDEO_VIDEO_RECEIVE_STREAM_H_
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