video/video_receive_stream.h - src - Git at Google

 /*
  *  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 <memory>
 #include <vector>

 #include "api/sequence_checker.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/video/recordable_encoded_frame.h"
 #include "call/rtp_packet_sink_interface.h"
 #include "call/syncable.h"
 #include "call/video_receive_stream.h"
 #include "modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "modules/rtp_rtcp/source/source_tracker.h"
 #include "modules/video_coding/frame_buffer2.h"
 #include "modules/video_coding/video_receiver2.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/system/no_unique_address.h"
 #include "rtc_base/task_queue.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 ProcessThread;
 class RtpStreamReceiverInterface;
 class RtpStreamReceiverControllerInterface;
 class RtxReceiveStream;
 class VCMTiming;

 namespace internal {

 class VideoReceiveStream : public webrtc::DEPRECATED_VideoReceiveStream,
                            public rtc::VideoSinkInterface<VideoFrame>,
                            public NackSender,
                            public OnCompleteFrameCallback,
                            public Syncable,
                            public CallStatsObserver {
  public:
   // The default number of milliseconds to pass before re-requesting a key frame
   // to be sent.
   static constexpr int kMaxWaitForKeyFrameMs = 200;

   VideoReceiveStream(TaskQueueFactory* task_queue_factory,
                      RtpStreamReceiverControllerInterface* receiver_controller,
                      int num_cpu_cores,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
                      ProcessThread* process_thread,
                      CallStats* call_stats,
                      Clock* clock,
                      VCMTiming* timing);
   VideoReceiveStream(TaskQueueFactory* task_queue_factory,
                      RtpStreamReceiverControllerInterface* receiver_controller,
                      int num_cpu_cores,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
                      ProcessThread* process_thread,
                      CallStats* call_stats,
                      Clock* clock);
   ~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;

   void AddSecondarySink(RtpPacketSinkInterface* sink) override;
   void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;

   // SetBaseMinimumPlayoutDelayMs and GetBaseMinimumPlayoutDelayMs are called
   // from webrtc/api level and requested by user code. For e.g. blink/js layer
   // in Chromium.
   bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
   int GetBaseMinimumPlayoutDelayMs() const override;

   void SetFrameDecryptor(
       rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
   void SetDepacketizerToDecoderFrameTransformer(
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;

   // Implements rtc::VideoSinkInterface<VideoFrame>.
   void OnFrame(const VideoFrame& video_frame) override;

   // Implements NackSender.
   // For this particular override of the interface,
   // only (buffering_allowed == true) is acceptable.
   void SendNack(const std::vector<uint16_t>& sequence_numbers,
                 bool buffering_allowed) override;

   // Implements OnCompleteFrameCallback.
   void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) override;

   // Implements CallStatsObserver::OnRttUpdate
   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;

   // Implements Syncable.
   uint32_t id() const override;
   absl::optional<Syncable::Info> GetInfo() const override;
   bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
                               int64_t* time_ms) const override;
   void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
                                          int64_t time_ms) override;

   // SetMinimumPlayoutDelay is only called by A/V sync.
   bool SetMinimumPlayoutDelay(int delay_ms) override;

   std::vector<webrtc::RtpSource> GetSources() const override;

   RecordingState SetAndGetRecordingState(RecordingState state,
                                          bool generate_key_frame) override;
   void GenerateKeyFrame() override;

  private:
   int64_t GetWaitMs() const;
   void StartNextDecode() RTC_RUN_ON(decode_queue_);
   void HandleEncodedFrame(std::unique_ptr<EncodedFrame> frame)
       RTC_RUN_ON(decode_queue_);
   void HandleFrameBufferTimeout() RTC_RUN_ON(decode_queue_);
   void UpdatePlayoutDelays() const
       RTC_EXCLUSIVE_LOCKS_REQUIRED(playout_delay_lock_);
   void RequestKeyFrame(int64_t timestamp_ms) RTC_RUN_ON(decode_queue_);
   void HandleKeyFrameGeneration(bool received_frame_is_keyframe, int64_t now_ms)
       RTC_RUN_ON(decode_queue_);
   bool IsReceivingKeyFrame(int64_t timestamp_ms) const
       RTC_RUN_ON(decode_queue_);

   void UpdateHistograms();

   RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_sequence_checker_;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker module_process_sequence_checker_;
   RTC_NO_UNIQUE_ADDRESS SequenceChecker network_sequence_checker_;

   TaskQueueFactory* const task_queue_factory_;

   TransportAdapter transport_adapter_;
   const VideoReceiveStream::Config config_;
   const int num_cpu_cores_;
   ProcessThread* const process_thread_;
   Clock* const clock_;

   CallStats* const call_stats_;

   bool decoder_running_ RTC_GUARDED_BY(worker_sequence_checker_) = false;
   bool decoder_stopped_ RTC_GUARDED_BY(decode_queue_) = true;

   SourceTracker source_tracker_;
   ReceiveStatisticsProxy stats_proxy_;
   // Shared by media and rtx stream receivers, since the latter has no RtpRtcp
   // module of its own.
   const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;

   std::unique_ptr<VCMTiming> timing_;  // Jitter buffer experiment.
   VideoReceiver2 video_receiver_;
   std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
   RtpVideoStreamReceiver rtp_video_stream_receiver_;
   std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
   RtpStreamsSynchronizer rtp_stream_sync_;

   // TODO(nisse, philipel): Creation and ownership of video encoders should be
   // moved to the new VideoStreamDecoder.
   std::vector<std::unique_ptr<VideoDecoder>> video_decoders_;

   // Members for the new jitter buffer experiment.
   std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;

   std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
   std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
   std::unique_ptr<RtpStreamReceiverInterface> 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;
   int64_t last_complete_frame_time_ms_ = 0;

   // Keyframe request intervals are configurable through field trials.
   const int max_wait_for_keyframe_ms_;
   const int max_wait_for_frame_ms_;

   mutable Mutex playout_delay_lock_;

   // All of them tries to change current min_playout_delay on |timing_| but
   // source of the change request is different in each case. Among them the
   // biggest delay is used. -1 means use default value from the |timing_|.
   //
   // Minimum delay as decided by the RTP playout delay extension.
   int frame_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
   // Minimum delay as decided by the setLatency function in "webrtc/api".
   int base_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
   // Minimum delay as decided by the A/V synchronization feature.
   int syncable_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) =
       -1;

   // Maximum delay as decided by the RTP playout delay extension.
   int frame_maximum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;

   // Function that is triggered with encoded frames, if not empty.
   std::function<void(const RecordableEncodedFrame&)>
       encoded_frame_buffer_function_ RTC_GUARDED_BY(decode_queue_);
   // Set to true while we're requesting keyframes but not yet received one.
   bool keyframe_generation_requested_ RTC_GUARDED_BY(decode_queue_) = false;

   // Defined last so they are destroyed before all other members.
   rtc::TaskQueue decode_queue_;
 };
 }  // namespace internal
 }  // namespace webrtc

 #endif  // VIDEO_VIDEO_RECEIVE_STREAM_H_
	/*
	* 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 <memory>
	#include <vector>

	#include "api/sequence_checker.h"
	#include "api/task_queue/task_queue_factory.h"
	#include "api/video/recordable_encoded_frame.h"
	#include "call/rtp_packet_sink_interface.h"
	#include "call/syncable.h"
	#include "call/video_receive_stream.h"
	#include "modules/rtp_rtcp/include/flexfec_receiver.h"
	#include "modules/rtp_rtcp/source/source_tracker.h"
	#include "modules/video_coding/frame_buffer2.h"
	#include "modules/video_coding/video_receiver2.h"
	#include "rtc_base/synchronization/mutex.h"
	#include "rtc_base/system/no_unique_address.h"
	#include "rtc_base/task_queue.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 ProcessThread;
	class RtpStreamReceiverInterface;
	class RtpStreamReceiverControllerInterface;
	class RtxReceiveStream;
	class VCMTiming;

	namespace internal {

	class VideoReceiveStream : public webrtc::DEPRECATED_VideoReceiveStream,
	public rtc::VideoSinkInterface<VideoFrame>,
	public NackSender,
	public OnCompleteFrameCallback,
	public Syncable,
	public CallStatsObserver {
	public:
	// The default number of milliseconds to pass before re-requesting a key frame
	// to be sent.
	static constexpr int kMaxWaitForKeyFrameMs = 200;

	VideoReceiveStream(TaskQueueFactory* task_queue_factory,
	RtpStreamReceiverControllerInterface* receiver_controller,
	int num_cpu_cores,
	PacketRouter* packet_router,
	VideoReceiveStream::Config config,
	ProcessThread* process_thread,
	CallStats* call_stats,
	Clock* clock,
	VCMTiming* timing);
	VideoReceiveStream(TaskQueueFactory* task_queue_factory,
	RtpStreamReceiverControllerInterface* receiver_controller,
	int num_cpu_cores,
	PacketRouter* packet_router,
	VideoReceiveStream::Config config,
	ProcessThread* process_thread,
	CallStats* call_stats,
	Clock* clock);
	~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;

	void AddSecondarySink(RtpPacketSinkInterface* sink) override;
	void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;

	// SetBaseMinimumPlayoutDelayMs and GetBaseMinimumPlayoutDelayMs are called
	// from webrtc/api level and requested by user code. For e.g. blink/js layer
	// in Chromium.
	bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
	int GetBaseMinimumPlayoutDelayMs() const override;

	void SetFrameDecryptor(
	rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
	void SetDepacketizerToDecoderFrameTransformer(
	rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;

	// Implements rtc::VideoSinkInterface<VideoFrame>.
	void OnFrame(const VideoFrame& video_frame) override;

	// Implements NackSender.
	// For this particular override of the interface,
	// only (buffering_allowed == true) is acceptable.
	void SendNack(const std::vector<uint16_t>& sequence_numbers,
	bool buffering_allowed) override;

	// Implements OnCompleteFrameCallback.
	void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) override;

	// Implements CallStatsObserver::OnRttUpdate
	void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;

	// Implements Syncable.
	uint32_t id() const override;
	absl::optional<Syncable::Info> GetInfo() const override;
	bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
	int64_t* time_ms) const override;
	void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
	int64_t time_ms) override;

	// SetMinimumPlayoutDelay is only called by A/V sync.
	bool SetMinimumPlayoutDelay(int delay_ms) override;

	std::vector<webrtc::RtpSource> GetSources() const override;

	RecordingState SetAndGetRecordingState(RecordingState state,
	bool generate_key_frame) override;
	void GenerateKeyFrame() override;

	private:
	int64_t GetWaitMs() const;
	void StartNextDecode() RTC_RUN_ON(decode_queue_);
	void HandleEncodedFrame(std::unique_ptr<EncodedFrame> frame)
	RTC_RUN_ON(decode_queue_);
	void HandleFrameBufferTimeout() RTC_RUN_ON(decode_queue_);
	void UpdatePlayoutDelays() const
	RTC_EXCLUSIVE_LOCKS_REQUIRED(playout_delay_lock_);
	void RequestKeyFrame(int64_t timestamp_ms) RTC_RUN_ON(decode_queue_);
	void HandleKeyFrameGeneration(bool received_frame_is_keyframe, int64_t now_ms)
	RTC_RUN_ON(decode_queue_);
	bool IsReceivingKeyFrame(int64_t timestamp_ms) const
	RTC_RUN_ON(decode_queue_);

	void UpdateHistograms();

	RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_sequence_checker_;
	RTC_NO_UNIQUE_ADDRESS SequenceChecker module_process_sequence_checker_;
	RTC_NO_UNIQUE_ADDRESS SequenceChecker network_sequence_checker_;

	TaskQueueFactory* const task_queue_factory_;

	TransportAdapter transport_adapter_;
	const VideoReceiveStream::Config config_;
	const int num_cpu_cores_;
	ProcessThread* const process_thread_;
	Clock* const clock_;

	CallStats* const call_stats_;

	bool decoder_running_ RTC_GUARDED_BY(worker_sequence_checker_) = false;
	bool decoder_stopped_ RTC_GUARDED_BY(decode_queue_) = true;

	SourceTracker source_tracker_;
	ReceiveStatisticsProxy stats_proxy_;
	// Shared by media and rtx stream receivers, since the latter has no RtpRtcp
	// module of its own.
	const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;

	std::unique_ptr<VCMTiming> timing_; // Jitter buffer experiment.
	VideoReceiver2 video_receiver_;
	std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
	RtpVideoStreamReceiver rtp_video_stream_receiver_;
	std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
	RtpStreamsSynchronizer rtp_stream_sync_;

	// TODO(nisse, philipel): Creation and ownership of video encoders should be
	// moved to the new VideoStreamDecoder.
	std::vector<std::unique_ptr<VideoDecoder>> video_decoders_;

	// Members for the new jitter buffer experiment.
	std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;

	std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
	std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
	std::unique_ptr<RtpStreamReceiverInterface> 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;
	int64_t last_complete_frame_time_ms_ = 0;

	// Keyframe request intervals are configurable through field trials.
	const int max_wait_for_keyframe_ms_;
	const int max_wait_for_frame_ms_;

	mutable Mutex playout_delay_lock_;

	// All of them tries to change current min_playout_delay on \|timing_\| but
	// source of the change request is different in each case. Among them the
	// biggest delay is used. -1 means use default value from the \|timing_\|.
	//
	// Minimum delay as decided by the RTP playout delay extension.
	int frame_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
	// Minimum delay as decided by the setLatency function in "webrtc/api".
	int base_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
	// Minimum delay as decided by the A/V synchronization feature.
	int syncable_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) =
	-1;

	// Maximum delay as decided by the RTP playout delay extension.
	int frame_maximum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;

	// Function that is triggered with encoded frames, if not empty.
	std::function<void(const RecordableEncodedFrame&)>
	encoded_frame_buffer_function_ RTC_GUARDED_BY(decode_queue_);
	// Set to true while we're requesting keyframes but not yet received one.
	bool keyframe_generation_requested_ RTC_GUARDED_BY(decode_queue_) = false;

	// Defined last so they are destroyed before all other members.
	rtc::TaskQueue decode_queue_;
	};
	} // namespace internal
	} // namespace webrtc

	#endif // VIDEO_VIDEO_RECEIVE_STREAM_H_