video/video_receive_stream2.h

/*
 *  Copyright (c) 2020 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_STREAM2_H_
#define VIDEO_VIDEO_RECEIVE_STREAM2_H_

#include <memory>
#include <vector>

#include "api/task_queue/task_queue_factory.h"
#include "api/units/timestamp.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/sequence_checker.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "system_wrappers/include/clock.h"
#include "video/receive_statistics_proxy2.h"
#include "video/rtp_streams_synchronizer2.h"
#include "video/rtp_video_stream_receiver2.h"
#include "video/transport_adapter.h"
#include "video/video_stream_decoder2.h"

namespace webrtc {

class ProcessThread;
class RtpStreamReceiverInterface;
class RtpStreamReceiverControllerInterface;
class RtxReceiveStream;
class VCMTiming;

namespace internal {

class CallStats;

// Utility struct for grabbing metadata from a VideoFrame and processing it
// asynchronously without needing the actual frame data.
// Additionally the caller can bundle information from the current clock
// when the metadata is captured, for accurate reporting and not needeing
// multiple calls to clock->Now().
struct VideoFrameMetaData {
  VideoFrameMetaData(const webrtc::VideoFrame& frame, Timestamp now)
      : rtp_timestamp(frame.timestamp()),
        timestamp_us(frame.timestamp_us()),
        ntp_time_ms(frame.ntp_time_ms()),
        width(frame.width()),
        height(frame.height()),
        decode_timestamp(now) {}

  int64_t render_time_ms() const {
    return timestamp_us / rtc::kNumMicrosecsPerMillisec;
  }

  const uint32_t rtp_timestamp;
  const int64_t timestamp_us;
  const int64_t ntp_time_ms;
  const int width;
  const int height;

  const Timestamp decode_timestamp;
};

class VideoReceiveStream2 : public webrtc::VideoReceiveStream,
                            public rtc::VideoSinkInterface<VideoFrame>,
                            public NackSender,
                            public video_coding::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;

  VideoReceiveStream2(TaskQueueFactory* task_queue_factory,
                      TaskQueueBase* current_queue,
                      RtpStreamReceiverControllerInterface* receiver_controller,
                      int num_cpu_cores,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
                      ProcessThread* process_thread,
                      CallStats* call_stats,
                      Clock* clock,
                      VCMTiming* timing);
  ~VideoReceiveStream2() 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 video_coding::OnCompleteFrameCallback.
  void OnCompleteFrame(
      std::unique_ptr<video_coding::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 GetMaxWaitMs() const RTC_RUN_ON(decode_queue_);
  void StartNextDecode() RTC_RUN_ON(decode_queue_);
  void HandleEncodedFrame(std::unique_ptr<video_coding::EncodedFrame> frame)
      RTC_RUN_ON(decode_queue_);
  void HandleFrameBufferTimeout(int64_t now_ms, int64_t wait_ms)
      RTC_RUN_ON(worker_sequence_checker_);
  void UpdatePlayoutDelays() const
      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_sequence_checker_);
  void RequestKeyFrame(int64_t timestamp_ms)
      RTC_RUN_ON(worker_sequence_checker_);
  void HandleKeyFrameGeneration(bool received_frame_is_keyframe,
                                int64_t now_ms,
                                bool always_request_key_frame,
                                bool keyframe_request_is_due)
      RTC_RUN_ON(worker_sequence_checker_);
  bool IsReceivingKeyFrame(int64_t timestamp_ms) const
      RTC_RUN_ON(worker_sequence_checker_);

  void UpdateHistograms();

  SequenceChecker worker_sequence_checker_;
  SequenceChecker module_process_sequence_checker_;

  TaskQueueFactory* const task_queue_factory_;

  TransportAdapter transport_adapter_;
  const VideoReceiveStream::Config config_;
  const int num_cpu_cores_;
  TaskQueueBase* const worker_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_;
  RtpVideoStreamReceiver2 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_ RTC_GUARDED_BY(decode_queue_) = true;

  // If we have successfully decoded any frame.
  bool frame_decoded_ RTC_GUARDED_BY(decode_queue_) = false;

  int64_t last_keyframe_request_ms_ RTC_GUARDED_BY(decode_queue_) = 0;
  int64_t last_complete_frame_time_ms_
      RTC_GUARDED_BY(worker_sequence_checker_) = 0;

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

  // 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(worker_sequence_checker_) =
      -1;
  // Minimum delay as decided by the setLatency function in "webrtc/api".
  int base_minimum_playout_delay_ms_ RTC_GUARDED_BY(worker_sequence_checker_) =
      -1;
  // Minimum delay as decided by the A/V synchronization feature.
  int syncable_minimum_playout_delay_ms_
      RTC_GUARDED_BY(worker_sequence_checker_) = -1;

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

  // Set by the field trial WebRTC-LowLatencyRenderer. The parameter |enabled|
  // determines if the low-latency renderer algorithm should be used for the
  // case min playout delay=0 and max playout delay>0.
  FieldTrialParameter<bool> low_latency_renderer_enabled_;
  // Set by the field trial WebRTC-LowLatencyRenderer. The parameter
  // |include_predecode_buffer| determines if the predecode buffer should be
  // taken into account when calculating maximum number of frames in composition
  // queue.
  FieldTrialParameter<bool> low_latency_renderer_include_predecode_buffer_;

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

  // Used to signal destruction to potentially pending tasks.
  ScopedTaskSafety task_safety_;
};
}  // namespace internal
}  // namespace webrtc

#endif  // VIDEO_VIDEO_RECEIVE_STREAM2_H_