Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 345 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/video_receive_stream2.h b/third_party/libwebrtc/video/video_receive_stream2.h
new file mode 100644
index 0000000000..44e2228dab
--- /dev/null
+++ b/third_party/libwebrtc/video/video_receive_stream2.h
@@ -0,0 +1,345 @@
+/*
+ *  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 <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/recordable_encoded_frame.h"
+#include "call/call.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "call/syncable.h"
+#include "call/video_receive_stream.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
+#include "modules/video_coding/nack_requester.h"
+#include "modules/video_coding/video_receiver2.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.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_buffer_controller.h"
+#include "video/video_stream_decoder2.h"
+
+namespace webrtc {
+
+class RtpStreamReceiverInterface;
+class RtpStreamReceiverControllerInterface;
+class RtxReceiveStream;
+class VCMTiming;
+
+constexpr TimeDelta kMaxWaitForKeyFrame = TimeDelta::Millis(200);
+constexpr TimeDelta kMaxWaitForFrame = TimeDelta::Seconds(3);
+
+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 needing
+// 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::VideoReceiveStreamInterface,
+      public rtc::VideoSinkInterface<VideoFrame>,
+      public RtpVideoStreamReceiver2::OnCompleteFrameCallback,
+      public Syncable,
+      public CallStatsObserver,
+      public FrameSchedulingReceiver {
+ public:
+  // The maximum number of buffered encoded frames when encoded output is
+  // configured.
+  static constexpr size_t kBufferedEncodedFramesMaxSize = 60;
+
+  VideoReceiveStream2(TaskQueueFactory* task_queue_factory,
+                      Call* call,
+                      int num_cpu_cores,
+                      PacketRouter* packet_router,
+                      VideoReceiveStreamInterface::Config config,
+                      CallStats* call_stats,
+                      Clock* clock,
+                      std::unique_ptr<VCMTiming> timing,
+                      NackPeriodicProcessor* nack_periodic_processor,
+                      DecodeSynchronizer* decode_sync,
+                      RtcEventLog* event_log);
+  // Destruction happens on the worker thread. Prior to destruction the caller
+  // must ensure that a registration with the transport has been cleared. See
+  // `RegisterWithTransport` for details.
+  // TODO(tommi): As a further improvement to this, performing the full
+  // destruction on the network thread could be made the default.
+  ~VideoReceiveStream2() override;
+
+  // Called on `packet_sequence_checker_` to register/unregister with the
+  // network transport.
+  void RegisterWithTransport(
+      RtpStreamReceiverControllerInterface* receiver_controller);
+  // If registration has previously been done (via `RegisterWithTransport`) then
+  // `UnregisterFromTransport` must be called prior to destruction, on the
+  // network thread.
+  void UnregisterFromTransport();
+
+  // Accessor for the a/v sync group. This value may change and the caller
+  // must be on the packet delivery thread.
+  const std::string& sync_group() const;
+
+  // Getters for const remote SSRC values that won't change throughout the
+  // object's lifetime.
+  uint32_t remote_ssrc() const { return config_.rtp.remote_ssrc; }
+  uint32_t rtx_ssrc() const { return config_.rtp.rtx_ssrc; }
+
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
+
+  void SetSync(Syncable* audio_syncable);
+
+  // Updates the `rtp_video_stream_receiver_`'s `local_ssrc` when the default
+  // sender has been created, changed or removed.
+  void SetLocalSsrc(uint32_t local_ssrc);
+
+  // Implements webrtc::VideoReceiveStreamInterface.
+  void Start() override;
+  void Stop() override;
+
+  void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
+  RtpHeaderExtensionMap GetRtpExtensionMap() const override;
+  void SetRtcpMode(RtcpMode mode) override;
+  void SetFlexFecProtection(RtpPacketSinkInterface* flexfec_sink) override;
+  void SetLossNotificationEnabled(bool enabled) override;
+  void SetNackHistory(TimeDelta history) override;
+  void SetProtectionPayloadTypes(int red_payload_type,
+                                 int ulpfec_payload_type) override;
+  void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) override;
+  void SetAssociatedPayloadTypes(
+      std::map<int, int> associated_payload_types) override;
+
+  webrtc::VideoReceiveStreamInterface::Stats GetStats() const 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 RtpVideoStreamReceiver2::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:
+  // FrameSchedulingReceiver implementation.
+  // Called on packet sequence.
+  void OnEncodedFrame(std::unique_ptr<EncodedFrame> frame) override;
+  // Called on packet sequence.
+  void OnDecodableFrameTimeout(TimeDelta wait) override;
+
+  void CreateAndRegisterExternalDecoder(const Decoder& decoder);
+
+  struct DecodeFrameResult {
+    // True if the decoder returned code WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME,
+    // or if the decoder failed and a keyframe is required. When true, a
+    // keyframe request should be sent even if a keyframe request was sent
+    // recently.
+    bool force_request_key_frame;
+
+    // The picture id of the frame that was decoded, or nullopt if the frame was
+    // not decoded.
+    absl::optional<int64_t> decoded_frame_picture_id;
+
+    // True if the next frame decoded must be a keyframe. This value will set
+    // the value of `keyframe_required_`, which will force the frame buffer to
+    // drop all frames that are not keyframes.
+    bool keyframe_required;
+  };
+
+  DecodeFrameResult HandleEncodedFrameOnDecodeQueue(
+      std::unique_ptr<EncodedFrame> frame,
+      bool keyframe_request_is_due,
+      bool keyframe_required) RTC_RUN_ON(decode_queue_);
+  void UpdatePlayoutDelays() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_sequence_checker_);
+  void RequestKeyFrame(Timestamp now) RTC_RUN_ON(packet_sequence_checker_);
+  void HandleKeyFrameGeneration(bool received_frame_is_keyframe,
+                                Timestamp now,
+                                bool always_request_key_frame,
+                                bool keyframe_request_is_due)
+      RTC_RUN_ON(packet_sequence_checker_);
+  bool IsReceivingKeyFrame(Timestamp timestamp) const
+      RTC_RUN_ON(packet_sequence_checker_);
+  int DecodeAndMaybeDispatchEncodedFrame(std::unique_ptr<EncodedFrame> frame)
+      RTC_RUN_ON(decode_queue_);
+
+  void UpdateHistograms();
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_sequence_checker_;
+  // TODO(bugs.webrtc.org/11993): This checker conceptually represents
+  // operations that belong to the network thread. The Call class is currently
+  // moving towards handling network packets on the network thread and while
+  // that work is ongoing, this checker may in practice represent the worker
+  // thread, but still serves as a mechanism of grouping together concepts
+  // that belong to the network thread. Once the packets are fully delivered
+  // on the network thread, this comment will be deleted.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
+
+  TaskQueueFactory* const task_queue_factory_;
+
+  TransportAdapter transport_adapter_;
+  const VideoReceiveStreamInterface::Config config_;
+  const int num_cpu_cores_;
+  Call* const call_;
+  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_;
+
+  std::unique_ptr<VideoStreamBufferController> buffer_;
+
+  std::unique_ptr<RtpStreamReceiverInterface> media_receiver_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+  std::unique_ptr<RtxReceiveStream> rtx_receive_stream_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+  std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+
+  // 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(packet_sequence_checker_) = true;
+
+  // If we have successfully decoded any frame.
+  bool frame_decoded_ RTC_GUARDED_BY(decode_queue_) = false;
+
+  absl::optional<Timestamp> last_keyframe_request_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+
+  // Keyframe request intervals are configurable through field trials.
+  TimeDelta max_wait_for_keyframe_ RTC_GUARDED_BY(packet_sequence_checker_);
+  TimeDelta max_wait_for_frame_ RTC_GUARDED_BY(packet_sequence_checker_);
+
+  // 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.
+  absl::optional<TimeDelta> frame_minimum_playout_delay_
+      RTC_GUARDED_BY(worker_sequence_checker_);
+  // Minimum delay as decided by the setLatency function in "webrtc/api".
+  absl::optional<TimeDelta> base_minimum_playout_delay_
+      RTC_GUARDED_BY(worker_sequence_checker_);
+  // Minimum delay as decided by the A/V synchronization feature.
+  absl::optional<TimeDelta> syncable_minimum_playout_delay_
+      RTC_GUARDED_BY(worker_sequence_checker_);
+
+  // Maximum delay as decided by the RTP playout delay extension.
+  absl::optional<TimeDelta> frame_maximum_playout_delay_
+      RTC_GUARDED_BY(worker_sequence_checker_);
+
+  // 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(packet_sequence_checker_) =
+      false;
+  // Lock to avoid unnecessary per-frame idle wakeups in the code.
+  webrtc::Mutex pending_resolution_mutex_;
+  // Signal from decode queue to OnFrame callback to fill pending_resolution_.
+  // absl::nullopt - no resolution needed. 0x0 - next OnFrame to fill with
+  // received resolution. Not 0x0 - OnFrame has filled a resolution.
+  absl::optional<RecordableEncodedFrame::EncodedResolution> pending_resolution_
+      RTC_GUARDED_BY(pending_resolution_mutex_);
+  // Buffered encoded frames held while waiting for decoded resolution.
+  std::vector<std::unique_ptr<EncodedFrame>> buffered_encoded_frames_
+      RTC_GUARDED_BY(decode_queue_);
+
+  // 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_