Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 180 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/video_send_stream_impl.h b/third_party/libwebrtc/video/video_send_stream_impl.h
new file mode 100644
index 0000000000..f145450655
--- /dev/null
+++ b/third_party/libwebrtc/video/video_send_stream_impl.h
@@ -0,0 +1,180 @@
+/*
+ *  Copyright 2018 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_SEND_STREAM_IMPL_H_
+#define VIDEO_VIDEO_SEND_STREAM_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <atomic>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "api/video_codecs/video_encoder.h"
+#include "call/bitrate_allocator.h"
+#include "call/rtp_config.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "call/rtp_video_sender_interface.h"
+#include "modules/include/module_common_types.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/utility/maybe_worker_thread.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+#include "video/config/video_encoder_config.h"
+#include "video/send_statistics_proxy.h"
+#include "video/video_stream_encoder_interface.h"
+
+namespace webrtc {
+namespace internal {
+
+// Pacing buffer config; overridden by ALR config if provided.
+struct PacingConfig {
+  explicit PacingConfig(const FieldTrialsView& field_trials);
+  PacingConfig(const PacingConfig&);
+  PacingConfig& operator=(const PacingConfig&) = default;
+  ~PacingConfig();
+  FieldTrialParameter<double> pacing_factor;
+  FieldTrialParameter<TimeDelta> max_pacing_delay;
+};
+
+// VideoSendStreamImpl implements internal::VideoSendStream.
+// It is created and destroyed on `rtp_transport_queue`. The intent is to
+// decrease the need for locking and to ensure methods are called in sequence.
+// Public methods except `DeliverRtcp` must be called on `rtp_transport_queue`.
+// DeliverRtcp is called on the libjingle worker thread or a network thread.
+// An encoder may deliver frames through the EncodedImageCallback on an
+// arbitrary thread.
+class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
+                            public VideoStreamEncoderInterface::EncoderSink {
+ public:
+  VideoSendStreamImpl(Clock* clock,
+                      SendStatisticsProxy* stats_proxy,
+                      RtpTransportControllerSendInterface* transport,
+                      BitrateAllocatorInterface* bitrate_allocator,
+                      VideoStreamEncoderInterface* video_stream_encoder,
+                      const VideoSendStream::Config* config,
+                      int initial_encoder_max_bitrate,
+                      double initial_encoder_bitrate_priority,
+                      VideoEncoderConfig::ContentType content_type,
+                      RtpVideoSenderInterface* rtp_video_sender,
+                      const FieldTrialsView& field_trials);
+  ~VideoSendStreamImpl() override;
+
+  void DeliverRtcp(const uint8_t* packet, size_t length);
+  void StartPerRtpStream(std::vector<bool> active_layers);
+  void Stop();
+
+  // TODO(holmer): Move these to RtpTransportControllerSend.
+  std::map<uint32_t, RtpState> GetRtpStates() const;
+
+  std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
+
+  const absl::optional<float>& configured_pacing_factor() const {
+    return configured_pacing_factor_;
+  }
+
+ private:
+  // Implements BitrateAllocatorObserver.
+  uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
+
+  // Implements VideoStreamEncoderInterface::EncoderSink
+  void OnEncoderConfigurationChanged(
+      std::vector<VideoStream> streams,
+      bool is_svc,
+      VideoEncoderConfig::ContentType content_type,
+      int min_transmit_bitrate_bps) override;
+
+  void OnBitrateAllocationUpdated(
+      const VideoBitrateAllocation& allocation) override;
+  void OnVideoLayersAllocationUpdated(
+      VideoLayersAllocation allocation) override;
+
+  // Implements EncodedImageCallback. The implementation routes encoded frames
+  // to the `payload_router_` and `config.pre_encode_callback` if set.
+  // Called on an arbitrary encoder callback thread.
+  EncodedImageCallback::Result OnEncodedImage(
+      const EncodedImage& encoded_image,
+      const CodecSpecificInfo* codec_specific_info) override;
+
+  // Implements EncodedImageCallback.
+  void OnDroppedFrame(EncodedImageCallback::DropReason reason) override;
+
+  // Starts monitoring and sends a keyframe.
+  void StartupVideoSendStream();
+  // Removes the bitrate observer, stops monitoring and notifies the video
+  // encoder of the bitrate update.
+  void StopVideoSendStream() RTC_RUN_ON(rtp_transport_queue_);
+
+  void ConfigureProtection();
+  void ConfigureSsrcs();
+  void SignalEncoderTimedOut();
+  void SignalEncoderActive();
+  MediaStreamAllocationConfig GetAllocationConfig() const
+      RTC_RUN_ON(rtp_transport_queue_);
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
+  Clock* const clock_;
+  const bool has_alr_probing_;
+  const PacingConfig pacing_config_;
+
+  SendStatisticsProxy* const stats_proxy_;
+  const VideoSendStream::Config* const config_;
+
+  MaybeWorkerThread* const rtp_transport_queue_;
+
+  RepeatingTaskHandle check_encoder_activity_task_
+      RTC_GUARDED_BY(rtp_transport_queue_);
+
+  std::atomic_bool activity_;
+  bool timed_out_ RTC_GUARDED_BY(rtp_transport_queue_);
+
+  RtpTransportControllerSendInterface* const transport_;
+  BitrateAllocatorInterface* const bitrate_allocator_;
+
+  bool disable_padding_;
+  int max_padding_bitrate_;
+  int encoder_min_bitrate_bps_;
+  uint32_t encoder_max_bitrate_bps_;
+  uint32_t encoder_target_rate_bps_;
+  double encoder_bitrate_priority_;
+
+  VideoStreamEncoderInterface* const video_stream_encoder_;
+
+  RtcpBandwidthObserver* const bandwidth_observer_;
+  RtpVideoSenderInterface* const rtp_video_sender_;
+
+  rtc::scoped_refptr<PendingTaskSafetyFlag> transport_queue_safety_ =
+      PendingTaskSafetyFlag::CreateDetached();
+
+  // Context for the most recent and last sent video bitrate allocation. Used to
+  // throttle sending of similar bitrate allocations.
+  struct VbaSendContext {
+    VideoBitrateAllocation last_sent_allocation;
+    absl::optional<VideoBitrateAllocation> throttled_allocation;
+    int64_t last_send_time_ms;
+  };
+  absl::optional<VbaSendContext> video_bitrate_allocation_context_
+      RTC_GUARDED_BY(rtp_transport_queue_);
+  const absl::optional<float> configured_pacing_factor_;
+};
+}  // namespace internal
+}  // namespace webrtc
+#endif  // VIDEO_VIDEO_SEND_STREAM_IMPL_H_