From 2aa4a82499d4becd2284cdb482213d541b8804dd Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Sun, 28 Apr 2024 16:29:10 +0200
Subject: Adding upstream version 86.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../libwebrtc/webrtc/video/video_send_stream.h     | 115 +++++++++++++++++++++
 1 file changed, 115 insertions(+)
 create mode 100644 third_party/libwebrtc/webrtc/video/video_send_stream.h

(limited to 'third_party/libwebrtc/webrtc/video/video_send_stream.h')

diff --git a/third_party/libwebrtc/webrtc/video/video_send_stream.h b/third_party/libwebrtc/webrtc/video/video_send_stream.h
new file mode 100644
index 0000000000..998250cd6e
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/video/video_send_stream.h
@@ -0,0 +1,115 @@
+/*
+ *  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_SEND_STREAM_H_
+#define VIDEO_VIDEO_SEND_STREAM_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "call/bitrate_allocator.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/video_coding/protection_bitrate_calculator.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue.h"
+#include "video/encoder_rtcp_feedback.h"
+#include "video/send_delay_stats.h"
+#include "video/send_statistics_proxy.h"
+#include "video/video_stream_encoder.h"
+
+namespace webrtc {
+
+class CallStats;
+class SendSideCongestionController;
+class IvfFileWriter;
+class ProcessThread;
+class RtpRtcp;
+class RtpTransportControllerSendInterface;
+class RtcEventLog;
+
+namespace internal {
+
+class VideoSendStreamImpl;
+
+// VideoSendStream implements webrtc::VideoSendStream.
+// Internally, it delegates all public methods to VideoSendStreamImpl and / or
+// VideoStreamEncoder. VideoSendStreamInternal is created and deleted on
+// |worker_queue|.
+class VideoSendStream : public webrtc::VideoSendStream {
+ public:
+  VideoSendStream(
+      int num_cpu_cores,
+      ProcessThread* module_process_thread,
+      rtc::TaskQueue* worker_queue,
+      CallStats* call_stats,
+      RtpTransportControllerSendInterface* transport,
+      BitrateAllocator* bitrate_allocator,
+      SendDelayStats* send_delay_stats,
+      RtcEventLog* event_log,
+      VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config,
+      const std::map<uint32_t, RtpState>& suspended_ssrcs,
+      const std::map<uint32_t, RtpPayloadState>& suspended_payload_states);
+
+  ~VideoSendStream() override;
+
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
+
+  // webrtc::VideoSendStream implementation.
+  void Start() override;
+  void Stop() override;
+
+  void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+                 const DegradationPreference& degradation_preference) override;
+
+  void ReconfigureVideoEncoder(VideoEncoderConfig) override;
+  Stats GetStats() override;
+
+  typedef std::map<uint32_t, RtpState> RtpStateMap;
+  typedef std::map<uint32_t, RtpPayloadState> RtpPayloadStateMap;
+
+  // Takes ownership of each file, is responsible for closing them later.
+  // Calling this method will close and finalize any current logs.
+  // Giving rtc::kInvalidPlatformFileValue in any position disables logging
+  // for the corresponding stream.
+  // If a frame to be written would make the log too large the write fails and
+  // the log is closed and finalized. A |byte_limit| of 0 means no limit.
+  void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
+                                   size_t byte_limit) override;
+
+  void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
+                                      RtpPayloadStateMap* payload_state_map);
+
+  void SetTransportOverhead(size_t transport_overhead_per_packet);
+
+ private:
+  class ConstructionTask;
+  class DestructAndGetRtpStateTask;
+
+  rtc::ThreadChecker thread_checker_;
+  rtc::TaskQueue* const worker_queue_;
+  rtc::Event thread_sync_event_;
+
+  SendStatisticsProxy stats_proxy_;
+  const VideoSendStream::Config config_;
+  const VideoEncoderConfig::ContentType content_type_;
+  std::unique_ptr<VideoSendStreamImpl> send_stream_;
+  std::unique_ptr<VideoStreamEncoder> video_stream_encoder_;
+};
+
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // VIDEO_VIDEO_SEND_STREAM_H_
-- 
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