Repo created

TMessagesProj/jni/voip/webrtc/call/rtp_video_sender_interface.h

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+/*
+ *  Copyright (c) 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 CALL_RTP_VIDEO_SENDER_INTERFACE_H_
+#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
+
+#include <map>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
+#include "api/fec_controller_override.h"
+#include "api/video/video_layers_allocation.h"
+#include "call/rtp_config.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+class VideoBitrateAllocation;
+struct FecProtectionParams;
+
+class RtpVideoSenderInterface : public EncodedImageCallback,
+                                public FecControllerOverride {
+ public:
+  // Sets weather or not RTP packets is allowed to be sent on this sender.
+  virtual void SetSending(bool enabled) = 0;
+  virtual bool IsActive() = 0;
+
+  virtual void OnNetworkAvailability(bool network_available) = 0;
+  virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
+  virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
+
+  virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 0;
+
+  virtual void OnBitrateAllocationUpdated(
+      const VideoBitrateAllocation& bitrate) = 0;
+  virtual void OnVideoLayersAllocationUpdated(
+      const VideoLayersAllocation& allocation) = 0;
+  virtual void OnBitrateUpdated(BitrateAllocationUpdate update,
+                                int framerate) = 0;
+  virtual void OnTransportOverheadChanged(
+      size_t transport_overhead_bytes_per_packet) = 0;
+  virtual uint32_t GetPayloadBitrateBps() const = 0;
+  virtual uint32_t GetProtectionBitrateBps() const = 0;
+  virtual void SetEncodingData(size_t width,
+                               size_t height,
+                               size_t num_temporal_layers) = 0;
+  virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      uint32_t ssrc,
+      rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
+
+  // Implements FecControllerOverride.
+  void SetFecAllowed(bool fec_allowed) override = 0;
+};
+}  // namespace webrtc
+#endif  // CALL_RTP_VIDEO_SENDER_INTERFACE_H_