Repo created

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

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+/*
+ *  Copyright (c) 2016 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_FLEXFEC_RECEIVE_STREAM_H_
+#define CALL_FLEXFEC_RECEIVE_STREAM_H_
+
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "api/call/transport.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "call/receive_stream.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+
+namespace webrtc {
+
+class FlexfecReceiveStream : public RtpPacketSinkInterface,
+                             public ReceiveStreamInterface {
+ public:
+  ~FlexfecReceiveStream() override = default;
+
+  struct Config {
+    explicit Config(Transport* rtcp_send_transport);
+    Config(const Config&);
+    ~Config();
+
+    std::string ToString() const;
+
+    // Returns true if all RTP information is available in order to
+    // enable receiving FlexFEC.
+    bool IsCompleteAndEnabled() const;
+
+    // Payload type for FlexFEC.
+    int payload_type = -1;
+
+    ReceiveStreamRtpConfig rtp;
+
+    // Vector containing a single element, corresponding to the SSRC of the
+    // media stream being protected by this FlexFEC stream. The vector MUST have
+    // size 1.
+    //
+    // TODO(brandtr): Update comment above when we support multistream
+    // protection.
+    std::vector<uint32_t> protected_media_ssrcs;
+
+    // What RTCP mode to use in the reports.
+    RtcpMode rtcp_mode = RtcpMode::kCompound;
+
+    // Transport for outgoing RTCP packets.
+    Transport* rtcp_send_transport = nullptr;
+  };
+
+  // TODO(tommi): FlexfecReceiveStream inherits from ReceiveStreamInterface,
+  // not VideoReceiveStreamInterface where there's also a SetRtcpMode method.
+  // Perhaps this should be in ReceiveStreamInterface and apply to audio streams
+  // as well (although there's no logic that would use it at present).
+  virtual void SetRtcpMode(RtcpMode mode) = 0;
+
+  // Called to change the payload type after initialization.
+  virtual void SetPayloadType(int payload_type) = 0;
+  virtual int payload_type() const = 0;
+
+  virtual const ReceiveStatistics* GetStats() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_FLEXFEC_RECEIVE_STREAM_H_