diff options
Diffstat (limited to 'third_party/libwebrtc/call/rtp_stream_receiver_controller_interface.h')
-rw-r--r-- | third_party/libwebrtc/call/rtp_stream_receiver_controller_interface.h | 43 |
1 files changed, 43 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/rtp_stream_receiver_controller_interface.h b/third_party/libwebrtc/call/rtp_stream_receiver_controller_interface.h new file mode 100644 index 0000000000..793d0bc145 --- /dev/null +++ b/third_party/libwebrtc/call/rtp_stream_receiver_controller_interface.h @@ -0,0 +1,43 @@ +/* + * Copyright (c) 2017 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_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ +#define CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ + +#include <memory> + +#include "call/rtp_packet_sink_interface.h" + +namespace webrtc { + +// An RtpStreamReceiver is responsible for the rtp-specific but +// media-independent state needed for receiving an RTP stream. +// TODO(bugs.webrtc.org/7135): Currently, only owns the association between ssrc +// and the stream's RtpPacketSinkInterface. Ownership of corresponding objects +// from modules/rtp_rtcp/ should move to this class (or rather, the +// corresponding implementation class). We should add methods for getting rtp +// receive stats, and for sending RTCP messages related to the receive stream. +class RtpStreamReceiverInterface { + public: + virtual ~RtpStreamReceiverInterface() {} +}; + +// This class acts as a factory for RtpStreamReceiver objects. +class RtpStreamReceiverControllerInterface { + public: + virtual ~RtpStreamReceiverControllerInterface() {} + + virtual std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver( + uint32_t ssrc, + RtpPacketSinkInterface* sink) = 0; +}; + +} // namespace webrtc + +#endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ |