Chromium Code Reviews| Index: webrtc/call/rtp_stream_receiver_controller_interface.h |
| diff --git a/webrtc/call/rtp_stream_receiver_controller_interface.h b/webrtc/call/rtp_stream_receiver_controller_interface.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..51d25a525e3c26ad514d9dd0fda8d7f66c46c6f7 |
| --- /dev/null |
| +++ b/webrtc/call/rtp_stream_receiver_controller_interface.h |
| @@ -0,0 +1,47 @@ |
| +/* |
| + * 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 WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ |
| +#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ |
| + |
| +#include <memory> |
| + |
| +#include "webrtc/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(nisse): 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 |
|
the sun
2017/06/16 13:37:39
I thought RTCP messages were usually sent for a co
nisse-webrtc
2017/06/16 14:33:26
The intended interface would like "I want a key fr
|
| +// 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; |
| + // For registering additional sinks, needed for FlexFEC. |
| + virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0; |
| + virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0; |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ |