Chromium Code Reviews| Index: webrtc/api/rtptransportinterface.h |
| diff --git a/webrtc/api/rtptransportinterface.h b/webrtc/api/rtptransportinterface.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..939903bd87626ddb91b8a49cd2a4cf91781230ca |
| --- /dev/null |
| +++ b/webrtc/api/rtptransportinterface.h |
| @@ -0,0 +1,71 @@ |
| +/* |
| + * Copyright 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_API_RTPTRANSPORTINTERFACE_H_ |
| +#define WEBRTC_API_RTPTRANSPORTINTERFACE_H_ |
| + |
| +#include <string> |
| + |
| +#include "webrtc/api/packettransportinterface.h" |
| +#include "webrtc/api/rtcerror.h" |
| +#include "webrtc/base/optional.h" |
| + |
| +namespace webrtc { |
| + |
| +class RtpTransportShim; |
| + |
| +struct RtcpParameters { |
| + // The SSRC to be used in the "SSRC of packet sender" field. |
| + // If not set, one will be chosen by the implementation. |
| + // TODO(deadbeef): Not implemented. |
| + rtc::Optional<uint32_t> ssrc; |
| + |
| + // RTCP CNAME; if empty, one will be chosen by the implementation. |
| + std::string cname; |
| + |
| + // Send reduced-size RTCP? |
| + bool reduced_size = false; |
| + |
| + // Send RTCP multiplexed on the RTP transport? |
| + bool mux = true; |
| +}; |
| + |
| +// Base class for different types of RTP transports that can be created by an |
| +// OrtcFactory. Used by RtpSender/RtpReceivers. |
| +class RtpTransportInterface { |
| + public: |
| + virtual ~RtpTransportInterface() {} |
| + |
| + // Returns packet transport that's used to send RTP packets. |
| + virtual PacketTransportInterface* GetRtpPacketTransport() const = 0; |
| + |
| + // Returns separate packet transport that's used to send RTCP packets. |
| + // If RTCP multiplexing is being used, returns nullptr. |
| + virtual PacketTransportInterface* GetRtcpPacketTransport() const = 0; |
| + |
| + // Set/get RTCP params. Changing "mux" from "true" to "false" is not allowed. |
| + virtual RTCError SetRtcpParameters(const RtcpParameters& parameters) = 0; |
| + virtual RtcpParameters GetRtcpParameters() const = 0; |
| + |
| + protected: |
| + // Only for internal use. |
| + // Returns a pointer to the internal (non-public) interface. |
| + virtual RtpTransportShim* GetInternal() = 0; |
|
pthatcher1
2017/02/08 01:33:49
Again, I'd like to understand this model better.
Taylor Brandstetter
2017/02/10 00:19:45
Ok, so suppose the user of the API passes an "RtpT
|
| + |
| + // Classes that can use this internal interface. |
| + friend class OrtcFactory; |
| + friend class RtpSenderShim; |
| + friend class RtpReceiverShim; |
| + friend class RtpTransportControllerShim; |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_API_RTPTRANSPORTINTERFACE_H_ |