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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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107 | 107 |
108 virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, | 108 virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, |
109 size_t incoming_packet_length) = 0; | 109 size_t incoming_packet_length) = 0; |
110 | 110 |
111 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; | 111 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; |
112 | 112 |
113 // ************************************************************************** | 113 // ************************************************************************** |
114 // Sender | 114 // Sender |
115 // ************************************************************************** | 115 // ************************************************************************** |
116 | 116 |
117 // Sets MTU. | 117 // TODO(nisse): Deprecated. Kept temporarily, as an alias for the |
118 // |size| - Max transfer unit in bytes, default is 1500. | 118 // new function which has slighly different semantics. Delete as |
119 // Returns -1 on failure else 0. | 119 // soon as known applications are updated. |
120 virtual int32_t SetMaxTransferUnit(uint16_t size) = 0; | 120 virtual int32_t SetMaxTransferUnit(uint16_t size) { |
| 121 SetMaxRtpPacketSize(size); |
| 122 return 0; |
| 123 } |
121 | 124 |
122 // Sets transport overhead. Default is IPv4 and UDP with no encryption. | 125 // Sets the maximum size of an RTP packet, including RTP headers. |
123 // |tcp| - true for TCP false UDP. | 126 virtual void SetMaxRtpPacketSize(size_t size) = 0; |
124 // |ipv6| - true for IP version 6 false for version 4. | |
125 // |authentication_overhead| - number of bytes to leave for an authentication | |
126 // header. | |
127 // Returns -1 on failure else 0 | |
128 // TODO(michaelt): deprecate the function. | |
129 virtual int32_t SetTransportOverhead(bool tcp, | |
130 bool ipv6, | |
131 uint8_t authentication_overhead = 0) = 0; | |
132 | 127 |
133 // Sets transport overhead per packet. | 128 // Returns max payload length. |
134 virtual void SetTransportOverhead(int transport_overhead_per_packet) = 0; | |
135 | |
136 // Returns max payload length, which is a combination of the configuration | |
137 // MaxTransferUnit and TransportOverhead. | |
138 // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is | 129 // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is |
139 // enabled). | 130 // enabled). |
140 virtual uint16_t MaxPayloadLength() const = 0; | 131 virtual size_t MaxPayloadSize() const = 0; |
141 | 132 |
142 // Returns max data payload length, which is a combination of the | 133 // Returns max RTP packet size. Takes into account RTP headers and |
143 // configuration MaxTransferUnit, headers and TransportOverhead. | 134 // FEC/ULP/RED overhead (when FEC is enabled). |
144 // Takes into account RTP headers and FEC/ULP/RED overhead (when FEC is | 135 virtual size_t MaxRtpPacketSize() const = 0; |
145 // enabled). | |
146 virtual uint16_t MaxDataPayloadLength() const = 0; | |
147 | 136 |
148 // Sets codec name and payload type. Returns -1 on failure else 0. | 137 // Sets codec name and payload type. Returns -1 on failure else 0. |
149 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; | 138 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; |
150 | 139 |
151 // Sets codec name and payload type. Return -1 on failure else 0. | 140 // Sets codec name and payload type. Return -1 on failure else 0. |
152 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; | 141 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; |
153 | 142 |
154 virtual void RegisterVideoSendPayload(int payload_type, | 143 virtual void RegisterVideoSendPayload(int payload_type, |
155 const char* payload_name) = 0; | 144 const char* payload_name) = 0; |
156 | 145 |
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481 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 470 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
482 | 471 |
483 // Sends a request for a keyframe. | 472 // Sends a request for a keyframe. |
484 // Returns -1 on failure else 0. | 473 // Returns -1 on failure else 0. |
485 virtual int32_t RequestKeyFrame() = 0; | 474 virtual int32_t RequestKeyFrame() = 0; |
486 }; | 475 }; |
487 | 476 |
488 } // namespace webrtc | 477 } // namespace webrtc |
489 | 478 |
490 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 479 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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