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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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109 | 109 |
110 virtual void IncomingRtcpPacket(const uint8_t* incoming_packet, | 110 virtual void IncomingRtcpPacket(const uint8_t* incoming_packet, |
111 size_t incoming_packet_length) = 0; | 111 size_t incoming_packet_length) = 0; |
112 | 112 |
113 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; | 113 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; |
114 | 114 |
115 // ************************************************************************** | 115 // ************************************************************************** |
116 // Sender | 116 // Sender |
117 // ************************************************************************** | 117 // ************************************************************************** |
118 | 118 |
119 // TODO(nisse): Deprecated. Kept temporarily, as an alias for the | |
120 // new function which has slighly different semantics. Delete as | |
121 // soon as known applications are updated. | |
122 virtual int32_t SetMaxTransferUnit(uint16_t size) { | |
123 SetMaxRtpPacketSize(size); | |
124 return 0; | |
125 } | |
126 | |
127 // Sets the maximum size of an RTP packet, including RTP headers. | 119 // Sets the maximum size of an RTP packet, including RTP headers. |
128 virtual void SetMaxRtpPacketSize(size_t size) = 0; | 120 virtual void SetMaxRtpPacketSize(size_t size) = 0; |
129 | 121 |
130 // Returns max RTP packet size. Takes into account RTP headers and | 122 // Returns max RTP packet size. Takes into account RTP headers and |
131 // FEC/ULP/RED overhead (when FEC is enabled). | 123 // FEC/ULP/RED overhead (when FEC is enabled). |
132 virtual size_t MaxRtpPacketSize() const = 0; | 124 virtual size_t MaxRtpPacketSize() const = 0; |
133 | 125 |
134 // Sets codec name and payload type. Returns -1 on failure else 0. | 126 // Sets codec name and payload type. Returns -1 on failure else 0. |
135 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; | 127 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; |
136 | 128 |
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453 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 445 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
454 | 446 |
455 // Sends a request for a keyframe. | 447 // Sends a request for a keyframe. |
456 // Returns -1 on failure else 0. | 448 // Returns -1 on failure else 0. |
457 virtual int32_t RequestKeyFrame() = 0; | 449 virtual int32_t RequestKeyFrame() = 0; |
458 }; | 450 }; |
459 | 451 |
460 } // namespace webrtc | 452 } // namespace webrtc |
461 | 453 |
462 #endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 454 #endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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