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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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118 // new function which has slighly different semantics. Delete as | 118 // new function which has slighly different semantics. Delete as |
119 // soon as known applications are updated. | 119 // soon as known applications are updated. |
120 virtual int32_t SetMaxTransferUnit(uint16_t size) { | 120 virtual int32_t SetMaxTransferUnit(uint16_t size) { |
121 SetMaxRtpPacketSize(size); | 121 SetMaxRtpPacketSize(size); |
122 return 0; | 122 return 0; |
123 } | 123 } |
124 | 124 |
125 // Sets the maximum size of an RTP packet, including RTP headers. | 125 // Sets the maximum size of an RTP packet, including RTP headers. |
126 virtual void SetMaxRtpPacketSize(size_t size) = 0; | 126 virtual void SetMaxRtpPacketSize(size_t size) = 0; |
127 | 127 |
128 // Returns max payload length. | |
129 // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is | |
130 // enabled). | |
131 virtual size_t MaxPayloadSize() const = 0; | |
132 | |
133 // Returns max RTP packet size. Takes into account RTP headers and | 128 // Returns max RTP packet size. Takes into account RTP headers and |
134 // FEC/ULP/RED overhead (when FEC is enabled). | 129 // FEC/ULP/RED overhead (when FEC is enabled). |
135 virtual size_t MaxRtpPacketSize() const = 0; | 130 virtual size_t MaxRtpPacketSize() const = 0; |
136 | 131 |
137 // Sets codec name and payload type. Returns -1 on failure else 0. | 132 // Sets codec name and payload type. Returns -1 on failure else 0. |
138 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; | 133 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; |
139 | 134 |
140 // Sets codec name and payload type. Return -1 on failure else 0. | 135 // Sets codec name and payload type. Return -1 on failure else 0. |
141 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; | 136 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; |
142 | 137 |
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466 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 461 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
467 | 462 |
468 // Sends a request for a keyframe. | 463 // Sends a request for a keyframe. |
469 // Returns -1 on failure else 0. | 464 // Returns -1 on failure else 0. |
470 virtual int32_t RequestKeyFrame() = 0; | 465 virtual int32_t RequestKeyFrame() = 0; |
471 }; | 466 }; |
472 | 467 |
473 } // namespace webrtc | 468 } // namespace webrtc |
474 | 469 |
475 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 470 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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