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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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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. | 128 // Returns max payload length. |
129 // 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 |
130 // enabled). | 130 // enabled). |
131 virtual size_t MaxPayloadSize() const = 0; | 131 // |extension_sizes| is used to to calculate the RTP header extension |
132 // overhead and should have an entry for each header extension used. | |
133 virtual size_t MaxPayloadSize( | |
danilchap
2017/05/09 12:33:49
This function look unused [and only slightly teste
erikvarga1
2017/05/09 13:30:30
Alright, I've removed it.
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134 rtc::ArrayView<const RtpExtensionSize> extension_sizes) const = 0; | |
132 | 135 |
133 // Returns max RTP packet size. Takes into account RTP headers and | 136 // Returns max RTP packet size. Takes into account RTP headers and |
134 // FEC/ULP/RED overhead (when FEC is enabled). | 137 // FEC/ULP/RED overhead (when FEC is enabled). |
135 virtual size_t MaxRtpPacketSize() const = 0; | 138 virtual size_t MaxRtpPacketSize() const = 0; |
136 | 139 |
137 // Sets codec name and payload type. Returns -1 on failure else 0. | 140 // Sets codec name and payload type. Returns -1 on failure else 0. |
138 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; | 141 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; |
139 | 142 |
140 // Sets codec name and payload type. Return -1 on failure else 0. | 143 // Sets codec name and payload type. Return -1 on failure else 0. |
141 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; | 144 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; |
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466 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 469 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
467 | 470 |
468 // Sends a request for a keyframe. | 471 // Sends a request for a keyframe. |
469 // Returns -1 on failure else 0. | 472 // Returns -1 on failure else 0. |
470 virtual int32_t RequestKeyFrame() = 0; | 473 virtual int32_t RequestKeyFrame() = 0; |
471 }; | 474 }; |
472 | 475 |
473 } // namespace webrtc | 476 } // namespace webrtc |
474 | 477 |
475 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 478 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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