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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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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| 110 | 110 |
| 111 // Decoders for every payload that we can receive. | 111 // Decoders for every payload that we can receive. |
| 112 std::vector<Decoder> decoders; | 112 std::vector<Decoder> decoders; |
| 113 | 113 |
| 114 // Receive-stream specific RTP settings. | 114 // Receive-stream specific RTP settings. |
| 115 struct Rtp { | 115 struct Rtp { |
| 116 std::string ToString() const; | 116 std::string ToString() const; |
| 117 | 117 |
| 118 // Synchronization source (stream identifier) to be received. | 118 // Synchronization source (stream identifier) to be received. |
| 119 uint32_t remote_ssrc = 0; | 119 uint32_t remote_ssrc = 0; |
| 120 |
| 120 // Sender SSRC used for sending RTCP (such as receiver reports). | 121 // Sender SSRC used for sending RTCP (such as receiver reports). |
| 121 uint32_t local_ssrc = 0; | 122 uint32_t local_ssrc = 0; |
| 122 | 123 |
| 123 // See RtcpMode for description. | 124 // See RtcpMode for description. |
| 124 RtcpMode rtcp_mode = RtcpMode::kCompound; | 125 RtcpMode rtcp_mode = RtcpMode::kCompound; |
| 125 | 126 |
| 126 // Extended RTCP settings. | 127 // Extended RTCP settings. |
| 127 struct RtcpXr { | 128 struct RtcpXr { |
| 128 // True if RTCP Receiver Reference Time Report Block extension | 129 // True if RTCP Receiver Reference Time Report Block extension |
| 129 // (RFC 3611) should be enabled. | 130 // (RFC 3611) should be enabled. |
| 130 bool receiver_reference_time_report = false; | 131 bool receiver_reference_time_report = false; |
| 131 } rtcp_xr; | 132 } rtcp_xr; |
| 132 | 133 |
| 133 // See draft-alvestrand-rmcat-remb for information. | 134 // See draft-alvestrand-rmcat-remb for information. |
| 134 bool remb = false; | 135 bool remb = false; |
| 135 | 136 |
| 136 // See draft-holmer-rmcat-transport-wide-cc-extensions for details. | 137 // See draft-holmer-rmcat-transport-wide-cc-extensions for details. |
| 137 bool transport_cc = false; | 138 bool transport_cc = false; |
| 138 | 139 |
| 139 // See NackConfig for description. | 140 // See NackConfig for description. |
| 140 NackConfig nack; | 141 NackConfig nack; |
| 141 | 142 |
| 142 // See UlpfecConfig for description. | 143 // See UlpfecConfig for description. |
| 143 UlpfecConfig ulpfec; | 144 UlpfecConfig ulpfec; |
| 144 | 145 |
| 145 // RTX settings for incoming video payloads that may be received. RTX is | 146 // SSRC for retransmissions. |
| 146 // disabled if there's no config present. | 147 uint32_t rtx_ssrc = 0; |
| 147 struct Rtx { | |
| 148 // SSRCs to use for the RTX streams. | |
| 149 uint32_t ssrc = 0; | |
| 150 | 148 |
| 151 // Payload type to use for the RTX stream. | 149 // Map from video payload type (apt) -> RTX payload type (pt). |
| 152 int payload_type = 0; | 150 // For RTX to be enabled, both an SSRC and this mapping are needed. |
| 153 }; | 151 std::map<int, int> rtx_payload_types; |
| 154 | |
| 155 // Map from video RTP payload type -> RTX config. | |
| 156 typedef std::map<int, Rtx> RtxMap; | |
| 157 RtxMap rtx; | |
| 158 | 152 |
| 159 // RTP header extensions used for the received stream. | 153 // RTP header extensions used for the received stream. |
| 160 std::vector<RtpExtension> extensions; | 154 std::vector<RtpExtension> extensions; |
| 161 } rtp; | 155 } rtp; |
| 162 | 156 |
| 163 // Transport for outgoing packets (RTCP). | 157 // Transport for outgoing packets (RTCP). |
| 164 Transport* rtcp_send_transport = nullptr; | 158 Transport* rtcp_send_transport = nullptr; |
| 165 | 159 |
| 166 // Must not be 'nullptr' when the stream is started. | 160 // Must not be 'nullptr' when the stream is started. |
| 167 rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr; | 161 rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr; |
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| 219 EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0); | 213 EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0); |
| 220 } | 214 } |
| 221 | 215 |
| 222 protected: | 216 protected: |
| 223 virtual ~VideoReceiveStream() {} | 217 virtual ~VideoReceiveStream() {} |
| 224 }; | 218 }; |
| 225 | 219 |
| 226 } // namespace webrtc | 220 } // namespace webrtc |
| 227 | 221 |
| 228 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ | 222 #endif // WEBRTC_VIDEO_RECEIVE_STREAM_H_ |
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