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| 1 /* |
| 2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 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 |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #include "webrtc/ortc/testrtpparameters.h" |
| 12 |
| 13 #include <algorithm> |
| 14 #include <utility> |
| 15 |
| 16 namespace webrtc { |
| 17 |
| 18 RtpParameters MakeMinimalOpusParameters() { |
| 19 RtpParameters parameters; |
| 20 RtpCodecParameters opus_codec; |
| 21 opus_codec.name = "opus"; |
| 22 opus_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| 23 opus_codec.payload_type = 111; |
| 24 opus_codec.clock_rate.emplace(48000); |
| 25 opus_codec.num_channels.emplace(2); |
| 26 parameters.codecs.push_back(std::move(opus_codec)); |
| 27 RtpEncodingParameters encoding; |
| 28 encoding.codec_payload_type.emplace(111); |
| 29 parameters.encodings.push_back(std::move(encoding)); |
| 30 return parameters; |
| 31 } |
| 32 |
| 33 RtpParameters MakeMinimalIsacParameters() { |
| 34 RtpParameters parameters; |
| 35 RtpCodecParameters isac_codec; |
| 36 isac_codec.name = "ISAC"; |
| 37 isac_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| 38 isac_codec.payload_type = 103; |
| 39 isac_codec.clock_rate.emplace(16000); |
| 40 parameters.codecs.push_back(std::move(isac_codec)); |
| 41 RtpEncodingParameters encoding; |
| 42 encoding.codec_payload_type.emplace(111); |
| 43 parameters.encodings.push_back(std::move(encoding)); |
| 44 return parameters; |
| 45 } |
| 46 |
| 47 RtpParameters MakeMinimalOpusParametersWithSsrc(uint32_t ssrc) { |
| 48 RtpParameters parameters = MakeMinimalOpusParameters(); |
| 49 parameters.encodings[0].ssrc.emplace(ssrc); |
| 50 return parameters; |
| 51 } |
| 52 |
| 53 RtpParameters MakeMinimalIsacParametersWithSsrc(uint32_t ssrc) { |
| 54 RtpParameters parameters = MakeMinimalIsacParameters(); |
| 55 parameters.encodings[0].ssrc.emplace(ssrc); |
| 56 return parameters; |
| 57 } |
| 58 |
| 59 RtpParameters MakeMinimalVideoParameters(const char* codec_name) { |
| 60 RtpParameters parameters; |
| 61 RtpCodecParameters vp8_codec; |
| 62 vp8_codec.name = codec_name; |
| 63 vp8_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 64 vp8_codec.payload_type = 96; |
| 65 parameters.codecs.push_back(std::move(vp8_codec)); |
| 66 RtpEncodingParameters encoding; |
| 67 encoding.codec_payload_type.emplace(96); |
| 68 parameters.encodings.push_back(std::move(encoding)); |
| 69 return parameters; |
| 70 } |
| 71 |
| 72 RtpParameters MakeMinimalVp8Parameters() { |
| 73 return MakeMinimalVideoParameters("VP8"); |
| 74 } |
| 75 |
| 76 RtpParameters MakeMinimalVp9Parameters() { |
| 77 return MakeMinimalVideoParameters("VP9"); |
| 78 } |
| 79 |
| 80 RtpParameters MakeMinimalVp8ParametersWithSsrc(uint32_t ssrc) { |
| 81 RtpParameters parameters = MakeMinimalVp8Parameters(); |
| 82 parameters.encodings[0].ssrc.emplace(ssrc); |
| 83 return parameters; |
| 84 } |
| 85 |
| 86 RtpParameters MakeMinimalVp9ParametersWithSsrc(uint32_t ssrc) { |
| 87 RtpParameters parameters = MakeMinimalVp9Parameters(); |
| 88 parameters.encodings[0].ssrc.emplace(ssrc); |
| 89 return parameters; |
| 90 } |
| 91 |
| 92 // Make audio parameters with all the available properties configured and |
| 93 // features used, and with multiple codecs offered. Obtained by taking a |
| 94 // snapshot of a default PeerConnection offer (and adding other things, like |
| 95 // bitrate limit). |
| 96 // |
| 97 // See "MakeFullOpusParameters"/"MakeFullIsacParameters" below. |
| 98 RtpParameters MakeFullAudioParameters(int preferred_payload_type) { |
| 99 RtpParameters parameters; |
| 100 |
| 101 RtpCodecParameters opus_codec; |
| 102 opus_codec.name = "opus"; |
| 103 opus_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| 104 opus_codec.payload_type = 111; |
| 105 opus_codec.clock_rate.emplace(48000); |
| 106 opus_codec.num_channels.emplace(2); |
| 107 opus_codec.parameters["minptime"] = "10"; |
| 108 opus_codec.parameters["useinbandfec"] = "1"; |
| 109 opus_codec.parameters["usedtx"] = "1"; |
| 110 opus_codec.parameters["stereo"] = "1"; |
| 111 opus_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC); |
| 112 parameters.codecs.push_back(std::move(opus_codec)); |
| 113 |
| 114 RtpCodecParameters isac_codec; |
| 115 isac_codec.name = "ISAC"; |
| 116 isac_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| 117 isac_codec.payload_type = 103; |
| 118 isac_codec.clock_rate.emplace(16000); |
| 119 parameters.codecs.push_back(std::move(isac_codec)); |
| 120 |
| 121 RtpCodecParameters cn_codec; |
| 122 cn_codec.name = "CN"; |
| 123 cn_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| 124 cn_codec.payload_type = 106; |
| 125 cn_codec.clock_rate.emplace(32000); |
| 126 parameters.codecs.push_back(std::move(cn_codec)); |
| 127 |
| 128 RtpCodecParameters dtmf_codec; |
| 129 dtmf_codec.name = "telephone-event"; |
| 130 dtmf_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| 131 dtmf_codec.payload_type = 126; |
| 132 dtmf_codec.clock_rate.emplace(8000); |
| 133 parameters.codecs.push_back(std::move(dtmf_codec)); |
| 134 |
| 135 // "codec_payload_type" isn't implemented, so we need to reorder codecs to |
| 136 // cause one to be used. |
| 137 // TODO(deadbeef): Remove this when it becomes unnecessary. |
| 138 std::sort(parameters.codecs.begin(), parameters.codecs.end(), |
| 139 [preferred_payload_type](const RtpCodecParameters& a, |
| 140 const RtpCodecParameters& b) { |
| 141 return a.payload_type == preferred_payload_type; |
| 142 }); |
| 143 |
| 144 // Intentionally leave out SSRC so one's chosen automatically. |
| 145 RtpEncodingParameters encoding; |
| 146 encoding.codec_payload_type.emplace(preferred_payload_type); |
| 147 encoding.dtx.emplace(DtxStatus::ENABLED); |
| 148 // 20 kbps. |
| 149 encoding.max_bitrate_bps.emplace(20000); |
| 150 parameters.encodings.push_back(std::move(encoding)); |
| 151 |
| 152 parameters.header_extensions.emplace_back( |
| 153 "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1); |
| 154 return parameters; |
| 155 } |
| 156 |
| 157 RtpParameters MakeFullOpusParameters() { |
| 158 return MakeFullAudioParameters(111); |
| 159 } |
| 160 |
| 161 RtpParameters MakeFullIsacParameters() { |
| 162 return MakeFullAudioParameters(103); |
| 163 } |
| 164 |
| 165 // Make video parameters with all the available properties configured and |
| 166 // features used, and with multiple codecs offered. Obtained by taking a |
| 167 // snapshot of a default PeerConnection offer (and adding other things, like |
| 168 // bitrate limit). |
| 169 // |
| 170 // See "MakeFullVp8Parameters"/"MakeFullVp9Parameters" below. |
| 171 RtpParameters MakeFullVideoParameters(int preferred_payload_type) { |
| 172 RtpParameters parameters; |
| 173 |
| 174 RtpCodecParameters vp8_codec; |
| 175 vp8_codec.name = "VP8"; |
| 176 vp8_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 177 vp8_codec.payload_type = 100; |
| 178 vp8_codec.clock_rate.emplace(90000); |
| 179 vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::CCM, |
| 180 RtcpFeedbackMessageType::FIR); |
| 181 vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK, |
| 182 RtcpFeedbackMessageType::GENERIC_NACK); |
| 183 vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK, |
| 184 RtcpFeedbackMessageType::PLI); |
| 185 vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::REMB); |
| 186 vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC); |
| 187 parameters.codecs.push_back(std::move(vp8_codec)); |
| 188 |
| 189 RtpCodecParameters vp8_rtx_codec; |
| 190 vp8_rtx_codec.name = "rtx"; |
| 191 vp8_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 192 vp8_rtx_codec.payload_type = 96; |
| 193 vp8_rtx_codec.clock_rate.emplace(90000); |
| 194 vp8_rtx_codec.parameters["apt"] = "100"; |
| 195 parameters.codecs.push_back(std::move(vp8_rtx_codec)); |
| 196 |
| 197 RtpCodecParameters vp9_codec; |
| 198 vp9_codec.name = "VP9"; |
| 199 vp9_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 200 vp9_codec.payload_type = 101; |
| 201 vp9_codec.clock_rate.emplace(90000); |
| 202 vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::CCM, |
| 203 RtcpFeedbackMessageType::FIR); |
| 204 vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK, |
| 205 RtcpFeedbackMessageType::GENERIC_NACK); |
| 206 vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK, |
| 207 RtcpFeedbackMessageType::PLI); |
| 208 vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::REMB); |
| 209 vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC); |
| 210 parameters.codecs.push_back(std::move(vp9_codec)); |
| 211 |
| 212 RtpCodecParameters vp9_rtx_codec; |
| 213 vp9_rtx_codec.name = "rtx"; |
| 214 vp9_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 215 vp9_rtx_codec.payload_type = 97; |
| 216 vp9_rtx_codec.clock_rate.emplace(90000); |
| 217 vp9_rtx_codec.parameters["apt"] = "101"; |
| 218 parameters.codecs.push_back(std::move(vp9_rtx_codec)); |
| 219 |
| 220 RtpCodecParameters red_codec; |
| 221 red_codec.name = "red"; |
| 222 red_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 223 red_codec.payload_type = 116; |
| 224 red_codec.clock_rate.emplace(90000); |
| 225 parameters.codecs.push_back(std::move(red_codec)); |
| 226 |
| 227 RtpCodecParameters red_rtx_codec; |
| 228 red_rtx_codec.name = "rtx"; |
| 229 red_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 230 red_rtx_codec.payload_type = 98; |
| 231 red_rtx_codec.clock_rate.emplace(90000); |
| 232 red_rtx_codec.parameters["apt"] = "116"; |
| 233 parameters.codecs.push_back(std::move(red_rtx_codec)); |
| 234 |
| 235 RtpCodecParameters ulpfec_codec; |
| 236 ulpfec_codec.name = "ulpfec"; |
| 237 ulpfec_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| 238 ulpfec_codec.payload_type = 117; |
| 239 ulpfec_codec.clock_rate.emplace(90000); |
| 240 parameters.codecs.push_back(std::move(ulpfec_codec)); |
| 241 |
| 242 // "codec_payload_type" isn't implemented, so we need to reorder codecs to |
| 243 // cause one to be used. |
| 244 // TODO(deadbeef): Remove this when it becomes unnecessary. |
| 245 std::sort(parameters.codecs.begin(), parameters.codecs.end(), |
| 246 [preferred_payload_type](const RtpCodecParameters& a, |
| 247 const RtpCodecParameters& b) { |
| 248 return a.payload_type == preferred_payload_type; |
| 249 }); |
| 250 |
| 251 // Intentionally leave out SSRC so one's chosen automatically. |
| 252 RtpEncodingParameters encoding; |
| 253 encoding.codec_payload_type.emplace(preferred_payload_type); |
| 254 encoding.fec.emplace(FecMechanism::RED_AND_ULPFEC); |
| 255 // Will create default RtxParameters, with unset SSRC. |
| 256 encoding.rtx.emplace(); |
| 257 // 100 kbps. |
| 258 encoding.max_bitrate_bps.emplace(100000); |
| 259 parameters.encodings.push_back(std::move(encoding)); |
| 260 |
| 261 parameters.header_extensions.emplace_back( |
| 262 "urn:ietf:params:rtp-hdrext:toffset", 2); |
| 263 parameters.header_extensions.emplace_back( |
| 264 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 3); |
| 265 parameters.header_extensions.emplace_back("urn:3gpp:video-orientation", 4); |
| 266 parameters.header_extensions.emplace_back( |
| 267 "http://www.ietf.org/id/" |
| 268 "draft-holmer-rmcat-transport-wide-cc-extensions-01", |
| 269 5); |
| 270 parameters.header_extensions.emplace_back( |
| 271 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", 6); |
| 272 return parameters; |
| 273 } |
| 274 |
| 275 RtpParameters MakeFullVp8Parameters() { |
| 276 return MakeFullVideoParameters(100); |
| 277 } |
| 278 |
| 279 RtpParameters MakeFullVp9Parameters() { |
| 280 return MakeFullVideoParameters(101); |
| 281 } |
| 282 |
| 283 } // namespace webrtc |
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