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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 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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1247 bool WebRtcSession::SetAudioRtpReceiveParameters( | 1247 bool WebRtcSession::SetAudioRtpReceiveParameters( |
1248 uint32_t ssrc, | 1248 uint32_t ssrc, |
1249 const RtpParameters& parameters) { | 1249 const RtpParameters& parameters) { |
1250 ASSERT(signaling_thread()->IsCurrent()); | 1250 ASSERT(signaling_thread()->IsCurrent()); |
1251 if (!voice_channel_) { | 1251 if (!voice_channel_) { |
1252 return false; | 1252 return false; |
1253 } | 1253 } |
1254 return voice_channel_->SetRtpReceiveParameters(ssrc, parameters); | 1254 return voice_channel_->SetRtpReceiveParameters(ssrc, parameters); |
1255 } | 1255 } |
1256 | 1256 |
1257 bool WebRtcSession::SetSource( | |
1258 uint32_t ssrc, | |
1259 rtc::VideoSourceInterface<cricket::VideoFrame>* source) { | |
1260 ASSERT(signaling_thread()->IsCurrent()); | |
1261 | |
1262 if (!video_channel_) { | |
1263 // |video_channel_| doesnt't exist. Probably because the remote end doesnt't | |
1264 // support video. | |
1265 LOG(LS_WARNING) << "Video not used in this call."; | |
1266 return false; | |
1267 } | |
1268 video_channel_->SetSource(ssrc, source); | |
1269 return true; | |
1270 } | |
1271 | |
1272 void WebRtcSession::SetVideoPlayout( | 1257 void WebRtcSession::SetVideoPlayout( |
1273 uint32_t ssrc, | 1258 uint32_t ssrc, |
1274 bool enable, | 1259 bool enable, |
1275 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) { | 1260 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) { |
1276 ASSERT(signaling_thread()->IsCurrent()); | 1261 ASSERT(signaling_thread()->IsCurrent()); |
1277 if (!video_channel_) { | 1262 if (!video_channel_) { |
1278 LOG(LS_WARNING) << "SetVideoPlayout: No video channel exists."; | 1263 LOG(LS_WARNING) << "SetVideoPlayout: No video channel exists."; |
1279 return; | 1264 return; |
1280 } | 1265 } |
1281 if (!video_channel_->SetSink(ssrc, enable ? sink : NULL)) { | 1266 if (!video_channel_->SetSink(ssrc, enable ? sink : NULL)) { |
1282 // Allow that SetSink fail if |sink| is NULL but assert otherwise. | 1267 // Allow that SetSink fail if |sink| is NULL but assert otherwise. |
1283 // This in the normal case when the underlying media channel has already | 1268 // This in the normal case when the underlying media channel has already |
1284 // been deleted. | 1269 // been deleted. |
1285 ASSERT(sink == NULL); | 1270 ASSERT(sink == NULL); |
1286 } | 1271 } |
1287 } | 1272 } |
1288 | 1273 |
1289 void WebRtcSession::SetVideoSend(uint32_t ssrc, | 1274 void WebRtcSession::SetVideoSend( |
1290 bool enable, | 1275 uint32_t ssrc, |
1291 const cricket::VideoOptions* options) { | 1276 bool enable, |
| 1277 const cricket::VideoOptions* options, |
| 1278 rtc::VideoSourceInterface<cricket::VideoFrame>* source) { |
1292 ASSERT(signaling_thread()->IsCurrent()); | 1279 ASSERT(signaling_thread()->IsCurrent()); |
1293 if (!video_channel_) { | 1280 if (!video_channel_) { |
1294 LOG(LS_WARNING) << "SetVideoSend: No video channel exists."; | 1281 LOG(LS_WARNING) << "SetVideoSend: No video channel exists."; |
1295 return; | 1282 return; |
1296 } | 1283 } |
1297 if (!video_channel_->SetVideoSend(ssrc, enable, options)) { | 1284 if (!video_channel_->SetVideoSend(ssrc, enable, options, source)) { |
1298 // Allow that MuteStream fail if |enable| is false but assert otherwise. | 1285 // Allow that MuteStream fail if |enable| is false and |source| is NULL but |
1299 // This in the normal case when the underlying media channel has already | 1286 // assert otherwise. This in the normal case when the underlying media |
1300 // been deleted. | 1287 // channel has already been deleted. |
1301 ASSERT(enable == false); | 1288 ASSERT(enable == false && source == nullptr); |
1302 } | 1289 } |
1303 } | 1290 } |
1304 | 1291 |
1305 RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const { | 1292 RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const { |
1306 ASSERT(signaling_thread()->IsCurrent()); | 1293 ASSERT(signaling_thread()->IsCurrent()); |
1307 if (video_channel_) { | 1294 if (video_channel_) { |
1308 return video_channel_->GetRtpSendParameters(ssrc); | 1295 return video_channel_->GetRtpSendParameters(ssrc); |
1309 } | 1296 } |
1310 return RtpParameters(); | 1297 return RtpParameters(); |
1311 } | 1298 } |
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2155 ssl_cipher_suite); | 2142 ssl_cipher_suite); |
2156 } | 2143 } |
2157 } | 2144 } |
2158 | 2145 |
2159 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) { | 2146 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) { |
2160 RTC_DCHECK(worker_thread()->IsCurrent()); | 2147 RTC_DCHECK(worker_thread()->IsCurrent()); |
2161 media_controller_->call_w()->OnSentPacket(sent_packet); | 2148 media_controller_->call_w()->OnSentPacket(sent_packet); |
2162 } | 2149 } |
2163 | 2150 |
2164 } // namespace webrtc | 2151 } // namespace webrtc |
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