OLD | NEW |
---|---|
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 |
(...skipping 1285 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
1296 _rtpRtcpModule->DeRegisterSendPayload(payload_type); | 1296 _rtpRtcpModule->DeRegisterSendPayload(payload_type); |
1297 if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) { | 1297 if (_rtpRtcpModule->RegisterSendPayload(lies) != 0) { |
1298 WEBRTC_TRACE( | 1298 WEBRTC_TRACE( |
1299 kTraceError, kTraceVoice, VoEId(_instanceId, _channelId), | 1299 kTraceError, kTraceVoice, VoEId(_instanceId, _channelId), |
1300 "SetEncoder() failed to register codec to RTP/RTCP module"); | 1300 "SetEncoder() failed to register codec to RTP/RTCP module"); |
1301 return false; | 1301 return false; |
1302 } | 1302 } |
1303 } | 1303 } |
1304 | 1304 |
1305 audio_coding_->SetEncoder(std::move(encoder)); | 1305 audio_coding_->SetEncoder(std::move(encoder)); |
1306 codec_manager_.UnsetCodecInst(); | |
1306 return true; | 1307 return true; |
1307 } | 1308 } |
1308 | 1309 |
1310 void Channel::ModifyEncoder( | |
1311 rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { | |
1312 audio_coding_->ModifyEncoder(modifier); | |
1313 } | |
1314 | |
1309 int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) { | 1315 int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) { |
1310 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), | 1316 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
1311 "Channel::RegisterVoiceEngineObserver()"); | 1317 "Channel::RegisterVoiceEngineObserver()"); |
1312 rtc::CritScope cs(&_callbackCritSect); | 1318 rtc::CritScope cs(&_callbackCritSect); |
1313 | 1319 |
1314 if (_voiceEngineObserverPtr) { | 1320 if (_voiceEngineObserverPtr) { |
1315 _engineStatisticsPtr->SetLastError( | 1321 _engineStatisticsPtr->SetLastError( |
1316 VE_INVALID_OPERATION, kTraceError, | 1322 VE_INVALID_OPERATION, kTraceError, |
1317 "RegisterVoiceEngineObserver() observer already enabled"); | 1323 "RegisterVoiceEngineObserver() observer already enabled"); |
1318 return -1; | 1324 return -1; |
(...skipping 11 matching lines...) Expand all Loading... | |
1330 _engineStatisticsPtr->SetLastError( | 1336 _engineStatisticsPtr->SetLastError( |
1331 VE_INVALID_OPERATION, kTraceWarning, | 1337 VE_INVALID_OPERATION, kTraceWarning, |
1332 "DeRegisterVoiceEngineObserver() observer already disabled"); | 1338 "DeRegisterVoiceEngineObserver() observer already disabled"); |
1333 return 0; | 1339 return 0; |
1334 } | 1340 } |
1335 _voiceEngineObserverPtr = NULL; | 1341 _voiceEngineObserverPtr = NULL; |
1336 return 0; | 1342 return 0; |
1337 } | 1343 } |
1338 | 1344 |
1339 int32_t Channel::GetSendCodec(CodecInst& codec) { | 1345 int32_t Channel::GetSendCodec(CodecInst& codec) { |
1340 auto send_codec = codec_manager_.GetCodecInst(); | 1346 { |
ossu
2017/04/26 15:45:36
At first, I had this code just ask the ACM directl
kwiberg-webrtc
2017/04/26 20:03:36
Yes. Hopefully we can remove that soon-ish.
| |
1341 if (send_codec) { | 1347 const CodecInst* send_codec = codec_manager_.GetCodecInst(); |
1342 codec = *send_codec; | 1348 if (send_codec) { |
1349 codec = *send_codec; | |
1350 return 0; | |
1351 } | |
1352 } | |
1353 rtc::Optional<CodecInst> acm_send_codec = audio_coding_->SendCodec(); | |
1354 if (acm_send_codec) { | |
1355 codec = *acm_send_codec; | |
1343 return 0; | 1356 return 0; |
1344 } | 1357 } |
1345 return -1; | 1358 return -1; |
1346 } | 1359 } |
1347 | 1360 |
1348 int32_t Channel::GetRecCodec(CodecInst& codec) { | 1361 int32_t Channel::GetRecCodec(CodecInst& codec) { |
1349 return (audio_coding_->ReceiveCodec(&codec)); | 1362 return (audio_coding_->ReceiveCodec(&codec)); |
1350 } | 1363 } |
1351 | 1364 |
1352 int32_t Channel::SetSendCodec(const CodecInst& codec) { | 1365 int32_t Channel::SetSendCodec(const CodecInst& codec) { |
(...skipping 1755 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
3108 int64_t min_rtt = 0; | 3121 int64_t min_rtt = 0; |
3109 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3122 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3110 0) { | 3123 0) { |
3111 return 0; | 3124 return 0; |
3112 } | 3125 } |
3113 return rtt; | 3126 return rtt; |
3114 } | 3127 } |
3115 | 3128 |
3116 } // namespace voe | 3129 } // namespace voe |
3117 } // namespace webrtc | 3130 } // namespace webrtc |
OLD | NEW |