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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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1501 int success = -1; | 1501 int success = -1; |
1502 audio_coding_->QueryEncoder([&](AudioEncoder const* encoder) { | 1502 audio_coding_->QueryEncoder([&](AudioEncoder const* encoder) { |
1503 if (encoder) { | 1503 if (encoder) { |
1504 *enabled = encoder->GetDtx(); | 1504 *enabled = encoder->GetDtx(); |
1505 success = 0; | 1505 success = 0; |
1506 } | 1506 } |
1507 }); | 1507 }); |
1508 return success; | 1508 return success; |
1509 } | 1509 } |
1510 | 1510 |
1511 bool Channel::EnableAudioNetworkAdaptor(const std::string& config_string) { | 1511 bool Channel::EnableAudioNetworkAdaptor(const std::string& config_string, |
| 1512 int min_receiver_frame_length_ms, |
| 1513 int max_receiver_frame_length_ms) { |
1512 bool success = false; | 1514 bool success = false; |
1513 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { | 1515 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
1514 if (*encoder) { | 1516 if (*encoder) { |
1515 success = (*encoder)->EnableAudioNetworkAdaptor( | 1517 success = (*encoder)->EnableAudioNetworkAdaptor( |
1516 config_string, Clock::GetRealTimeClock()); | 1518 config_string, min_receiver_frame_length_ms, |
| 1519 max_receiver_frame_length_ms, Clock::GetRealTimeClock()); |
1517 } | 1520 } |
1518 }); | 1521 }); |
1519 return success; | 1522 return success; |
1520 } | 1523 } |
1521 | 1524 |
1522 void Channel::DisableAudioNetworkAdaptor() { | 1525 void Channel::DisableAudioNetworkAdaptor() { |
1523 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { | 1526 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
1524 if (*encoder) | 1527 if (*encoder) |
1525 (*encoder)->DisableAudioNetworkAdaptor(); | 1528 (*encoder)->DisableAudioNetworkAdaptor(); |
1526 }); | 1529 }); |
1527 } | 1530 } |
1528 | 1531 |
1529 void Channel::SetReceiverFrameLengthRange(int min_frame_length_ms, | |
1530 int max_frame_length_ms) { | |
1531 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { | |
1532 if (*encoder) { | |
1533 (*encoder)->SetReceiverFrameLengthRange(min_frame_length_ms, | |
1534 max_frame_length_ms); | |
1535 } | |
1536 }); | |
1537 } | |
1538 | |
1539 int32_t Channel::RegisterExternalTransport(Transport* transport) { | 1532 int32_t Channel::RegisterExternalTransport(Transport* transport) { |
1540 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), | 1533 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
1541 "Channel::RegisterExternalTransport()"); | 1534 "Channel::RegisterExternalTransport()"); |
1542 | 1535 |
1543 rtc::CritScope cs(&_callbackCritSect); | 1536 rtc::CritScope cs(&_callbackCritSect); |
1544 if (_externalTransport) { | 1537 if (_externalTransport) { |
1545 _engineStatisticsPtr->SetLastError( | 1538 _engineStatisticsPtr->SetLastError( |
1546 VE_INVALID_OPERATION, kTraceError, | 1539 VE_INVALID_OPERATION, kTraceError, |
1547 "RegisterExternalTransport() external transport already enabled"); | 1540 "RegisterExternalTransport() external transport already enabled"); |
1548 return -1; | 1541 return -1; |
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3243 int64_t min_rtt = 0; | 3236 int64_t min_rtt = 0; |
3244 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3237 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3245 0) { | 3238 0) { |
3246 return 0; | 3239 return 0; |
3247 } | 3240 } |
3248 return rtt; | 3241 return rtt; |
3249 } | 3242 } |
3250 | 3243 |
3251 } // namespace voe | 3244 } // namespace voe |
3252 } // namespace webrtc | 3245 } // namespace webrtc |
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