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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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704 capture_start_ntp_time_ms_ = | 704 capture_start_ntp_time_ms_ = |
705 audioFrame->ntp_time_ms_ - audioFrame->elapsed_time_ms_; | 705 audioFrame->ntp_time_ms_ - audioFrame->elapsed_time_ms_; |
706 } | 706 } |
707 } | 707 } |
708 } | 708 } |
709 | 709 |
710 return muted ? MixerParticipant::AudioFrameInfo::kMuted | 710 return muted ? MixerParticipant::AudioFrameInfo::kMuted |
711 : MixerParticipant::AudioFrameInfo::kNormal; | 711 : MixerParticipant::AudioFrameInfo::kNormal; |
712 } | 712 } |
713 | 713 |
| 714 AudioMixer::Source::AudioFrameWithInfo Channel::GetAudioFrameWithInfo( |
| 715 int sample_rate_hz) { |
| 716 mix_audio_frame_.sample_rate_hz_ = sample_rate_hz; |
| 717 |
| 718 const auto frame_info = GetAudioFrameWithMuted(-1, &mix_audio_frame_); |
| 719 |
| 720 using FrameInfo = AudioMixer::Source::AudioFrameInfo; |
| 721 FrameInfo new_audio_frame_info = FrameInfo::kError; |
| 722 switch (frame_info) { |
| 723 case MixerParticipant::AudioFrameInfo::kNormal: |
| 724 new_audio_frame_info = FrameInfo::kNormal; |
| 725 break; |
| 726 case MixerParticipant::AudioFrameInfo::kMuted: |
| 727 new_audio_frame_info = FrameInfo::kMuted; |
| 728 break; |
| 729 case MixerParticipant::AudioFrameInfo::kError: |
| 730 new_audio_frame_info = FrameInfo::kError; |
| 731 break; |
| 732 } |
| 733 return {&mix_audio_frame_, new_audio_frame_info}; |
| 734 } |
| 735 |
714 int32_t Channel::NeededFrequency(int32_t id) const { | 736 int32_t Channel::NeededFrequency(int32_t id) const { |
715 WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId), | 737 WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId), |
716 "Channel::NeededFrequency(id=%d)", id); | 738 "Channel::NeededFrequency(id=%d)", id); |
717 | 739 |
718 int highestNeeded = 0; | 740 int highestNeeded = 0; |
719 | 741 |
720 // Determine highest needed receive frequency | 742 // Determine highest needed receive frequency |
721 int32_t receiveFrequency = audio_coding_->ReceiveFrequency(); | 743 int32_t receiveFrequency = audio_coding_->ReceiveFrequency(); |
722 | 744 |
723 // Return the bigger of playout and receive frequency in the ACM. | 745 // Return the bigger of playout and receive frequency in the ACM. |
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3243 int64_t min_rtt = 0; | 3265 int64_t min_rtt = 0; |
3244 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3266 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3245 0) { | 3267 0) { |
3246 return 0; | 3268 return 0; |
3247 } | 3269 } |
3248 return rtt; | 3270 return rtt; |
3249 } | 3271 } |
3250 | 3272 |
3251 } // namespace voe | 3273 } // namespace voe |
3252 } // namespace webrtc | 3274 } // namespace webrtc |
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