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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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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919 return true; | 919 return true; |
920 } | 920 } |
921 | 921 |
922 int WebRtcVoiceEngine::GetInputLevel() { | 922 int WebRtcVoiceEngine::GetInputLevel() { |
923 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 923 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
924 unsigned int ulevel; | 924 unsigned int ulevel; |
925 return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ? | 925 return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ? |
926 static_cast<int>(ulevel) : -1; | 926 static_cast<int>(ulevel) : -1; |
927 } | 927 } |
928 | 928 |
929 const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() { | 929 const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const { |
930 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); | 930 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
931 return codecs_; | 931 return codecs_; |
932 } | 932 } |
| 933 |
| 934 const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const { |
| 935 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
| 936 return codecs_; |
| 937 } |
933 | 938 |
934 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { | 939 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
935 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); | 940 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
936 RtpCapabilities capabilities; | 941 RtpCapabilities capabilities; |
937 capabilities.header_extensions.push_back(RtpHeaderExtension( | 942 capabilities.header_extensions.push_back(RtpHeaderExtension( |
938 kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId)); | 943 kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId)); |
939 capabilities.header_extensions.push_back( | 944 capabilities.header_extensions.push_back( |
940 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, | 945 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, |
941 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); | 946 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); |
942 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == | 947 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
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2559 } | 2564 } |
2560 } else { | 2565 } else { |
2561 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2566 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
2562 engine()->voe()->base()->StopPlayout(channel); | 2567 engine()->voe()->base()->StopPlayout(channel); |
2563 } | 2568 } |
2564 return true; | 2569 return true; |
2565 } | 2570 } |
2566 } // namespace cricket | 2571 } // namespace cricket |
2567 | 2572 |
2568 #endif // HAVE_WEBRTC_VOICE | 2573 #endif // HAVE_WEBRTC_VOICE |
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