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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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