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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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95 } | 95 } |
96 | 96 |
97 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { | 97 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
98 // TODO(solenberg): Tests call this function on a network thread, libjingle | 98 // TODO(solenberg): Tests call this function on a network thread, libjingle |
99 // calls on the worker thread. We should move towards always using a network | 99 // calls on the worker thread. We should move towards always using a network |
100 // thread. Then this check can be enabled. | 100 // thread. Then this check can be enabled. |
101 // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); | 101 // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
102 return false; | 102 return false; |
103 } | 103 } |
104 | 104 |
| 105 bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event, |
| 106 uint32_t duration_ms) { |
| 107 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| 108 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && |
| 109 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); |
| 110 } |
| 111 |
105 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | 112 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
106 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 113 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
107 webrtc::AudioSendStream::Stats stats; | 114 webrtc::AudioSendStream::Stats stats; |
108 stats.local_ssrc = config_.rtp.ssrc; | 115 stats.local_ssrc = config_.rtp.ssrc; |
109 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); | 116 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); |
110 ScopedVoEInterface<VoECodec> codec(voice_engine()); | 117 ScopedVoEInterface<VoECodec> codec(voice_engine()); |
111 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); | 118 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); |
112 | 119 |
113 webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); | 120 webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); |
114 stats.bytes_sent = call_stats.bytesSent; | 121 stats.bytes_sent = call_stats.bytesSent; |
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192 | 199 |
193 VoiceEngine* AudioSendStream::voice_engine() const { | 200 VoiceEngine* AudioSendStream::voice_engine() const { |
194 internal::AudioState* audio_state = | 201 internal::AudioState* audio_state = |
195 static_cast<internal::AudioState*>(audio_state_.get()); | 202 static_cast<internal::AudioState*>(audio_state_.get()); |
196 VoiceEngine* voice_engine = audio_state->voice_engine(); | 203 VoiceEngine* voice_engine = audio_state->voice_engine(); |
197 RTC_DCHECK(voice_engine); | 204 RTC_DCHECK(voice_engine); |
198 return voice_engine; | 205 return voice_engine; |
199 } | 206 } |
200 } // namespace internal | 207 } // namespace internal |
201 } // namespace webrtc | 208 } // namespace webrtc |
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