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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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| 1263 UpdateSendState(); | 1263 UpdateSendState(); |
| 1264 } | 1264 } |
| 1265 | 1265 |
| 1266 // AudioSource::Sink implementation. | 1266 // AudioSource::Sink implementation. |
| 1267 // This method is called on the audio thread. | 1267 // This method is called on the audio thread. |
| 1268 void OnData(const void* audio_data, | 1268 void OnData(const void* audio_data, |
| 1269 int bits_per_sample, | 1269 int bits_per_sample, |
| 1270 int sample_rate, | 1270 int sample_rate, |
| 1271 size_t number_of_channels, | 1271 size_t number_of_channels, |
| 1272 size_t number_of_frames) override { | 1272 size_t number_of_frames) override { |
| 1273 RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread()); | |
|
Henrik Grunell WebRTC
2016/09/14 07:27:04
Just for the record: this is not the check that fa
| |
| 1274 RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread()); | 1273 RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread()); |
| 1275 RTC_DCHECK(voe_audio_transport_); | 1274 RTC_DCHECK(voe_audio_transport_); |
| 1276 voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data, | 1275 voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data, |
| 1277 bits_per_sample, sample_rate, | 1276 bits_per_sample, sample_rate, |
| 1278 number_of_channels, number_of_frames); | 1277 number_of_channels, number_of_frames); |
| 1279 } | 1278 } |
| 1280 | 1279 |
| 1281 // Callback from the |source_| when it is going away. In case Start() has | 1280 // Callback from the |source_| when it is going away. In case Start() has |
| 1282 // never been called, this callback won't be triggered. | 1281 // never been called, this callback won't be triggered. |
| 1283 void OnClose() override { | 1282 void OnClose() override { |
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| 2661 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2660 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2662 const auto it = send_streams_.find(ssrc); | 2661 const auto it = send_streams_.find(ssrc); |
| 2663 if (it != send_streams_.end()) { | 2662 if (it != send_streams_.end()) { |
| 2664 return it->second->channel(); | 2663 return it->second->channel(); |
| 2665 } | 2664 } |
| 2666 return -1; | 2665 return -1; |
| 2667 } | 2666 } |
| 2668 } // namespace cricket | 2667 } // namespace cricket |
| 2669 | 2668 |
| 2670 #endif // HAVE_WEBRTC_VOICE | 2669 #endif // HAVE_WEBRTC_VOICE |
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