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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 1782053002: Relanding https://codereview.webrtc.org/1715883002/ in pieces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 months ago
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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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1171 if (stream_) { 1171 if (stream_) {
1172 call_->DestroyAudioSendStream(stream_); 1172 call_->DestroyAudioSendStream(stream_);
1173 stream_ = nullptr; 1173 stream_ = nullptr;
1174 } 1174 }
1175 config_.rtp.extensions = extensions; 1175 config_.rtp.extensions = extensions;
1176 RTC_DCHECK(!stream_); 1176 RTC_DCHECK(!stream_);
1177 stream_ = call_->CreateAudioSendStream(config_); 1177 stream_ = call_->CreateAudioSendStream(config_);
1178 RTC_CHECK(stream_); 1178 RTC_CHECK(stream_);
1179 } 1179 }
1180 1180
1181 bool SendTelephoneEvent(int payload_type, uint8_t event, 1181 bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
1182 uint32_t duration_ms) {
1183 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1182 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1184 RTC_DCHECK(stream_); 1183 RTC_DCHECK(stream_);
1185 return stream_->SendTelephoneEvent(payload_type, event, duration_ms); 1184 return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
1186 } 1185 }
1187 1186
1188 void SetSend(bool send) { 1187 void SetSend(bool send) {
1189 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1188 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1190 send_ = send; 1189 send_ = send;
1191 UpdateSendState(); 1190 UpdateSendState();
1192 } 1191 }
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2536 } 2535 }
2537 } else { 2536 } else {
2538 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2537 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2539 engine()->voe()->base()->StopPlayout(channel); 2538 engine()->voe()->base()->StopPlayout(channel);
2540 } 2539 }
2541 return true; 2540 return true;
2542 } 2541 }
2543 } // namespace cricket 2542 } // namespace cricket
2544 2543
2545 #endif // HAVE_WEBRTC_VOICE 2544 #endif // HAVE_WEBRTC_VOICE
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