Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(227)

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

Issue 1776243003: Revert of - Clean up unused voice engine DTMF code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_1
Patch Set: Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 1160 matching lines...) Expand 10 before | Expand all | Expand 10 after
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, int event, int duration_ms) { 1181 bool SendTelephoneEvent(int payload_type, uint8_t event,
1182 uint32_t duration_ms) {
1182 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1183 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1183 RTC_DCHECK(stream_); 1184 RTC_DCHECK(stream_);
1184 return stream_->SendTelephoneEvent(payload_type, event, duration_ms); 1185 return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
1185 } 1186 }
1186 1187
1187 void SetSend(bool send) { 1188 void SetSend(bool send) {
1188 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1189 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1189 send_ = send; 1190 send_ = send;
1190 UpdateSendState(); 1191 UpdateSendState();
1191 } 1192 }
(...skipping 1343 matching lines...) Expand 10 before | Expand all | Expand 10 after
2535 } 2536 }
2536 } else { 2537 } else {
2537 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2538 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2538 engine()->voe()->base()->StopPlayout(channel); 2539 engine()->voe()->base()->StopPlayout(channel);
2539 } 2540 }
2540 return true; 2541 return true;
2541 } 2542 }
2542 } // namespace cricket 2543 } // namespace cricket
2543 2544
2544 #endif // HAVE_WEBRTC_VOICE 2545 #endif // HAVE_WEBRTC_VOICE
OLDNEW
« no previous file with comments | « webrtc/media/engine/fakewebrtccall.cc ('k') | webrtc/modules/audio_coding/test/TwoWayCommunication.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698