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Issue 1588693002: Revert of Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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1313 if (!voice_channel_) { 1313 if (!voice_channel_) {
1314 LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists."; 1314 LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
1315 return; 1315 return;
1316 } 1316 }
1317 1317
1318 if (!voice_channel_->SetOutputVolume(ssrc, volume)) { 1318 if (!voice_channel_->SetOutputVolume(ssrc, volume)) {
1319 ASSERT(false); 1319 ASSERT(false);
1320 } 1320 }
1321 } 1321 }
1322 1322
1323 void WebRtcSession::SetRawAudioSink( 1323 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
1324 uint32_t ssrc, 1324 rtc::scoped_ptr<AudioSinkInterface> sink) {
1325 const rtc::scoped_refptr<AudioSinkInterface>& sink) {
1326 ASSERT(signaling_thread()->IsCurrent()); 1325 ASSERT(signaling_thread()->IsCurrent());
1327 if (!voice_channel_) 1326 if (!voice_channel_)
1328 return; 1327 return;
1329 1328
1330 voice_channel_->SetRawAudioSink(ssrc, sink); 1329 voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
1331 } 1330 }
1332 1331
1333 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc, 1332 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
1334 cricket::VideoCapturer* camera) { 1333 cricket::VideoCapturer* camera) {
1335 ASSERT(signaling_thread()->IsCurrent()); 1334 ASSERT(signaling_thread()->IsCurrent());
1336 1335
1337 if (!video_channel_) { 1336 if (!video_channel_) {
1338 // |video_channel_| doesnt't exist. Probably because the remote end doesnt't 1337 // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
1339 // support video. 1338 // support video.
1340 LOG(LS_WARNING) << "Video not used in this call."; 1339 LOG(LS_WARNING) << "Video not used in this call.";
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2197 } 2196 }
2198 } 2197 }
2199 2198
2200 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel, 2199 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
2201 const rtc::SentPacket& sent_packet) { 2200 const rtc::SentPacket& sent_packet) {
2202 RTC_DCHECK(worker_thread()->IsCurrent()); 2201 RTC_DCHECK(worker_thread()->IsCurrent());
2203 media_controller_->call_w()->OnSentPacket(sent_packet); 2202 media_controller_->call_w()->OnSentPacket(sent_packet);
2204 } 2203 }
2205 2204
2206 } // namespace webrtc 2205 } // namespace webrtc
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