Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(84)

Side by Side Diff: talk/app/webrtc/remoteaudiosource.cc

Issue 1517253004: Fix build on clang/win. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years 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
« no previous file with comments | « no previous file | no next file » | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2014 Google Inc. 3 * Copyright 2014 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,
(...skipping 79 matching lines...) Expand 10 before | Expand all | Expand 10 after
90 RTC_DCHECK(sinks_.empty()); 90 RTC_DCHECK(sinks_.empty());
91 } 91 }
92 92
93 void RemoteAudioSource::Initialize(uint32_t ssrc, 93 void RemoteAudioSource::Initialize(uint32_t ssrc,
94 AudioProviderInterface* provider) { 94 AudioProviderInterface* provider) {
95 RTC_DCHECK(main_thread_->IsCurrent()); 95 RTC_DCHECK(main_thread_->IsCurrent());
96 // To make sure we always get notified when the provider goes out of scope, 96 // To make sure we always get notified when the provider goes out of scope,
97 // we register for callbacks here and not on demand in AddSink. 97 // we register for callbacks here and not on demand in AddSink.
98 if (provider) { // May be null in tests. 98 if (provider) { // May be null in tests.
99 provider->SetRawAudioSink( 99 provider->SetRawAudioSink(
100 ssrc, std::move(rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)))); 100 ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)));
101 } 101 }
102 } 102 }
103 103
104 MediaSourceInterface::SourceState RemoteAudioSource::state() const { 104 MediaSourceInterface::SourceState RemoteAudioSource::state() const {
105 RTC_DCHECK(main_thread_->IsCurrent()); 105 RTC_DCHECK(main_thread_->IsCurrent());
106 return state_; 106 return state_;
107 } 107 }
108 108
109 void RemoteAudioSource::SetVolume(double volume) { 109 void RemoteAudioSource::SetVolume(double volume) {
110 RTC_DCHECK(volume >= 0 && volume <= 10); 110 RTC_DCHECK(volume >= 0 && volume <= 10);
(...skipping 51 matching lines...) Expand 10 before | Expand all | Expand 10 after
162 } 162 }
163 163
164 void RemoteAudioSource::OnMessage(rtc::Message* msg) { 164 void RemoteAudioSource::OnMessage(rtc::Message* msg) {
165 RTC_DCHECK(main_thread_->IsCurrent()); 165 RTC_DCHECK(main_thread_->IsCurrent());
166 sinks_.clear(); 166 sinks_.clear();
167 state_ = MediaSourceInterface::kEnded; 167 state_ = MediaSourceInterface::kEnded;
168 FireOnChanged(); 168 FireOnChanged();
169 } 169 }
170 170
171 } // namespace webrtc 171 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698