1 files changed, 72 insertions, 0 deletions
diff --git a/talk/app/webrtc/remoteaudiosource.cc b/talk/app/webrtc/remoteaudiosource.cc
new file mode 100644
index 0000000000..41f3d8798a
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+++ b/talk/app/webrtc/remoteaudiosource.cc
@@ -0,0 +1,72 @@
+/*
+ * libjingle
+ * Copyright 2014 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/remoteaudiosource.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() {
+  return new rtc::RefCountedObject<RemoteAudioSource>();
+}
+
+RemoteAudioSource::RemoteAudioSource() {
+}
+
+RemoteAudioSource::~RemoteAudioSource() {
+  ASSERT(audio_observers_.empty());
+}
+
+MediaSourceInterface::SourceState RemoteAudioSource::state() const {
+  return MediaSourceInterface::kLive;
+}
+
+void RemoteAudioSource::SetVolume(double volume) {
+  ASSERT(volume >= 0 && volume <= 10);
+  for (AudioObserverList::iterator it = audio_observers_.begin();
+       it != audio_observers_.end(); ++it) {
+    (*it)->OnSetVolume(volume);
+  }
+}
+
+void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
+  ASSERT(observer != NULL);
+  ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(),
+                   observer) == audio_observers_.end());
+  audio_observers_.push_back(observer);
+}
+
+void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
+  ASSERT(observer != NULL);
+  audio_observers_.remove(observer);
+}
+
+}  // namespace webrtc