1 files changed, 40 insertions, 0 deletions
diff --git a/chromium/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc b/chromium/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc
new file mode 100644
index 00000000000..593f4d95777
--- /dev/null
+++ b/chromium/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc
@@ -0,0 +1,40 @@
+// Copyright 2014 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "base/logging.h"
+#include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+
+namespace content {
+
+WebRtcAudioSinkAdapter::WebRtcAudioSinkAdapter(
+    webrtc::AudioTrackSinkInterface* sink)
+    : sink_(sink) {
+  DCHECK(sink);
+}
+
+WebRtcAudioSinkAdapter::~WebRtcAudioSinkAdapter() {
+}
+
+bool WebRtcAudioSinkAdapter::IsEqual(
+    const webrtc::AudioTrackSinkInterface* other) const {
+  return (other == sink_);
+}
+
+void WebRtcAudioSinkAdapter::OnData(const int16* audio_data,
+                                    int sample_rate,
+                                    int number_of_channels,
+                                    int number_of_frames) {
+  sink_->OnData(audio_data, 16, sample_rate, number_of_channels,
+                number_of_frames);
+}
+
+void WebRtcAudioSinkAdapter::OnSetFormat(
+    const media::AudioParameters& params) {
+  // No need to forward the OnSetFormat() callback to
+  // webrtc::AudioTrackSinkInterface sink since the sink will handle the
+  // format change in OnData().
+}
+
+}  // namespace content