1 files changed, 46 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
new file mode 100644
index 00000000000..5743c3641de
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+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object receiving raw output audio from test
+// applications.
+class AudioSink {
+ public:
+  AudioSink() {}
+  virtual ~AudioSink() {}
+
+  // Writes |num_samples| from |audio| to the AudioSink. Returns true if
+  // successful, otherwise false.
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
+
+  // Writes |audio_frame| to the AudioSink. Returns true if successful,
+  // otherwise false.
+  bool WriteAudioFrame(const AudioFrame& audio_frame) {
+    return WriteArray(
+        audio_frame.data_,
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+  }
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(AudioSink);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_