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+/*
+ * Copyright (c) 2012 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_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioFrame;
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+ // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
+ // operation, meaning src_audio and dst_audio must point to different
+ // buffers. It is the caller's responsibility to ensure that |dst_audio| is
+ // sufficiently large.
+ static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
+ int16_t* dst_audio);
+ // |frame.num_channels_| will be updated. This version checks for sufficient
+ // buffer size and that |num_channels_| is mono.
+ static int MonoToStereo(AudioFrame* frame);
+
+ // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
+ // operation, meaning |src_audio| and |dst_audio| may point to the same
+ // buffer.
+ static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
+ int16_t* dst_audio);
+ // |frame.num_channels_| will be updated. This version checks that
+ // |num_channels_| is stereo.
+ static int StereoToMono(AudioFrame* frame);
+
+ // Swap the left and right channels of |frame|. Fails silently if |frame| is
+ // not stereo.
+ static void SwapStereoChannels(AudioFrame* frame);
+
+ // Zeros out the audio and sets |frame.energy| to zero.
+ static void Mute(AudioFrame& frame);
+
+ static int Scale(float left, float right, AudioFrame& frame);
+
+ static int ScaleWithSat(float scale, AudioFrame& frame);
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_