Move functionality out from AudioFrame and into AudioFrameOperations.

webrtc/audio/utility/audio_frame_operations.h

Index: webrtc/audio/utility/audio_frame_operations.h
diff --git a/webrtc/modules/utility/include/audio_frame_operations.h b/webrtc/audio/utility/audio_frame_operations.h
similarity index 65%
copy from webrtc/modules/utility/include/audio_frame_operations.h
copy to webrtc/audio/utility/audio_frame_operations.h
index e12e3e561be8d439fe5a59709149a875fc2b0509..b272db4538e30f34dbe0ba4aa6460f7986eb92d9 100644
--- a/webrtc/modules/utility/include/audio_frame_operations.h
+++ b/webrtc/audio/utility/audio_frame_operations.h
@@ -8,25 +8,34 @@
  *  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_
+#ifndef WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
 
+#include "webrtc/modules/include/module_common_types.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:
+  // Add samples in |frame_to_add| with samples in |result_frame|
+  // putting the results in |results_frame|.  The fields
+  // |vad_activity_| and |speech_type_| of the result frame are
+  // updated. If |result_frame| is empty (|samples_per_channel_|==0),
+  // the samples in |frame_to_add| are added to it.  The number of
+  // channels and number of samples per channel must match except when
+  // |result_frame| is empty.
+  static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
+
   // 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,
+  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.
@@ -35,7 +44,8 @@ class AudioFrameOperations {
   // 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,
+  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.
@@ -50,14 +60,23 @@ class AudioFrameOperations {
   //  |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
   // !|previous_frame_muted| &&  |current_frame_muted| - Fade-out at frame end.
   // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
-  static void Mute(AudioFrame* frame, bool previous_frame_muted,
+  static void Mute(AudioFrame* frame,
+                   bool previous_frame_muted,
                    bool current_frame_muted);
 
+  // Zero out contents of frame.
+  static void Mute(AudioFrame* frame);
+
+  // Halve samples in |frame|.
+  static void ApplyHalfGain(AudioFrame* frame);
+
   static int Scale(float left, float right, AudioFrame& frame);
 
   static int ScaleWithSat(float scale, AudioFrame& frame);
 };
 
+int16_t ClampToInt16(int32_t input);
+
 }  // namespace webrtc
 
-#endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#endif  // WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_