Move functionality out from AudioFrame and into AudioFrameOperations.

webrtc/modules/utility/include/audio_frame_operations.h

Index: webrtc/modules/utility/include/audio_frame_operations.h
diff --git a/webrtc/modules/utility/include/audio_frame_operations.h b/webrtc/modules/utility/include/audio_frame_operations.h
index e12e3e561be8d439fe5a59709149a875fc2b0509..5ae5ae1472360ed75910e8352dd4a7fef45a4854 100644
--- a/webrtc/modules/utility/include/audio_frame_operations.h
+++ b/webrtc/modules/utility/include/audio_frame_operations.h
@@ -11,17 +11,26 @@
 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
 #define WEBRTC_MODULES_UTILITY_INCLUDE_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.

[the sun, 2016/10/19 09:23:17]

Perhaps address that comment in a follow-up? Would indeed be nice to remove the
class and make this namespace audio_frame_operations { instead.

[aleloi, 2016/10/20 08:27:05]

Acknowledged.

 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.

[aleloi, 2016/10/18 11:38:04]

I'll try to improve this comment somewhat.

+  static void AddFrames(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
@@ -53,11 +62,33 @@ class AudioFrameOperations {
   static void Mute(AudioFrame* frame, bool previous_frame_muted,
                    bool current_frame_muted);
 
+  // Zero out contents of frame.
+  static void Mute(AudioFrame* frame);
+
+  // Reset all members to their default state (except do not modify
+  // the contents of |frame->data_|).
+  static void Reset(AudioFrame* frame);
+
+  // Apply a right shift by 1 on the samples in |frame|.
+  static void ShiftDown(AudioFrame* frame);

[the sun, 2016/10/19 09:23:17]

Consider renaming to "HalfGain()", "ApplyHalfGain()", "ScaleToHalf()" or perhaps
"ScaledDown6dB()"

[aleloi, 2016/10/20 08:27:05]

Done.

+
   static int Scale(float left, float right, AudioFrame& frame);
 
   static int ScaleWithSat(float scale, AudioFrame& frame);
+
+  static void UpdateFrame(int id,
+                          uint32_t timestamp,
+                          const int16_t* data,
+                          size_t samples_per_channel,
+                          int sample_rate_hz,
+                          AudioFrame::SpeechType speech_type,
+                          AudioFrame::VADActivity vad_activity,
+                          size_t num_channels,
+                          AudioFrame* frame);
 };
 
+int16_t ClampToInt16(int32_t input);

[the sun, 2016/10/19 09:23:17]

Add TODO to move this into a separate file (we probably have similar utilities
which could benefit from being in the same place).

[aleloi, 2016/10/20 08:27:05]

Done.

+
 }  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_