Upgrade AudioBus to support wrapping, interleaving.

media/audio/audio_util.cc

 // Copyright (c) 2012 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.
 
 // Software adjust volume of samples, allows each audio stream its own
 // volume without impacting master volume for chrome and other applications.
 
 // Implemented as templates to allow 8, 16 and 32 bit implementations.
 // 8 bit is unsigned and biased by 128.
 
@@ skipping 164 matching lines @@
           reinterpret_cast<int32*>(buf),
           sample_count,
           volume,
           channels);
       return true;
     }
   }
   return false;
 }
 
+// TODO(dalecurtis): Delete once everywhere is using the AudioBus version:
+// http://crbug.com/120319.
 bool DeinterleaveAudioChannel(void* source,
                               float* destination,
                               int channels,
                               int channel_index,
                               int bytes_per_sample,
                               size_t number_of_frames) {
   switch (bytes_per_sample) {
     case 1:
     {
       uint8* source8 = reinterpret_cast<uint8*>(source) + channel_index;
@@ skipping 26 matching lines @@
       }
       return true;
     }
 
     default:
      break;
   }
   return false;
 }
 
-// |Format| is the destination type, |Fixed| is a type larger than |Format|
-// such that operations can be made without overflowing.
-template<class Format, class Fixed>
-static void InterleaveFloatToInt(const AudioBus* source,
-                                 void* dst_bytes, size_t number_of_frames) {
-  Format* destination = reinterpret_cast<Format*>(dst_bytes);
-  Fixed max_value = std::numeric_limits<Format>::max();
-  Fixed min_value = std::numeric_limits<Format>::min();
-
-  Format bias = 0;
-  if (!std::numeric_limits<Format>::is_signed) {
-    bias = max_value / 2;
-    max_value = bias;
-    min_value = -(bias - 1);
-  }
-
-  int channels = source->channels();
-  for (int i = 0; i < channels; ++i) {
-    const float* channel_data = source->channel(i);
-    for (size_t j = 0; j < number_of_frames; ++j) {
-      Fixed sample = max_value * channel_data[j];
-      if (sample > max_value)
-        sample = max_value;
-      else if (sample < min_value)
-        sample = min_value;
-
-      destination[j * channels + i] = static_cast<Format>(sample) + bias;
-    }
-  }
-}
-
-void InterleaveFloatToInt(const AudioBus* source, void* dst,
-                          size_t number_of_frames, int bytes_per_sample) {
-  switch (bytes_per_sample) {
-    case 1:
-      InterleaveFloatToInt<uint8, int32>(source, dst, number_of_frames);
-      break;
-    case 2:
-      InterleaveFloatToInt<int16, int32>(source, dst, number_of_frames);
-      break;
-    case 4:
-      InterleaveFloatToInt<int32, int64>(source, dst, number_of_frames);
-      break;
-    default:
-      break;
-  }
-}
-
 // TODO(enal): use template specialization and size-specific intrinsics.
 //             Call is on the time-critical path, and by using SSE/AVX
 //             instructions we can speed things up by ~4-8x, more for the case
 //             when we have to adjust volume as well.
 template<class Format, class Fixed, int min_value, int max_value, int bias>
 static void MixStreams(Format* dst, Format* src, int count, float volume) {
   if (volume == 0.0f)
     return;
   if (volume == 1.0f) {
     // Most common case -- no need to adjust volume.
@@ skipping 258 matching lines @@
   //    out performance was degraded compared to XP.
   //  - The regression was fixed in Windows 7 and most configurations will work
   //    with 2, but some (e.g., some Sound Blasters) still need 3.
   //  - Some XP configurations (even multi-processor ones) also need 3.
   return (base::win::GetVersion() == base::win::VERSION_VISTA) ? 4 : 3;
 }
 
 #endif
 
 }  // namespace media