Switch AudioRenderSink::Callback to use AudioBus.

media/audio/audio_output_device.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.
 
 #include "media/audio/audio_output_device.h"
 
 #include "base/debug/trace_event.h"
 #include "base/message_loop.h"
 #include "base/threading/thread_restrictions.h"
 #include "base/time.h"
@@ skipping 248 matching lines @@
     return;
   }
 
   // Convert the number of pending bytes in the render buffer
   // into milliseconds.
   int audio_delay_milliseconds = pending_data / bytes_per_ms_;
 
   TRACE_EVENT0("audio", "AudioOutputDevice::FireRenderCallback");
 
   // Update the audio-delay measurement then ask client to render audio.
-  size_t num_frames = render_callback_->Render(audio_data_,
-      audio_parameters_.frames_per_buffer(), audio_delay_milliseconds);
+  size_t num_frames = render_callback_->Render(
+      audio_bus_.get(), audio_bus_->frames(), audio_delay_milliseconds);
 
   // Interleave, scale, and clip to int.
-  // TODO(crogers/vrk): Figure out a way to avoid the float -> int -> float
-  // conversions that happen in the <audio> and WebRTC scenarios.
-  InterleaveFloatToInt(audio_data_, shared_memory_.memory(),
-      num_frames, audio_parameters_.bits_per_sample() / 8);
+  // TODO(dalecurtis): Remove this when we have float everywhere.
+  InterleaveFloatToInt(
+      *audio_bus_.get(), shared_memory_.memory(), audio_bus_->frames(),
+      audio_parameters_.bits_per_sample() / 8);
 
   // Let the host know we are done.
-  SetActualDataSizeInBytes(&shared_memory_, memory_length_,
+  SetActualDataSizeInBytes(
+      &shared_memory_, memory_length_,
       num_frames * audio_parameters_.GetBytesPerFrame());
 }
 
 }  // namespace media.