Modifies AudioInputCallback::OnData and use media::AudioBus instead of plain byte vector (Relanding)

media/audio/audio_input_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_input_device.h"
 
 #include "base/basictypes.h"
 #include "base/bind.h"
+#include "base/memory/scoped_vector.h"
 #include "base/threading/thread_restrictions.h"
 #include "base/time/time.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/base/audio_bus.h"
 
 namespace media {
 
 // The number of shared memory buffer segments indicated to browser process
 // in order to avoid data overwriting. This number can be any positive number,
 // dependent how fast the renderer process can pick up captured data from
@@ skipping 13 matching lines @@
                       CaptureCallback* capture_callback);
   virtual ~AudioThreadCallback();
 
   virtual void MapSharedMemory() OVERRIDE;
 
   // Called whenever we receive notifications about pending data.
   virtual void Process(int pending_data) OVERRIDE;
 
  private:
   int current_segment_id_;
+  ScopedVector<media::AudioBus> audio_buses_;
   CaptureCallback* capture_callback_;
-  scoped_ptr<AudioBus> audio_bus_;
+
   DISALLOW_COPY_AND_ASSIGN(AudioThreadCallback);
 };
 
 AudioInputDevice::AudioInputDevice(
     scoped_ptr<AudioInputIPC> ipc,
     const scoped_refptr<base::SingleThreadTaskRunner>& io_task_runner)
     : ScopedTaskRunnerObserver(io_task_runner),
       callback_(NULL),
       ipc_(ipc.Pass()),
       state_(IDLE),
@@ skipping 212 matching lines @@
 AudioInputDevice::AudioThreadCallback::AudioThreadCallback(
     const AudioParameters& audio_parameters,
     base::SharedMemoryHandle memory,
     int memory_length,
     int total_segments,
     CaptureCallback* capture_callback)
     : AudioDeviceThread::Callback(audio_parameters, memory, memory_length,
                                   total_segments),
       current_segment_id_(0),
       capture_callback_(capture_callback) {
-  audio_bus_ = AudioBus::Create(audio_parameters_);
 }
 
 AudioInputDevice::AudioThreadCallback::~AudioThreadCallback() {
 }
 
 void AudioInputDevice::AudioThreadCallback::MapSharedMemory() {
   shared_memory_.Map(memory_length_);
+
+  // Create vector of audio buses by wrapping existing blocks of memory.
+  uint8* ptr = static_cast<uint8*>(shared_memory_.memory());
+  for (int i = 0; i < total_segments_; ++i) {
+    media::AudioInputBuffer* buffer =
+        reinterpret_cast<media::AudioInputBuffer*>(ptr);
+    scoped_ptr<media::AudioBus> audio_bus =
+        media::AudioBus::WrapMemory(audio_parameters_, buffer->audio);
+    audio_buses_.push_back(audio_bus.release());
+    ptr += segment_length_;
+  }
 }
 
 void AudioInputDevice::AudioThreadCallback::Process(int pending_data) {
   // The shared memory represents parameters, size of the data buffer and the
   // actual data buffer containing audio data. Map the memory into this
   // structure and parse out parameters and the data area.
   uint8* ptr = static_cast<uint8*>(shared_memory_.memory());
   ptr += current_segment_id_ * segment_length_;
   AudioInputBuffer* buffer = reinterpret_cast<AudioInputBuffer*>(ptr);
   // Usually this will be equal but in the case of low sample rate (e.g. 8kHz,
   // the buffer may be bigger (on mac at least)).
   DCHECK_GE(buffer->params.size,
             segment_length_ - sizeof(AudioInputBufferParameters));
   double volume = buffer->params.volume;
   bool key_pressed = buffer->params.key_pressed;
 
+  // Use pre-allocated audio bus wrapping existing block of shared memory.
+  media::AudioBus* audio_bus = audio_buses_[current_segment_id_];
+
+  // Deliver captured data to the client in floating point format
+  // and update the audio-delay measurement.
   int audio_delay_milliseconds = pending_data / bytes_per_ms_;
-  int16* memory = reinterpret_cast<int16*>(&buffer->audio[0]);
-  const int bytes_per_sample = sizeof(memory[0]);
+  capture_callback_->Capture(
+      audio_bus, audio_delay_milliseconds, volume, key_pressed);
 
   if (++current_segment_id_ >= total_segments_)
     current_segment_id_ = 0;
-
-  // Deinterleave each channel and convert to 32-bit floating-point
-  // with nominal range -1.0 -> +1.0.
-  audio_bus_->FromInterleaved(memory, audio_bus_->frames(), bytes_per_sample);
-
-  // Deliver captured data to the client in floating point format
-  // and update the audio-delay measurement.
-  capture_callback_->Capture(
-      audio_bus_.get(), audio_delay_milliseconds, volume, key_pressed);
 }
 
 }  // namespace media