Low-latency AudioInputStream implementation based on WASAPI for Windows.

content/renderer/media/audio_input_device.cc

 // Copyright (c) 2011 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 "content/renderer/media/audio_input_device.h"
 
 #include "base/bind.h"
 #include "base/message_loop.h"
 #include "base/time.h"
 #include "content/common/child_process.h"
@@ skipping 12 matching lines @@
       audio_delay_milliseconds_(0),
       volume_(1.0),
       stream_id_(0),
       session_id_(0),
       pending_device_ready_(false) {
   filter_ = RenderThreadImpl::current()->audio_input_message_filter();
   audio_data_.reserve(channels);
 #if defined(OS_MACOSX)
   VLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Mac OS X.";
   audio_parameters_.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
+#elif defined(OS_WIN)
+  // Use Core Audio (WASAPI) on Vista and higher. Use Wave for XP and lower.

[scherkus (not reviewing), 2011/10/18 21:12:30]

shouldn't this comment go in the windows code?

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Done.

+  VLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Windows.";
+  audio_parameters_.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
 #else
+  // TODO(henrika): add support for AUDIO_PCM_LOW_LATENCY on Linux as well.
   audio_parameters_.format = AudioParameters::AUDIO_PCM_LINEAR;
 #endif
   audio_parameters_.channels = channels;
   audio_parameters_.sample_rate = static_cast<int>(sample_rate);
   audio_parameters_.bits_per_sample = 16;
   audio_parameters_.samples_per_packet = buffer_size;
   for (int i = 0; i < channels; ++i) {
     float* channel_data = new float[buffer_size];
     audio_data_.push_back(channel_data);
   }
@@ skipping 271 matching lines @@
                                     bytes_per_sample,
                                     number_of_frames);
   }
 
   // Deliver captured data to the client in floating point format
   // and update the audio-delay measurement.
   callback_->Capture(audio_data_,
                      number_of_frames,
                      audio_delay_milliseconds_);
 }