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1 // Copyright (c) 2011 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2011 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/audio_input_device.h" | 5 #include "content/renderer/media/audio_input_device.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/message_loop.h" | 8 #include "base/message_loop.h" |
9 #include "base/time.h" | 9 #include "base/time.h" |
10 #include "content/common/child_process.h" | 10 #include "content/common/child_process.h" |
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23 audio_delay_milliseconds_(0), | 23 audio_delay_milliseconds_(0), |
24 volume_(1.0), | 24 volume_(1.0), |
25 stream_id_(0), | 25 stream_id_(0), |
26 session_id_(0), | 26 session_id_(0), |
27 pending_device_ready_(false) { | 27 pending_device_ready_(false) { |
28 filter_ = RenderThreadImpl::current()->audio_input_message_filter(); | 28 filter_ = RenderThreadImpl::current()->audio_input_message_filter(); |
29 audio_data_.reserve(channels); | 29 audio_data_.reserve(channels); |
30 #if defined(OS_MACOSX) | 30 #if defined(OS_MACOSX) |
31 VLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Mac OS X."; | 31 VLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Mac OS X."; |
32 audio_parameters_.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; | 32 audio_parameters_.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; |
| 33 #elif defined(OS_WIN) |
| 34 VLOG(1) << "Using AUDIO_PCM_LOW_LATENCY as input mode on Windows."; |
| 35 audio_parameters_.format = AudioParameters::AUDIO_PCM_LOW_LATENCY; |
33 #else | 36 #else |
| 37 // TODO(henrika): add support for AUDIO_PCM_LOW_LATENCY on Linux as well. |
34 audio_parameters_.format = AudioParameters::AUDIO_PCM_LINEAR; | 38 audio_parameters_.format = AudioParameters::AUDIO_PCM_LINEAR; |
35 #endif | 39 #endif |
36 audio_parameters_.channels = channels; | 40 audio_parameters_.channels = channels; |
37 audio_parameters_.sample_rate = static_cast<int>(sample_rate); | 41 audio_parameters_.sample_rate = static_cast<int>(sample_rate); |
38 audio_parameters_.bits_per_sample = 16; | 42 audio_parameters_.bits_per_sample = 16; |
39 audio_parameters_.samples_per_packet = buffer_size; | 43 audio_parameters_.samples_per_packet = buffer_size; |
40 for (int i = 0; i < channels; ++i) { | 44 for (int i = 0; i < channels; ++i) { |
41 float* channel_data = new float[buffer_size]; | 45 float* channel_data = new float[buffer_size]; |
42 audio_data_.push_back(channel_data); | 46 audio_data_.push_back(channel_data); |
43 } | 47 } |
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315 bytes_per_sample, | 319 bytes_per_sample, |
316 number_of_frames); | 320 number_of_frames); |
317 } | 321 } |
318 | 322 |
319 // Deliver captured data to the client in floating point format | 323 // Deliver captured data to the client in floating point format |
320 // and update the audio-delay measurement. | 324 // and update the audio-delay measurement. |
321 callback_->Capture(audio_data_, | 325 callback_->Capture(audio_data_, |
322 number_of_frames, | 326 number_of_frames, |
323 audio_delay_milliseconds_); | 327 audio_delay_milliseconds_); |
324 } | 328 } |
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