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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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/webrtc_audio_capturer.h" | 5 #include "content/renderer/media/webrtc_audio_capturer.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/logging.h" | 8 #include "base/logging.h" |
9 #include "base/metrics/histogram.h" | 9 #include "base/metrics/histogram.h" |
10 #include "base/string_util.h" | 10 #include "base/string_util.h" |
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78 if (!buffer_size) { | 78 if (!buffer_size) { |
79 DLOG(ERROR) << "Unsupported sample-rate: " << sample_rate; | 79 DLOG(ERROR) << "Unsupported sample-rate: " << sample_rate; |
80 return false; | 80 return false; |
81 } | 81 } |
82 | 82 |
83 media::AudioParameters::Format format = | 83 media::AudioParameters::Format format = |
84 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; | 84 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; |
85 | 85 |
86 // bits_per_sample is always 16 for now. | 86 // bits_per_sample is always 16 for now. |
87 int bits_per_sample = 16; | 87 int bits_per_sample = 16; |
88 int channels = ChannelLayoutToChannelCount(channel_layout); | 88 |
89 params_.Reset(format, channel_layout, channels, 0, | 89 params_.Reset(format, channel_layout, 0, sample_rate, bits_per_sample, |
90 sample_rate, bits_per_sample, buffer_size); | 90 buffer_size); |
91 buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]); | 91 buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]); |
92 | 92 |
93 return true; | 93 return true; |
94 } | 94 } |
95 | 95 |
96 int16* buffer() const { return buffer_.get(); } | 96 int16* buffer() const { return buffer_.get(); } |
97 const media::AudioParameters& params() const { return params_; } | 97 const media::AudioParameters& params() const { return params_; } |
98 | 98 |
99 private: | 99 private: |
100 ~ConfiguredBuffer() {} | 100 ~ConfiguredBuffer() {} |
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394 for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it) | 394 for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it) |
395 (*it)->OnCaptureDeviceStopped(); | 395 (*it)->OnCaptureDeviceStopped(); |
396 } | 396 } |
397 | 397 |
398 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const { | 398 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const { |
399 base::AutoLock auto_lock(lock_); | 399 base::AutoLock auto_lock(lock_); |
400 return buffer_->params(); | 400 return buffer_->params(); |
401 } | 401 } |
402 | 402 |
403 } // namespace content | 403 } // namespace content |
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