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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/webrtc_audio_device_impl.h" | 5 #include "content/renderer/media/webrtc_audio_device_impl.h" |
| 6 | 6 |
| 7 #include "base/bind.h" | 7 #include "base/bind.h" |
| 8 #include "base/string_util.h" | 8 #include "base/string_util.h" |
| 9 #include "base/win/windows_version.h" | 9 #include "base/win/windows_version.h" |
| 10 #include "content/renderer/media/audio_hardware.h" | 10 #include "content/renderer/media/audio_hardware.h" |
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| 52 } | 52 } |
| 53 | 53 |
| 54 int32_t WebRtcAudioDeviceImpl::Release() { | 54 int32_t WebRtcAudioDeviceImpl::Release() { |
| 55 int ret = base::subtle::Barrier_AtomicIncrement(&ref_count_, -1); | 55 int ret = base::subtle::Barrier_AtomicIncrement(&ref_count_, -1); |
| 56 if (ret == 0) { | 56 if (ret == 0) { |
| 57 delete this; | 57 delete this; |
| 58 } | 58 } |
| 59 return ret; | 59 return ret; |
| 60 } | 60 } |
| 61 | 61 |
| 62 void WebRtcAudioDeviceImpl::Render( | 62 size_t WebRtcAudioDeviceImpl::Render( |
| 63 const std::vector<float*>& audio_data, | 63 const std::vector<float*>& audio_data, |
| 64 size_t number_of_frames, | 64 size_t number_of_frames, |
| 65 size_t audio_delay_milliseconds) { | 65 size_t audio_delay_milliseconds) { |
| 66 DCHECK_LE(number_of_frames, output_buffer_size_); | 66 DCHECK_LE(number_of_frames, output_buffer_size_); |
| 67 | 67 |
| 68 // Store the reported audio delay locally. | 68 // Store the reported audio delay locally. |
| 69 output_delay_ms_ = audio_delay_milliseconds; | 69 output_delay_ms_ = audio_delay_milliseconds; |
| 70 | 70 |
| 71 const int channels = audio_data.size(); | 71 const int channels = audio_data.size(); |
| 72 DCHECK_LE(channels, output_channels_); | 72 DCHECK_LE(channels, output_channels_); |
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| 104 // with nominal range -1.0 -> +1.0 to match the callback format. | 104 // with nominal range -1.0 -> +1.0 to match the callback format. |
| 105 for (int channel_index = 0; channel_index < channels; ++channel_index) { | 105 for (int channel_index = 0; channel_index < channels; ++channel_index) { |
| 106 media::DeinterleaveAudioChannel( | 106 media::DeinterleaveAudioChannel( |
| 107 output_buffer_.get(), | 107 output_buffer_.get(), |
| 108 audio_data[channel_index], | 108 audio_data[channel_index], |
| 109 channels, | 109 channels, |
| 110 channel_index, | 110 channel_index, |
| 111 bytes_per_sample_, | 111 bytes_per_sample_, |
| 112 number_of_frames); | 112 number_of_frames); |
| 113 } | 113 } |
| 114 return number_of_frames; |
| 114 } | 115 } |
| 115 | 116 |
| 116 void WebRtcAudioDeviceImpl::Capture( | 117 void WebRtcAudioDeviceImpl::Capture( |
| 117 const std::vector<float*>& audio_data, | 118 const std::vector<float*>& audio_data, |
| 118 size_t number_of_frames, | 119 size_t number_of_frames, |
| 119 size_t audio_delay_milliseconds) { | 120 size_t audio_delay_milliseconds) { |
| 120 DCHECK_LE(number_of_frames, input_buffer_size_); | 121 DCHECK_LE(number_of_frames, input_buffer_size_); |
| 121 | 122 |
| 122 // Store the reported audio delay locally. | 123 // Store the reported audio delay locally. |
| 123 input_delay_ms_ = audio_delay_milliseconds; | 124 input_delay_ms_ = audio_delay_milliseconds; |
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| 966 } | 967 } |
| 967 | 968 |
| 968 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const { | 969 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const { |
| 969 NOTIMPLEMENTED(); | 970 NOTIMPLEMENTED(); |
| 970 return -1; | 971 return -1; |
| 971 } | 972 } |
| 972 | 973 |
| 973 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) { | 974 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) { |
| 974 session_id_ = session_id; | 975 session_id_ = session_id; |
| 975 } | 976 } |
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