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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 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
7 #pragma once | 7 #pragma once |
8 | 8 |
9 #include <string> | 9 #include <string> |
10 #include <vector> | 10 #include <vector> |
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298 | 298 |
299 // Provides access to the native audio output layer in the browser process. | 299 // Provides access to the native audio output layer in the browser process. |
300 scoped_refptr<AudioDevice> audio_output_device_; | 300 scoped_refptr<AudioDevice> audio_output_device_; |
301 | 301 |
302 // Weak reference to the audio callback. | 302 // Weak reference to the audio callback. |
303 // The webrtc client defines |audio_transport_callback_| by calling | 303 // The webrtc client defines |audio_transport_callback_| by calling |
304 // RegisterAudioCallback(). | 304 // RegisterAudioCallback(). |
305 webrtc::AudioTransport* audio_transport_callback_; | 305 webrtc::AudioTransport* audio_transport_callback_; |
306 | 306 |
307 // Cached values of utilized audio parameters. Platform dependent. | 307 // Cached values of utilized audio parameters. Platform dependent. |
308 AudioParameters input_audio_parameters_; | 308 media::AudioParameters input_audio_parameters_; |
309 AudioParameters output_audio_parameters_; | 309 media::AudioParameters output_audio_parameters_; |
310 | 310 |
311 // Cached value of the current audio delay on the input/capture side. | 311 // Cached value of the current audio delay on the input/capture side. |
312 int input_delay_ms_; | 312 int input_delay_ms_; |
313 | 313 |
314 // Cached value of the current audio delay on the output/renderer side. | 314 // Cached value of the current audio delay on the output/renderer side. |
315 int output_delay_ms_; | 315 int output_delay_ms_; |
316 | 316 |
317 // Buffers used for temporary storage during capture/render callbacks. | 317 // Buffers used for temporary storage during capture/render callbacks. |
318 // Allocated during initialization to save stack. | 318 // Allocated during initialization to save stack. |
319 scoped_array<int16> input_buffer_; | 319 scoped_array<int16> input_buffer_; |
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333 int bytes_per_sample_; | 333 int bytes_per_sample_; |
334 | 334 |
335 bool initialized_; | 335 bool initialized_; |
336 bool playing_; | 336 bool playing_; |
337 bool recording_; | 337 bool recording_; |
338 | 338 |
339 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); | 339 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); |
340 }; | 340 }; |
341 | 341 |
342 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 342 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
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