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Side by Side Diff: content/renderer/media/webrtc_audio_device_impl.h

Issue 9805001: Move media/audio files into media namespace (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Fix various compiler errors Created 8 years, 9 months ago
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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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