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

Issue 9965076: Revert 130180 - Move media/audio files into media namespace (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src/
Patch Set: Created 8 years, 8 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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409 409
410 // Provides access to the native audio output layer in the browser process. 410 // Provides access to the native audio output layer in the browser process.
411 scoped_refptr<AudioDevice> audio_output_device_; 411 scoped_refptr<AudioDevice> audio_output_device_;
412 412
413 // Weak reference to the audio callback. 413 // Weak reference to the audio callback.
414 // The webrtc client defines |audio_transport_callback_| by calling 414 // The webrtc client defines |audio_transport_callback_| by calling
415 // RegisterAudioCallback(). 415 // RegisterAudioCallback().
416 webrtc::AudioTransport* audio_transport_callback_; 416 webrtc::AudioTransport* audio_transport_callback_;
417 417
418 // Cached values of utilized audio parameters. Platform dependent. 418 // Cached values of utilized audio parameters. Platform dependent.
419 media::AudioParameters input_audio_parameters_; 419 AudioParameters input_audio_parameters_;
420 media::AudioParameters output_audio_parameters_; 420 AudioParameters output_audio_parameters_;
421 421
422 // Cached value of the current audio delay on the input/capture side. 422 // Cached value of the current audio delay on the input/capture side.
423 int input_delay_ms_; 423 int input_delay_ms_;
424 424
425 // Cached value of the current audio delay on the output/renderer side. 425 // Cached value of the current audio delay on the output/renderer side.
426 int output_delay_ms_; 426 int output_delay_ms_;
427 427
428 // Buffers used for temporary storage during capture/render callbacks. 428 // Buffers used for temporary storage during capture/render callbacks.
429 // Allocated during initialization to save stack. 429 // Allocated during initialization to save stack.
430 scoped_array<int16> input_buffer_; 430 scoped_array<int16> input_buffer_;
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447 bool playing_; 447 bool playing_;
448 bool recording_; 448 bool recording_;
449 449
450 // Local copy of the current Automatic Gain Control state. 450 // Local copy of the current Automatic Gain Control state.
451 bool agc_is_enabled_; 451 bool agc_is_enabled_;
452 452
453 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); 453 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
454 }; 454 };
455 455
456 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ 456 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
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