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

Issue 21183002: Adding key press detection in the browser process. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 7 years, 4 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 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 7
8 #include <string> 8 #include <string>
9 #include <vector> 9 #include <vector>
10 10
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216 // audio processing. 216 // audio processing.
217 // The return value is the new microphone volume, in the range of |0, 255]. 217 // The return value is the new microphone volume, in the range of |0, 255].
218 // When the volume does not need to be updated, it returns 0. 218 // When the volume does not need to be updated, it returns 0.
219 virtual int CaptureData(const std::vector<int>& channels, 219 virtual int CaptureData(const std::vector<int>& channels,
220 const int16* audio_data, 220 const int16* audio_data,
221 int sample_rate, 221 int sample_rate,
222 int number_of_channels, 222 int number_of_channels,
223 int number_of_frames, 223 int number_of_frames,
224 int audio_delay_milliseconds, 224 int audio_delay_milliseconds,
225 int current_volume, 225 int current_volume,
226 bool need_audio_processing) = 0; 226 bool need_audio_processing,
227 bool key_pressed) = 0;
227 228
228 // Set the format for the capture audio parameters. 229 // Set the format for the capture audio parameters.
229 virtual void SetCaptureFormat(const media::AudioParameters& params) = 0; 230 virtual void SetCaptureFormat(const media::AudioParameters& params) = 0;
230 231
231 protected: 232 protected:
232 virtual ~WebRtcAudioCapturerSink() {} 233 virtual ~WebRtcAudioCapturerSink() {}
233 }; 234 };
234 235
235 // Note that this class inherits from webrtc::AudioDeviceModule but due to 236 // Note that this class inherits from webrtc::AudioDeviceModule but due to
236 // the high number of non-implemented methods, we move the cruft over to the 237 // the high number of non-implemented methods, we move the cruft over to the
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332 // WebRtcAudioCapturerSink implementation. 333 // WebRtcAudioCapturerSink implementation.
333 334
334 // Called on the AudioInputDevice worker thread. 335 // Called on the AudioInputDevice worker thread.
335 virtual int CaptureData(const std::vector<int>& channels, 336 virtual int CaptureData(const std::vector<int>& channels,
336 const int16* audio_data, 337 const int16* audio_data,
337 int sample_rate, 338 int sample_rate,
338 int number_of_channels, 339 int number_of_channels,
339 int number_of_frames, 340 int number_of_frames,
340 int audio_delay_milliseconds, 341 int audio_delay_milliseconds,
341 int current_volume, 342 int current_volume,
342 bool need_audio_processing) OVERRIDE; 343 bool need_audio_processing,
344 bool key_pressed) OVERRIDE;
343 345
344 // Called on the main render thread. 346 // Called on the main render thread.
345 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; 347 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
346 348
347 // WebRtcAudioRendererSource implementation. 349 // WebRtcAudioRendererSource implementation.
348 350
349 // Called on the AudioInputDevice worker thread. 351 // Called on the AudioInputDevice worker thread.
350 virtual void RenderData(uint8* audio_data, 352 virtual void RenderData(uint8* audio_data,
351 int number_of_channels, 353 int number_of_channels,
352 int number_of_frames, 354 int number_of_frames,
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400 // Stores latest microphone volume received in a CaptureData() callback. 402 // Stores latest microphone volume received in a CaptureData() callback.
401 // Range is [0, 255]. 403 // Range is [0, 255].
402 uint32_t microphone_volume_; 404 uint32_t microphone_volume_;
403 405
404 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); 406 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
405 }; 407 };
406 408
407 } // namespace content 409 } // namespace content
408 410
409 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ 411 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
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