OLD | NEW |
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> |
11 | 11 |
12 #include "base/basictypes.h" | 12 #include "base/basictypes.h" |
13 #include "base/compiler_specific.h" | 13 #include "base/compiler_specific.h" |
14 #include "base/memory/ref_counted.h" | 14 #include "base/memory/ref_counted.h" |
15 #include "base/memory/scoped_ptr.h" | 15 #include "base/memory/scoped_ptr.h" |
16 #include "base/message_loop_proxy.h" | 16 #include "base/message_loop_proxy.h" |
17 #include "base/time.h" | 17 #include "base/time.h" |
18 #include "content/common/content_export.h" | 18 #include "content/common/content_export.h" |
19 #include "content/renderer/media/audio_device.h" | |
20 #include "content/renderer/media/audio_input_device.h" | 19 #include "content/renderer/media/audio_input_device.h" |
| 20 #include "media/base/audio_renderer_sink.h" |
21 #include "third_party/webrtc/modules/audio_device/main/interface/audio_device.h" | 21 #include "third_party/webrtc/modules/audio_device/main/interface/audio_device.h" |
22 | 22 |
23 // A WebRtcAudioDeviceImpl instance implements the abstract interface | 23 // A WebRtcAudioDeviceImpl instance implements the abstract interface |
24 // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc:: | 24 // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc:: |
25 // VoiceEngine) to register this class as an external AudioDeviceModule (ADM). | 25 // VoiceEngine) to register this class as an external AudioDeviceModule (ADM). |
26 // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the | 26 // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the |
27 // session id that tells which device to use. The user can either get the | 27 // session id that tells which device to use. The user can either get the |
28 // session id from the MediaStream or use a value of 1 (AudioInputDeviceManager | 28 // session id from the MediaStream or use a value of 1 (AudioInputDeviceManager |
29 // ::kFakeOpenSessionId), the later will open the default device without going | 29 // ::kFakeOpenSessionId), the later will open the default device without going |
30 // through the MediaStream. The user can then call WebRtcAudioDeviceImpl:: | 30 // through the MediaStream. The user can then call WebRtcAudioDeviceImpl:: |
(...skipping 182 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
213 | 213 |
214 // webrtc::RefCountedModule implementation. | 214 // webrtc::RefCountedModule implementation. |
215 // The creator must call AddRef() after construction and use Release() | 215 // The creator must call AddRef() after construction and use Release() |
216 // to release the reference and delete this object. | 216 // to release the reference and delete this object. |
217 virtual int32_t AddRef() OVERRIDE; | 217 virtual int32_t AddRef() OVERRIDE; |
218 virtual int32_t Release() OVERRIDE; | 218 virtual int32_t Release() OVERRIDE; |
219 | 219 |
220 // We need this one to support runnable method tasks. | 220 // We need this one to support runnable method tasks. |
221 static bool ImplementsThreadSafeReferenceCounting() { return true; } | 221 static bool ImplementsThreadSafeReferenceCounting() { return true; } |
222 | 222 |
223 // AudioDevice::RenderCallback implementation. | 223 // media::AudioRendererSink::RenderCallback implementation. |
224 virtual int Render(const std::vector<float*>& audio_data, | 224 virtual int Render(const std::vector<float*>& audio_data, |
225 int number_of_frames, | 225 int number_of_frames, |
226 int audio_delay_milliseconds) OVERRIDE; | 226 int audio_delay_milliseconds) OVERRIDE; |
227 virtual void OnRenderError() OVERRIDE; | 227 virtual void OnRenderError() OVERRIDE; |
228 | 228 |
229 // AudioInputDevice::CaptureCallback implementation. | 229 // AudioInputDevice::CaptureCallback implementation. |
230 virtual void Capture(const std::vector<float*>& audio_data, | 230 virtual void Capture(const std::vector<float*>& audio_data, |
231 int number_of_frames, | 231 int number_of_frames, |
232 int audio_delay_milliseconds, | 232 int audio_delay_milliseconds, |
233 double volume) OVERRIDE; | 233 double volume) OVERRIDE; |
(...skipping 166 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
400 int ref_count_; | 400 int ref_count_; |
401 | 401 |
402 // Gives access to the message loop of the render thread on which this | 402 // Gives access to the message loop of the render thread on which this |
403 // object is created. | 403 // object is created. |
404 scoped_refptr<base::MessageLoopProxy> render_loop_; | 404 scoped_refptr<base::MessageLoopProxy> render_loop_; |
405 | 405 |
406 // Provides access to the native audio input layer in the browser process. | 406 // Provides access to the native audio input layer in the browser process. |
407 scoped_refptr<AudioInputDevice> audio_input_device_; | 407 scoped_refptr<AudioInputDevice> audio_input_device_; |
408 | 408 |
409 // Provides access to the native audio output layer in the browser process. | 409 // Provides access to the native audio output layer in the browser process. |
410 scoped_refptr<AudioDevice> audio_output_device_; | 410 scoped_refptr<media::AudioRendererSink> audio_output_device_; |
411 | 411 |
412 // Weak reference to the audio callback. | 412 // Weak reference to the audio callback. |
413 // The webrtc client defines |audio_transport_callback_| by calling | 413 // The webrtc client defines |audio_transport_callback_| by calling |
414 // RegisterAudioCallback(). | 414 // RegisterAudioCallback(). |
415 webrtc::AudioTransport* audio_transport_callback_; | 415 webrtc::AudioTransport* audio_transport_callback_; |
416 | 416 |
417 // Cached values of utilized audio parameters. Platform dependent. | 417 // Cached values of utilized audio parameters. Platform dependent. |
418 media::AudioParameters input_audio_parameters_; | 418 media::AudioParameters input_audio_parameters_; |
419 media::AudioParameters output_audio_parameters_; | 419 media::AudioParameters output_audio_parameters_; |
420 | 420 |
(...skipping 29 matching lines...) Expand all Loading... |
450 bool agc_is_enabled_; | 450 bool agc_is_enabled_; |
451 | 451 |
452 // Used for histograms of total recording and playout times. | 452 // Used for histograms of total recording and playout times. |
453 base::Time start_capture_time_; | 453 base::Time start_capture_time_; |
454 base::Time start_render_time_; | 454 base::Time start_render_time_; |
455 | 455 |
456 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); | 456 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); |
457 }; | 457 }; |
458 | 458 |
459 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 459 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
OLD | NEW |