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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 | 9 |
| 10 #include "base/basictypes.h" | 10 #include "base/basictypes.h" |
| 11 #include "base/compiler_specific.h" | 11 #include "base/compiler_specific.h" |
| 12 #include "base/logging.h" | 12 #include "base/logging.h" |
| 13 #include "base/memory/ref_counted.h" | 13 #include "base/memory/ref_counted.h" |
| 14 #include "base/memory/scoped_ptr.h" | 14 #include "base/memory/scoped_ptr.h" |
| 15 #include "base/threading/thread_checker.h" | 15 #include "base/threading/thread_checker.h" |
| 16 #include "content/common/content_export.h" | 16 #include "content/common/content_export.h" |
| 17 #include "content/renderer/media/webrtc_audio_capturer.h" | 17 #include "content/renderer/media/webrtc_audio_capturer.h" |
| 18 #include "content/renderer/media/webrtc_audio_device_not_impl.h" | 18 #include "content/renderer/media/webrtc_audio_device_not_impl.h" |
| 19 #include "content/renderer/media/webrtc_audio_renderer.h" | 19 #include "content/renderer/media/webrtc_audio_renderer.h" |
| 20 #include "media/base/audio_capturer_source.h" | 20 #include "media/base/audio_capturer_source.h" |
| 21 #include "media/base/audio_renderer_sink.h" | 21 #include "media/base/audio_renderer_sink.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 then call |
| 28 // session id from the MediaStream or use a value of 1 (AudioInputDeviceManager | 28 // WebRtcAudioDeviceImpl::StartPlayout() and |
| 29 // ::kFakeOpenSessionId), the later will open the default device without going | 29 // WebRtcAudioDeviceImpl::StartRecording() from the render process to initiate |
| 30 // through the MediaStream. The user can then call WebRtcAudioDeviceImpl:: | 30 // and start audio rendering and capturing in the browser process. IPC is |
| 31 // StartPlayout() and WebRtcAudioDeviceImpl::StartRecording() from the render | 31 // utilized to set up the media streams. |
| 32 // process to initiate and start audio rendering and capturing in the browser | |
| 33 // process. IPC is utilized to set up the media streams. | |
| 34 // | 32 // |
| 35 // Usage example: | 33 // Usage example: |
| 36 // | 34 // |
| 37 // using namespace webrtc; | 35 // using namespace webrtc; |
| 38 // | 36 // |
| 39 // { | 37 // { |
| 40 // scoped_refptr<WebRtcAudioDeviceImpl> external_adm; | 38 // scoped_refptr<WebRtcAudioDeviceImpl> external_adm; |
| 41 // external_adm = new WebRtcAudioDeviceImpl(); | 39 // external_adm = new WebRtcAudioDeviceImpl(); |
| 42 // external_adm->SetSessionId(1); | 40 // external_adm->SetSessionId(session_id); |
| 43 // VoiceEngine* voe = VoiceEngine::Create(); | 41 // VoiceEngine* voe = VoiceEngine::Create(); |
| 44 // VoEBase* base = VoEBase::GetInterface(voe); | 42 // VoEBase* base = VoEBase::GetInterface(voe); |
| 45 // base->Init(external_adm); | 43 // base->Init(external_adm); |
| 46 // int ch = base->CreateChannel(); | 44 // int ch = base->CreateChannel(); |
| 47 // ... | 45 // ... |
| 48 // base->StartReceive(ch) | 46 // base->StartReceive(ch) |
| 49 // base->StartPlayout(ch); | 47 // base->StartPlayout(ch); |
| 50 // base->StartSending(ch); | 48 // base->StartSending(ch); |
| 51 // ... | 49 // ... |
| 52 // <== full-duplex audio session with AGC enabled ==> | 50 // <== full-duplex audio session with AGC enabled ==> |
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| 382 // Stores latest microphone volume received in a CaptureData() callback. | 380 // Stores latest microphone volume received in a CaptureData() callback. |
| 383 // Range is [0, 255]. | 381 // Range is [0, 255]. |
| 384 uint32_t microphone_volume_; | 382 uint32_t microphone_volume_; |
| 385 | 383 |
| 386 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); | 384 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); |
| 387 }; | 385 }; |
| 388 | 386 |
| 389 } // namespace content | 387 } // namespace content |
| 390 | 388 |
| 391 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 389 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
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