| Index: content/renderer/media/media_stream_audio_source.h
|
| diff --git a/content/renderer/media/media_stream_audio_source.h b/content/renderer/media/media_stream_audio_source.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..006bd3ddf0b484621fed4328a527073ba8069319
|
| --- /dev/null
|
| +++ b/content/renderer/media/media_stream_audio_source.h
|
| @@ -0,0 +1,73 @@
|
| +// Copyright 2014 The Chromium Authors. All rights reserved.
|
| +// Use of this source code is governed by a BSD-style license that can be
|
| +// found in the LICENSE file.
|
| +
|
| +#ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
|
| +#define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
|
| +
|
| +#include "base/compiler_specific.h"
|
| +#include "content/common/content_export.h"
|
| +#include "content/renderer/media/media_stream_source.h"
|
| +#include "content/renderer/media/webrtc_audio_capturer.h"
|
| +#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| +
|
| +namespace content {
|
| +
|
| +class CONTENT_EXPORT MediaStreamAudioSource
|
| + : NON_EXPORTED_BASE(public MediaStreamSource) {
|
| + public:
|
| + MediaStreamAudioSource(const StreamDeviceInfo& device_info,
|
| + const SourceStopCallback& stop_callback);
|
| + MediaStreamAudioSource();
|
| + virtual ~MediaStreamAudioSource();
|
| +
|
| + virtual void ApplyConstraints(const blink::WebMediaStreamTrack& track,
|
| + const blink::WebMediaConstraints& constraints,
|
| + const ConstraintsCallback& callback) OVERRIDE;
|
| +
|
| + void SetLocalAudioSource(webrtc::AudioSourceInterface* source) {
|
| + local_audio_source_ = source;
|
| + }
|
| +
|
| + void SetAudioCapturer(WebRtcAudioCapturer* capturer) {
|
| + DCHECK(!audio_capturer_);
|
| + audio_capturer_ = capturer;
|
| + }
|
| +
|
| + WebRtcAudioCapturer* GetAudioCapturer() const {
|
| + // TODO(perkj): |audio_capturer_| can currently be reconfigured to use
|
| + // another microphone even after it has been created since only one
|
| + // capturer is supported. See issue crbug/262117.
|
| + // It would make more sense if a WebRtcAudioCapturer represent one and only
|
| + // one audio source.
|
| + if (audio_capturer_ &&
|
| + device_info_.session_id == audio_capturer_->session_id()) {
|
| + return audio_capturer_;
|
| + }
|
| + return NULL;
|
| + }
|
| +
|
| + webrtc::AudioSourceInterface* local_audio_source() {
|
| + return local_audio_source_.get();
|
| + }
|
| +
|
| + protected:
|
| + virtual void DoStopSource() OVERRIDE;
|
| +
|
| + private:
|
| + StreamDeviceInfo device_info_;
|
| +
|
| + // This member holds an instance of webrtc::LocalAudioSource. This is used
|
| + // as a container for audio options.
|
| + // TODO(hclam): This should be merged with |audio_source_| such that it
|
| + // carries audio options.
|
| + scoped_refptr<webrtc::AudioSourceInterface> local_audio_source_;
|
| +
|
| + scoped_refptr<WebRtcAudioCapturer> audio_capturer_;
|
| +
|
| + DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource);
|
| +};
|
| +
|
| +} // namespace content
|
| +
|
| +#endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
|
|
|