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// 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/peer_connection_dependency_factory.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(int render_frame_id,
const StreamDeviceInfo& device_info,
const SourceStoppedCallback& stop_callback,
PeerConnectionDependencyFactory* factory);
MediaStreamAudioSource();
virtual ~MediaStreamAudioSource();
void AddTrack(const blink::WebMediaStreamTrack& track,
const blink::WebMediaConstraints& constraints,
const ConstraintsCallback& callback);
void SetLocalAudioSource(webrtc::AudioSourceInterface* source) {
local_audio_source_ = source;
}
void SetAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer) {
DCHECK(!audio_capturer_.get());
audio_capturer_ = capturer;
}
const scoped_refptr<WebRtcAudioCapturer>& GetAudioCapturer() {
return audio_capturer_;
}
webrtc::AudioSourceInterface* local_audio_source() {
return local_audio_source_.get();
}
protected:
void DoStopSource() override;
private:
const int render_frame_id_;
PeerConnectionDependencyFactory* const factory_;
// This member holds an instance of webrtc::LocalAudioSource. This is used
// as a container for 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_
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