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// Copyright (c) 2012 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_WEBAUDIO_CAPTURER_SOURCE_H_
#define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
#include "base/memory/ref_counted.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_capturer_source.h"
#include "media/base/audio_fifo.h"
#include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h"
#include "third_party/WebKit/public/platform/WebVector.h"
namespace content {
class WebRtcAudioCapturer;
class WebRtcLocalAudioTrack;
// WebAudioCapturerSource is the missing link between
// WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
//
// 1. WebKit calls the setFormat() method setting up the basic stream format
// (channels, and sample-rate).
// 2. consumeAudio() is called periodically by WebKit which dispatches the
// audio stream to the WebRtcLocalAudioTrack::Capture() method.
class WebAudioCapturerSource
: public base::RefCountedThreadSafe<WebAudioCapturerSource>,
public blink::WebAudioDestinationConsumer {
public:
WebAudioCapturerSource();
// WebAudioDestinationConsumer implementation.
// setFormat() is called early on, so that we can configure the audio track.
virtual void setFormat(size_t number_of_channels, float sample_rate) OVERRIDE;
// MediaStreamAudioDestinationNode periodically calls consumeAudio().
// Called on the WebAudio audio thread.
virtual void consumeAudio(const blink::WebVector<const float*>& audio_data,
size_t number_of_frames) OVERRIDE;
// Called when the WebAudioCapturerSource is hooking to a media audio track.
// |track| is the sink of the data flow. |source_provider| is the source of
// the data flow where stream information like delay, volume, key_pressed,
// is stored.
void Start(WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer);
// Called when the media audio track is stopping.
void Stop();
protected:
friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
virtual ~WebAudioCapturerSource();
private:
// Used to DCHECK that some methods are called on the correct thread.
base::ThreadChecker thread_checker_;
// The audio track this WebAudioCapturerSource is feeding data to.
// WebRtcLocalAudioTrack is reference counted, and owning this object.
// To avoid circular reference, a raw pointer is kept here.
WebRtcLocalAudioTrack* track_;
// A raw pointer to the capturer to get audio processing params like
// delay, volume, key_pressed information.
// This |capturer_| is guaranteed to outlive this object.
WebRtcAudioCapturer* capturer_;
media::AudioParameters params_;
// Flag to help notify the |track_| when the audio format has changed.
bool audio_format_changed_;
// Wraps data coming from HandleCapture().
scoped_ptr<media::AudioBus> wrapper_bus_;
// Bus for reading from FIFO and calling the CaptureCallback.
scoped_ptr<media::AudioBus> capture_bus_;
// Handles mismatch between WebAudio buffer size and WebRTC.
scoped_ptr<media::AudioFifo> fifo_;
// Buffer to pass audio data to WebRtc.
scoped_ptr<int16[]> audio_data_;
// Synchronizes HandleCapture() with AudioCapturerSource calls.
base::Lock lock_;
bool started_;
DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
};
} // namespace content
#endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
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