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// Copyright 2013 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_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
#include <stddef.h>
#include <memory>
#include <vector>
#include "base/macros.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "base/time/time.h"
#include "content/common/content_export.h"
#include "content/public/renderer/media_stream_audio_sink.h"
#include "media/base/audio_converter.h"
#include "third_party/WebKit/public/platform/WebAudioSourceProvider.h"
#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
#include "third_party/WebKit/public/platform/WebVector.h"
namespace media {
class AudioBus;
class AudioConverter;
class AudioFifo;
class AudioParameters;
}
namespace blink {
class WebAudioSourceProviderClient;
}
namespace content {
// TODO(miu): This implementation should be renamed to WebAudioMediaStreamSink,
// as it should work as a provider for WebAudio from ANY MediaStreamAudioTrack.
// http://crbug.com/577874
//
// WebRtcLocalAudioSourceProvider provides a bridge between classes:
// MediaStreamAudioTrack ---> blink::WebAudioSourceProvider
//
// WebRtcLocalAudioSourceProvider works as a sink to the MediaStreamAudioTrack
// and stores the capture data to a FIFO. When the media stream is connected to
// WebAudio MediaStreamAudioSourceNode as a source provider,
// MediaStreamAudioSourceNode will periodically call provideInput() to get the
// data from the FIFO.
//
// All calls are protected by a lock.
class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
: public blink::WebAudioSourceProvider,
public media::AudioConverter::InputCallback,
public MediaStreamAudioSink {
public:
static const size_t kWebAudioRenderBufferSize;
explicit WebRtcLocalAudioSourceProvider(
const blink::WebMediaStreamTrack& track);
~WebRtcLocalAudioSourceProvider() override;
// MediaStreamAudioSink implementation.
void OnData(const media::AudioBus& audio_bus,
base::TimeTicks estimated_capture_time) override;
void OnSetFormat(const media::AudioParameters& params) override;
void OnReadyStateChanged(
blink::WebMediaStreamSource::ReadyState state) override;
// blink::WebAudioSourceProvider implementation.
void SetClient(blink::WebAudioSourceProviderClient* client) override;
void ProvideInput(const blink::WebVector<float*>& audio_data,
size_t number_of_frames) override;
// media::AudioConverter::Inputcallback implementation.
// This function is triggered by provideInput()on the WebAudio audio thread,
// so it has been under the protection of |lock_|.
double ProvideInput(media::AudioBus* audio_bus,
uint32_t frames_delayed) override;
// Method to allow the unittests to inject its own sink parameters to avoid
// query the hardware.
// TODO(xians,tommi): Remove and instead offer a way to inject the sink
// parameters so that the implementation doesn't rely on the global default
// hardware config but instead gets the parameters directly from the sink
// (WebAudio in this case). Ideally the unit test should be able to use that
// same mechanism to inject the sink parameters for testing.
void SetSinkParamsForTesting(const media::AudioParameters& sink_params);
private:
// Used to DCHECK that some methods are called on the capture audio thread.
base::ThreadChecker capture_thread_checker_;
std::unique_ptr<media::AudioConverter> audio_converter_;
std::unique_ptr<media::AudioFifo> fifo_;
std::unique_ptr<media::AudioBus> output_wrapper_;
bool is_enabled_;
media::AudioParameters source_params_;
media::AudioParameters sink_params_;
// Protects all the member variables above.
base::Lock lock_;
// Used to report the correct delay to |webaudio_source_|.
base::TimeTicks last_fill_;
// The audio track that this source provider is connected to.
blink::WebMediaStreamTrack track_;
// Flag to tell if the track has been stopped or not.
bool track_stopped_;
DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
};
} // namespace content
#endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
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