blob: d6e971a1a652a3f1355b155f8da90b3c7207a0ac [file] [log] [blame]
/*
* Copyright (C) 2017 Apple Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted, provided that the following conditions
* are required to be met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
* ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
* CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#pragma once
#if USE(LIBWEBRTC)
#include "AudioSampleDataSource.h"
#include "RealtimeOutgoingAudioSource.h"
namespace webrtc {
class AudioTrackInterface;
class AudioTrackSinkInterface;
}
namespace WebCore {
class RealtimeOutgoingAudioSourceCocoa final : public RealtimeOutgoingAudioSource {
public:
static Ref<RealtimeOutgoingAudioSourceCocoa> create(Ref<MediaStreamTrackPrivate>&& audioSource) { return adoptRef(*new RealtimeOutgoingAudioSourceCocoa(WTFMove(audioSource))); }
private:
explicit RealtimeOutgoingAudioSourceCocoa(Ref<MediaStreamTrackPrivate>&&);
~RealtimeOutgoingAudioSourceCocoa();
void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t) final;
bool isReachingBufferedAudioDataHighLimit() final;
bool isReachingBufferedAudioDataLowLimit() final;
bool hasBufferedEnoughData() final;
void sourceUpdated() final;
void pullAudioData();
Ref<AudioSampleDataSource> m_sampleConverter;
CAAudioStreamDescription m_inputStreamDescription;
CAAudioStreamDescription m_outputStreamDescription;
Vector<uint8_t> m_audioBuffer;
uint64_t m_readCount { 0 };
uint64_t m_writeCount { 0 };
bool m_skippingAudioData { false };
};
} // namespace WebCore
#endif // USE(LIBWEBRTC)