RealtimeOutgoingAudioSourceLibWebRTC.h [plain text]
#pragma once
#if USE(LIBWEBRTC)
#include "GStreamerAudioStreamDescription.h"
#include "GStreamerCommon.h"
#include "RealtimeOutgoingAudioSource.h"
#include <gst/audio/audio.h>
namespace WebCore {
class RealtimeOutgoingAudioSourceLibWebRTC final : public RealtimeOutgoingAudioSource {
public:
static Ref<RealtimeOutgoingAudioSourceLibWebRTC> create(Ref<MediaStreamTrackPrivate>&& audioTrackPrivate)
{
return adoptRef(*new RealtimeOutgoingAudioSourceLibWebRTC(WTFMove(audioTrackPrivate)));
}
private:
explicit RealtimeOutgoingAudioSourceLibWebRTC(Ref<MediaStreamTrackPrivate>&&);
~RealtimeOutgoingAudioSourceLibWebRTC();
void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t) final;
bool isReachingBufferedAudioDataHighLimit() final;
bool isReachingBufferedAudioDataLowLimit() final;
bool hasBufferedEnoughData() final;
void pullAudioData();
GUniquePtr<GstAudioConverter> m_sampleConverter;
std::unique_ptr<GStreamerAudioStreamDescription> m_inputStreamDescription;
std::unique_ptr<GStreamerAudioStreamDescription> m_outputStreamDescription;
Lock m_adapterMutex;
GRefPtr<GstAdapter> m_adapter;
Vector<uint8_t> m_audioBuffer;
};
}
#endif // USE(LIBWEBRTC)