1/*
2 * Copyright (C) 2017 Igalia S.L
3 *
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
8 *
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
13 *
14 * You should have received a copy of the GNU Library General Public License
15 * aint with this library; see the file COPYING.LIB. If not, write to
16 * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20#pragma once
21
22#if USE(LIBWEBRTC)
23
24#include "GStreamerAudioStreamDescription.h"
25#include "GStreamerCommon.h"
26#include "RealtimeOutgoingAudioSource.h"
27
28#include <gst/audio/audio.h>
29
30namespace WebCore {
31
32class RealtimeOutgoingAudioSourceLibWebRTC final : public RealtimeOutgoingAudioSource {
33public:
34 static Ref<RealtimeOutgoingAudioSourceLibWebRTC> create(Ref<MediaStreamTrackPrivate>&& audioTrackPrivate)
35 {
36 return adoptRef(*new RealtimeOutgoingAudioSourceLibWebRTC(WTFMove(audioTrackPrivate)));
37 }
38
39private:
40 explicit RealtimeOutgoingAudioSourceLibWebRTC(Ref<MediaStreamTrackPrivate>&&);
41 ~RealtimeOutgoingAudioSourceLibWebRTC();
42
43 void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t) final;
44
45 bool isReachingBufferedAudioDataHighLimit() final;
46 bool isReachingBufferedAudioDataLowLimit() final;
47 bool hasBufferedEnoughData() final;
48
49 void pullAudioData() final;
50
51 GUniquePtr<GstAudioConverter> m_sampleConverter;
52 std::unique_ptr<GStreamerAudioStreamDescription> m_inputStreamDescription;
53 std::unique_ptr<GStreamerAudioStreamDescription> m_outputStreamDescription;
54
55 Lock m_adapterMutex;
56 GRefPtr<GstAdapter> m_adapter;
57 Vector<uint8_t> m_audioBuffer;
58};
59
60} // namespace WebCore
61
62#endif // USE(LIBWEBRTC)
63