1/*
2 * Copyright (C) 2018 Metrological Group B.V.
3 * Author: Thibault Saunier <tsaunier@igalia.com>
4 * Author: Alejandro G. Castro <alex@igalia.com>
5 *
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
10 *
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
15 *
16 * You should have received a copy of the GNU Library General Public License
17 * aint with this library; see the file COPYING.LIB. If not, write to
18 * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
20 */
21
22#pragma once
23
24#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
25#include "CaptureDevice.h"
26#include "GStreamerAudioCapturer.h"
27#include "GStreamerCaptureDevice.h"
28#include "RealtimeMediaSource.h"
29
30namespace WebCore {
31
32class GStreamerAudioCaptureSource : public RealtimeMediaSource {
33public:
34 static CaptureSourceOrError create(String&& deviceID, String&& hashSalt, const MediaConstraints*);
35 WEBCORE_EXPORT static AudioCaptureFactory& factory();
36
37 const RealtimeMediaSourceCapabilities& capabilities() override;
38 const RealtimeMediaSourceSettings& settings() override;
39
40 GstElement* pipeline() { return m_capturer->pipeline(); }
41 GStreamerCapturer* capturer() { return m_capturer.get(); }
42
43protected:
44 GStreamerAudioCaptureSource(GStreamerCaptureDevice, String&& hashSalt);
45 GStreamerAudioCaptureSource(String&& deviceID, String&& name, String&& hashSalt);
46 virtual ~GStreamerAudioCaptureSource();
47 void startProducingData() override;
48 void stopProducingData() override;
49 CaptureDevice::DeviceType deviceType() const override { return CaptureDevice::DeviceType::Microphone; }
50
51 mutable Optional<RealtimeMediaSourceCapabilities> m_capabilities;
52 mutable Optional<RealtimeMediaSourceSettings> m_currentSettings;
53
54private:
55 bool isCaptureSource() const final { return true; }
56 void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag>) final;
57
58 std::unique_ptr<GStreamerAudioCapturer> m_capturer;
59
60 static GstFlowReturn newSampleCallback(GstElement*, GStreamerAudioCaptureSource*);
61 void triggerSampleAvailable(GstSample*);
62};
63
64} // namespace WebCore
65
66#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
67