| /* |
| * Copyright (C) 2018 Metrological Group B.V. |
| * Author: Thibault Saunier <tsaunier@igalia.com> |
| * Author: Alejandro G. Castro <alex@igalia.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public License |
| * aint with this library; see the file COPYING.LIB. If not, write to |
| * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #pragma once |
| |
| #if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER) |
| #include "CaptureDevice.h" |
| #include "GStreamerAudioCapturer.h" |
| #include "GStreamerCaptureDevice.h" |
| #include "RealtimeMediaSource.h" |
| |
| namespace WebCore { |
| |
| class GStreamerAudioCaptureSource : public RealtimeMediaSource { |
| public: |
| static CaptureSourceOrError create(String&& deviceID, String&& hashSalt, const MediaConstraints*); |
| WEBCORE_EXPORT static AudioCaptureFactory& factory(); |
| |
| const RealtimeMediaSourceCapabilities& capabilities() override; |
| const RealtimeMediaSourceSettings& settings() override; |
| |
| GstElement* pipeline() { return m_capturer->pipeline(); } |
| GStreamerCapturer* capturer() { return m_capturer.get(); } |
| |
| protected: |
| GStreamerAudioCaptureSource(GStreamerCaptureDevice, String&& hashSalt); |
| GStreamerAudioCaptureSource(String&& deviceID, String&& name, String&& hashSalt); |
| virtual ~GStreamerAudioCaptureSource(); |
| void startProducingData() override; |
| void stopProducingData() override; |
| CaptureDevice::DeviceType deviceType() const override { return CaptureDevice::DeviceType::Microphone; } |
| |
| mutable Optional<RealtimeMediaSourceCapabilities> m_capabilities; |
| mutable Optional<RealtimeMediaSourceSettings> m_currentSettings; |
| |
| private: |
| bool isCaptureSource() const final { return true; } |
| void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag>) final; |
| |
| std::unique_ptr<GStreamerAudioCapturer> m_capturer; |
| |
| static GstFlowReturn newSampleCallback(GstElement*, GStreamerAudioCaptureSource*); |
| void triggerSampleAvailable(GstSample*); |
| }; |
| |
| } // namespace WebCore |
| |
| #endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER) |