| /* |
| * Copyright (C) 2014 Igalia S.L |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser 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 |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with this library; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #ifndef AudioSourceProviderGStreamer_h |
| #define AudioSourceProviderGStreamer_h |
| |
| #if ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER) |
| |
| #include "AudioSourceProvider.h" |
| #include "GRefPtrGStreamer.h" |
| #include "MainThreadNotifier.h" |
| #include <gst/gst.h> |
| #include <wtf/Forward.h> |
| #include <wtf/Noncopyable.h> |
| |
| #if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) |
| #include "GStreamerAudioStreamDescription.h" |
| #include "MediaStreamTrackPrivate.h" |
| #include "WebAudioSourceProvider.h" |
| #endif |
| |
| typedef struct _GstAdapter GstAdapter; |
| typedef struct _GstAppSink GstAppSink; |
| |
| namespace WebCore { |
| |
| #if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) |
| class AudioSourceProviderGStreamer final : public WebAudioSourceProvider { |
| public: |
| static Ref<AudioSourceProviderGStreamer> create(MediaStreamTrackPrivate& source) |
| { |
| return adoptRef(*new AudioSourceProviderGStreamer(source)); |
| } |
| AudioSourceProviderGStreamer(MediaStreamTrackPrivate&); |
| #else |
| class AudioSourceProviderGStreamer : public AudioSourceProvider { |
| WTF_MAKE_FAST_ALLOCATED; |
| WTF_MAKE_NONCOPYABLE(AudioSourceProviderGStreamer); |
| public: |
| #endif |
| |
| AudioSourceProviderGStreamer(); |
| ~AudioSourceProviderGStreamer(); |
| |
| void configureAudioBin(GstElement* audioBin, GstElement* teePredecessor); |
| |
| void provideInput(AudioBus*, size_t framesToProcess) override; |
| void setClient(AudioSourceProviderClient*) override; |
| const AudioSourceProviderClient* client() const { return m_client; } |
| |
| void handleNewDeinterleavePad(GstPad*); |
| void deinterleavePadsConfigured(); |
| void handleRemovedDeinterleavePad(GstPad*); |
| |
| GstFlowReturn handleAudioBuffer(GstAppSink*); |
| GstElement* getAudioBin() const { return m_audioSinkBin.get(); } |
| void clearAdapters(); |
| |
| private: |
| GRefPtr<GstElement> m_pipeline; |
| enum MainThreadNotification { |
| DeinterleavePadsConfigured = 1 << 0, |
| }; |
| Ref<MainThreadNotifier<MainThreadNotification>> m_notifier; |
| GRefPtr<GstElement> m_audioSinkBin; |
| AudioSourceProviderClient* m_client; |
| int m_deinterleaveSourcePads; |
| GstAdapter* m_frontLeftAdapter; |
| GstAdapter* m_frontRightAdapter; |
| unsigned long m_deinterleavePadAddedHandlerId; |
| unsigned long m_deinterleaveNoMorePadsHandlerId; |
| unsigned long m_deinterleavePadRemovedHandlerId; |
| Lock m_adapterMutex; |
| }; |
| |
| } |
| #endif // ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER) |
| |
| #endif // AudioSourceProviderGStreamer_h |