| /* |
| * Copyright (C) 2017 Igalia S.L |
| * |
| * 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. |
| */ |
| |
| #include "config.h" |
| |
| #if USE(LIBWEBRTC) && USE(GSTREAMER) |
| #include "RealtimeOutgoingAudioSourceLibWebRTC.h" |
| |
| namespace WebCore { |
| |
| RealtimeOutgoingAudioSourceLibWebRTC::RealtimeOutgoingAudioSourceLibWebRTC(Ref<MediaStreamTrackPrivate>&& audioSource) |
| : RealtimeOutgoingAudioSource(WTFMove(audioSource)) |
| { |
| } |
| |
| RealtimeOutgoingAudioSourceLibWebRTC::~RealtimeOutgoingAudioSourceLibWebRTC() |
| { |
| } |
| |
| Ref<RealtimeOutgoingAudioSource> RealtimeOutgoingAudioSource::create(Ref<MediaStreamTrackPrivate>&& audioSource) |
| { |
| return RealtimeOutgoingAudioSourceLibWebRTC::create(WTFMove(audioSource)); |
| } |
| |
| void RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, |
| size_t /* sampleCount */) |
| { |
| } |
| |
| void RealtimeOutgoingAudioSourceLibWebRTC::handleMutedIfNeeded() |
| { |
| } |
| |
| void RealtimeOutgoingAudioSourceLibWebRTC::sendSilence() |
| { |
| } |
| |
| void RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData() |
| { |
| } |
| |
| bool RealtimeOutgoingAudioSourceLibWebRTC::isReachingBufferedAudioDataHighLimit() |
| { |
| return false; |
| } |
| |
| bool RealtimeOutgoingAudioSourceLibWebRTC::isReachingBufferedAudioDataLowLimit() |
| { |
| return false; |
| } |
| |
| bool RealtimeOutgoingAudioSourceLibWebRTC::hasBufferedEnoughData() |
| { |
| return false; |
| } |
| |
| } // namespace WebCore |
| |
| #endif // USE(LIBWEBRTC) |