/* * Copyright (C) 2017 Apple Inc. All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer * in the documentation and/or other materials provided with the * distribution. * 3. Neither the name of Google Inc. nor the names of its contributors * may be used to endorse or promote products derived from this * software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #include "config.h" #include "RealtimeIncomingAudioSource.h" #if USE(LIBWEBRTC) #include "LibWebRTCAudioFormat.h" namespace WebCore { RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId) : RealtimeMediaSource(WTFMove(audioTrackId), RealtimeMediaSource::Type::Audio, String()) , m_audioTrack(WTFMove(audioTrack)) { notifyMutedChange(!m_audioTrack); } RealtimeIncomingAudioSource::~RealtimeIncomingAudioSource() { stop(); } void RealtimeIncomingAudioSource::startProducingData() { if (m_audioTrack) m_audioTrack->AddSink(this); } void RealtimeIncomingAudioSource::stopProducingData() { if (m_audioTrack) m_audioTrack->RemoveSink(this); } void RealtimeIncomingAudioSource::setSourceTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& track) { ASSERT(!m_audioTrack); ASSERT(track); m_audioTrack = WTFMove(track); notifyMutedChange(!m_audioTrack); if (isProducingData()) m_audioTrack->AddSink(this); } const RealtimeMediaSourceCapabilities& RealtimeIncomingAudioSource::capabilities() const { return RealtimeMediaSourceCapabilities::emptyCapabilities(); } const RealtimeMediaSourceSettings& RealtimeIncomingAudioSource::settings() const { return m_currentSettings; } } #endif // USE(LIBWEBRTC)