blob: 84e73ad262d6bf9735df9853be11770df6477798 [file] [log] [blame]
/*
* Copyright (C) 2017-2019 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"
#include "Logging.h"
namespace WebCore {
RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
: RealtimeMediaSource(RealtimeMediaSource::Type::Audio, "remote audio"_s, WTFMove(audioTrackId))
, m_audioTrack(WTFMove(audioTrack))
{
ASSERT(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);
}
const RealtimeMediaSourceCapabilities& RealtimeIncomingAudioSource::capabilities()
{
return RealtimeMediaSourceCapabilities::emptyCapabilities();
}
const RealtimeMediaSourceSettings& RealtimeIncomingAudioSource::settings()
{
return m_currentSettings;
}
}
#endif // USE(LIBWEBRTC)