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/*
* Copyright (C) 2017-2022 Igalia S.L. All rights reserved.
* Copyright (C) 2022 Metrological Group B.V.
*
* 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
*/
#include "config.h"
#if USE(GSTREAMER_WEBRTC)
#include "RealtimeIncomingAudioSourceGStreamer.h"
#include "GStreamerAudioData.h"
#include "GStreamerAudioStreamDescription.h"
#include <wtf/text/MakeString.h>
GST_DEBUG_CATEGORY(webkit_webrtc_incoming_audio_debug);
#define GST_CAT_DEFAULT webkit_webrtc_incoming_audio_debug
namespace WebCore {
RealtimeIncomingAudioSourceGStreamer::RealtimeIncomingAudioSourceGStreamer(AtomString&& audioTrackId)
: RealtimeIncomingSourceGStreamer(CaptureDevice { WTFMove(audioTrackId), CaptureDevice::DeviceType::Microphone, emptyString() })
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_incoming_audio_debug, "webkitwebrtcincomingaudio", 0, "WebKit WebRTC incoming audio");
});
}
RealtimeIncomingAudioSourceGStreamer::~RealtimeIncomingAudioSourceGStreamer()
{
stop();
}
const RealtimeMediaSourceSettings& RealtimeIncomingAudioSourceGStreamer::settings()
{
return m_currentSettings;
}
void RealtimeIncomingAudioSourceGStreamer::dispatchSample(GRefPtr<GstSample>&& sample)
{
ASSERT(isMainThread());
auto presentationTime = MediaTime(GST_TIME_AS_USECONDS(GST_BUFFER_PTS(gst_sample_get_buffer(sample.get()))), G_USEC_PER_SEC);
GStreamerAudioStreamDescription description;
GStreamerAudioData frames(WTFMove(sample), description.getInfo());
audioSamplesAvailable(presentationTime, frames, description, 0);
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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