haikuwebkit/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceGStre...

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3.3 KiB
C++

/*
* Copyright (C) 2020 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"
#include "MediaStreamAudioSource.h"
#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
#include "AudioBus.h"
#include "GStreamerAudioData.h"
#include "GStreamerAudioStreamDescription.h"
#include "Logging.h"
namespace WebCore {
static Vector<size_t> copyBusData(AudioBus& bus, GstBuffer* buffer, bool isMuted)
{
Vector<size_t> offsets;
GstMappedBuffer mappedBuffer(buffer, GST_MAP_WRITE);
if (isMuted) {
memset(mappedBuffer.data(), 0, mappedBuffer.size());
return offsets;
}
DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
offsets.reserveInitialCapacity(sizeof(size_t) * bus.numberOfChannels());
size_t size = mappedBuffer.size() / bus.numberOfChannels();
for (size_t channelIndex = 0; channelIndex < bus.numberOfChannels(); ++channelIndex) {
const auto& channel = *bus.channel(channelIndex);
auto offset = channelIndex * size;
memcpy(mappedBuffer.data() + offset, channel.data(), sizeof(float) * channel.length());
offsets.uncheckedAppend(offset);
}
return offsets;
}
void MediaStreamAudioSource::consumeAudio(AudioBus& bus, size_t numberOfFrames)
{
if (!bus.numberOfChannels() || bus.numberOfChannels() > 2) {
RELEASE_LOG_ERROR(Media, "MediaStreamAudioSource::consumeAudio(%p) trying to consume bus with %u channels", this, bus.numberOfChannels());
return;
}
MediaTime mediaTime((m_numberOfFrames * G_USEC_PER_SEC) / m_currentSettings.sampleRate(), G_USEC_PER_SEC);
m_numberOfFrames += numberOfFrames;
GstAudioInfo info;
gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, m_currentSettings.sampleRate(), bus.numberOfChannels(), nullptr);
GST_AUDIO_INFO_LAYOUT(&info) = GST_AUDIO_LAYOUT_NON_INTERLEAVED;
size_t size = GST_AUDIO_INFO_BPF(&info) * bus.numberOfChannels() * numberOfFrames;
auto caps = adoptGRef(gst_audio_info_to_caps(&info));
auto buffer = adoptGRef(gst_buffer_new_allocate(nullptr, size, nullptr));
auto offsets = copyBusData(bus, buffer.get(), muted());
#if GST_CHECK_VERSION(1, 16, 0)
gst_buffer_add_audio_meta(buffer.get(), &info, numberOfFrames, offsets.data());
#else
UNUSED_VARIABLE(offsets);
#endif
auto sample = adoptGRef(gst_sample_new(buffer.get(), caps.get(), nullptr, nullptr));
GStreamerAudioData audioBuffer(WTFMove(sample), info);
GStreamerAudioStreamDescription description(&info);
audioSamplesAvailable(mediaTime, audioBuffer, description, numberOfFrames);
}
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)