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/*
* 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) && ENABLE(WEB_AUDIO)
#include "AudioBus.h"
#include "GStreamerAudioData.h"
#include "GStreamerAudioStreamDescription.h"
#include "Logging.h"
namespace WebCore {
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;
// Lazily initialize caps, the settings don't change so this is OK.
if (!m_caps || GST_AUDIO_INFO_CHANNELS(&m_info) != static_cast<int>(bus.numberOfChannels())) {
gst_audio_info_set_format(&m_info, GST_AUDIO_FORMAT_F32LE, m_currentSettings.sampleRate(), bus.numberOfChannels(), nullptr);
GST_AUDIO_INFO_LAYOUT(&m_info) = GST_AUDIO_LAYOUT_NON_INTERLEAVED;
m_caps = adoptGRef(gst_audio_info_to_caps(&m_info));
}
auto buffer = adoptGRef(gst_buffer_new());
GST_BUFFER_PTS(buffer.get()) = toGstClockTime(mediaTime);
GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_LIVE);
for (size_t channelIndex = 0; channelIndex < bus.numberOfChannels(); ++channelIndex) {
auto& channel = *bus.channel(channelIndex);
auto dataSize = sizeof(float) * channel.length();
bus.ref();
gst_buffer_append_memory(buffer.get(), gst_memory_new_wrapped(GST_MEMORY_FLAG_READONLY, channel.mutableData(), dataSize, 0, dataSize, &bus, reinterpret_cast<GDestroyNotify>(+[](gpointer data) {
auto bus = reinterpret_cast<AudioBus*>(data);
bus->deref();
})));
}
gst_buffer_add_audio_meta(buffer.get(), &m_info, numberOfFrames, nullptr);
#if GST_CHECK_VERSION(1, 20, 0)
if (bus.isSilent())
gst_buffer_add_audio_level_meta(buffer.get(), 127, FALSE);
#endif
auto sample = adoptGRef(gst_sample_new(buffer.get(), m_caps.get(), nullptr, nullptr));
GStreamerAudioData audioBuffer(WTFMove(sample), m_info);
GStreamerAudioStreamDescription description(&m_info);
audioSamplesAvailable(mediaTime, audioBuffer, description, numberOfFrames);
}
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER) && ENABLE(WEB_AUDIO)
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