| /* |
| * 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) |