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/*
* Copyright (C) 2017 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"
#if USE(LIBWEBRTC) && USE(GSTREAMER)
#include "RealtimeOutgoingAudioSourceLibWebRTC.h"
#include "GStreamerAudioData.h"
#include "GStreamerAudioStreamDescription.h"
#include "LibWebRTCAudioFormat.h"
#include "LibWebRTCProvider.h"
#include "NotImplemented.h"
#include <wtf/TZoneMallocInlines.h>
namespace WebCore {
WTF_MAKE_TZONE_ALLOCATED_IMPL(RealtimeOutgoingAudioSourceLibWebRTC);
RealtimeOutgoingAudioSourceLibWebRTC::RealtimeOutgoingAudioSourceLibWebRTC(Ref<MediaStreamTrackPrivate>&& audioSource)
: RealtimeOutgoingAudioSource(WTFMove(audioSource))
{
m_adapter = adoptGRef(gst_adapter_new()),
m_sampleConverter = nullptr;
}
RealtimeOutgoingAudioSourceLibWebRTC::~RealtimeOutgoingAudioSourceLibWebRTC()
{
m_sampleConverter = nullptr;
}
Ref<RealtimeOutgoingAudioSource> RealtimeOutgoingAudioSource::create(Ref<MediaStreamTrackPrivate>&& audioSource)
{
return RealtimeOutgoingAudioSourceLibWebRTC::create(WTFMove(audioSource));
}
static inline GstAudioInfo libwebrtcAudioFormat(int sampleRate, size_t channelCount)
{
GstAudioFormat format = gst_audio_format_build_integer(
LibWebRTCAudioFormat::isSigned,
LibWebRTCAudioFormat::isBigEndian ? G_BIG_ENDIAN : G_LITTLE_ENDIAN,
LibWebRTCAudioFormat::sampleSize,
LibWebRTCAudioFormat::sampleSize);
GstAudioInfo info;
size_t libWebRTCChannelCount = channelCount >= 2 ? 2 : channelCount;
gst_audio_info_set_format(&info, format, sampleRate, libWebRTCChannelCount, nullptr);
return info;
}
void RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable(const MediaTime&, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t /* sampleCount */)
{
DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
auto data = static_cast<const GStreamerAudioData&>(audioData);
auto desc = static_cast<const GStreamerAudioStreamDescription&>(streamDescription);
if (m_sampleConverter && !gst_audio_info_is_equal(&m_inputStreamDescription, &desc.getInfo())) {
GST_ERROR_OBJECT(this, "FIXME - Audio format renegotiation is not possible yet!");
m_sampleConverter = nullptr;
}
if (!m_sampleConverter) {
m_inputStreamDescription = desc.getInfo();
m_outputStreamDescription = libwebrtcAudioFormat(LibWebRTCAudioFormat::sampleRate, desc.numberOfChannels());
m_sampleConverter.reset(gst_audio_converter_new(GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE, &m_inputStreamDescription,
&m_outputStreamDescription, nullptr));
}
{
Locker locker { m_adapterLock };
const auto& sample = data.getSample();
auto* buffer = gst_sample_get_buffer(sample.get());
gst_adapter_push(m_adapter.get(), gst_buffer_ref(buffer));
}
LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = Ref { *this }] {
protectedThis->pullAudioData();
});
}
void RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData()
{
if (!GST_AUDIO_INFO_IS_VALID(&m_inputStreamDescription) || !GST_AUDIO_INFO_IS_VALID(&m_outputStreamDescription)) {
GST_INFO("No stream description set yet.");
return;
}
size_t outChunkSampleCount = LibWebRTCAudioFormat::chunkSampleCount;
size_t outBufferSize = outChunkSampleCount * m_outputStreamDescription.bpf;
Locker locker { m_adapterLock };
size_t inChunkSampleCount = gst_audio_converter_get_in_frames(m_sampleConverter.get(), outChunkSampleCount);
size_t inBufferSize = inChunkSampleCount * m_inputStreamDescription.bpf;
while (gst_adapter_available(m_adapter.get()) > inBufferSize) {
auto inBuffer = adoptGRef(gst_adapter_take_buffer(m_adapter.get(), inBufferSize));
m_audioBuffer.grow(outBufferSize);
if (isSilenced())
webkitGstAudioFormatFillSilence(m_outputStreamDescription.finfo, m_audioBuffer.data(), outBufferSize);
else {
GstMappedBuffer inMap(inBuffer.get(), GST_MAP_READ);
gpointer in[1] = { inMap.data() };
gpointer out[1] = { m_audioBuffer.data() };
if (!gst_audio_converter_samples(m_sampleConverter.get(), static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
GST_ERROR("Could not convert samples.");
return;
}
}
sendAudioFrames(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, GST_AUDIO_INFO_RATE(&m_outputStreamDescription),
GST_AUDIO_INFO_CHANNELS(&m_outputStreamDescription), outChunkSampleCount);
}
}
bool RealtimeOutgoingAudioSourceLibWebRTC::isReachingBufferedAudioDataHighLimit()
{
notImplemented();
return false;
}
bool RealtimeOutgoingAudioSourceLibWebRTC::isReachingBufferedAudioDataLowLimit()
{
notImplemented();
return false;
}
bool RealtimeOutgoingAudioSourceLibWebRTC::hasBufferedEnoughData()
{
notImplemented();
return false;
}
} // namespace WebCore
#endif // USE(LIBWEBRTC)
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