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/*
* Copyright (C) 2018 Metrological Group B.V.
* Copyright (C) 2020 Igalia S.L.
* Author: Thibault Saunier <tsaunier@igalia.com>
* Author: Alejandro G. Castro <alex@igalia.com>
*
* 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 ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
#include "MockRealtimeAudioSourceGStreamer.h"
#include "GStreamerCaptureDeviceManager.h"
#include "MockRealtimeMediaSourceCenter.h"
#include <gst/app/gstappsrc.h>
namespace WebCore {
static const double s_Tau = 2 * M_PI;
static const double s_BipBopDuration = 0.07;
static const double s_BipBopVolume = 0.5;
static const double s_BipFrequency = 1500;
static const double s_BopFrequency = 500;
static const double s_HumFrequency = 150;
static const double s_HumVolume = 0.1;
static const double s_NoiseFrequency = 3000;
static const double s_NoiseVolume = 0.05;
static HashSet<MockRealtimeAudioSource*>& allMockRealtimeAudioSourcesStorage()
{
static MainThreadNeverDestroyed<HashSet<MockRealtimeAudioSource*>> audioSources;
return audioSources;
}
const HashSet<MockRealtimeAudioSource*>& MockRealtimeAudioSourceGStreamer::allMockRealtimeAudioSources()
{
return allMockRealtimeAudioSourcesStorage();
}
CaptureSourceOrError MockRealtimeAudioSource::create(String&& deviceID, AtomString&& name, MediaDeviceHashSalts&& hashSalts, const MediaConstraints* constraints, PageIdentifier)
{
#ifndef NDEBUG
auto device = MockRealtimeMediaSourceCenter::mockDeviceWithPersistentID(deviceID);
ASSERT(device);
if (!device)
return CaptureSourceOrError({ "No mock microphone device"_s, MediaAccessDenialReason::PermissionDenied });
#endif
auto source = adoptRef(*new MockRealtimeAudioSourceGStreamer(WTFMove(deviceID), WTFMove(name), WTFMove(hashSalts)));
if (constraints) {
if (auto error = source->applyConstraints(*constraints))
return CaptureSourceOrError({ WTFMove(error->badConstraint), MediaAccessDenialReason::InvalidConstraint });
}
return CaptureSourceOrError(WTFMove(source));
}
Ref<MockRealtimeAudioSource> MockRealtimeAudioSourceGStreamer::createForMockAudioCapturer(String&& deviceID, AtomString&& name, MediaDeviceHashSalts&& hashSalts)
{
return adoptRef(*new MockRealtimeAudioSourceGStreamer(WTFMove(deviceID), WTFMove(name), WTFMove(hashSalts)));
}
MockRealtimeAudioSourceGStreamer::MockRealtimeAudioSourceGStreamer(String&& deviceID, AtomString&& name, MediaDeviceHashSalts&& hashSalts)
: MockRealtimeAudioSource(WTFMove(deviceID), WTFMove(name), WTFMove(hashSalts), { })
{
ensureGStreamerInitialized();
allMockRealtimeAudioSourcesStorage().add(this);
auto& singleton = GStreamerAudioCaptureDeviceManager::singleton();
auto device = singleton.gstreamerDeviceWithUID(this->captureDevice().persistentId());
ASSERT(device);
if (!device)
return;
device->setIsMockDevice(true);
m_capturer = adoptRef(*new GStreamerAudioCapturer(WTFMove(*device)));
m_capturer->addObserver(*this);
m_capturer->setupPipeline();
m_capturer->setSinkAudioCallback([this](auto&& sample, auto&& presentationTime) {
const auto& info = m_streamFormat->getInfo();
auto samplesCount = gst_buffer_get_size(gst_sample_get_buffer(sample.get())) / m_streamFormat->bytesPerFrame();
GStreamerAudioData data(WTFMove(sample), info);
audioSamplesAvailable(presentationTime, data, *m_streamFormat, samplesCount);
});
singleton.registerCapturer(m_capturer);
}
MockRealtimeAudioSourceGStreamer::~MockRealtimeAudioSourceGStreamer()
{
allMockRealtimeAudioSourcesStorage().remove(this);
m_capturer->stop();
m_capturer->removeObserver(*this);
auto& singleton = GStreamerAudioCaptureDeviceManager::singleton();
singleton.unregisterCapturer(*m_capturer);
}
void MockRealtimeAudioSourceGStreamer::startProducingData()
{
m_capturer->start();
MockRealtimeAudioSource::startProducingData();
}
void MockRealtimeAudioSourceGStreamer::stopProducingData()
{
m_capturer->stop();
MockRealtimeAudioSource::stopProducingData();
}
void MockRealtimeAudioSourceGStreamer::captureEnded()
{
captureFailed();
}
void MockRealtimeAudioSourceGStreamer::render(Seconds delta)
{
if (!m_bipBopBuffer.size())
reconfigure();
uint32_t totalFrameCount = GST_ROUND_UP_16(static_cast<size_t>(delta.seconds() * sampleRate()));
uint32_t frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
while (frameCount) {
uint32_t bipBopStart = m_samplesRendered % m_bipBopBuffer.size();
uint32_t bipBopRemain = m_bipBopBuffer.size() - bipBopStart;
uint32_t bipBopCount = std::min(frameCount, bipBopRemain);
ASSERT(m_streamFormat);
const auto& info = m_streamFormat->getInfo();
GRefPtr<GstBuffer> buffer = adoptGRef(gst_buffer_new_allocate(nullptr, bipBopCount * m_streamFormat->bytesPerFrame(), nullptr));
{
GstMappedBuffer map(buffer.get(), GST_MAP_WRITE);
if (muted())
webkitGstAudioFormatFillSilence(info.finfo, map.data(), map.size());
else {
memcpy(map.data(), &m_bipBopBuffer[bipBopStart], sizeof(float) * bipBopCount);
addHum(s_HumVolume, s_HumFrequency, sampleRate(), m_samplesRendered, reinterpret_cast<float*>(map.data()), bipBopCount);
}
}
m_samplesRendered += bipBopCount;
totalFrameCount -= bipBopCount;
frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
MediaTime presentationTime((m_samplesRendered * G_USEC_PER_SEC) / sampleRate(), G_USEC_PER_SEC);
GST_BUFFER_PTS(buffer.get()) = toGstClockTime(presentationTime);
GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_LIVE);
auto sample = adoptGRef(gst_sample_new(buffer.get(), m_caps.get(), nullptr, nullptr));
// Mock GstDevice is an appsrc, see webkitMockDeviceCreateElement().
ASSERT(GST_IS_APP_SRC(m_capturer->source()));
gst_app_src_push_sample(GST_APP_SRC_CAST(m_capturer->source()), sample.get());
}
}
void MockRealtimeAudioSourceGStreamer::addHum(float amplitude, float frequency, float sampleRate, uint64_t start, float *p, uint64_t count)
{
float humPeriod = sampleRate / frequency;
for (uint64_t i = start, end = start + count; i < end; ++i) {
float a = amplitude * sin(i * s_Tau / humPeriod);
a += *p;
*p++ = a;
}
}
void MockRealtimeAudioSourceGStreamer::reconfigure()
{
GstAudioInfo info;
auto rate = sampleRate();
size_t sampleCount = 2 * rate;
m_maximiumFrameCount = WTF::roundUpToPowerOfTwo(renderInterval().seconds() * sampleRate());
gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, rate, 1, nullptr);
m_streamFormat = GStreamerAudioStreamDescription(info);
m_caps = adoptGRef(gst_audio_info_to_caps(&info));
m_bipBopBuffer.resize(sampleCount);
m_bipBopBuffer.fill(0);
size_t bipBopSampleCount = ceil(s_BipBopDuration * rate);
size_t bipStart = 0;
size_t bopStart = rate;
addHum(s_BipBopVolume, s_BipFrequency, rate, 0, m_bipBopBuffer.data() + bipStart, bipBopSampleCount);
addHum(s_BipBopVolume, s_BopFrequency, rate, 0, m_bipBopBuffer.data() + bopStart, bipBopSampleCount);
if (!echoCancellation())
addHum(s_NoiseVolume, s_NoiseFrequency, rate, 0, m_bipBopBuffer.data(), sampleCount);
}
void MockRealtimeAudioSourceGStreamer::setInterruptedForTesting(bool isInterrupted)
{
m_isInterrupted = isInterrupted;
MockRealtimeAudioSource::setInterruptedForTesting(isInterrupted);
}
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
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