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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 "GStreamerAudioCaptureSource.h"
#include "GStreamerAudioData.h"
#include "GStreamerAudioStreamDescription.h"
#include "GStreamerCaptureDeviceManager.h"
#include <gst/app/gstappsink.h>
#include <gst/gst.h>
#include <wtf/NeverDestroyed.h>
#include <wtf/text/MakeString.h>
namespace WebCore {
static DoubleCapabilityRange defaultVolumeCapability()
{
return { 0.0, 1.0 };
}
const static RealtimeMediaSourceCapabilities::EchoCancellation defaultEchoCancellationCapability = RealtimeMediaSourceCapabilities::EchoCancellation::OnOrOff;
GST_DEBUG_CATEGORY(webkit_audio_capture_source_debug);
#define GST_CAT_DEFAULT webkit_audio_capture_source_debug
class GStreamerAudioCaptureSourceFactory : public AudioCaptureFactory {
public:
CaptureSourceOrError createAudioCaptureSource(const CaptureDevice& device, MediaDeviceHashSalts&& hashSalts, const MediaConstraints* constraints, std::optional<PageIdentifier>) final
{
return GStreamerAudioCaptureSource::create(String { device.persistentId() }, WTFMove(hashSalts), constraints);
}
private:
CaptureDeviceManager& audioCaptureDeviceManager() final { return GStreamerAudioCaptureDeviceManager::singleton(); }
const Vector<CaptureDevice>& speakerDevices() const final { return GStreamerAudioCaptureDeviceManager::singleton().speakerDevices(); }
};
static GStreamerAudioCaptureSourceFactory& libWebRTCAudioCaptureSourceFactory()
{
static NeverDestroyed<GStreamerAudioCaptureSourceFactory> factory;
return factory.get();
}
CaptureSourceOrError GStreamerAudioCaptureSource::create(String&& deviceID, MediaDeviceHashSalts&& hashSalts, const MediaConstraints* constraints)
{
auto device = GStreamerAudioCaptureDeviceManager::singleton().gstreamerDeviceWithUID(deviceID);
if (!device) {
auto errorMessage = makeString("GStreamerAudioCaptureSource::create(): GStreamer did not find the device: "_s, deviceID, '.');
return CaptureSourceOrError({ WTFMove(errorMessage), MediaAccessDenialReason::PermissionDenied });
}
auto source = adoptRef(*new GStreamerAudioCaptureSource(WTFMove(*device), WTFMove(hashSalts)));
if (constraints) {
if (auto result = source->applyConstraints(*constraints))
return CaptureSourceOrError(CaptureSourceError { result->invalidConstraint });
}
return CaptureSourceOrError(WTFMove(source));
}
AudioCaptureFactory& GStreamerAudioCaptureSource::factory()
{
return libWebRTCAudioCaptureSourceFactory();
}
GStreamerAudioCaptureSource::GStreamerAudioCaptureSource(GStreamerCaptureDevice&& device, MediaDeviceHashSalts&& hashSalts)
: RealtimeMediaSource(device, WTFMove(hashSalts))
, m_capturer(adoptRef(*new GStreamerAudioCapturer(WTFMove(device))))
{
ensureGStreamerInitialized();
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_audio_capture_source_debug, "webkitaudiocapturesource", 0, "WebKit Audio Capture Source.");
});
auto& singleton = GStreamerAudioCaptureDeviceManager::singleton();
singleton.registerCapturer(m_capturer.copyRef());
}
GStreamerAudioCaptureSource::~GStreamerAudioCaptureSource()
{
auto& singleton = GStreamerAudioCaptureDeviceManager::singleton();
singleton.unregisterCapturer(*m_capturer);
}
void GStreamerAudioCaptureSource::captureEnded()
{
m_capturer->stop();
captureFailed();
}
std::pair<GstClockTime, GstClockTime> GStreamerAudioCaptureSource::queryCaptureLatency() const
{
if (!m_capturer)
return { GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE };
return m_capturer->queryLatency();
}
void GStreamerAudioCaptureSource::startProducingData()
{
m_capturer->setupPipeline();
m_capturer->setSampleRate(sampleRate());
m_capturer->setSinkAudioCallback([this](auto&& sample, auto&& presentationTime) {
auto bufferSize = gst_buffer_get_size(gst_sample_get_buffer(sample.get()));
GStreamerAudioData frames(WTFMove(sample));
GStreamerAudioStreamDescription description(frames.getAudioInfo());
audioSamplesAvailable(presentationTime, frames, description, bufferSize / description.getInfo().bpf);
});
m_capturer->start();
}
void GStreamerAudioCaptureSource::stopProducingData()
{
m_capturer->stop();
}
const RealtimeMediaSourceCapabilities& GStreamerAudioCaptureSource::capabilities()
{
if (m_capabilities)
return m_capabilities.value();
uint i;
auto caps = m_capturer->caps();
int minSampleRate = 0, maxSampleRate = 0;
for (i = 0; i < gst_caps_get_size(caps.get()); i++) {
int capabilityMinSampleRate = 0, capabilityMaxSampleRate = 0;
GstStructure* str = gst_caps_get_structure(caps.get(), i);
// Only accept raw audio for now.
if (!gst_structure_has_name(str, "audio/x-raw"))
continue;
gst_structure_get(str, "rate", GST_TYPE_INT_RANGE, &capabilityMinSampleRate, &capabilityMaxSampleRate, nullptr);
if (i > 0) {
minSampleRate = std::min(minSampleRate, capabilityMinSampleRate);
maxSampleRate = std::max(maxSampleRate, capabilityMaxSampleRate);
} else {
minSampleRate = capabilityMinSampleRate;
maxSampleRate = capabilityMaxSampleRate;
}
}
RealtimeMediaSourceCapabilities capabilities(settings().supportedConstraints());
capabilities.setDeviceId(hashedId());
capabilities.setEchoCancellation(defaultEchoCancellationCapability);
capabilities.setVolume(defaultVolumeCapability());
capabilities.setSampleRate({ minSampleRate, maxSampleRate });
m_capabilities = WTFMove(capabilities);
return m_capabilities.value();
}
void GStreamerAudioCaptureSource::settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag> settings)
{
if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate))
m_capturer->setSampleRate(sampleRate());
}
const RealtimeMediaSourceSettings& GStreamerAudioCaptureSource::settings()
{
if (!m_currentSettings) {
RealtimeMediaSourceSettings settings;
settings.setDeviceId(hashedId());
RealtimeMediaSourceSupportedConstraints supportedConstraints;
supportedConstraints.setSupportsDeviceId(true);
supportedConstraints.setSupportsEchoCancellation(true);
supportedConstraints.setSupportsVolume(true);
supportedConstraints.setSupportsSampleRate(true);
settings.setSupportedConstraints(supportedConstraints);
m_currentSettings = WTFMove(settings);
}
m_currentSettings->setVolume(volume());
m_currentSettings->setSampleRate(sampleRate());
m_currentSettings->setEchoCancellation(echoCancellation());
return m_currentSettings.value();
}
bool GStreamerAudioCaptureSource::interrupted() const
{
if (m_capturer->pipeline())
return m_capturer->isInterrupted() || RealtimeMediaSource::interrupted();
return RealtimeMediaSource::interrupted();
}
void GStreamerAudioCaptureSource::setInterruptedForTesting(bool isInterrupted)
{
m_capturer->setInterrupted(isInterrupted);
RealtimeMediaSource::setInterruptedForTesting(isInterrupted);
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
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