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/*
* Copyright (C) 2021-2022 Igalia S.L. All rights reserved.
* Copyright (C) 2022 Metrological Group B.V.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "GStreamerIceTransportBackend.h"
#if ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GStreamerWebRTCUtils.h"
#include "NotImplemented.h"
#include <JavaScriptCore/ArrayBuffer.h>
#include <wtf/TZoneMallocInlines.h>
#include <wtf/glib/GUniquePtr.h>
namespace WebCore {
GST_DEBUG_CATEGORY(webkit_webrtc_ice_transport_debug);
#define GST_CAT_DEFAULT webkit_webrtc_ice_transport_debug
WTF_MAKE_TZONE_ALLOCATED_IMPL(GStreamerIceTransportBackend);
GStreamerIceTransportBackend::GStreamerIceTransportBackend(GRefPtr<GstWebRTCDTLSTransport>&& transport)
: m_backend(WTFMove(transport))
{
ASSERT(m_backend);
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_ice_transport_debug, "webkitwebrtcice", 0, "WebKit WebRTC ICE Transport");
});
iceTransportChanged();
g_signal_connect_swapped(m_backend.get(), "notify::transport", G_CALLBACK(+[](GStreamerIceTransportBackend* backend) {
backend->iceTransportChanged();
}), this);
}
GStreamerIceTransportBackend::~GStreamerIceTransportBackend()
{
if (G_IS_OBJECT(m_iceTransport.get()))
g_signal_handlers_disconnect_by_data(m_iceTransport.get(), this);
g_signal_handlers_disconnect_by_data(m_backend.get(), this);
}
void GStreamerIceTransportBackend::iceTransportChanged()
{
if (G_IS_OBJECT(m_iceTransport.get()))
g_signal_handlers_disconnect_by_data(m_iceTransport.get(), this);
g_object_get(m_backend.get(), "transport", &m_iceTransport.outPtr(), nullptr);
// Setting same libnice socket size options as LibWebRTC. 1MB for incoming streams and 256Kb for outgoing streams.
if (gstObjectHasProperty(GST_OBJECT_CAST(m_iceTransport.get()), "receive-buffer-size"))
g_object_set(m_iceTransport.get(), "receive-buffer-size", 1048576, nullptr);
if (gstObjectHasProperty(GST_OBJECT_CAST(m_iceTransport.get()), "send-buffer-size"))
g_object_set(m_iceTransport.get(), "send-buffer-size", 262144, nullptr);
g_signal_connect_swapped(m_iceTransport.get(), "notify::state", G_CALLBACK(+[](GStreamerIceTransportBackend* backend) {
backend->stateChanged();
}), this);
g_signal_connect_swapped(m_iceTransport.get(), "notify::gathering-state", G_CALLBACK(+[](GStreamerIceTransportBackend* backend) {
backend->gatheringStateChanged();
}), this);
g_signal_connect_swapped(m_iceTransport.get(), "on-selected-candidate-pair-change", G_CALLBACK(+[](GStreamerIceTransportBackend* backend) {
backend->selectedCandidatePairChanged();
}), this);
}
void GStreamerIceTransportBackend::registerClient(RTCIceTransportBackendClient& client)
{
ASSERT(!m_client);
m_client = client;
GstWebRTCICEConnectionState transportState;
GstWebRTCICEGatheringState gatheringState;
g_object_get(m_iceTransport.get(), "state", &transportState, "gathering-state", &gatheringState, nullptr);
callOnMainThread([weakThis = WeakPtr { *this }, transportState, gatheringState] {
if (!weakThis || !weakThis->m_client)
return;
#ifndef GST_DISABLE_GST_DEBUG
GUniquePtr<char> desc(g_enum_to_string(GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, transportState));
GST_DEBUG_OBJECT(weakThis->m_backend.get(), "Initial ICE transport state: %s", desc.get());
#endif
// We start observing a bit late and might miss the checking state. Synthesize it as needed.
if (transportState > GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING && transportState != GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED)
weakThis->m_client->onStateChanged(RTCIceTransportState::Checking);
weakThis->m_client->onStateChanged(toRTCIceTransportState(transportState));
weakThis->m_client->onGatheringStateChanged(toRTCIceGatheringState(gatheringState));
});
}
void GStreamerIceTransportBackend::unregisterClient()
{
ASSERT(m_client);
m_client.clear();
}
void GStreamerIceTransportBackend::stateChanged() const
{
if (!m_client)
return;
GstWebRTCICEConnectionState transportState;
g_object_get(m_iceTransport.get(), "state", &transportState, nullptr);
#ifndef GST_DISABLE_GST_DEBUG
GUniquePtr<char> desc(g_enum_to_string(GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, transportState));
GST_DEBUG_OBJECT(m_backend.get(), "ICE transport state changed to %s", desc.get());
#endif
callOnMainThread([weakThis = WeakPtr { *this }, transportState] {
if (!weakThis || !weakThis->m_client)
return;
weakThis->m_client->onStateChanged(toRTCIceTransportState(transportState));
});
}
void GStreamerIceTransportBackend::gatheringStateChanged() const
{
if (!m_client)
return;
GstWebRTCICEGatheringState gatheringState;
g_object_get(m_iceTransport.get(), "gathering-state", &gatheringState, nullptr);
callOnMainThread([weakThis = WeakPtr { *this }, gatheringState] {
if (!weakThis || !weakThis->m_client)
return;
weakThis->m_client->onGatheringStateChanged(toRTCIceGatheringState(gatheringState));
});
}
void GStreamerIceTransportBackend::selectedCandidatePairChanged()
{
// FIXME: call m_client->onSelectedCandidatePairChanged(). See also
// https://github.com/WebKit/WebKit/commit/0692fae10c8e53deba214fd080a35f7c54bd6985
notImplemented();
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
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