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/*
* Copyright (C) 2024 Igalia S.L.
*
* 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"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerWebRTCLogSink.h"
namespace WebCore {
GStreamerWebRTCLogSink::GStreamerWebRTCLogSink(LogCallback&& callback)
: m_callback(WTFMove(callback))
, m_isGstDebugActive(gst_debug_is_active())
{
}
GStreamerWebRTCLogSink::~GStreamerWebRTCLogSink() = default;
static String toWebRTCLogLevel(GstDebugLevel level)
{
switch (level) {
case GST_LEVEL_NONE:
return "none"_s;
case GST_LEVEL_ERROR:
return "error"_s;
case GST_LEVEL_WARNING:
return "warning"_s;
case GST_LEVEL_FIXME:
return "fixme"_s;
case GST_LEVEL_INFO:
return "info"_s;
case GST_LEVEL_DEBUG:
return "debug"_s;
case GST_LEVEL_LOG:
return "log"_s;
case GST_LEVEL_TRACE:
return "trace"_s;
case GST_LEVEL_MEMDUMP:
return "memdump"_s;
default:
break;
}
ASSERT_NOT_REACHED();
return ""_s;
}
void GStreamerWebRTCLogSink::start()
{
#ifdef GST_DISABLE_GST_DEBUG
static std::once_flag onceFlag;
std::call_once(onceFlag, [] {
WTFLogAlways("GST_DEBUG is disabled in this build. gatherLogs() will report only WebRTC stats logs.");
});
#else
if (!m_isGstDebugActive)
gst_debug_remove_log_function(gst_debug_log_default);
gst_debug_add_log_function(static_cast<GstLogFunction>(+[](GstDebugCategory*, GstDebugLevel level, const char*, const char*, int, GObject*, GstDebugMessage* debugMessage, gpointer userData) G_GNUC_NO_INSTRUMENT {
const char* message = gst_debug_message_get(debugMessage);
if (!message)
return;
auto self = reinterpret_cast<GStreamerWebRTCLogSink*>(userData);
self->m_callback(toWebRTCLogLevel(level), String::fromUTF8(message));
}), this, nullptr);
// Do not include webrtcstats in the list, because stats are logged using a different code path by the endpoint.
gst_debug_set_threshold_from_string("webrtcbin:5,webrtcdatachannel:5,webrtctransport*:5,webrtcsctp*:5,nice*:6", FALSE);
#endif
}
void GStreamerWebRTCLogSink::stop()
{
#ifndef GST_DISABLE_GST_DEBUG
gst_debug_remove_log_function_by_data(this);
if (!m_isGstDebugActive)
gst_debug_add_log_function(gst_debug_log_default, nullptr, nullptr);
#endif
}
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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