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/*
* Copyright (C) 2017-2022 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 "RealtimeOutgoingMediaSourceGStreamer.h"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerCommon.h"
#include "GStreamerMediaStreamSource.h"
#include "GStreamerWebRTCUtils.h"
#include "MediaStreamTrack.h"
#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
#undef GST_USE_UNSTABLE_API
GST_DEBUG_CATEGORY_EXTERN(webkit_webrtc_endpoint_debug);
#define GST_CAT_DEFAULT webkit_webrtc_endpoint_debug
namespace WebCore {
RealtimeOutgoingMediaSourceGStreamer::RealtimeOutgoingMediaSourceGStreamer(const String& mediaStreamId, MediaStreamTrack& track)
: m_mediaStreamId(mediaStreamId)
, m_trackId(track.id())
{
m_bin = gst_bin_new(nullptr);
m_valve = gst_element_factory_make("valve", nullptr);
gst_util_set_object_arg(G_OBJECT(m_valve.get()), "drop-mode", "forward-sticky-events");
m_preEncoderQueue = gst_element_factory_make("queue", nullptr);
m_postEncoderQueue = gst_element_factory_make("queue", nullptr);
m_capsFilter = gst_element_factory_make("capsfilter", nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_valve.get(), m_preEncoderQueue.get(),
m_postEncoderQueue.get(), m_capsFilter.get(), nullptr);
auto srcPad = adoptGRef(gst_element_get_static_pad(m_capsFilter.get(), "src"));
gst_element_add_pad(m_bin.get(), gst_ghost_pad_new("src", srcPad.get()));
setSource(track.privateTrack());
}
RealtimeOutgoingMediaSourceGStreamer::~RealtimeOutgoingMediaSourceGStreamer()
{
if (m_transceiver)
g_signal_handlers_disconnect_by_data(m_transceiver.get(), this);
stop();
if (m_webrtcSinkPad) {
auto srcPad = adoptGRef(gst_element_get_static_pad(m_bin.get(), "src"));
if (gst_pad_unlink(srcPad.get(), m_webrtcSinkPad.get())) {
GST_DEBUG_OBJECT(m_bin.get(), "Removing webrtcbin pad %" GST_PTR_FORMAT, m_webrtcSinkPad.get());
if (auto parent = adoptGRef(gst_pad_get_parent_element(m_webrtcSinkPad.get())))
gst_element_release_request_pad(parent.get(), m_webrtcSinkPad.get());
}
}
gst_element_set_locked_state(m_bin.get(), TRUE);
gst_element_set_state(m_bin.get(), GST_STATE_NULL);
if (auto pipeline = adoptGRef(gst_element_get_parent(m_bin.get())))
gst_bin_remove(GST_BIN_CAST(pipeline.get()), m_bin.get());
gst_element_set_locked_state(m_bin.get(), FALSE);
}
const GRefPtr<GstCaps>& RealtimeOutgoingMediaSourceGStreamer::allowedCaps() const
{
if (m_allowedCaps)
return m_allowedCaps;
auto sdpMsIdLine = makeString(m_mediaStreamId, ' ', m_trackId);
m_allowedCaps = capsFromRtpCapabilities(rtpCapabilities(), [&sdpMsIdLine](GstStructure* structure) {
gst_structure_set(structure, "a-msid", G_TYPE_STRING, sdpMsIdLine.ascii().data(), nullptr);
});
GST_DEBUG_OBJECT(m_bin.get(), "Allowed caps: %" GST_PTR_FORMAT, m_allowedCaps.get());
return m_allowedCaps;
}
void RealtimeOutgoingMediaSourceGStreamer::setSource(Ref<MediaStreamTrackPrivate>&& newSource)
{
if (m_source && !m_initialSettings)
m_initialSettings = m_source.value()->settings();
if (m_source)
m_source.value()->removeObserver(*this);
m_source = WTFMove(newSource);
initializeFromTrack();
}
void RealtimeOutgoingMediaSourceGStreamer::start()
{
m_source.value()->addObserver(*this);
m_isStopped = false;
gst_element_link(m_outgoingSource.get(), m_valve.get());
gst_element_sync_state_with_parent(m_bin.get());
}
void RealtimeOutgoingMediaSourceGStreamer::stop()
{
m_isStopped = true;
if (!m_source)
return;
m_source.value()->removeObserver(*this);
webkitMediaStreamSrcSignalEndOfStream(WEBKIT_MEDIA_STREAM_SRC(m_outgoingSource.get()));
gst_element_set_locked_state(m_outgoingSource.get(), TRUE);
gst_element_set_state(m_outgoingSource.get(), GST_STATE_NULL);
gst_bin_remove(GST_BIN_CAST(m_bin.get()), m_outgoingSource.get());
gst_element_set_locked_state(m_outgoingSource.get(), FALSE);
m_source.reset();
}
void RealtimeOutgoingMediaSourceGStreamer::sourceMutedChanged()
{
if (!m_source)
return;
ASSERT(m_muted != m_source.value()->muted());
m_muted = m_source.value()->muted();
}
void RealtimeOutgoingMediaSourceGStreamer::sourceEnabledChanged()
{
if (!m_source)
return;
m_enabled = m_source.value()->enabled();
if (m_valve)
g_object_set(m_valve.get(), "drop", !m_enabled, nullptr);
}
void RealtimeOutgoingMediaSourceGStreamer::initializeFromTrack()
{
m_muted = m_source.value()->muted();
m_enabled = m_source.value()->enabled();
m_outgoingSource = webkitMediaStreamSrcNew();
gst_bin_add(GST_BIN_CAST(m_bin.get()), m_outgoingSource.get());
webkitMediaStreamSrcAddTrack(WEBKIT_MEDIA_STREAM_SRC(m_outgoingSource.get()), m_source->ptr(), true);
}
void RealtimeOutgoingMediaSourceGStreamer::link()
{
GST_DEBUG_OBJECT(m_bin.get(), "Linking webrtcbin pad %" GST_PTR_FORMAT, m_webrtcSinkPad.get());
gst_element_link(m_postEncoderQueue.get(), m_capsFilter.get());
auto srcPad = adoptGRef(gst_element_get_static_pad(m_bin.get(), "src"));
gst_pad_link(srcPad.get(), m_webrtcSinkPad.get());
}
void RealtimeOutgoingMediaSourceGStreamer::setSinkPad(GRefPtr<GstPad>&& pad)
{
m_webrtcSinkPad = WTFMove(pad);
if (m_transceiver)
g_signal_handlers_disconnect_by_data(m_transceiver.get(), this);
g_object_get(m_webrtcSinkPad.get(), "transceiver", &m_transceiver.outPtr(), nullptr);
g_signal_connect_swapped(m_transceiver.get(), "notify::codec-preferences", G_CALLBACK(+[](RealtimeOutgoingMediaSourceGStreamer* source, GParamSpec*, GstWebRTCRTPTransceiver* transceiver) {
GRefPtr<GstCaps> codecPreferences;
g_object_get(transceiver, "codec-preferences", &codecPreferences.outPtr(), nullptr);
callOnMainThreadAndWait([&] {
source->codecPreferencesChanged(codecPreferences);
});
}), this);
g_object_get(m_transceiver.get(), "sender", &m_sender.outPtr(), nullptr);
}
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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