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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 "RealtimeOutgoingVideoSourceGStreamer.h"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerRegistryScanner.h"
#include "GStreamerVideoEncoder.h"
#include <wtf/glib/WTFGType.h>
GST_DEBUG_CATEGORY_EXTERN(webkit_webrtc_endpoint_debug);
#define GST_CAT_DEFAULT webkit_webrtc_endpoint_debug
namespace WebCore {
struct RealtimeOutgoingVideoSourceHolder {
RefPtr<RealtimeOutgoingVideoSourceGStreamer> source;
};
WEBKIT_DEFINE_ASYNC_DATA_STRUCT(RealtimeOutgoingVideoSourceHolder)
RealtimeOutgoingVideoSourceGStreamer::RealtimeOutgoingVideoSourceGStreamer(const String& mediaStreamId, MediaStreamTrack& track)
: RealtimeOutgoingMediaSourceGStreamer(mediaStreamId, track)
{
registerWebKitGStreamerElements();
static Atomic<uint64_t> sourceCounter = 0;
gst_element_set_name(m_bin.get(), makeString("outgoing-video-source-", sourceCounter.exchangeAdd(1)).ascii().data());
m_stats.reset(gst_structure_new_empty("webrtc-outgoing-video-stats"));
auto holder = createRealtimeOutgoingVideoSourceHolder();
holder->source = this;
auto pad = adoptGRef(gst_element_get_static_pad(m_bin.get(), "src"));
gst_pad_add_probe(pad.get(), GST_PAD_PROBE_TYPE_BUFFER, [](GstPad*, GstPadProbeInfo* info, gpointer userData) -> GstPadProbeReturn {
auto* holder = static_cast<RealtimeOutgoingVideoSourceHolder*>(userData);
auto* buffer = GST_PAD_PROBE_INFO_BUFFER(info);
holder->source->updateStats(buffer);
return GST_PAD_PROBE_OK;
}, holder, reinterpret_cast<GDestroyNotify>(destroyRealtimeOutgoingVideoSourceHolder));
m_videoConvert = makeGStreamerElement("videoconvert", nullptr);
m_videoFlip = makeGStreamerElement("videoflip", nullptr);
gst_util_set_object_arg(G_OBJECT(m_videoFlip.get()), "method", "automatic");
m_encoder = gst_element_factory_make("webrtcvideoencoder", nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_videoFlip.get(), m_videoConvert.get(), m_encoder.get(), nullptr);
}
RTCRtpCapabilities RealtimeOutgoingVideoSourceGStreamer::rtpCapabilities() const
{
auto& registryScanner = GStreamerRegistryScanner::singleton();
return registryScanner.videoRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
}
void RealtimeOutgoingVideoSourceGStreamer::updateStats(GstBuffer*)
{
uint64_t framesSent = 0;
gst_structure_get_uint64(m_stats.get(), "frames-sent", &framesSent);
framesSent++;
if (m_encoder) {
uint32_t bitrate;
g_object_get(m_encoder.get(), "bitrate", &bitrate, nullptr);
gst_structure_set(m_stats.get(), "bitrate", G_TYPE_DOUBLE, static_cast<double>(bitrate * 1024), nullptr);
}
gst_structure_set(m_stats.get(), "frames-sent", G_TYPE_UINT64, framesSent, "frames-encoded", G_TYPE_UINT64, framesSent, nullptr);
}
bool RealtimeOutgoingVideoSourceGStreamer::setPayloadType(const GRefPtr<GstCaps>& caps)
{
GST_DEBUG_OBJECT(m_bin.get(), "Setting payload caps: %" GST_PTR_FORMAT, caps.get());
auto* structure = gst_caps_get_structure(caps.get(), 0);
const char* encodingName = gst_structure_get_string(structure, "encoding-name");
if (!encodingName) {
GST_ERROR_OBJECT(m_bin.get(), "encoding-name not found");
return false;
}
auto encoding = makeString(encodingName).convertToASCIILowercase();
m_payloader = makeGStreamerElement(makeString("rtp"_s, encoding, "pay"_s).ascii().data(), nullptr);
if (UNLIKELY(!m_payloader)) {
GST_ERROR_OBJECT(m_bin.get(), "RTP payloader not found for encoding %s", encodingName);
return false;
}
GRefPtr<GstCaps> encoderCaps;
if (encoding == "vp8"_s) {
encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-vp8"));
} else if (encoding == "vp9"_s) {
encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-vp9"));
if (const char* vp9Profile = gst_structure_get_string(structure, "vp9-profile-id"))
gst_caps_set_simple(encoderCaps.get(), "profile", G_TYPE_STRING, vp9Profile, nullptr);
} else if (encoding == "h264"_s) {
encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-h264"));
// FIXME: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/893
// gst_util_set_object_arg(G_OBJECT(m_payloader.get()), "aggregate-mode", "zero-latency");
// g_object_set(m_payloader.get(), "config-interval", -1, nullptr);
const char* profile = gst_structure_get_string(structure, "profile");
if (!profile)
profile = "baseline";
gst_caps_set_simple(encoderCaps.get(), "profile", G_TYPE_STRING, profile, nullptr);
} else if (encoding == "h265"_s) {
encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-h265"));
// FIXME: profile tier level?
} else {
GST_ERROR_OBJECT(m_bin.get(), "Unsupported outgoing video encoding: %s", encodingName);
return false;
}
// Align MTU with libwebrtc implementation, also helping to reduce packet fragmentation.
g_object_set(m_payloader.get(), "mtu", 1200, nullptr);
if (!webrtcVideoEncoderSetFormat(WEBKIT_WEBRTC_VIDEO_ENCODER(m_encoder.get()), WTFMove(encoderCaps))) {
GST_ERROR_OBJECT(m_bin.get(), "Unable to set encoder format");
return false;
}
int payloadType;
if (gst_structure_get_int(structure, "payload", &payloadType))
g_object_set(m_payloader.get(), "pt", payloadType, nullptr);
g_object_set(m_capsFilter.get(), "caps", caps.get(), nullptr);
gst_bin_add(GST_BIN_CAST(m_bin.get()), m_payloader.get());
auto encoderSinkPad = adoptGRef(gst_element_get_static_pad(m_encoder.get(), "sink"));
if (!gst_pad_is_linked(encoderSinkPad.get()))
gst_element_link_many(m_outgoingSource.get(), m_valve.get(), m_videoFlip.get(), m_videoConvert.get(), m_preEncoderQueue.get(), m_encoder.get(), nullptr);
gst_element_link_many(m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
gst_bin_sync_children_states(GST_BIN_CAST(m_bin.get()));
return true;
}
void RealtimeOutgoingVideoSourceGStreamer::codecPreferencesChanged(const GRefPtr<GstCaps>& codecPreferences)
{
gst_element_set_locked_state(m_bin.get(), TRUE);
if (m_payloader) {
gst_element_set_locked_state(m_payloader.get(), TRUE);
gst_element_set_state(m_payloader.get(), GST_STATE_NULL);
gst_element_unlink_many(m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
gst_bin_remove(GST_BIN_CAST(m_bin.get()), m_payloader.get());
m_payloader.clear();
}
setPayloadType(codecPreferences);
gst_element_set_locked_state(m_bin.get(), FALSE);
gst_element_sync_state_with_parent(m_bin.get());
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_bin.get()), GST_DEBUG_GRAPH_SHOW_ALL, "outgoing-video-new-codec-prefs");
}
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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