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/*
* Copyright (C) 2019-2022 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 "GStreamerRtpTransceiverBackend.h"
#if ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GStreamerRtpReceiverBackend.h"
#include "GStreamerRtpSenderBackend.h"
#include "GStreamerWebRTCUtils.h"
#include "RTCRtpCodecCapability.h"
#include <wtf/TZoneMallocInlines.h>
#include <wtf/glib/GUniquePtr.h>
GST_DEBUG_CATEGORY(webkit_webrtc_transceiver_debug);
#define GST_CAT_DEFAULT webkit_webrtc_transceiver_debug
namespace WebCore {
WTF_MAKE_TZONE_ALLOCATED_IMPL(GStreamerRtpTransceiverBackend);
GStreamerRtpTransceiverBackend::GStreamerRtpTransceiverBackend(GRefPtr<GstWebRTCRTPTransceiver>&& rtcTransceiver)
: m_rtcTransceiver(WTFMove(rtcTransceiver))
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_transceiver_debug, "webkitwebrtcrtptransceiver", 0, "WebKit WebRTC RTP transceiver");
});
GstWebRTCKind kind;
g_object_get(m_rtcTransceiver.get(), "kind", &kind, nullptr);
// FIXME: The ulp/red encoders drop MID extension headers. See also:
// https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/923
// gst_util_set_object_arg(G_OBJECT(m_rtcTransceiver.get()), "fec-type", "ulp-red");
// Enable nack only for video transceivers, so that RTX payloads are not signaled in SDP
// offer/answer. Those are confusing some media servers... Internally webrtcbin will always
// setup RTX, RED and FEC anyway.
if (kind != GST_WEBRTC_KIND_VIDEO)
return;
g_object_set(m_rtcTransceiver.get(), "do-nack", TRUE, nullptr);
}
std::unique_ptr<GStreamerRtpReceiverBackend> GStreamerRtpTransceiverBackend::createReceiverBackend()
{
return WTF::makeUnique<GStreamerRtpReceiverBackend>(GRefPtr(m_rtcTransceiver));
}
std::unique_ptr<GStreamerRtpSenderBackend> GStreamerRtpTransceiverBackend::createSenderBackend(GStreamerPeerConnectionBackend& backend, GStreamerRtpSenderBackend::Source&& source, GUniquePtr<GstStructure>&& initData)
{
GRefPtr<GstWebRTCRTPSender> sender;
g_object_get(m_rtcTransceiver.get(), "sender", &sender.outPtr(), nullptr);
return WTF::makeUnique<GStreamerRtpSenderBackend>(backend, WTFMove(sender), WTFMove(source), WTFMove(initData));
}
RTCRtpTransceiverDirection GStreamerRtpTransceiverBackend::direction() const
{
GstWebRTCRTPTransceiverDirection gstDirection;
g_object_get(m_rtcTransceiver.get(), "direction", &gstDirection, nullptr);
return toRTCRtpTransceiverDirection(gstDirection);
}
std::optional<RTCRtpTransceiverDirection> GStreamerRtpTransceiverBackend::currentDirection() const
{
GstWebRTCRTPTransceiverDirection gstDirection;
g_object_get(m_rtcTransceiver.get(), "current-direction", &gstDirection, nullptr);
if (!gstDirection)
return std::nullopt;
return toRTCRtpTransceiverDirection(gstDirection);
}
void GStreamerRtpTransceiverBackend::setDirection(RTCRtpTransceiverDirection direction)
{
auto gstDirection = fromRTCRtpTransceiverDirection(direction);
#ifndef GST_DISABLE_GST_DEBUG
GUniquePtr<char> directionString(g_enum_to_string(GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, gstDirection));
GST_DEBUG_OBJECT(m_rtcTransceiver.get(), "Setting direction to %s", directionString.get());
#endif
g_object_set(m_rtcTransceiver.get(), "direction", gstDirection, nullptr);
}
String GStreamerRtpTransceiverBackend::mid()
{
GUniqueOutPtr<char> mid;
g_object_get(m_rtcTransceiver.get(), "mid", &mid.outPtr(), nullptr);
return String::fromUTF8(mid.get());
}
void GStreamerRtpTransceiverBackend::stop()
{
// Ideally we should also stop webrtcbin transceivers but it's not supported yet.
m_isStopped = true;
}
bool GStreamerRtpTransceiverBackend::stopped() const
{
// Ideally this should be queried on webrtcbin, but its transceivers can't be stopped yet.
return m_isStopped;
}
static inline WARN_UNUSED_RETURN ExceptionOr<GRefPtr<GstCaps>> toRtpCodecCapability(const RTCRtpCodecCapability& codec, int& dynamicPayloadType, const String& msid)
{
if (!codec.mimeType.startsWith("video/"_s) && !codec.mimeType.startsWith("audio/"_s))
return Exception { ExceptionCode::InvalidModificationError, "RTCRtpCodecCapability bad mimeType"_s };
auto components = codec.mimeType.split('/');
const auto mediaType = components[0];
const auto codecName = components[1];
int payloadType = payloadTypeForEncodingName(codecName).value_or(dynamicPayloadType++);
auto caps = adoptGRef(gst_caps_new_simple("application/x-rtp", "media", G_TYPE_STRING, mediaType.ascii().data(), "encoding-name", G_TYPE_STRING, codecName.ascii().data(), "clock-rate", G_TYPE_INT, codec.clockRate, "payload", G_TYPE_INT, payloadType, nullptr));
if (codec.channels)
gst_caps_set_simple(caps.get(), "channels", G_TYPE_INT, *codec.channels, nullptr);
if (!codec.sdpFmtpLine.isEmpty()) {
// Forward each fmtp attribute as <fmtp-name> in the caps so that the downstream
// webkitvideoencoder can take those into account when configuring the encoder. For instance
// VP9 profile 2 requires a 10bit pixel input format, so a conversion might be needed just
// before encoding. This is taken care of in the webkitvideoencoder itself.
for (auto& attribute : codec.sdpFmtpLine.split(';')) {
auto components = attribute.split('=');
gst_caps_set_simple(caps.get(), components[0].ascii().data(), G_TYPE_STRING, components[1].ascii().data(), nullptr);
}
}
if (!msid.isEmpty())
gst_caps_set_simple(caps.get(), "a-msid", G_TYPE_STRING, msid.ascii().data(), nullptr);
GST_DEBUG("Codec capability: %" GST_PTR_FORMAT, caps.get());
return caps;
}
ExceptionOr<void> GStreamerRtpTransceiverBackend::setCodecPreferences(const Vector<RTCRtpCodecCapability>& codecs)
{
GRefPtr<GstCaps> currentCaps;
g_object_get(m_rtcTransceiver.get(), "codec-preferences", ¤tCaps.outPtr(), nullptr);
GST_TRACE_OBJECT(m_rtcTransceiver.get(), "Current codec preferences: %" GST_PTR_FORMAT, currentCaps.get());
String msid;
HashMap<String, String> extensions;
if (gst_caps_get_size(currentCaps.get()) > 0) {
auto structure = gst_caps_get_structure(currentCaps.get(), 0);
if (auto msIdValue = gstStructureGetString(structure, "a-msid"_s))
msid = msIdValue.toString();
gstStructureForeach(structure, [&](auto id, const auto& value) -> bool {
auto key = gstIdToString(id);
if (!key.startsWith("extmap-"_s))
return true;
extensions.add(key.toString(), String::fromLatin1(g_value_get_string(value)));
return true;
});
}
auto gstCodecs = adoptGRef(gst_caps_new_empty());
int dynamicPayloadType = 96;
for (auto& codec : codecs) {
auto result = toRtpCodecCapability(codec, dynamicPayloadType, msid);
if (result.hasException())
return result.releaseException();
auto codecCaps = result.releaseReturnValue();
// Restore extensions data on the first codec. It might be useful to do in the others too.
if (!extensions.isEmpty()) {
for (auto& [extensionId, url] : extensions)
gst_caps_set_simple(codecCaps.get(), extensionId.ascii().data(), G_TYPE_STRING, url.ascii().data(), nullptr);
extensions.clear();
}
gst_caps_append(gstCodecs.get(), codecCaps.leakRef());
}
GST_DEBUG_OBJECT(m_rtcTransceiver.get(), "Setting codec preferences to %" GST_PTR_FORMAT, gstCodecs.get());
g_object_set(m_rtcTransceiver.get(), "codec-preferences", gstCodecs.get(), nullptr);
return { };
}
void GStreamerRtpTransceiverBackend::tearDown()
{
m_rtcTransceiver.clear();
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
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