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/*
* Copyright (C) 2024 Igalia S.L. All rights reserved.
* Copyright (C) 2024 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 "GStreamerAudioRTPPacketizer.h"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerCommon.h"
#include "GStreamerRegistryScanner.h"
#include <gst/rtp/rtp.h>
#include <wtf/text/MakeString.h>
#include <wtf/text/StringToIntegerConversion.h>
namespace WebCore {
GST_DEBUG_CATEGORY(webkit_webrtc_audio_rtp_packetizer_debug);
#define GST_CAT_DEFAULT webkit_webrtc_audio_rtp_packetizer_debug
RefPtr<GStreamerAudioRTPPacketizer> GStreamerAudioRTPPacketizer::create(RefPtr<UniqueSSRCGenerator> ssrcGenerator, const GstStructure* codecParameters, GUniquePtr<GstStructure>&& encodingParameters)
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_audio_rtp_packetizer_debug, "webkitwebrtcrtppacketizeraudio", 0, "WebKit WebRTC Audio RTP Packetizer");
});
GST_DEBUG("Creating packetizer for codec: %" GST_PTR_FORMAT " and encoding parameters %" GST_PTR_FORMAT, codecParameters, encodingParameters.get());
String encoding;
if (auto encodingName = gstStructureGetString(codecParameters, "encoding-name"_s))
encoding = encodingName.convertToASCIILowercase();
else {
GST_ERROR("encoding-name not found");
return nullptr;
}
auto& registryScanner = GStreamerRegistryScanner::singleton();
auto lookupResult = registryScanner.isRtpPacketizerSupported(encoding);
if (!lookupResult) {
GST_ERROR("RTP payloader not found for encoding %s", encoding.ascii().data());
return nullptr;
}
GRefPtr<GstElement> payloader = gst_element_factory_create(lookupResult.factory.get(), nullptr);
GST_DEBUG("Using %" GST_PTR_FORMAT " for %s RTP packetizing", payloader.get(), encoding.ascii().data());
auto inputCaps = adoptGRef(gst_caps_new_any());
GUniquePtr<GstStructure> structure(gst_structure_copy(codecParameters));
auto ssrc = ssrcGenerator->generateSSRC();
if (ssrc != std::numeric_limits<uint32_t>::max())
gst_structure_set(structure.get(), "ssrc", G_TYPE_UINT, ssrc, nullptr);
GRefPtr<GstElement> encoder;
if (encoding == "opus"_s) {
encoder = makeGStreamerElement("opusenc", nullptr);
if (!encoder)
return nullptr;
gst_structure_set(structure.get(), "encoding-name", G_TYPE_STRING, "OPUS", nullptr);
// FIXME: Enable dtx too?
gst_util_set_object_arg(G_OBJECT(encoder.get()), "audio-type", "voice");
g_object_set(encoder.get(), "perfect-timestamp", TRUE, nullptr);
if (auto useInbandFec = gstStructureGetString(structure.get(), "useinbandfec"_s)) {
if (useInbandFec == "1"_s)
g_object_set(encoder.get(), "inband-fec", TRUE, nullptr);
gst_structure_remove_field(structure.get(), "useinbandfec");
}
if (auto isStereo = gstStructureGetString(structure.get(), "stereo"_s)) {
if (isStereo == "1"_s)
inputCaps = adoptGRef(gst_caps_new_simple("audio/x-raw", "channels", G_TYPE_INT, 2, nullptr));
gst_structure_remove_field(structure.get(), "stereo");
}
if (gst_caps_is_any(inputCaps.get())) {
if (auto encodingParameters = gstStructureGetString(structure.get(), "encoding-params"_s)) {
if (auto channels = parseIntegerAllowingTrailingJunk<int>(encodingParameters))
inputCaps = adoptGRef(gst_caps_new_simple("audio/x-raw", "channels", G_TYPE_INT, *channels, nullptr));
}
}
} else if (encoding == "g722"_s)
encoder = makeGStreamerElement("avenc_g722", nullptr);
else if (encoding == "pcma"_s)
encoder = makeGStreamerElement("alawenc", nullptr);
else if (encoding == "pcmu"_s)
encoder = makeGStreamerElement("mulawenc", nullptr);
else {
GST_ERROR("Unsupported outgoing audio encoding: %s", encoding.ascii().data());
return nullptr;
}
if (!encoder) {
GST_ERROR("Encoder not found for encoding %s", encoding.ascii().data());
return nullptr;
}
// Align MTU with libwebrtc implementation, also helping to reduce packet fragmentation.
g_object_set(payloader.get(), "auto-header-extension", TRUE, "mtu", 1200, nullptr);
if (auto minPTime = gstStructureGetString(structure.get(), "minptime"_s)) {
if (auto value = parseIntegerAllowingTrailingJunk<int64_t>(minPTime)) {
if (gstObjectHasProperty(payloader.get(), "min-ptime"))
g_object_set(payloader.get(), "min-ptime", *value * GST_MSECOND, nullptr);
else
GST_WARNING_OBJECT(payloader.get(), "min-ptime property not supported");
}
gst_structure_remove_field(structure.get(), "minptime");
}
auto payloadType = gstStructureGet<int>(codecParameters, "payload"_s);
if (!payloadType)
payloadType = gstStructureGet<int>(encodingParameters.get(), "payload"_s);
auto rtpCaps = adoptGRef(gst_caps_new_empty());
// When not present in caps, the vad support of the ssrc-audio-level extension should be
// enabled. In order to prevent caps negotiation issues with downstream, explicitely set it.
setSsrcAudioLevelVadOn(structure.get());
gst_caps_append_structure(rtpCaps.get(), structure.release());
return adoptRef(*new GStreamerAudioRTPPacketizer(WTFMove(inputCaps), WTFMove(encoder), WTFMove(payloader), WTFMove(encodingParameters), WTFMove(rtpCaps), WTFMove(payloadType)));
}
GStreamerAudioRTPPacketizer::GStreamerAudioRTPPacketizer(GRefPtr<GstCaps>&& inputCaps, GRefPtr<GstElement>&& encoder, GRefPtr<GstElement>&& payloader, GUniquePtr<GstStructure>&& encodingParameters, GRefPtr<GstCaps>&& rtpCaps, std::optional<int>&& payloadType)
: GStreamerRTPPacketizer(WTFMove(encoder), WTFMove(payloader), WTFMove(encodingParameters), WTFMove(payloadType))
{
g_object_set(m_capsFilter.get(), "caps", rtpCaps.get(), nullptr);
GST_DEBUG_OBJECT(m_bin.get(), "RTP caps: %" GST_PTR_FORMAT, rtpCaps.get());
m_audioconvert = makeGStreamerElement("audioconvert", nullptr);
m_audioresample = makeGStreamerElement("audioresample", nullptr);
m_inputCapsFilter = gst_element_factory_make("capsfilter", nullptr);
g_object_set(m_inputCapsFilter.get(), "caps", inputCaps.get(), nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_audioconvert.get(), m_audioresample.get(), m_inputCapsFilter.get(), nullptr);
gst_element_link_many(m_inputQueue.get(), m_audioconvert.get(), m_audioresample.get(), m_inputCapsFilter.get(), m_encoder.get(), m_payloader.get(), m_capsFilter.get(), m_outputQueue.get(), m_valve.get(), nullptr);
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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