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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 "RealtimeOutgoingAudioSourceGStreamer.h"
#if USE(GSTREAMER_WEBRTC)
#include "ContextDestructionObserverInlines.h"
#include "GStreamerAudioRTPPacketizer.h"
#include "GStreamerCommon.h"
#include "GStreamerMediaStreamSource.h"
#include "GStreamerRegistryScanner.h"
#include "GStreamerWebRTCCommon.h"
#include "MediaStreamTrack.h"
#include "NotImplemented.h"
#include <wtf/text/MakeString.h>
GST_DEBUG_CATEGORY(webkit_webrtc_outgoing_audio_debug);
#define GST_CAT_DEFAULT webkit_webrtc_outgoing_audio_debug
namespace WebCore {
RealtimeOutgoingAudioSourceGStreamer::RealtimeOutgoingAudioSourceGStreamer(const RefPtr<UniqueSSRCGenerator>& ssrcGenerator, const String& mediaStreamId, MediaStreamTrack& track)
: RealtimeOutgoingMediaSourceGStreamer(RealtimeOutgoingMediaSourceGStreamer::Type::Audio, ssrcGenerator, mediaStreamId, track)
{
initialize();
}
RealtimeOutgoingAudioSourceGStreamer::RealtimeOutgoingAudioSourceGStreamer(const RefPtr<UniqueSSRCGenerator>& ssrcGenerator)
: RealtimeOutgoingMediaSourceGStreamer(RealtimeOutgoingMediaSourceGStreamer::Type::Audio, ssrcGenerator)
{
initialize();
m_outgoingSource = gst_element_factory_make("audiotestsrc", nullptr);
gst_util_set_object_arg(G_OBJECT(m_outgoingSource.get()), "wave", "silence");
g_object_set(m_outgoingSource.get(), "is-live", TRUE, "do-timestamp", TRUE, nullptr);
gst_bin_add(GST_BIN_CAST(m_bin.get()), m_outgoingSource.get());
}
RealtimeOutgoingAudioSourceGStreamer::~RealtimeOutgoingAudioSourceGStreamer() = default;
void RealtimeOutgoingAudioSourceGStreamer::initialize()
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_outgoing_audio_debug, "webkitwebrtcoutgoingaudio", 0, "WebKit WebRTC outgoing audio");
});
static Atomic<uint64_t> sourceCounter = 0;
gst_element_set_name(m_bin.get(), makeString("outgoing-audio-source-"_s, sourceCounter.exchangeAdd(1)).ascii().data());
}
void RealtimeOutgoingAudioSourceGStreamer::setInitialParameters(GUniquePtr<GstStructure>&& parameters)
{
for (const auto& codec : gstStructureGetList<const GstStructure*>(parameters.get(), "codecs"_s)) {
auto encodingName = gstStructureGetString(codec, "mime-type");
if (encodingName.isEmpty() || encodingName.isNull())
continue;
if (encodingName != "audio/telephone-event"_s)
continue;
auto pt = gstStructureGet<unsigned>(codec, "pt");
if (!pt) [[unlikely]]
continue;
// We're picking up only the first encoding. Maybe we should check clock-rate too.
setupDTMFSource(*pt);
break;
}
RealtimeOutgoingMediaSourceGStreamer::setInitialParameters(WTFMove(parameters));
}
void RealtimeOutgoingAudioSourceGStreamer::setupDTMFSource(int pt)
{
m_dtmfSource = makeGStreamerElement("rtpdtmfsrc"_s, "dtmfSource"_s);
if (!m_dtmfSource) {
gst_printerrln("RTP DTMF element(s) missing, DTMF tones sending support disabled.");
return;
}
gstPayloaderSetPayloadType(m_dtmfSource, pt);
gst_bin_add(GST_BIN_CAST(m_bin.get()), m_dtmfSource.get());
auto srcPad = adoptGRef(gst_element_get_static_pad(m_dtmfSource.get(), "src"));
auto sinkPad = adoptGRef(gst_element_request_pad_simple(m_rtpFunnel.get(), "sink_%u"));
gst_pad_link(srcPad.get(), sinkPad.get());
}
RTCRtpCapabilities RealtimeOutgoingAudioSourceGStreamer::rtpCapabilities() const
{
auto& registryScanner = GStreamerRegistryScanner::singleton();
return registryScanner.audioRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
}
GRefPtr<GstPad> RealtimeOutgoingAudioSourceGStreamer::outgoingSourcePad() const
{
if (WEBKIT_IS_MEDIA_STREAM_SRC(m_outgoingSource.get()))
return adoptGRef(gst_element_get_static_pad(m_outgoingSource.get(), "audio_src0"));
return adoptGRef(gst_element_get_static_pad(m_outgoingSource.get(), "src"));
}
RefPtr<GStreamerRTPPacketizer> RealtimeOutgoingAudioSourceGStreamer::createPacketizer(RefPtr<UniqueSSRCGenerator> ssrcGenerator, const GstStructure* codecParameters, GUniquePtr<GstStructure>&& encodingParameters)
{
return GStreamerAudioRTPPacketizer::create(ssrcGenerator, codecParameters, WTFMove(encodingParameters));
}
void RealtimeOutgoingAudioSourceGStreamer::dispatchBitrateRequest(uint32_t)
{
notImplemented();
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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