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/*
* Copyright (C) 2019-2025 Igalia S.L.
* Copyright (C) 2025 Comcast Inc.
*
* 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 "GStreamerDTMFSenderBackend.h"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerCommon.h"
#include <wtf/HashFunctions.h>
#include <wtf/HashTraits.h>
#include <wtf/TZoneMallocInlines.h>
#include <wtf/WorkQueue.h>
namespace WebCore {
WTF_MAKE_TZONE_ALLOCATED_IMPL(GStreamerDTMFSenderBackend);
GST_DEBUG_CATEGORY(webkit_webrtc_dtmf_sender_debug);
#define GST_CAT_DEFAULT webkit_webrtc_dtmf_sender_debug
RefPtr<GStreamerDTMFSenderPrivate> GStreamerDTMFSenderPrivate::create()
{
auto dtmfSrc = makeGStreamerElement("dtmfsrc"_s);
if (!dtmfSrc)
return nullptr;
return adoptRef(*new GStreamerDTMFSenderPrivate(WTFMove(dtmfSrc)));
}
GStreamerDTMFSenderPrivate::GStreamerDTMFSenderPrivate(GRefPtr<GstElement>&& dtmfSrc)
: m_dtmfSrc(WTFMove(dtmfSrc))
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_dtmf_sender_debug, "webkitwebrtcdtmfsender", 0, "WebKit WebRTC DTMF Sender");
});
static uint32_t nPipeline = 0;
auto pipelineName = makeString("webkit-dtmf-playout-pipeline-"_s, nPipeline);
m_pipeline = gst_pipeline_new(pipelineName.ascii().data());
registerActivePipeline(m_pipeline);
connectSimpleBusMessageCallback(m_pipeline.get());
GST_DEBUG_OBJECT(m_pipeline.get(), "DTMF sender backend created");
auto audioconvert = makeGStreamerElement("audioconvert"_s);
auto audioresample = makeGStreamerElement("audioresample"_s);
auto queue = gst_element_factory_make("queue", nullptr);
auto sink = createPlatformAudioSink("dtmf"_s);
gst_bin_add_many(GST_BIN_CAST(m_pipeline.get()), m_dtmfSrc.get(), audioconvert, audioresample, queue, sink, nullptr);
gst_element_link_many(m_dtmfSrc.get(), audioconvert, audioresample, queue, sink, nullptr);
gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
}
GStreamerDTMFSenderPrivate::~GStreamerDTMFSenderPrivate()
{
if (!m_pipeline)
return;
unregisterPipeline(m_pipeline);
disconnectSimpleBusMessageCallback(m_pipeline.get());
gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);
}
void GStreamerDTMFSenderPrivate::playTone(const RefPtr<RealtimeOutgoingAudioSourceGStreamer>& source, const char tone, size_t duration)
{
static HashMap<char, int, WTF::IntHash<char>, WTF::UnsignedWithZeroKeyHashTraits<char>> tones = {
{ '0', 0 },
{ '1', 1 },
{ '2', 2 },
{ '3', 3 },
{ '4', 4 },
{ '5', 5 },
{ '6', 6 },
{ '7', 7 },
{ '8', 8 },
{ '9', 9 },
{ '*', 10 },
{ '#', 11 },
{ 'A', 12 },
{ 'B', 13 },
{ 'C', 14 },
{ 'D', 15 }
};
auto element = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(source->bin().get()), "dtmfSource"));
auto pad = adoptGRef(gst_element_get_static_pad(element.get(), "src"));
if (tone == ',') {
GST_DEBUG_OBJECT(element.get(), "Playing silence for 2 seconds");
sleep(2_s);
m_onTonePlayed();
GST_DEBUG_OBJECT(element.get(), "Playing tone %c DONE", tone);
return;
}
auto toneNumber = tones.get(tone);
GST_DEBUG_OBJECT(pad.get(), "Playing tone %c for %zu milliseconds", tone, duration);
sendEvent(pad, toneNumber, 25, true);
RunLoop::mainSingleton().dispatchAfter(Seconds::fromMilliseconds(duration), [toneNumber, pad = WTFMove(pad), weakThis = ThreadSafeWeakPtr { *this }] {
RefPtr self = weakThis.get();
if (!self)
return;
self->stopTone(pad, toneNumber);
});
}
void GStreamerDTMFSenderPrivate::sendEvent(const GRefPtr<GstPad>& pad, int number, int volume, bool start)
{
auto event = adoptGRef(gst_event_new_custom(GST_EVENT_CUSTOM_UPSTREAM, gst_structure_new("dtmf-event", "type", G_TYPE_INT, 1, "number", G_TYPE_INT, number, "volume", G_TYPE_INT, volume, "start", G_TYPE_BOOLEAN, start, nullptr)));
gst_pad_send_event(pad.get(), event.ref());
gst_element_send_event(m_dtmfSrc.get(), event.leakRef());
}
void GStreamerDTMFSenderPrivate::stopTone(const GRefPtr<GstPad>& pad, int tone)
{
sendEvent(pad, tone, 0, false);
GST_DEBUG_OBJECT(pad.get(), "Playing tone %c DONE", tone);
m_onTonePlayed();
}
GStreamerDTMFSenderBackend::GStreamerDTMFSenderBackend(ThreadSafeWeakPtr<RealtimeOutgoingAudioSourceGStreamer>&& source)
: m_source(WTFMove(source))
{
RefPtr strongSource = m_source.get();
if (!strongSource) {
m_canInsertDTMF = false;
return;
}
m_senderPrivate = GStreamerDTMFSenderPrivate::create();
if (!m_senderPrivate) {
m_canInsertDTMF = false;
return;
}
m_canInsertDTMF = true;
}
bool GStreamerDTMFSenderBackend::canInsertDTMF()
{
RefPtr source = m_source.get();
if (!source)
return false;
return m_canInsertDTMF;
}
void GStreamerDTMFSenderBackend::playTone(const char tone, size_t duration, size_t interToneGap)
{
RefPtr source = m_source.get();
if (!source)
return;
if (!m_senderPrivate) [[unlikely]]
return;
m_duration = duration;
m_interToneGap = interToneGap;
if (m_isFirstTone)
m_isFirstTone = false;
else
sleep(Seconds::fromMilliseconds(interToneGap));
m_senderPrivate->playTone(source, tone, duration);
m_tones.append(tone);
}
String GStreamerDTMFSenderBackend::tones() const
{
return m_tones.toStringPreserveCapacity();
}
void GStreamerDTMFSenderBackend::onTonePlayed(Function<void()>&& onTonePlayed)
{
if (!m_senderPrivate) [[unlikely]]
return;
m_senderPrivate->setOnTonePlayedCallback(WTFMove(onTonePlayed));
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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