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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
*/
#pragma once
#if ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GRefPtrGStreamer.h"
#include "RTCDTMFSenderBackend.h"
#include "RealtimeOutgoingAudioSourceGStreamer.h"
#include <wtf/TZoneMalloc.h>
#include <wtf/ThreadSafeRefCounted.h>
#include <wtf/WeakPtr.h>
namespace WebCore {
class GStreamerDTMFSenderPrivate : public ThreadSafeRefCountedAndCanMakeThreadSafeWeakPtr<GStreamerDTMFSenderPrivate, WTF::DestructionThread::Main> {
public:
static RefPtr<GStreamerDTMFSenderPrivate> create();
~GStreamerDTMFSenderPrivate();
using OnTonePlayedCallback = Function<void()>;
void setOnTonePlayedCallback(OnTonePlayedCallback&& callback) { m_onTonePlayed = WTFMove(callback); }
void playTone(const RefPtr<RealtimeOutgoingAudioSourceGStreamer>&, const char tone, size_t duration);
private:
explicit GStreamerDTMFSenderPrivate(GRefPtr<GstElement>&&);
void sendEvent(const GRefPtr<GstPad>&, int number, int volume, bool start);
void stopTone(const GRefPtr<GstPad>&, int);
OnTonePlayedCallback m_onTonePlayed;
GRefPtr<GstElement> m_pipeline;
GRefPtr<GstElement> m_dtmfSrc;
};
class GStreamerDTMFSenderBackend final : public RTCDTMFSenderBackend {
WTF_MAKE_TZONE_ALLOCATED(GStreamerDTMFSenderBackend);
WTF_MAKE_NONCOPYABLE(GStreamerDTMFSenderBackend);
public:
explicit GStreamerDTMFSenderBackend(ThreadSafeWeakPtr<RealtimeOutgoingAudioSourceGStreamer>&&);
~GStreamerDTMFSenderBackend() = default;
private:
// RTCDTMFSenderBackend
bool canInsertDTMF() final;
void playTone(const char tone, size_t duration, size_t interToneGap) final;
void onTonePlayed(Function<void()>&&) final;
String tones() const final;
size_t duration() const final { return m_duration; }
size_t interToneGap() const final { return m_interToneGap; }
ThreadSafeWeakPtr<RealtimeOutgoingAudioSourceGStreamer> m_source;
RefPtr<GStreamerDTMFSenderPrivate> m_senderPrivate;
bool m_isFirstTone { true };
bool m_canInsertDTMF { false };
StringBuilder m_tones;
size_t m_duration;
size_t m_interToneGap;
};
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
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