142 lines
4.9 KiB
C++
142 lines
4.9 KiB
C++
/*
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* Copyright (C) 2017-2019 Apple Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted, provided that the following conditions
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* are required to be met:
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*
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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* 3. Neither the name of Apple Inc. nor the names of
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* its contributors may be used to endorse or promote products derived
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* from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
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* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
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* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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* DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
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* ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
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* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
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* CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
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* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
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* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#pragma once
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#if USE(LIBWEBRTC)
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#include "LibWebRTCMacros.h"
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#include "MediaStreamTrackPrivate.h"
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#include "Timer.h"
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ALLOW_UNUSED_PARAMETERS_BEGIN
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#include <webrtc/api/media_stream_interface.h>
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ALLOW_UNUSED_PARAMETERS_END
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#include <wtf/Lock.h>
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#include <wtf/LoggerHelper.h>
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#include <wtf/ThreadSafeRefCounted.h>
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namespace webrtc {
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class AudioTrackInterface;
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class AudioTrackSinkInterface;
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}
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namespace WebCore {
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class RealtimeOutgoingAudioSource
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: public ThreadSafeRefCounted<RealtimeOutgoingAudioSource, WTF::DestructionThread::Main>
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, public webrtc::AudioSourceInterface
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, private MediaStreamTrackPrivate::Observer
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, private RealtimeMediaSource::AudioSampleObserver
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#if !RELEASE_LOG_DISABLED
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, private LoggerHelper
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#endif
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{
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public:
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static Ref<RealtimeOutgoingAudioSource> create(Ref<MediaStreamTrackPrivate>&& audioSource);
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~RealtimeOutgoingAudioSource();
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void start() { observeSource(); }
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void stop() { unobserveSource(); }
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void setSource(Ref<MediaStreamTrackPrivate>&&);
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MediaStreamTrackPrivate& source() const { return m_audioSource.get(); }
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protected:
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explicit RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&&);
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bool isSilenced() const { return m_muted || !m_enabled; }
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void sendAudioFrames(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames);
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#if !RELEASE_LOG_DISABLED
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// LoggerHelper API
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const Logger& logger() const final { return m_audioSource->logger(); }
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const void* logIdentifier() const final { return m_audioSource->logIdentifier(); }
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const char* logClassName() const final { return "RealtimeOutgoingAudioSource"; }
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WTFLogChannel& logChannel() const final;
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#endif
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private:
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// webrtc::AudioSourceInterface API
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void AddSink(webrtc::AudioTrackSinkInterface*) final;
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void RemoveSink(webrtc::AudioTrackSinkInterface*) final;
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void AddRef() const final { ref(); }
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rtc::RefCountReleaseStatus Release() const final
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{
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auto result = refCount() - 1;
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deref();
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return result ? rtc::RefCountReleaseStatus::kOtherRefsRemained : rtc::RefCountReleaseStatus::kDroppedLastRef;
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}
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SourceState state() const final { return kLive; }
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bool remote() const final { return false; }
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void RegisterObserver(webrtc::ObserverInterface*) final { }
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void UnregisterObserver(webrtc::ObserverInterface*) final { }
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void observeSource();
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void unobserveSource();
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void sourceMutedChanged();
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void sourceEnabledChanged();
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virtual bool isReachingBufferedAudioDataHighLimit() { return false; };
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virtual bool isReachingBufferedAudioDataLowLimit() { return false; };
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virtual bool hasBufferedEnoughData() { return false; };
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virtual void sourceUpdated() { }
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// MediaStreamTrackPrivate::Observer API
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void trackMutedChanged(MediaStreamTrackPrivate&) final { sourceMutedChanged(); }
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void trackEnabledChanged(MediaStreamTrackPrivate&) final { sourceEnabledChanged(); }
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void trackEnded(MediaStreamTrackPrivate&) final { }
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void trackSettingsChanged(MediaStreamTrackPrivate&) final { }
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void initializeConverter();
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Ref<MediaStreamTrackPrivate> m_audioSource;
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bool m_muted { false };
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bool m_enabled { true };
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mutable Lock m_sinksLock;
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HashSet<webrtc::AudioTrackSinkInterface*> m_sinks WTF_GUARDED_BY_LOCK(m_sinksLock);
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#if !RELEASE_LOG_DISABLED
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size_t m_chunksSent { 0 };
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#endif
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};
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} // namespace WebCore
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#endif // USE(LIBWEBRTC)
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