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/*
* Copyright (C) 2017-2019 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer
* in the documentation and/or other materials provided with the
* distribution.
* 3. Neither the name of Ericsson nor the names of its contributors
* may be used to endorse or promote products derived from this
* software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
* OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#pragma once
#if USE(LIBWEBRTC)
#include "LibWebRTCMacros.h"
#include "RealtimeMediaSource.h"
ALLOW_UNUSED_PARAMETERS_BEGIN
#include <webrtc/api/media_stream_interface.h>
ALLOW_UNUSED_PARAMETERS_END
#include <wtf/RetainPtr.h>
namespace WebCore {
class LibWebRTCAudioModule;
class RealtimeIncomingAudioSource
: public RealtimeMediaSource
, private webrtc::AudioTrackSinkInterface
, private webrtc::ObserverInterface
{
public:
static Ref<RealtimeIncomingAudioSource> create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
void setAudioModule(RefPtr<LibWebRTCAudioModule>&&);
LibWebRTCAudioModule* audioModule() { return m_audioModule.get(); }
protected:
RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
~RealtimeIncomingAudioSource();
#if !RELEASE_LOG_DISABLED
const char* logClassName() const final { return "RealtimeIncomingAudioSource"; }
#endif
// RealtimeMediaSource API
void startProducingData() override;
void stopProducingData() override;
private:
// webrtc::AudioTrackSinkInterface API
void OnData(const void* /* audioData */, int /* bitsPerSample */, int /* sampleRate */, size_t /* numberOfChannels */, size_t /* numberOfFrames */) override { };
// webrtc::ObserverInterface API
void OnChanged() final;
const RealtimeMediaSourceCapabilities& capabilities() final;
const RealtimeMediaSourceSettings& settings() final;
bool isIncomingAudioSource() const final { return true; }
RealtimeMediaSourceSettings m_currentSettings;
rtc::scoped_refptr<webrtc::AudioTrackInterface> m_audioTrack;
RefPtr<LibWebRTCAudioModule> m_audioModule;
#if !RELEASE_LOG_DISABLED
mutable RefPtr<const Logger> m_logger;
const void* m_logIdentifier;
#endif
};
} // namespace WebCore
SPECIALIZE_TYPE_TRAITS_BEGIN(WebCore::RealtimeIncomingAudioSource)
static bool isType(const WebCore::RealtimeMediaSource& source) { return source.isIncomingAudioSource(); }
SPECIALIZE_TYPE_TRAITS_END()
#endif // USE(LIBWEBRTC)