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/*
* Copyright (C) 2017 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.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS 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 APPLE INC. OR ITS 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 "Timer.h"
#include <wtf/MonotonicTime.h>
#include <wtf/WorkQueue.h>
ALLOW_UNUSED_PARAMETERS_BEGIN
ALLOW_COMMA_BEGIN
#include <webrtc/modules/audio_device/include/audio_device.h>
ALLOW_UNUSED_PARAMETERS_END
ALLOW_COMMA_END
namespace WebCore {
class BaseAudioMediaStreamTrackRendererUnit;
class IncomingAudioMediaStreamTrackRendererUnit;
// LibWebRTCAudioModule is pulling streamed data to ensure audio data is passed to the audio track.
class LibWebRTCAudioModule : public webrtc::AudioDeviceModule {
WTF_MAKE_FAST_ALLOCATED;
public:
LibWebRTCAudioModule();
~LibWebRTCAudioModule();
static constexpr unsigned PollSamplesCount = 1;
void ref() { AddRef(); }
void deref() { Release(); }
#if PLATFORM(COCOA)
void startIncomingAudioRendering() { m_isRenderingIncomingAudio = true; }
void stopIncomingAudioRendering() { m_isRenderingIncomingAudio = false; }
BaseAudioMediaStreamTrackRendererUnit& incomingAudioMediaStreamTrackRendererUnit();
uint64_t currentAudioSampleCount() const { return m_currentAudioSampleCount; }
#endif
private:
template<typename U> U shouldNotBeCalled(U value) const
{
ASSERT_NOT_REACHED();
return value;
}
// webrtc::AudioDeviceModule API
int32_t StartPlayout() final;
int32_t StopPlayout() final;
int32_t RegisterAudioCallback(webrtc::AudioTransport*) final;
bool Playing() const final { return m_isPlaying; }
int32_t ActiveAudioLayer(AudioLayer*) const final { return shouldNotBeCalled(-1); }
int32_t Init() final { return 0; }
int32_t Terminate() final { return 0; }
bool Initialized() const final { return true; }
int16_t PlayoutDevices() final { return 0; }
int16_t RecordingDevices() final { return 0; }
int32_t PlayoutDeviceName(uint16_t, char[webrtc::kAdmMaxDeviceNameSize], char[webrtc::kAdmMaxGuidSize]) final { return 0; }
int32_t RecordingDeviceName(uint16_t, char[webrtc::kAdmMaxDeviceNameSize], char[webrtc::kAdmMaxGuidSize]) final { return 0; }
int32_t SetPlayoutDevice(uint16_t) final { return 0; }
int32_t SetPlayoutDevice(WindowsDeviceType) final { return 0; }
int32_t SetRecordingDevice(uint16_t) final { return 0; }
int32_t SetRecordingDevice(WindowsDeviceType) final { return 0; }
int32_t PlayoutIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t InitPlayout() final { return 0; }
bool PlayoutIsInitialized() const final { return true; }
int32_t RecordingIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t InitRecording() final { return 0; }
bool RecordingIsInitialized() const final { return false; }
int32_t StartRecording() final { return 0; }
int32_t StopRecording() final { return 0; }
bool Recording() const final { return 0; }
int32_t InitSpeaker() final { return 0; }
bool SpeakerIsInitialized() const final { return false; }
int32_t InitMicrophone() final { return 0; }
bool MicrophoneIsInitialized() const final { return false; }
int32_t MicrophoneVolumeIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t SpeakerVolumeIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t SetSpeakerVolume(uint32_t) final { return shouldNotBeCalled(-1); }
int32_t SpeakerVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t MaxSpeakerVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t MinSpeakerVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t SetMicrophoneVolume(uint32_t) final { return shouldNotBeCalled(-1); }
int32_t MicrophoneVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t MaxMicrophoneVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t MinMicrophoneVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t SpeakerMuteIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t SetSpeakerMute(bool) final { return shouldNotBeCalled(-1); }
int32_t SpeakerMute(bool*) const final { return shouldNotBeCalled(-1); }
int32_t MicrophoneMuteIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t SetMicrophoneMute(bool) final { return shouldNotBeCalled(-1); }
int32_t MicrophoneMute(bool*) const final { return shouldNotBeCalled(-1); }
int32_t StereoPlayoutIsAvailable(bool* available) const final { *available = false; return 0; }
int32_t SetStereoPlayout(bool) final { return 0; }
int32_t StereoPlayout(bool*) const final { return shouldNotBeCalled(-1); }
int32_t StereoRecordingIsAvailable(bool* available) const final { *available = false; return 0; }
int32_t SetStereoRecording(bool) final { return 0; }
int32_t StereoRecording(bool*) const final { return shouldNotBeCalled(-1); }
int32_t PlayoutDelay(uint16_t* delay) const final { *delay = 0; return 0; }
bool BuiltInAECIsAvailable() const final { return false; }
bool BuiltInAGCIsAvailable() const final { return false; }
bool BuiltInNSIsAvailable() const final { return false; }
int32_t EnableBuiltInAEC(bool) final { return shouldNotBeCalled(-1); }
int32_t EnableBuiltInAGC(bool) final { return shouldNotBeCalled(-1); }
int32_t EnableBuiltInNS(bool) final { return shouldNotBeCalled(-1); }
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(webrtc::AudioParameters*) const final { return shouldNotBeCalled(-1); }
int GetRecordAudioParameters(webrtc::AudioParameters*) const final { return shouldNotBeCalled(-1); }
#endif
private:
void pollAudioData();
void pollFromSource();
void logTimerFired();
Seconds computeDelayUntilNextPolling();
static constexpr Seconds logTimerInterval = 2_s;
Ref<WorkQueue> m_queue;
bool m_isPlaying { false };
webrtc::AudioTransport* m_audioTransport { nullptr };
MonotonicTime m_pollingTime;
std::unique_ptr<Timer> m_logTimer;
int m_timeSpent { 0 };
#if PLATFORM(COCOA)
uint64_t m_currentAudioSampleCount { 0 };
bool m_isRenderingIncomingAudio { false };
std::unique_ptr<IncomingAudioMediaStreamTrackRendererUnit> m_incomingAudioMediaStreamTrackRendererUnit;
#endif
};
} // namespace WebCore
#endif // USE(LIBWEBRTC)
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