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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 <webrtc/base/messagehandler.h>
#include <webrtc/base/thread.h>
#include <webrtc/modules/audio_device/include/audio_device.h>
namespace WebCore {
// LibWebRTCAudioModule is pulling streamed data to ensure audio data is passed to the audio track.
class LibWebRTCAudioModule final : public webrtc::AudioDeviceModule, private rtc::MessageHandler {
public:
LibWebRTCAudioModule();
private:
template<typename U> U shouldNotBeCalled(U value) const
{
ASSERT_NOT_REACHED();
return value;
}
int32_t AddRef() const final { return 1; }
int32_t Release() const final { return 1; }
void OnMessage(rtc::Message*);
// webrtc::AudioDeviceModule API
int32_t StartPlayout() final;
int32_t StopPlayout() final;
int32_t RegisterAudioCallback(webrtc::AudioTransport*) final;
bool Playing() const final { return m_isPlaying; }
// Set it to a big value so that we are not called often but not close to int64_t max so that it does not create integer overflows.
int64_t TimeUntilNextProcess() final { return 100000000; }
void Process() final { }
int32_t ActiveAudioLayer(AudioLayer*) const final { return shouldNotBeCalled(-1); }
ErrorCode LastError() const final { return kAdmErrNone; }
int32_t RegisterEventObserver(webrtc::AudioDeviceObserver*) final { return 0; }
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 SetAGC(bool) final { return 0; }
bool AGC() const final { return shouldNotBeCalled(0); }
int32_t SetWaveOutVolume(uint16_t, uint16_t) final { return shouldNotBeCalled(-1); }
int32_t WaveOutVolume(uint16_t*, uint16_t*) const final { return shouldNotBeCalled(-1); }
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 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 SpeakerVolumeStepSize(uint16_t*) const final { return shouldNotBeCalled(-1); }
int32_t MicrophoneVolumeIsAvailable(bool*) 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 MicrophoneVolumeStepSize(uint16_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 MicrophoneBoostIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
int32_t SetMicrophoneBoost(bool) final { return shouldNotBeCalled(-1); }
int32_t MicrophoneBoost(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 SetRecordingChannel(const ChannelType) final { return 0; }
int32_t RecordingChannel(ChannelType*) const final { return shouldNotBeCalled(-1); }
int32_t SetPlayoutBuffer(const BufferType, uint16_t) final { return shouldNotBeCalled(-1); }
int32_t PlayoutBuffer(BufferType*, uint16_t*) const final { return shouldNotBeCalled(-1); }
int32_t PlayoutDelay(uint16_t* delay) const final { *delay = 0; return 0; }
int32_t RecordingDelay(uint16_t*) const final { return shouldNotBeCalled(-1); }
int32_t CPULoad(uint16_t*) const final { return shouldNotBeCalled(-1); }
int32_t StartRawOutputFileRecording(const char[webrtc::kAdmMaxFileNameSize]) final { return shouldNotBeCalled(-1); }
int32_t StopRawOutputFileRecording() final { return shouldNotBeCalled(-1); }
int32_t StartRawInputFileRecording(const char[webrtc::kAdmMaxFileNameSize]) final { return shouldNotBeCalled(-1); }
int32_t StopRawInputFileRecording() final { return shouldNotBeCalled(-1); }
int32_t SetRecordingSampleRate(const uint32_t) final { return shouldNotBeCalled(-1); }
int32_t RecordingSampleRate(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t SetPlayoutSampleRate(const uint32_t) final { return shouldNotBeCalled(-1); }
int32_t PlayoutSampleRate(uint32_t*) const final { return shouldNotBeCalled(-1); }
int32_t ResetAudioDevice() final { return shouldNotBeCalled(-1); }
int32_t SetLoudspeakerStatus(bool) final { return shouldNotBeCalled(-1); }
int32_t GetLoudspeakerStatus(bool*) const final { return shouldNotBeCalled(-1); }
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 StartPlayoutOnAudioThread();
void PollFromSource();
std::unique_ptr<rtc::Thread> m_audioTaskRunner;
bool m_isPlaying = false;
webrtc::AudioTransport* m_audioTransport = nullptr;
};
} // namespace WebCore
#endif // USE(LIBWEBRTC)
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