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o3de/Gems/Microphone/Code/Source/Platform/iOS/MicrophoneSystemComponent_iOS.mm
T
Esteban Papp 1f9b284de2 Merge branch 'development' into cmake/SPEC-7179
Signed-off-by: Esteban Papp <81431996+amznestebanpapp@users.noreply.github.com>

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/*
* Copyright (c) Contributors to the Open 3D Engine Project.
* For complete copyright and license terms please see the LICENSE at the root of this distribution.
*
* SPDX-License-Identifier: Apache-2.0 OR MIT
*
*/
#import <Foundation/Foundation.h>
#import <AudioToolbox/AudioToolbox.h>
#include "MicrophoneSystemComponent.h"
#include "SimpleDownsample.h"
#include <AzCore/std/smart_ptr/unique_ptr.h>
#include <AzCore/EBus/EBus.h>
#include <AudioRingBuffer.h>
#include <IAudioInterfacesCommonData.h>
#if defined(USE_LIBSAMPLERATE)
#include <samplerate.h>
#endif // USE_LIBSAMPLERATE
namespace Audio
{
class MicrophoneSystemComponentIOS : public MicrophoneSystemComponent::Implementation
{
public:
bool InitializeDevice() override
{
m_isCapturing = false;
OSStatus status;
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
AudioComponent inputComponent = AudioComponentFindNext(nullptr, &desc);
status = AudioComponentInstanceNew(inputComponent, &m_audioUnit);
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error creating audio component: %d", status);
return false;
}
AZ::u32 flag = 1;
status = AudioUnitSetProperty(m_audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBus,
&flag,
sizeof(flag));
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error enabling audio input IO: %d", status);
return false;
}
// disable output
flag = 0;
status = AudioUnitSetProperty(m_audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBus,
&flag,
sizeof(flag));
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error enabling audio input IO: %d", status);
return false;
}
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
m_config.m_sampleRate = audioFormat.mSampleRate;
m_config.m_numChannels = audioFormat.mChannelsPerFrame;
m_config.m_bitsPerSample = audioFormat.mBitsPerChannel;
m_config.SetBufferSizeFromFrameCount(8192);
status = AudioUnitSetProperty(m_audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error setting microphone inuput configuration: %d", status);
return false;
}
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = this;
status = AudioUnitSetProperty(m_audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error setting microphone callback: %d", status);
return false;
}
// This disables buffer allocation as we provide our own
flag = 0;
status = AudioUnitSetProperty(m_audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
// Initialise
status = AudioUnitInitialize(m_audioUnit);
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error initializing microphone: %d", status);
return false;
}
return true;
}
void ShutdownDevice() override
{
m_isCapturing = false;
AudioUnitUninitialize(m_audioUnit);
}
bool StartSession() override
{
m_captureData.reset(aznew Audio::RingBuffer<AZ::s16>(m_config.GetSampleCountFromBufferSize()));
OSStatus status = AudioOutputUnitStart(m_audioUnit);
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error starting microphone: %d", status);
return false;
}
m_isCapturing = true;
return true;
}
void EndSession() override
{
OSStatus status = AudioOutputUnitStop(m_audioUnit);
if(status != 0)
{
AZ_Error("iOSMicrophone", false, "Error stopping microphone: %d", status);
}
m_isCapturing = false;
if(m_captureData)
{
m_captureData.reset();
}
}
bool IsCapturing() override
{
return m_isCapturing;
}
SAudioInputConfig GetFormatConfig() const override
{
return m_config;
}
AZStd::size_t GetData(void** outputData, AZStd::size_t numFrames, const SAudioInputConfig& targetConfig, bool shouldDeinterleave) override
{
bool changeSampleType = (targetConfig.m_sampleType != m_config.m_sampleType);
bool changeSampleRate = (targetConfig.m_sampleRate != m_config.m_sampleRate);
bool changeNumChannels = (targetConfig.m_numChannels != m_config.m_numChannels);
#if defined(USE_LIBSAMPLERATE)
// pending port of LIBSAMPLERATE to iOS
return {};
#else
if (changeSampleType || changeNumChannels)
{
// Without the SRC library, any change is unsupported!
return {};
}
else if (changeSampleRate)
{
if(targetConfig.m_sampleRate > m_config.m_sampleRate)
{
AZ_Error("MacOSMicrophone", false, "Target sample rate is larger than source sample rate, this is not supported");
return {};
}
auto sourceBuffer = new AZ::s16[numFrames];
AZStd::size_t targetSize = GetDownsampleSize(numFrames, m_config.m_sampleRate, targetConfig.m_sampleRate);
auto targetBuffer = new AZ::s16[targetSize];
numFrames = m_captureData->ConsumeData(reinterpret_cast<void**>(&sourceBuffer), numFrames, m_config.m_numChannels, false);
if(numFrames > 0)
{
Downsample(sourceBuffer, numFrames, m_config.m_sampleRate, targetBuffer, targetSize, targetConfig.m_sampleRate);
// swap target data to output
::memcpy(*outputData, targetBuffer, numFrames * targetConfig.m_numChannels * (targetConfig.m_bitsPerSample >> 3));
}
delete [] sourceBuffer;
delete [] targetBuffer;
return numFrames;
}
else
{
// No change to the data from Input to Output
return m_captureData->ConsumeData(outputData, numFrames, m_config.m_numChannels, shouldDeinterleave);
}
#endif
return {};
}
void ProcessAudio(AudioBufferList* bufferList)
{
AudioBuffer sourceBuffer = bufferList->mBuffers[0];
m_captureData->AddData((int16*)sourceBuffer.mData, sourceBuffer.mDataByteSize / 2, m_config.m_numChannels);
}
AudioComponentInstance m_audioUnit;
private:
static OSStatus recordingCallback(void* inRefCon,
AudioUnitRenderActionFlags* ioActionFlags,
const AudioTimeStamp* inTimeStamp,
AZ::u32 inBusNumber,
AZ::u32 inNumberFrames,
AudioBufferList* ioData) {
auto impl = reinterpret_cast<MicrophoneSystemComponentIOS*>(inRefCon);
// Samples are 16 bits = 2 bytes.
// 1 frame includes only 1 sample
// one channel as mono was selected in setup
AudioBuffer buffer;
buffer.mNumberChannels = 1;
buffer.mDataByteSize = inNumberFrames * 2;
buffer.mData = new AZ::u8[buffer.mDataByteSize];
// Put buffer in a AudioBufferList
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
// Obtain recorded samples
OSStatus status;
status = AudioUnitRender(impl->m_audioUnit,
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames,
&bufferList);
impl->ProcessAudio(&bufferList);
// release the malloc'ed data in the buffer we created earlier
delete [] (AZ::u8*)bufferList.mBuffers[0].mData;
return noErr;
}
Audio::SAudioInputConfig m_config;
bool m_isCapturing = false;
AZStd::unique_ptr<Audio::RingBufferBase> m_captureData = nullptr;
const AudioUnitElement kInputBus = 1;
const AudioUnitElement kOutputBus = 0;
};
///////////////////////////////////////////////////////////////////////////////////////////////
MicrophoneSystemComponent::Implementation* MicrophoneSystemComponent::Implementation::Create()
{
return aznew MicrophoneSystemComponentIOS();
}
}