Chiunque ha eseguito correttamente il rendering offline utilizzando core-Audio.?core offline rendering offline GenericOutput
Ho dovuto mixare due file audio e applicare riverbero (utilizzato 2 AudioFilePlayer, MultiChannelMixer, Reverb2 e RemoteIO). Ho funzionato. e potrei salvarlo mentre è in anteprima (su renderCallBack di RemoteIO).
Ho bisogno di salvarlo senza giocare (offline). Grazie in anticipo.
Rendering offline Ha funzionato per me utilizzando GenericOutput AudioUnit. Sto condividendo il codice di lavoro qui. Il framework core-audio sembra un po 'però. Ma piccole-piccole cose come ASBD, parametri ... ecc. Stanno facendo questi problemi. prova a farlo funzionare. Non arrenderti :-). core-audio è molto potente e utile mentre si parla di audio di basso livello. Questo è quello che ho imparato da queste ultime settimane. Godetevi :-D ....
Dichiarare questi in .h
//AUGraph
AUGraph mGraph;
//Audio Unit References
AudioUnit mFilePlayer;
AudioUnit mFilePlayer2;
AudioUnit mReverb;
AudioUnit mTone;
AudioUnit mMixer;
AudioUnit mGIO;
//Audio File Location
AudioFileID inputFile;
AudioFileID inputFile2;
//Audio file refereces for saving
ExtAudioFileRef extAudioFile;
//Standard sample rate
Float64 graphSampleRate;
AudioStreamBasicDescription stereoStreamFormat864;
Float64 MaxSampleTime;
// in classe .m
- (id) init
{
self = [super init];
graphSampleRate = 44100.0;
MaxSampleTime = 0.0;
UInt32 category = kAudioSessionCategory_MediaPlayback;
CheckError(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
sizeof(category),
&category),
"Couldn't set category on audio session");
[self initializeAUGraph];
return self;
}
// ASBD configurazione
- (void) setupStereoStream864 {
// The AudioUnitSampleType data type is the recommended type for sample data in audio
// units. This obtains the byte size of the type for use in filling in the ASBD.
size_t bytesPerSample = sizeof (AudioUnitSampleType);
// Fill the application audio format struct's fields to define a linear PCM,
// stereo, noninterleaved stream at the hardware sample rate.
stereoStreamFormat864.mFormatID = kAudioFormatLinearPCM;
stereoStreamFormat864.mFormatFlags = kAudioFormatFlagsAudioUnitCanonical;
stereoStreamFormat864.mBytesPerPacket = bytesPerSample;
stereoStreamFormat864.mFramesPerPacket = 1;
stereoStreamFormat864.mBytesPerFrame = bytesPerSample;
stereoStreamFormat864.mChannelsPerFrame = 2; // 2 indicates stereo
stereoStreamFormat864.mBitsPerChannel = 8 * bytesPerSample;
stereoStreamFormat864.mSampleRate = graphSampleRate;
}
// configurazione AUGraph
- (void)initializeAUGraph
{
[self setupStereoStream864];
// Setup the AUGraph, add AUNodes, and make connections
// create a new AUGraph
CheckError(NewAUGraph(&mGraph),"Couldn't create new graph");
// AUNodes represent AudioUnits on the AUGraph and provide an
// easy means for connecting audioUnits together.
AUNode filePlayerNode;
AUNode filePlayerNode2;
AUNode mixerNode;
AUNode reverbNode;
AUNode toneNode;
AUNode gOutputNode;
// file player component
AudioComponentDescription filePlayer_desc;
filePlayer_desc.componentType = kAudioUnitType_Generator;
filePlayer_desc.componentSubType = kAudioUnitSubType_AudioFilePlayer;
filePlayer_desc.componentFlags = 0;
filePlayer_desc.componentFlagsMask = 0;
filePlayer_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// file player component2
AudioComponentDescription filePlayer2_desc;
filePlayer2_desc.componentType = kAudioUnitType_Generator;
filePlayer2_desc.componentSubType = kAudioUnitSubType_AudioFilePlayer;
filePlayer2_desc.componentFlags = 0;
filePlayer2_desc.componentFlagsMask = 0;
filePlayer2_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Create AudioComponentDescriptions for the AUs we want in the graph
// mixer component
AudioComponentDescription mixer_desc;
mixer_desc.componentType = kAudioUnitType_Mixer;
mixer_desc.componentSubType = kAudioUnitSubType_MultiChannelMixer;
mixer_desc.componentFlags = 0;
mixer_desc.componentFlagsMask = 0;
mixer_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Create AudioComponentDescriptions for the AUs we want in the graph
// Reverb component
AudioComponentDescription reverb_desc;
reverb_desc.componentType = kAudioUnitType_Effect;
reverb_desc.componentSubType = kAudioUnitSubType_Reverb2;
reverb_desc.componentFlags = 0;
reverb_desc.componentFlagsMask = 0;
reverb_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
//tone component
AudioComponentDescription tone_desc;
tone_desc.componentType = kAudioUnitType_FormatConverter;
//tone_desc.componentSubType = kAudioUnitSubType_NewTimePitch;
tone_desc.componentSubType = kAudioUnitSubType_Varispeed;
tone_desc.componentFlags = 0;
tone_desc.componentFlagsMask = 0;
tone_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
AudioComponentDescription gOutput_desc;
gOutput_desc.componentType = kAudioUnitType_Output;
gOutput_desc.componentSubType = kAudioUnitSubType_GenericOutput;
gOutput_desc.componentFlags = 0;
gOutput_desc.componentFlagsMask = 0;
gOutput_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
//Add nodes to graph
// Add nodes to the graph to hold our AudioUnits,
// You pass in a reference to the AudioComponentDescription
// and get back an AudioUnit
AUGraphAddNode(mGraph, &filePlayer_desc, &filePlayerNode);
AUGraphAddNode(mGraph, &filePlayer2_desc, &filePlayerNode2);
AUGraphAddNode(mGraph, &mixer_desc, &mixerNode);
AUGraphAddNode(mGraph, &reverb_desc, &reverbNode);
AUGraphAddNode(mGraph, &tone_desc, &toneNode);
AUGraphAddNode(mGraph, &gOutput_desc, &gOutputNode);
//Open the graph early, initialize late
// open the graph AudioUnits are open but not initialized (no resource allocation occurs here)
CheckError(AUGraphOpen(mGraph),"Couldn't Open the graph");
//Reference to Nodes
// get the reference to the AudioUnit object for the file player graph node
AUGraphNodeInfo(mGraph, filePlayerNode, NULL, &mFilePlayer);
AUGraphNodeInfo(mGraph, filePlayerNode2, NULL, &mFilePlayer2);
AUGraphNodeInfo(mGraph, reverbNode, NULL, &mReverb);
AUGraphNodeInfo(mGraph, toneNode, NULL, &mTone);
AUGraphNodeInfo(mGraph, mixerNode, NULL, &mMixer);
AUGraphNodeInfo(mGraph, gOutputNode, NULL, &mGIO);
AUGraphConnectNodeInput(mGraph, filePlayerNode, 0, reverbNode, 0);
AUGraphConnectNodeInput(mGraph, reverbNode, 0, toneNode, 0);
AUGraphConnectNodeInput(mGraph, toneNode, 0, mixerNode,0);
AUGraphConnectNodeInput(mGraph, filePlayerNode2, 0, mixerNode, 1);
AUGraphConnectNodeInput(mGraph, mixerNode, 0, gOutputNode, 0);
UInt32 busCount = 2; // bus count for mixer unit input
//Setup mixer unit bus count
CheckError(AudioUnitSetProperty (
mMixer,
kAudioUnitProperty_ElementCount,
kAudioUnitScope_Input,
0,
&busCount,
sizeof (busCount)
),
"Couldn't set mixer unit's bus count");
//Enable metering mode to view levels input and output levels of mixer
UInt32 onValue = 1;
CheckError(AudioUnitSetProperty(mMixer,
kAudioUnitProperty_MeteringMode,
kAudioUnitScope_Input,
0,
&onValue,
sizeof(onValue)),
"error");
// Increase the maximum frames per slice allows the mixer unit to accommodate the
// larger slice size used when the screen is locked.
UInt32 maximumFramesPerSlice = 4096;
CheckError(AudioUnitSetProperty (
mMixer,
kAudioUnitProperty_MaximumFramesPerSlice,
kAudioUnitScope_Global,
0,
&maximumFramesPerSlice,
sizeof (maximumFramesPerSlice)
),
"Couldn't set mixer units maximum framers per slice");
// set the audio data format of tone Unit
AudioUnitSetProperty(mTone,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Global,
0,
&stereoStreamFormat864,
sizeof(AudioStreamBasicDescription));
// set the audio data format of reverb Unit
AudioUnitSetProperty(mReverb,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Global,
0,
&stereoStreamFormat864,
sizeof(AudioStreamBasicDescription));
// set initial reverb
AudioUnitParameterValue reverbTime = 2.5;
AudioUnitSetParameter(mReverb, 4, kAudioUnitScope_Global, 0, reverbTime, 0);
AudioUnitSetParameter(mReverb, 5, kAudioUnitScope_Global, 0, reverbTime, 0);
AudioUnitSetParameter(mReverb, 0, kAudioUnitScope_Global, 0, 0, 0);
AudioStreamBasicDescription auEffectStreamFormat;
UInt32 asbdSize = sizeof (auEffectStreamFormat);
memset (&auEffectStreamFormat, 0, sizeof (auEffectStreamFormat));
// get the audio data format from reverb
CheckError(AudioUnitGetProperty(mReverb,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&auEffectStreamFormat,
&asbdSize),
"Couldn't get aueffectunit ASBD");
auEffectStreamFormat.mSampleRate = graphSampleRate;
// set the audio data format of mixer Unit
CheckError(AudioUnitSetProperty(mMixer,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
0,
&auEffectStreamFormat, sizeof(auEffectStreamFormat)),
"Couldn't set ASBD on mixer output");
CheckError(AUGraphInitialize(mGraph),"Couldn't Initialize the graph");
[self setUpAUFilePlayer];
[self setUpAUFilePlayer2];
}
// impostazione riproduzione di file audio qui sto installando il file vocale
-(OSStatus) setUpAUFilePlayer{
NSString *songPath = [[NSBundle mainBundle] pathForResource: @"testVoice" ofType:@".m4a"];
CFURLRef songURL = (CFURLRef) [NSURL fileURLWithPath:songPath];
// open the input audio file
CheckError(AudioFileOpenURL(songURL, kAudioFileReadPermission, 0, &inputFile),
"setUpAUFilePlayer AudioFileOpenURL failed");
AudioStreamBasicDescription fileASBD;
// get the audio data format from the file
UInt32 propSize = sizeof(fileASBD);
CheckError(AudioFileGetProperty(inputFile, kAudioFilePropertyDataFormat,
&propSize, &fileASBD),
"setUpAUFilePlayer couldn't get file's data format");
// tell the file player unit to load the file we want to play
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduledFileIDs,
kAudioUnitScope_Global, 0, &inputFile, sizeof(inputFile)),
"setUpAUFilePlayer AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileIDs] failed");
UInt64 nPackets;
UInt32 propsize = sizeof(nPackets);
CheckError(AudioFileGetProperty(inputFile, kAudioFilePropertyAudioDataPacketCount,
&propsize, &nPackets),
"setUpAUFilePlayer AudioFileGetProperty[kAudioFilePropertyAudioDataPacketCount] failed");
// tell the file player AU to play the entire file
ScheduledAudioFileRegion rgn;
memset (&rgn.mTimeStamp, 0, sizeof(rgn.mTimeStamp));
rgn.mTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
rgn.mTimeStamp.mSampleTime = 0;
rgn.mCompletionProc = NULL;
rgn.mCompletionProcUserData = NULL;
rgn.mAudioFile = inputFile;
rgn.mLoopCount = -1;
rgn.mStartFrame = 0;
rgn.mFramesToPlay = nPackets * fileASBD.mFramesPerPacket;
if (MaxSampleTime < rgn.mFramesToPlay)
{
MaxSampleTime = rgn.mFramesToPlay;
}
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduledFileRegion,
kAudioUnitScope_Global, 0,&rgn, sizeof(rgn)),
"setUpAUFilePlayer1 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileRegion] failed");
// prime the file player AU with default values
UInt32 defaultVal = 0;
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduledFilePrime,
kAudioUnitScope_Global, 0, &defaultVal, sizeof(defaultVal)),
"setUpAUFilePlayer AudioUnitSetProperty[kAudioUnitProperty_ScheduledFilePrime] failed");
// tell the file player AU when to start playing (-1 sample time means next render cycle)
AudioTimeStamp startTime;
memset (&startTime, 0, sizeof(startTime));
startTime.mFlags = kAudioTimeStampSampleTimeValid;
startTime.mSampleTime = -1;
CheckError(AudioUnitSetProperty(mFilePlayer, kAudioUnitProperty_ScheduleStartTimeStamp,
kAudioUnitScope_Global, 0, &startTime, sizeof(startTime)),
"setUpAUFilePlayer AudioUnitSetProperty[kAudioUnitProperty_ScheduleStartTimeStamp]");
return noErr;
}
// impostazione riproduzione di file audio qui sto installando il file BGMusic
-(OSStatus) setUpAUFilePlayer2{
NSString *songPath = [[NSBundle mainBundle] pathForResource: @"BGmusic" ofType:@".mp3"];
CFURLRef songURL = (CFURLRef) [NSURL fileURLWithPath:songPath];
// open the input audio file
CheckError(AudioFileOpenURL(songURL, kAudioFileReadPermission, 0, &inputFile2),
"setUpAUFilePlayer2 AudioFileOpenURL failed");
AudioStreamBasicDescription fileASBD;
// get the audio data format from the file
UInt32 propSize = sizeof(fileASBD);
CheckError(AudioFileGetProperty(inputFile2, kAudioFilePropertyDataFormat,
&propSize, &fileASBD),
"setUpAUFilePlayer2 couldn't get file's data format");
// tell the file player unit to load the file we want to play
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduledFileIDs,
kAudioUnitScope_Global, 0, &inputFile2, sizeof(inputFile2)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileIDs] failed");
UInt64 nPackets;
UInt32 propsize = sizeof(nPackets);
CheckError(AudioFileGetProperty(inputFile2, kAudioFilePropertyAudioDataPacketCount,
&propsize, &nPackets),
"setUpAUFilePlayer2 AudioFileGetProperty[kAudioFilePropertyAudioDataPacketCount] failed");
// tell the file player AU to play the entire file
ScheduledAudioFileRegion rgn;
memset (&rgn.mTimeStamp, 0, sizeof(rgn.mTimeStamp));
rgn.mTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
rgn.mTimeStamp.mSampleTime = 0;
rgn.mCompletionProc = NULL;
rgn.mCompletionProcUserData = NULL;
rgn.mAudioFile = inputFile2;
rgn.mLoopCount = -1;
rgn.mStartFrame = 0;
rgn.mFramesToPlay = nPackets * fileASBD.mFramesPerPacket;
if (MaxSampleTime < rgn.mFramesToPlay)
{
MaxSampleTime = rgn.mFramesToPlay;
}
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduledFileRegion,
kAudioUnitScope_Global, 0,&rgn, sizeof(rgn)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFileRegion] failed");
// prime the file player AU with default values
UInt32 defaultVal = 0;
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduledFilePrime,
kAudioUnitScope_Global, 0, &defaultVal, sizeof(defaultVal)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduledFilePrime] failed");
// tell the file player AU when to start playing (-1 sample time means next render cycle)
AudioTimeStamp startTime;
memset (&startTime, 0, sizeof(startTime));
startTime.mFlags = kAudioTimeStampSampleTimeValid;
startTime.mSampleTime = -1;
CheckError(AudioUnitSetProperty(mFilePlayer2, kAudioUnitProperty_ScheduleStartTimeStamp,
kAudioUnitScope_Global, 0, &startTime, sizeof(startTime)),
"setUpAUFilePlayer2 AudioUnitSetProperty[kAudioUnitProperty_ScheduleStartTimeStamp]");
return noErr;
}
// Inizi a salvare file
- (void)startRecordingAAC{
AudioStreamBasicDescription destinationFormat;
memset(&destinationFormat, 0, sizeof(destinationFormat));
destinationFormat.mChannelsPerFrame = 2;
destinationFormat.mFormatID = kAudioFormatMPEG4AAC;
UInt32 size = sizeof(destinationFormat);
OSStatus result = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &destinationFormat);
if(result) printf("AudioFormatGetProperty %ld \n", result);
NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString *documentsDirectory = [paths objectAtIndex:0];
NSString *destinationFilePath = [[NSString alloc] initWithFormat: @"%@/output.m4a", documentsDirectory];
CFURLRef destinationURL = CFURLCreateWithFileSystemPath(kCFAllocatorDefault,
(CFStringRef)destinationFilePath,
kCFURLPOSIXPathStyle,
false);
[destinationFilePath release];
// specify codec Saving the output in .m4a format
result = ExtAudioFileCreateWithURL(destinationURL,
kAudioFileM4AType,
&destinationFormat,
NULL,
kAudioFileFlags_EraseFile,
&extAudioFile);
if(result) printf("ExtAudioFileCreateWithURL %ld \n", result);
CFRelease(destinationURL);
// This is a very important part and easiest way to set the ASBD for the File with correct format.
AudioStreamBasicDescription clientFormat;
UInt32 fSize = sizeof (clientFormat);
memset(&clientFormat, 0, sizeof(clientFormat));
// get the audio data format from the Output Unit
CheckError(AudioUnitGetProperty(mGIO,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
0,
&clientFormat,
&fSize),"AudioUnitGetProperty on failed");
// set the audio data format of mixer Unit
CheckError(ExtAudioFileSetProperty(extAudioFile,
kExtAudioFileProperty_ClientDataFormat,
sizeof(clientFormat),
&clientFormat),
"ExtAudioFileSetProperty kExtAudioFileProperty_ClientDataFormat failed");
// specify codec
UInt32 codec = kAppleHardwareAudioCodecManufacturer;
CheckError(ExtAudioFileSetProperty(extAudioFile,
kExtAudioFileProperty_CodecManufacturer,
sizeof(codec),
&codec),"ExtAudioFileSetProperty on extAudioFile Faild");
CheckError(ExtAudioFileWriteAsync(extAudioFile, 0, NULL),"ExtAudioFileWriteAsync Failed");
[self pullGenericOutput];
}
// Manuale di alimentazione e ottenere i dati/Buffer dal GenericOutput nodo.
-(void)pullGenericOutput{
AudioUnitRenderActionFlags flags = 0;
AudioTimeStamp inTimeStamp;
memset(&inTimeStamp, 0, sizeof(AudioTimeStamp));
inTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;
UInt32 busNumber = 0;
UInt32 numberFrames = 512;
inTimeStamp.mSampleTime = 0;
int channelCount = 2;
NSLog(@"Final numberFrames :%li",numberFrames);
int totFrms = MaxSampleTime;
while (totFrms > 0)
{
if (totFrms < numberFrames)
{
numberFrames = totFrms;
NSLog(@"Final numberFrames :%li",numberFrames);
}
else
{
totFrms -= numberFrames;
}
AudioBufferList *bufferList = (AudioBufferList*)malloc(sizeof(AudioBufferList)+sizeof(AudioBuffer)*(channelCount-1));
bufferList->mNumberBuffers = channelCount;
for (int j=0; j<channelCount; j++)
{
AudioBuffer buffer = {0};
buffer.mNumberChannels = 1;
buffer.mDataByteSize = numberFrames*sizeof(AudioUnitSampleType);
buffer.mData = calloc(numberFrames, sizeof(AudioUnitSampleType));
bufferList->mBuffers[j] = buffer;
}
CheckError(AudioUnitRender(mGIO,
&flags,
&inTimeStamp,
busNumber,
numberFrames,
bufferList),
"AudioUnitRender mGIO");
CheckError(ExtAudioFileWrite(extAudioFile, numberFrames, bufferList),("extaudiofilewrite fail"));
}
[self FilesSavingCompleted];
}
// FilesSavingCompleted
-(void)FilesSavingCompleted{
OSStatus status = ExtAudioFileDispose(extAudioFile);
printf("OSStatus(ExtAudioFileDispose): %ld\n", status);
}
Un modo per eseguire il rendering offline è rimuovere l'unità RemoteIO e chiamare esplicitamente lo AudioUnitRender nell'unità più a destra del grafico (il mixer o l'unità di riverbero in base alla topologia). Effettuando questo in un ciclo finché non si estinguono i campioni da entrambi i file sorgente e si scrivono i buffer di esempio risultanti con Extended Audio File Services, è possibile creare un file audio compresso del mixdown. Dovrai farlo su un thread in background per mantenere l'interfaccia utente reattiva, ma ho già usato questa tecnica con un po 'di successo.
Grazie a @warrenm, lo proverò e aggiornerò qui. :) –
Sopra il codice sta lavorando sul dispositivo iOS7 ma non funziona su dispositivi iOS 8 e su tutti i simulatori. avevo sostituito il seguente segmento di codice
UInt32 category = kAudioSessionCategory_MediaPlayback;
CheckError(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
sizeof(category),
&category),
"Couldn't set category on audio session");
con il seguente codice. Perché AudioSessionSetProperty è deprecato quindi ho sostituito il seguente codice.
AVAudioSession *session = [AVAudioSession sharedInstance];
NSError *setCategoryError = nil;
if (![session setCategory:AVAudioSessionCategoryPlayback
withOptions:AVAudioSessionCategoryOptionMixWithOthers
error:&setCategoryError]) {
// handle error
}
Ci deve essere qualche aggiornamento per iOS 8. che può essere nel codice sopra o in qualche altro.
Ho seguito l'approccio di Abdusha ma il mio file di output non aveva audio e la dimensione era molto piccola rispetto all'input. Dopo averlo esaminato, una correzione che ho apportato era nella funzione "pullGenericOutput". Dopo chiamata AudioUnitRender:
AudioUnitRender(genericOutputUnit,
&flags,
&inTimeStamp,
busNumber,
numberFrames,
bufferList);
inTimeStamp.mSampleTime++; //Updated
incremento il timestamp di 1. Dopo questa operazione, il file di output era perfetta, con effetti di lavoro. Grazie. La tua risposta ha aiutato molto.
Grazie! Il tuo codice è molto utile !! Ho passato un paio di giorni a provare a fare qualcosa del genere, e di quello che ho trovato - molto bello! – Seify