核心音频离线渲染GenericOutput [英] core audio offline rendering GenericOutput
本文介绍了核心音频离线渲染GenericOutput的处理方法，对大家解决问题具有一定的参考价值，需要的朋友们下面随着小编来一起学习吧！
问题描述
有人使用core-Audio成功完成离线渲染。？
Anybody successfully done offline rendering using core-Audio.?
我必须混合两个音频文件并应用混响（使用2个AudioFilePlayer，MultiChannelMixer，Reverb2和RemoteIO） 。
搞定了。我可以在预览时保存它（在RemoteIO的renderCallBack上）。
I had to mix two audio files and apply reverb(used 2 AudioFilePlayer,MultiChannelMixer,Reverb2 and RemoteIO). Got it working. and i could save it while its previewing(on renderCallBack of RemoteIO).
我需要保存它而不播放它（离线）。
提前致谢。
I need to save it without playing it (offline). Thanks in advance.
推荐答案
离线渲染使用GenericOutput AudioUnit为我工作。
我在这里分享工作代码。
核心音频框架看起来有点像。但像ASBD，参数......等小小的东西正在制造这些问题。努力工作吧。不要放弃:-)。核心音频在处理低级音频时非常强大和有用。这就是我从过去几周学到的东西。享受:-D ....
Offline rendering Worked for me using GenericOutput AudioUnit. I am sharing the working code here. core-audio framework seems a little though. But small-small things in it like ASBD, parameters ...etc are making these issues. try hard it will work. Don't give-up :-). core-audio is very powerful and useful while dealing with low-level audio. Thats what I learned from these last weeks. Enjoy :-D ....
在.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 .m class
//in .m class
- (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设置
- (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;
}
// AUGraph setup
//AUGraph setup
- (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];
}
//音频文件播放设置在这里设置语音文件
//Audio file playback setup here i am setting the voice file
-(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;
}
//音频文件播放设置在这里设置BGMusic文件
//Audio file playback setup here i am setting the BGMusic file
-(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;
}
//开始保存文件
- (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];
}
//手动从GenericOutput节点提供和获取数据/缓冲区。
// Manual Feeding and getting data/Buffer from the GenericOutput Node.
-(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
//FilesSavingCompleted
-(void)FilesSavingCompleted{
OSStatus status = ExtAudioFileDispose(extAudioFile);
printf("OSStatus(ExtAudioFileDispose): %ld\n", status);
}
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