核心audio离线渲染GenericOutput
任何人都成功地使用core-Audio进行离线渲染。
我必须混合两个audio文件并应用混响（使用2个AudioFilePlayer，MultiChannelMixer，Reverb2和RemoteIO）。 得到它的工作。 我可以在预览时（在RemoteIO的renderCallBack上）保存它。
我需要保存它而不玩（离线）。 提前致谢。
离线渲染使用GenericOutput AudioUnit为我工作。 我在这里分享工作代码。 核心audio框架似乎有点虽然。 但是像ASBD，参数等等的小东西就是在做这些事情。 努力工作会努力。 不要放弃:-)。 核心audio处理低级audio时非常强大有用。 那就是我从上个星期以来学到的东西。 享受:-D ….
在.h中声明这些
//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; //在.m类中
- (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设置
- (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]; }
/ /audio文件播放设置在这里我设置语音文件
-(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; }
//audio文件播放设置在这里我设置BGMusic文件
-(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节点中获取数据/缓冲区。
-(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); }
进行离线渲染的一种方法是移除RemoteIO单元，并显式调用graphics中最右侧单元（取决于拓扑的混音器或混响单元）上的AudioUnitRender。 通过循环操作，直到从两个源文件中抽取样本，然后使用扩展audio文件服务写入生成的样本缓冲区，可以创build一个压缩的压缩audio文件。 你会想在后台线程上做到这一点，以保持用户界面的响应，但我已经使用这种技术，取得了一些成功。
上面的代码是在iOS7设备上工作，但不能在iOS8设备和所有模拟器上工作。 我已经replace了以下代码段
UInt32 category = kAudioSessionCategory_MediaPlayback; CheckError(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category), "Couldn't set category on audio session");
用下面的代码。 因为AudioSessionSetProperty已被弃用，所以我已经取代了下面的代码。
AVAudioSession *session = [AVAudioSession sharedInstance]; NSError *setCategoryError = nil; if (![session setCategory:AVAudioSessionCategoryPlayback withOptions:AVAudioSessionCategoryOptionMixWithOthers error:&setCategoryError]) { // handle error }
iOS8必须有一些更新，可以在上面的代码或在其他地方。
我遵循Abdusha的方法，但是我的输出文件没有audio，加上与input相比尺寸非常小。 查看后，我做了一个修复是在“pullGenericOutput”function。 在AudioUnitRender调用之后：
AudioUnitRender(genericOutputUnit, &flags, &inTimeStamp, busNumber, numberFrames, bufferList); inTimeStamp.mSampleTime++; //Updated
增加timeStamp 1.这样做后，输出文件是完美的影响工作。 谢谢。 你的答案帮了很大忙。