iPhoneマイクからiLBCにオーディオを変換する

Question

iPhoneマイクからオーディオを録音し、iLBCに変換してからリモートサーバーにストリームします。しかし、私はいつもAudioConverterFillComplexBufferに1768846202を持っています。 私はこの意味はkAudioConverterErr_InvalidInputSizeですが、どの入力が間違っているのか分かりません。

Stream audio from iOS、Record audio on iPhone with smallest file size、AudioUnit PCM compression to iLBC and decompression to PCMなどの記事を検索しましたが、いずれも問題を解決していません。ここで

は私のCONVERT関数です：

-(AudioBuffer) doConvert: (AudioBuffer)pcmData 
{ 
char *outputBuffer = NULL; 
OSStatus status; 

UInt32 theOutputBufSize = pcmData.mDataByteSize;//32768; 
outputBuffer = (char*)malloc(sizeof(char) * theOutputBufSize); 

/* Create the output buffer list */ 
AudioBufferList outBufferList; 
outBufferList.mNumberBuffers = 1; 
outBufferList.mBuffers[0].mNumberChannels = 1; 
outBufferList.mBuffers[0].mDataByteSize = theOutputBufSize; 
outBufferList.mBuffers[0].mData   = outputBuffer; 

//Converting 
//UInt32 ioOutputDataPackets = numOutputPackets; 
UInt32 numOutputDataPackets = 1; 
AudioStreamPacketDescription outPacketDesc[1]; 
status = AudioConverterFillComplexBuffer(audioConverterDecode, 
             encodeProc, 
             &pcmData, 
             &numOutputDataPackets, 
             &outBufferList, 
             outPacketDesc); 
//outBufferList.mBuffers[0].mDataByteSize = theOutputBufSize; 
[self hasError:status:__FILE__:__LINE__]; 

/* Set the ouput data */ 
AudioBuffer outData; 
outData.mNumberChannels  = 1; 
outData.mData   = outBufferList.mBuffers[0].mData; 
outData.mDataByteSize = outBufferList.mBuffers[0].mDataByteSize; 

return outData; 
} 

そして、私の初期機能：

-(void)initDecoder 
{ 

NSLog(@"initDecoder"); 
AudioStreamBasicDescription srcFormat, dstFormat; 
//AudioConverterRef converter = NULL; 
char *outputBuffer = NULL; 
OSStatus status; 

//output format 
dstFormat.mSampleRate = 8000.0; 
dstFormat.mFormatID = kAudioFormatiLBC ; 
dstFormat.mChannelsPerFrame = 1; 
//dstFormat.mBitsPerChannel = 0; 
dstFormat.mBytesPerPacket = 38;//50; 
dstFormat.mFramesPerPacket = 160;//240; 
dstFormat.mBytesPerFrame = 0; 
dstFormat.mBitsPerChannel = 0; 
dstFormat.mFormatFlags = 0;// little-endian 

//source format 
srcFormat.mSampleRate   = SAMPLE_RATE; //This is 48000 
srcFormat.mFormatID   = kAudioFormatLinearPCM; 
srcFormat.mFormatFlags  = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; 
srcFormat.mFramesPerPacket = 1; 
srcFormat.mChannelsPerFrame = 1; 
srcFormat.mBitsPerChannel  = 16; 
srcFormat.mBytesPerPacket  = 2; 
srcFormat.mBytesPerFrame  = 2; 
srcFormat.mReserved   = 0; 



status = AudioConverterNew(&srcFormat, &dstFormat, &audioConverterDecode); 
[self hasError:status:__FILE__:__LINE__]; 
} 

Answer 1

私はthisにいくつかのコードを変更しました、と私は私の解決策を見つけました。

createAudioConvert：

AudioStreamBasicDescription inputFormat = *(CMAudioFormatDescriptionGetStreamBasicDescription(CMSampleBufferGetFormatDescription(sampleBuffer))); // 输入音频格式 
AudioStreamBasicDescription outputFormat; 

memset(&outputFormat, 0, sizeof(outputFormat)); 
outputFormat.mSampleRate  = 8000; 
outputFormat.mFormatID   = kAudioFormatiLBC; 
outputFormat.mChannelsPerFrame = 1; 

// use AudioFormat API to fill out the rest of the description 
UInt32 size = sizeof(outputFormat); 
AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &outputFormat); 


outputFormat.mBytesPerPacket = 50; 
outputFormat.mFramesPerPacket = 240; 


AudioClassDescription *desc = [self getAudioClassDescriptionWithType:kAudioFormatiLBC, fromManufacturer:kAppleSoftwareAudioCodecManufacturer]; 
if (AudioConverterNewSpecific(&inputFormat, &outputFormat, 1, desc, &m_converter) != noErr) 
{ 
    printf("AudioConverterNewSpecific failed\n"); 
    return NO; 
} 

return YES; 

encoderAAC

if ([self createAudioConvert:sampleBuffer] != YES) 
{ 
    return NO; 
} 

CMBlockBufferRef blockBuffer = nil; 
AudioBufferList inBufferList; 
if (CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, NULL, &inBufferList, sizeof(inBufferList), NULL, NULL, 0, &blockBuffer) != noErr) 
{ 
    printf("CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer failed"); 
    return NO; 
} 

AudioBufferList outBufferList; 
outBufferList.mNumberBuffers    = 1; 
outBufferList.mBuffers[0].mNumberChannels = 1;//2; 
outBufferList.mBuffers[0].mDataByteSize = *aacLen; 
outBufferList.mBuffers[0].mData   = aacData; 

UInt32 outputDataPacketSize    = 1; 



OSStatus err = AudioConverterFillComplexBuffer(m_converter, inputDataProc, &inBufferList, &outputDataPacketSize, &outBufferList, NULL); 
printf("AudioConverterFillComplexBuffer\n"); 


if (err != noErr) 
{ 
    printf("AudioConverterFillComplexBuffer failed\n"); 
    return NO; 
} 

*aacLen = outBufferList.mBuffers[0].mDataByteSize; 
CFRelease(blockBuffer); 
return YES; 

コールバック関数：

OSStatus inputDataProc(AudioConverterRef inConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData,AudioStreamPacketDescription **outDataPacketDescription, void *inUserData) { 

ioData->mNumberBuffers = 1; 


AudioBufferList bufferList = *(AudioBufferList*)inUserData; 
ioData->mBuffers[0].mNumberChannels = 1; 
ioData->mBuffers[0].mData   = bufferList.mBuffers[0].mData; 
ioData->mBuffers[0].mDataByteSize = bufferList.mBuffers[0].mDataByteSize; 

UInt32 maxPackets = bufferList.mBuffers[0].mDataByteSize/2; 
*ioNumberDataPackets = maxPackets; 



return noErr; 
}