交叉編譯faac共享庫

作者：咕唧咕唧liukun321

來自：http://blog.csdn.net/liukun321

     Advanced Audio Coding。一種專爲聲音數據設計的文件壓縮格式，與Mp3不同，它採用了全新的算法進行編碼，更加高效，具有更高的“性價比”。利用AAC格式，可使人感覺聲音質量沒有明顯降低的前提下，更加小巧。

     FAAC是在嵌入式系統中常用的AAC音頻編碼開源庫，關於AAC音頻格式可以看一下這篇博文作簡單瞭解：AAC音頻編碼格式簡析

FAAC開源工程源碼下載鏈接：FAAC源碼下載

得到FAAC工程源碼後首先執行 configure獲得Makefile，並指定目標平臺和交叉工具鏈

./configure--target=arm-linux--host=arm-none-linux-gnueabi

編譯：

make

安裝：

make install

最終會在指定安裝目錄獲得如下動態及靜態庫：

libfaac.a                         

libfaac.la                        

libfaac.so                       

libfaac.so.0                      

libfaac.so.0.0.0  

將獲得的動態鏈接庫放入開發板/usr/lib目錄即可

 

下面順帶附上一個將PCM 16bit 原始音頻數據編碼成AAC格式音頻數據的C++類，下面的代碼是從一個項目中抽取的，沒有單獨測試，僅做參考：

class AudioProcess {
public:
	
	AudioProcess (void)
		{
			
			nSampleRate = RATE;  // 採樣率
			nChannels = CHANNELS;         // 聲道數
			nPCMBitSize = SIZE; 
			nInputSamples = 0;
			nMaxOutputBytes = 0;
			AACDecoderInitFlag = 0;
			DecoderHandle = 0;
			ADTSFrameInBuf = NULL;
			PCMData = NULL;
			ppBuffer = NULL;
			
		}// var init

		~AudioProcess(void)
		{
		
		}// var init
	
private:
	ULONG nInputSamples ;
	ULONG nMaxOutputBytes ;
	faacEncHandle hEncoder;
	faacEncConfigurationPtr pConfiguration;
	BYTE* pbAACBuffer;
	int nRet;
public:
	int OutAACLength;
	
	ULONG nSampleRate;  // 採樣率
	UINT nChannels;         // 聲道數
	UINT nPCMBitSize; 
	unsigned char* ppBuffer;
	unsigned long pSizeOfDecoderSpecificInfo;
	int nBytesRead;
	int nPCMBufferSize;
	int nAACBufferSize;
	BYTE* pbPCMBuffer;
	
	BYTE* OutAACBuffer;
public:	
	int AACEncoderInit();
	int AACEncoding();
	int AACEncoderDestory();
};

int AudioProcess ::AACEncoderInit()
{
	   	hEncoder = faacEncOpen(nSampleRate, nChannels, &nInputSamples, &nMaxOutputBytes);
	    if(hEncoder == NULL)
	    {
	        printf("[ERROR] Failed to call faacEncOpen()\n");
	        return -1;
	    }
	    printf("nInputSamples = %d\n",nInputSamples);
	    nPCMBufferSize = nInputSamples * nPCMBitSize / 8;
	    pbPCMBuffer = new BYTE [nPCMBufferSize];
	    pbAACBuffer = new BYTE [nMaxOutputBytes];

	    //  Get current encoding configuration
	    pConfiguration = faacEncGetCurrentConfiguration(hEncoder);
	    pConfiguration->inputFormat = FAAC_INPUT_16BIT;//_16BIT;
		pConfiguration->mpegVersion = MPEG4;
		 pConfiguration->version = MPEG4;  // 1
		 pConfiguration->outputFormat =1;// ADTS_STREAM;
	
	 	 pConfiguration->aacObjectType = 2;//LOW;
	 	 pConfiguration->useTns = 0;//DEFAULT_TNS;
	  	 pConfiguration->shortctl =  0;//SHORTCTL_NORMAL;
	 	 pConfiguration->allowMidside = 1 ;
	   
	    //  Set encoding configuration
	    nRet = faacEncSetConfiguration(hEncoder, pConfiguration);
	    faacEncGetDecoderSpecificInfo(hEncoder,&(ppBuffer), &(pSizeOfDecoderSpecificInfo));
}

int AudioProcess ::AACEncoding()
{
	
	
        // 輸入樣本數，用實際讀入字節數計算，一般只有讀到文件尾時纔不是				//nPCMBufferSize/(nPCMBitSize/8);
		 nBytesRead = length;
		nInputSamples = nBytesRead / (nPCMBitSize / 8);
		printf("nInputSamples = %d\n",nInputSamples);
		
        //Encode
        nRet = faacEncEncode(hEncoder, (int*) pbPCMBuffer, nInputSamples, pbAACBuffer,nMaxOutputBytes);
	 	OutAACBuffer = pbAACBuffer;
	 	OutAACLength = nRet;
        
	return nRet;
}

void AudioProcess::AACEncoderDestroy()
{	
		nRet = faacEncClose(hEncoder);
		delete[] pbPCMBuffer;
		delete[] pbAACBuffer;
			
}