交叉编译faac共享库

作者：咕唧咕唧liukun321

来自：http://blog.csdn.net/liukun321

     Advanced Audio Coding。一种专为声音数据设计的文件压缩格式，与Mp3不同，它采用了全新的算法进行编码，更加高效，具有更高的“性价比”。利用AAC格式，可使人感觉声音质量没有明显降低的前提下，更加小巧。

     FAAC是在嵌入式系统中常用的AAC音频编码开源库，关于AAC音频格式可以看一下这篇博文作简单了解：AAC音频编码格式简析

FAAC开源工程源码下载链接：FAAC源码下载

得到FAAC工程源码后首先执行 configure获得Makefile，并指定目标平台和交叉工具链

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

编译：

make

安装：

make install

最终会在指定安装目录获得如下动态及静态库：

libfaac.a                         

libfaac.la                        

libfaac.so                       

libfaac.so.0                      

libfaac.so.0.0.0  

将获得的动态链接库放入开发板/usr/lib目录即可

 

下面顺带附上一个将PCM 16bit 原始音频数据编码成AAC格式音频数据的C++类，下面的代码是从一个项目中抽取的，没有单独测试，仅做参考：

class AudioProcess {public:AudioProcess (void){nSampleRate = RATE;  // 采样率nChannels = CHANNELS;         // 声道数nPCMBitSize = SIZE; nInputSamples = 0;nMaxOutputBytes = 0;AACDecoderInitFlag = 0;DecoderHandle = 0;ADTSFrameInBuf = NULL;PCMData = NULL;ppBuffer = NULL;}// var init~AudioProcess(void){}// var initprivate:ULONG nInputSamples ;ULONG nMaxOutputBytes ;faacEncHandle hEncoder;faacEncConfigurationPtr pConfiguration;BYTE* pbAACBuffer;int nRet;public:int OutAACLength;ULONG nSampleRate;  // 采样率UINT nChannels;         // 声道数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(){        // 输入样本数，用实际读入字节数计算，一般只有读到文件尾时才不是//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;}