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 init
private:
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;
}