音频重采样(8k到16k)

基本抄的这里

这段代码限制很大，只能采样率8k的16bit的音频变成采样率16k的16bit的声音

/**
 * 采样率改变.
 * （原理就是将值简单的复制一下）
 * \param pInAudioData 输入的音频数据（注意这里是short）
 * \param nInAudioLen 输入数据的长度
 * \param pOutAudioData 输出的音频数据
 * \param nOutAudioLen 输出的音频数据长度
 */
void Resample16K(short* pInAudioData, int nInAudioLen, short* pOutAudioData, int& nOutAudioLen)
{
	short* sSampleIn = pInAudioData;
	int nFrequency = 0;
	while (sSampleIn - pInAudioData < nInAudioLen)
	{
		memcpy((char*)(pOutAudioData + nFrequency), (char*)sSampleIn, 2);
		nFrequency++;
		memcpy((char*)(pOutAudioData + nFrequency), (char*)sSampleIn, 2);
		nFrequency++;
		sSampleIn++;
	}

	nOutAudioLen = nFrequency * 2;
}

int main()
{
	//pcm_rate_trans(16, "./robin_ori_8_1_16.pcm", 8000, "./robin_ori_16_1_16.pcm", 16000);


	char filename[] = "./robin_ori_8_1_16.pcm";
	FILE* fp = NULL;
	fopen_s(&fp,filename, "ab+");
	FILE* fp_down = NULL;
	fopen_s(&fp_down,"./robin_ori_16_1_16.pcm", "wb+");

	fseek(fp, 0, SEEK_END);
	int buff_len = ftell(fp);
	fseek(fp, 0, SEEK_SET);

	unsigned char* pData = (unsigned char*)malloc(buff_len);
	memset(pData, 0, buff_len);
	fread(pData, sizeof(unsigned char), buff_len, fp);

	char* pOutData = new char[buff_len * 2];
	memset(pOutData, 0, buff_len * 2);
	int nLOutLen = 0;
	Resample16K((short*)pData, buff_len / 2, (short*)pOutData, nLOutLen);


	fwrite(pOutData, 1, nLOutLen, fp_down);

	fclose(fp);
	fclose(fp_down);


    std::cout << "Hello World!\n";
}

类似ffmepg直接转的命令
pcm转pcm

./ffmpeg -ar 8000 -ac 1 -f s16le -i robin_app.pcm -ar 16000 -ac 1 -f s16le robin_16k_1_16.pcm

不过ffmpeg的转换命令功能要强大的多。

wav转pcm

 ./ffmpeg -ar 16000 -ac 1 -f s16le -i test.wav  -ar 16000 -ac 1 -f s16le  beijing_16_16.pcm