wav转amr以及简单调整音量

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我是靠谱客的博主幸福芒果，这篇文章主要介绍wav转amr以及简单调整音量，现在分享给大家，希望可以做个参考。

说一下wav转amr的方式。wav是PC上录制音频最容易生成的方式，但是缺点是生成的音频体积比较大。amr是手机上音频播放比较主流的格式，优点是音频体积小，易于传输。

转换的方式很简单，amr分两种，这里以nb为例。首先需要下载opencore-amr，将静态库和文件导入工程里。然后输入以下代码

复制代码

int wav2Amr( const char *infile, const char *outfile) {
enum Mode mode = MR122;
int ch, dtx = 0;
FILE *out;
void *wav, *amr;
int format, sampleRate, channels, bitsPerSample;
int inputSize;
uint8_t* inputBuf;
int modeRate = 4750;
mode = findMode(modeRate);
wav = wav_read_open(infile);
if (!wav) {
fprintf(stderr, "Unable to open wav file %sn", infile);
return 1;
}
if (!wav_get_header(wav, &format, &channels, &sampleRate, &bitsPerSample, NULL)) {
fprintf(stderr, "Bad wav file %sn", infile);
return 1;
}
if (format != 1) {
fprintf(stderr, "Unsupported WAV format %dn", format);
return 1;
}
if (bitsPerSample != 16) {
fprintf(stderr, "Unsupported WAV sample depth %dn", bitsPerSample);
return 1;
}
if (channels != 1)
fprintf(stderr, "Warning, only compressing one audio channeln");
if (sampleRate != 8000)
fprintf(stderr, "Warning, AMR-NB uses 8000 Hz sample rate (WAV file has %d Hz)n", sampleRate);
inputSize = channels*2*160;
inputBuf = (uint8_t*) malloc(inputSize);
amr = Encoder_Interface_init(dtx);
out = fopen(outfile, "wb");
if (!out) {
perror(outfile);
return 1;
}
fwrite("#!AMRn", 1, 6, out);
while (1) {
short buf[160];
uint8_t outbuf[500];
int read, i, n;
read = wav_read_data(wav, inputBuf, inputSize);
read /= channels;
read /= 2;
if (read < 160)
break;
for (i = 0; i < 160; i++) {
const uint8_t* in = &inputBuf[2*channels*i];
buf[i] = in[0] | (in[1] << 8);
}
n = Encoder_Interface_Encode(amr, mode, buf, outbuf, 0);
fwrite(outbuf, 1, n, out);
}
free(inputBuf);
fclose(out);
Encoder_Interface_exit(amr);
wav_read_close(wav);
return 0;
}

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int wav2Amr( const char *infile, const char *outfile) {
enum Mode mode = MR122;
int ch, dtx = 0;
FILE *out;
void *wav, *amr;
int format, sampleRate, channels, bitsPerSample;
int inputSize;
uint8_t* inputBuf;
int modeRate = 4750;
mode = findMode(modeRate);
wav = wav_read_open(infile);
if (!wav) {
fprintf(stderr, "Unable to open wav file %sn", infile);
return 1;
}
if (!wav_get_header(wav, &format, &channels, &sampleRate, &bitsPerSample, NULL)) {
fprintf(stderr, "Bad wav file %sn", infile);
return 1;
}
if (format != 1) {
fprintf(stderr, "Unsupported WAV format %dn", format);
return 1;
}
if (bitsPerSample != 16) {
fprintf(stderr, "Unsupported WAV sample depth %dn", bitsPerSample);
return 1;
}
if (channels != 1)
fprintf(stderr, "Warning, only compressing one audio channeln");
if (sampleRate != 8000)
fprintf(stderr, "Warning, AMR-NB uses 8000 Hz sample rate (WAV file has %d Hz)n", sampleRate);
inputSize = channels*2*160;
inputBuf = (uint8_t*) malloc(inputSize);
amr = Encoder_Interface_init(dtx);
out = fopen(outfile, "wb");
if (!out) {
perror(outfile);
return 1;
}
fwrite("#!AMRn", 1, 6, out);
while (1) {
short buf[160];
uint8_t outbuf[500];
int read, i, n;
read = wav_read_data(wav, inputBuf, inputSize);
read /= channels;
read /= 2;
if (read < 160)
break;
for (i = 0; i < 160; i++) {
const uint8_t* in = &inputBuf[2*channels*i];
buf[i] = in[0] | (in[1] << 8);
}
n = Encoder_Interface_Encode(amr, mode, buf, outbuf, 0);
fwrite(outbuf, 1, n, out);
}
free(inputBuf);
fclose(out);
Encoder_Interface_exit(amr);
wav_read_close(wav);
return 0;
}