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
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 %s\n", infile);return 1;}if (!wav_get_header(wav, &format, &channels, &sampleRate, &bitsPerSample, NULL)) {fprintf(stderr, "Bad wav file %s\n", infile);return 1;}if (format != 1) {fprintf(stderr, "Unsupported WAV format %d\n", format);return 1;}if (bitsPerSample != 16) {fprintf(stderr, "Unsupported WAV sample depth %d\n", bitsPerSample);return 1;}if (channels != 1)fprintf(stderr, "Warning, only compressing one audio channel\n");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("#!AMR\n", 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;}代码是opencore的测试用例上的。可以直接拿来使用。
接下来说说调整音量的相关处理。测试发现什么设置主音量，wave，，麦克风音量，合成器音量等等统统都不管用，起码我测试着效果不明显。既然录制时无法处理，只能在录制结束后对音频文件进行处理。这一篇博客有详细的说明，测试发现能正常运行，注意这里处理的音频，不包含音频头部：点击打开链接
测试发现音频如果将音量设置为大小一致是发不出声音的，所以具体调整需要另行设计算法，但是原理基本就是连接所说的。
成功不是将来才有的，而是从决定去做的那一刻起，持续累积而成。
