我正在pcm尝试mp3使用Lame. 虽然 pcm 数据确实被转换为 mp3 文件，但输出非常刺耳。在我发布问题之前，以下是我的代码：
/*
Sample program to generate a single sinusoid and encode it in mp3.
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <include/lame/lame.h>
#include <assert.h>
#include <string.h>
int main(int argc, char *argv[]) {
unsigned int sampleRate = 16000; /*assumed.*/
unsigned int nSecondsAudio = 4;
float *arr;
lame_global_flags *gfp;
unsigned char mp3buffer[2304]; /*some odd buffer sizes hard-coded.*/
int pcm_samples_1d[2*1152];
int pcm_samples_2d[2][1152];
int read = 0, write = 0;
int return_code = 1;
int mp3buf_size;
FILE *mp3;
FILE *pcm;
int framesize = 0;
int i = 0, j = 0, num_samples_encoded = 0;
/*Step 1. Generate sinusoid.*/
/*arr = (float *) malloc(sizeof(float) * nSecondsAudio * sampleRate);
arr = generateSinusoid(sampleRate, nSecondsAudio);*/
/*Step 2. See if encoder exists.*/
char *s = (char *) malloc(sizeof(char)*200);
s = get_lame_version();
printf("Lame version = %s\n", s);
/* Init lame flags.*/
gfp = lame_init();
if(!gfp) {
printf("Unable to initialize gfp object.");
} else {
printf("Able to initialize gfp object.\n");
}
/* set other parameters.*/
lame_set_num_channels(gfp, 1);
/*lame_set_num_samples(gfp, (nSecondsAudio * sampleRate));*/
lame_set_in_samplerate(gfp, sampleRate);
lame_set_quality(gfp, 5); /* set for high speed and good quality. */
lame_set_mode(gfp, 3); /* the input audio is mono */
lame_set_out_samplerate(gfp, sampleRate);
printf("Able to set a number of parameters too.");
framesize = lame_get_framesize(gfp);
printf("Framesize = %d\n", framesize);
assert(framesize <= 1152);
/* set more internal variables. check for failure.*/
if(lame_init_params(gfp) == -1) {
printf("Something failed in setting internal parameters.");
}
/* encode the pcm array as mp3.*
* Read the file. Encode whatever is read.
* As soon as end of file is reached, flush the buffers.
* Write everything to a file.
* Write headers too.
*/
/* Open PCM file for reading from.*/
pcm = fopen("out.pcm", "rb"); /*hard-coded to the only available pcm file.*/
if(!pcm) {
printf("Cannot open pcm file for reading.");
return 1;
}
mp3 = fopen("out.mp3", "wb+");
if(!mp3) {
printf("Cannot open file for writing.");
return 1;
}
do {
read = fread(pcm_samples_1d, sizeof(short), 2304, pcm); /*reads framesize shorts from pcm file.*/
printf("Read %d shorts from file.\n", read);
/* check for number of samples read. if 0, start flushing, else encode.*/
if(read > 0) {
/* got data in 1D array. convert it to 2D */
/* snippet below taken from lame source code. needs better understanding. pcm_samples_2d[0] = contents of buffer. pcm_samples_2d[1] = 0 since number of channels is always one.*/
memset(pcm_samples_2d[1], 0, 1152 * sizeof(int)); /*set all other samples with 0.*/
memset(pcm_samples_2d[0], 0, 1152 * sizeof(int));
i = 0, j = 0;
for(i = 0; i < 1152; i++) {
pcm_samples_2d[0][i] = pcm_samples_1d[i];
}
/* encode samples. */
num_samples_encoded = lame_encode_buffer_int(gfp, pcm_samples_2d[0], pcm_samples_2d[1], read, mp3buffer, sizeof(mp3buffer));
printf("number of samples encoded = %d\n", num_samples_encoded);
/* check for value returned.*/
if(num_samples_encoded > 1) {
printf("It seems the conversion was successful.\n");
} else if(num_samples_encoded == -1) {
printf("mp3buf was too small");
return 1;
} else if(num_samples_encoded == -2) {
printf("There was a malloc problem.");
return 1;
} else if(num_samples_encoded == -3) {
printf("lame_init_params() not called.");
return 1;
} else if(num_samples_encoded == -4) {
printf("Psycho acoustic problems.");
return 1;
} else {
printf("The conversion was not successful.");
return 1;
}
printf("Contents of mp3buffer = \n");
for(i = 0; i < 2304; i++) {
printf("mp3buffer[%d] = %d\n", i, mp3buffer[i]);
}
write = (int) fwrite(mp3buffer, sizeof(char), num_samples_encoded, mp3);
if(write != num_samples_encoded) {
printf("There seems to have been an error writing to mp3 within the loop.\n");
return 1;
} else {
printf("Writing of %d samples a success.\n", write);
}
}
} while(read > 0);
/* in case where the number of samples read is 0, or negative, start flushing.*/
read = lame_encode_flush(gfp, mp3buffer, sizeof(mp3buffer)); /*this may yield one more mp3 buffer.*/
if(read < 0) {
if(read == -1) {
printf("mp3buffer is probably not big enough.\n");
} else {
printf("MP3 internal error.\n");
}
return 1;
} else {
printf("Flushing stage yielded %d frames.\n", read);
}
write = (int) fwrite(mp3buffer, 1, read, mp3);
if(write != read) {
printf("There seems to have been an error writing to mp3.\n");
return 1;
}
/*samples have been written. write ID3 tag.*/
read = lame_get_id3v1_tag(gfp, mp3buffer, sizeof(mp3buffer));
if(sizeof(read) > sizeof(mp3buffer)) {
printf("Buffer too small to write ID3v1 tag.\n");
} else {
if(read > 0) {
write = (int) fwrite(mp3buffer, 1, read, mp3);
if(read != write) {
printf("more errors in writing id tag to mp3 file.\n");
}
}
}
lame_close(gfp);
fclose(pcm);
fclose(mp3);
return 0;
}
我的问题：
1. 我的输入 pcm 数据以 16kHz 采样，单声道并以 16 位编码。鉴于只有一个通道，lame_encode_buffer_int 的输入中的左右通道是什么？
2. 我不确定我是否理解从一维数组到二维数组（代码中的 pcm_samples_1d 到 pcm_samples_2d）的“转换”过程，正如这个问题中给出的那样。
3. 为什么我的声音很刺耳？在代码中，使用的库是使用--enable-debug标志从源代码编译的。但是，我无法使用gdb. 我还应该做什么？
到目前为止我所尝试的：
1. 阅读 LAME 项目的文档（或网络上可用的内容）。
2. 通读 SO 和其他论坛上发布的问题。
3. 浏览源代码：lame.h、frontend/main.cfrontend/get_audio.c`等。
非常欢迎对此提供任何帮助。