副标题#e#
libmad:是一个开源的高精度mpeg音频解码库,支持 MPEG-1(Layer I, Layer II 和 LayerIII(也就是 MP3)。LIBMAD 提供 24-bit 的 PCM 输出,完全是定点计较,很是适合没有浮点支持的平台上利用。利用 libmad 提供的一系列 API,就可以很是简朴地实现 MP3 数据解码事情。在 libmad 的源代码文件目次下的 mad.h 文件中,可以看到绝大部门该库的数据布局和 API 等。
PCM编码:即为脉冲代码调制编码。
PCM通过抽样,量化,编码三个步调将持续的模仿信号转换成数字编码。
libmad中的主要数据布局:
| 主要数据布局 | 浸染 |
| struct mad_stream | 存放解码前的Bitstream数据 |
| struct mad_synth | 存放解码合成滤波后的PCM数据 |
| struct mad_pcm | 界说了音频的采样率,声道个数和PCM采样数据,用来初始化音频 |
| struct mad_frame | 记录MPEG帧解码后PCM数据的数据布局,个中的mad_header用来记录MPEG帧的根基信息,好比MPEG层数、声道模式、流比特率、采样比特率。声道模式包罗单声道、双声道、连系立体混音道以及一般立体声。 |
MAD通过回调函数机制来实现解码,每个回调函数会返回一个列举范例mad_flow,通过mad_flow可以节制解码的进程。在未经处理惩罚的环境下,MAD一般输出32bit,以little endian名目存放在mad_fixed_t中的数据。可是大大都的声卡并能支持输出高达32bit精度的数据,因而还必需对mad_fixed_t举办量化,圆滑处理惩罚以及发抖,使到采样信号降到16bit精度。MAD认真的只是解码的进程,它事情进程是:从外部获取输入,逐帧解码,在解码的进程中返复书息,然后获得解码功效。开拓人员要手动配置输入输出。
在libmad中提供了一个解码源措施minimad.c,实现了将MP3文件解码成pcm数据,并将其数据显示在终端上。
此刻就以该源码措施为例,来写出我们本身的基于libmad的MP3播放器。
在我们打开我们的音频措施之时同时也打开我们的音频设备"/dev/dsp"。
static int sfd;
if((sfd = open("/dev/dsp", O_WRONLY)) < 0)
{
printf("can not open device!!!/n");
return 1;
}
#p#副标题#e#
一般来说,我们的MP3文件都是立体音,有2个声道,由于要把pcm采样后并处理惩罚的数据放入一个char型的数组,而并行的阁下声道的每个采样要在字符数组中处理惩罚成2个,所以字符数组中的数据的个数应该是pcm音频采样数的4倍。又因为把阁下声道的数据合在一个字符数组里串行处理惩罚,所以播放的速度应该是pcm音频采样率的两倍。
static
enum mad_flow output(void *data,
struct mad_header const *header, struct mad_pcm *pcm)
{
unsigned int nchannels, nsamples, n;
mad_fixed_t const *left_ch, *right_ch;
unsigned char Output[6912], *OutputPtr;
int fmt, wrote, speed;
nchannels = pcm->channels;
n = nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
fmt = AFMT_S16_LE;
speed = pcm->samplerate * 2; /*播放速度是采样率的两倍 */
ioctl(sfd, SNDCTL_DSP_SPEED, &(speed));
ioctl(sfd, SNDCTL_DSP_SETFMT, &fmt);
ioctl(sfd, SNDCTL_DSP_CHANNELS, &(pcm->channels));
OutputPtr = Output;
while (nsamples--) {
signed int sample;
sample = scale(*left_ch++);
*(OutputPtr++) = sample >> 0;
*(OutputPtr++) = sample >> 8;
if (nchannels == 2) {
sample = scale(*right_ch++);
*(OutputPtr++) = sample >> 0;
*(OutputPtr++) = sample >> 8;
}
}
n *= 4; /*数据长度为pcm音频采样的4倍 */
OutputPtr = Output;
while (n) {
wrote = write(sfd, OutputPtr, n);
OutputPtr += wrote;
n -= wrote;
}
OutputPtr = Output;
return MAD_FLOW_CONTINUE;
}
这样就可以实现我们的播放器了…..
下面就以一个简朴的实例来说明问题:
# include <stdio.h>
# include <stdlib.h>
# include <unistd.h>
# include <sys/stat.h>
# include <sys/mman.h>
# include <sys/soundcard.h>
# include <sys/ioctl.h>
# include <sys/fcntl.h>
# include <sys/types.h>
# include <mad.h>
struct buffer {
unsigned char const *start;
unsigned long length;
};
static int sfd; /*声音设备的描写符 */
static int decode(unsigned char const *, unsigned long);
int main(int argc, char *argv[])
{
struct stat stat;
void *fdm;
char const *file;
int fd;
file = argv[1];
fd = open(file, O_RDONLY);
if ((sfd = open("/dev/dsp", O_WRONLY)) < 0) {
printf("can not open device!!!/n");
return 5;
}
ioctl(sfd, SNDCTL_DSP_SYNC, 0); /*此句可以不要 */
if (fstat(fd, &stat) == -1 || stat.st_size == 0)
return 2;
fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);
if (fdm == MAP_FAILED)
return 3;
decode(fdm, stat.st_size);
if (munmap(fdm, stat.st_size) == -1)
return 4;
ioctl(sfd, SNDCTL_DSP_RESET, 0);
close(sfd);
return 0;
}
static
enum mad_flow input(void *data, struct mad_stream *stream)
{
struct buffer *buffer = data;
if (!buffer->length)
return MAD_FLOW_STOP;
mad_stream_buffer(stream, buffer->start, buffer->length);
buffer->length = 0;
return MAD_FLOW_CONTINUE;
}
/*这一段是处理惩罚采样后的pcm音频 */
static inline signed int scale(mad_fixed_t sample)
{
sample += (1L << (MAD_F_FRACBITS - 16));
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
return sample >> (MAD_F_FRACBITS + 1 - 16);
}
static
enum mad_flow output(void *data,
struct mad_header const *header, struct mad_pcm *pcm)
{
unsigned int nchannels, nsamples, n;
mad_fixed_t const *left_ch, *right_ch;
unsigned char Output[6912], *OutputPtr;
int fmt, wrote, speed;
nchannels = pcm->channels;
n = nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
fmt = AFMT_S16_LE;
speed = pcm->samplerate * 2; /*播放速度是采样率的两倍 */
ioctl(sfd, SNDCTL_DSP_SPEED, &(speed));
ioctl(sfd, SNDCTL_DSP_SETFMT, &fmt);
ioctl(sfd, SNDCTL_DSP_CHANNELS, &(pcm->channels));
OutputPtr = Output;
while (nsamples--) {
signed int sample;
sample = scale(*left_ch++);
*(OutputPtr++) = sample >> 0;
*(OutputPtr++) = sample >> 8;
if (nchannels == 2) {
sample = scale(*right_ch++);
*(OutputPtr++) = sample >> 0;
*(OutputPtr++) = sample >> 8;
}
}
n *= 4; /*数据长度为pcm音频采样的4倍 */
OutputPtr = Output;
while (n) {
wrote = write(sfd, OutputPtr, n);
OutputPtr += wrote;
n -= wrote;
}
OutputPtr = Output;
return MAD_FLOW_CONTINUE;
}
static
enum mad_flow error(void *data,
struct mad_stream *stream, struct mad_frame *frame)
{
return MAD_FLOW_CONTINUE;
}
static
int decode(unsigned char const *start, unsigned long length)
{
struct buffer buffer;
struct mad_decoder decoder;
int result;
buffer.start = start;
buffer.length = length;
mad_decoder_init(&decoder, &buffer, input, 0, 0, output, error, 0);
mad_decoder_options(&decoder, 0);
result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
mad_decoder_finish(&decoder);
return result;
}
本文出自 “驿落薄暮” 博客,请务必保存此出处http://yiluohuanghun.blog.51cto.com/3407300/867922
