據說這個更新到2004年2月的libmad是一種高品質的MPEG音頻解碼器,支持24-bit輸出,優點多多。
對其的詳細介紹請參考主頁:http://www.underbit.com/products/mad/
準備工作
x86_64平臺的編譯可直接運行configure,arm下
libmad: ./configure --host=arm-xxx(arm-xxx爲交叉編譯工具的前綴)
解碼流程
libmad將mp3文件解碼後生成pcm數據。libmad庫自帶一個非常簡潔的實例:minimad.c,分析其中的decode函數可看到其解碼流程非常簡單:
1、配置輸入回調函數、輸出回調函數、用戶數據、篩選回調函數、錯誤回調函數、消息回調函數,參考初始化函數源代碼。
回調函數中,輸入(讀取原始音頻數據)、輸出(對解碼後的音頻數據進行處理)是必需的。
void mad_decoder_init(struct mad_decoder *decoder, void *data,
enum mad_flow(*input_func)(void *, struct mad_stream *),
enum mad_flow(*header_func)(void *, struct mad_header const *),
enum mad_flow(*filter_func)(void *, struct mad_stream const *,
struct mad_frame *),
enum mad_flow(*output_func)(void *,
struct mad_header const *,
struct mad_pcm *),
enum mad_flow(*error_func)(void *,
struct mad_stream *,
struct mad_frame *),
enum mad_flow(*message_func)(void *,
void *, unsigned int *))
{
decoder->mode = -1;
decoder->options = 0;
decoder->async.pid = 0;
decoder->async.in = -1;
decoder->async.out = -1;
decoder->sync = 0;
decoder->cb_data = data;
decoder->input_func = input_func;
decoder->header_func = header_func;
decoder->filter_func = filter_func;
decoder->output_func = output_func;
decoder->error_func = error_func;
decoder->message_func = message_func;
}
其中的data參數是用戶需要傳給回調函數的自定義數據結構。比如,解碼一個文件,並且採用系統函數open函數打開,那麼可以定義一個對此描述的數據結構:
typedef struct _mp3_file
{
int *fd;//open("xx.mp3",O_RDONLY)
uint32_t flen;//mp3文件的長度
uint32_t fpos;//當前文件指針位置
uint8_t buf[BUFSIZE];
uint32_t buf_size;
} mp3_file;
數據成員buf、buf_size傳遞給mad_stream_buffer,用來設置流緩衝區指針。
錯誤處理在minimad中被忽略了,可參考madplay中處理方法(player.c:1776),此處貼出來大概的流程
/*
* NAME: decode->error()
* DESCRIPTION: handle a decoding error
*/
static
enum mad_flow decode_error(void *data, struct mad_stream *stream,
struct mad_frame *frame)
{
struct player *player = data;
signed long tagsize;
switch (stream->error)
{
case MAD_ERROR_BADDATAPTR:
/*do something*/
return MAD_FLOW_CONTINUE;
case MAD_ERROR_LOSTSYNC:
/* todo*/
default:
/*todo*/
}
if (stream->error == MAD_ERROR_BADCRC)
{
return MAD_FLOW_IGNORE;
}
return MAD_FLOW_CONTINUE;
}
錯誤處理的返回值定義如下:
enum mad_flow {
MAD_FLOW_CONTINUE = 0x0000, /* continue normally */
MAD_FLOW_STOP = 0x0010, /* stop decoding normally */
MAD_FLOW_BREAK = 0x0011, /* stop decoding and signal an error */
MAD_FLOW_IGNORE = 0x0020 /* ignore the current frame */
};
可視情況選擇繼續或者終止解碼。
2、調用mad_decoder_run開始解碼,支持兩種同步、異步兩種運行方式
enum mad_decoder_mode {
MAD_DECODER_MODE_SYNC = 0,
MAD_DECODER_MODE_ASYNC
};
3、解碼結束後調用mad_decoder_finish釋放解碼器
播放
在輸出回調函數中,將解碼數據直接寫入“/dev/dsp”即可實現mp3文件的播放,也可以生成pcm文件,供支持pcm解碼的音頻芯片使用。在稍微不算太古董的Linux系統中,都可以使用alsa來播放解碼後的數據。關於alsa請參考:
https://wiki.archlinux.org/index.php/Advanced_Linux_Sound_Architecture
http://www.equalarea.com/paul/alsa-audio.html (這是一篇alsa導學)
https://www.linuxjournal.com/article/6735 (這是另一篇導學)
以下代碼示範使用libmad解碼MP3,並使用alsa接口來播放:
#include <stdio.h>
#include <stdlib.h>
#include <errno.h>
#include <alsa/asoundlib.h>
#include <stdint.h>
#include <mad.h>
#include <id3tag.h>
int alsa_device_open(snd_pcm_t** ppcm, const char* dev)
{
int err;
if ((err = snd_pcm_open(ppcm, dev == 0?"default":dev, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
fprintf(stderr, "cannot open audio device %s (%s)\n",
dev,
snd_strerror(err));
return -1;
}
snd_pcm_t *pcm = *ppcm;
snd_pcm_hw_params_t *hw_params = 0;
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0)
{
fprintf(stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror(err));
return -1;
}
if ((err = snd_pcm_hw_params_any(pcm, hw_params)) < 0)
{
fprintf(stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror(err));
return -1;
}
if ((err = snd_pcm_hw_params_set_access(pcm, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
fprintf(stderr, "cannot set access type (%s)\n",
snd_strerror(err));
return -1;
}
if ((err = snd_pcm_hw_params_set_format(pcm, hw_params,
SND_PCM_FORMAT_S16/*SND_PCM_FORMAT_S16_LE*/)) < 0)
{
fprintf(stderr, "cannot set sample format (%s)\n",
snd_strerror(err));
return -1;
}
int rate = 44100;
if ((err = snd_pcm_hw_params_set_rate_near(pcm, hw_params, &rate, 0)) < 0)
{
fprintf(stderr, "cannot set sample rate (%s)\n",
snd_strerror(err));
return -1;
}
if ((err = snd_pcm_hw_params_set_channels(pcm, hw_params, 2)) < 0)
{
fprintf(stderr, "cannot set channel count (%s)\n",
snd_strerror(err));
return -1;
}
snd_pcm_uframes_t periodsize = 1024;//frag_size / 4;
err = snd_pcm_hw_params_set_period_size_near(pcm, hw_params,
&periodsize, 0);
if (err < 0)
{
printf("error on set_period_size (%d)\n", (int)periodsize);
return -1;
}
uint frag_count = 8;
err = snd_pcm_hw_params_set_periods_near(pcm, hw_params,
&frag_count, 0);
if (err < 0)
{
printf("error on set_periods (%d)\n", frag_count);
return -1;
}
if ((err = snd_pcm_hw_params(pcm, hw_params)) < 0)
{
fprintf(stderr, "cannot set parameters (%s)\n",
snd_strerror(err));
return -1;
}
snd_pcm_hw_params_free(hw_params);
if ((err = snd_pcm_prepare(pcm)) < 0)
{
fprintf(stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror(err));
return -1;
}
return 0;
}
int alsa_device_write(snd_pcm_t* pcm, uint8_t* data, int data_len)
{
int remain = data_len;
while (remain > 0)
{
//之所以用/4,是因爲雙聲道,16位,所以每個frame爲4byte
int ret = snd_pcm_writei(pcm, data, remain / 4);
if (ret == -EAGAIN)
{
snd_pcm_prepare(pcm);
continue;
}
if (ret != remain / 4)
{
printf("pcm write %u frame ret %d frames\n", remain / 4, ret);
}
if (ret > 0)
{
remain -= (ret * 4);
}
else
{
fprintf(stderr,
"error from writei: %s\n",
snd_strerror(ret));
return MAD_FLOW_STOP;
}
}
return data_len - remain;
}
int alsa_device_close(snd_pcm_t* pcm)
{
snd_pcm_drain(pcm);
return snd_pcm_close(pcm);
}
static snd_pcm_t* pcm =0;
int output_decode_data(uint8_t* data, int data_len)
{
if (pcm == 0)
{
return -1;
}
int write_count = alsa_device_write(pcm,data,data_len);
#ifdef _OUT_TEST
//如果沒有聲音,可以打開該開關,用sox工具play播放該數據文件,驗證解碼的數據是否正確。
static int fd = -1;
if (fd == -1)
{
fd = open("/tmp/t.pcm", O_WRONLY | O_CREAT);
}
if (fd != -1)
{
int ret = write(fd, data, write_count);
if (ret <= 0)
{
return MAD_FLOW_STOP;
}
//fsync(fd);
}
#endif
return write_count;
}
#define MAX_BUFF_SIZE 4096*10
typedef struct _DecodeData
{
int fd; //open("xx.mp3",O_RDONLY)
uint8_t *buf;
uint32_t buf_size;
uint8_t *decode_buf;
uint decode_buf_size;
} DecodeData;
static enum mad_flow decode_input(void *data, struct mad_stream *stream)
{
DecodeData *mp3 = (DecodeData *)data;
size_t remaining = 0;
if (stream->next_frame != 0)
{
remaining = stream->bufend - stream->next_frame;
if (remaining != 0)
{
memmove(mp3->buf, stream->next_frame, remaining);
printf("remaining %lu\n", remaining);
}
}
ssize_t ret = read(mp3->fd, mp3->buf + remaining, MAX_BUFF_SIZE - remaining);
if (ret <= 0)
{
return MAD_FLOW_STOP;
}
mp3->buf_size = ret + remaining;
mad_stream_buffer(stream, mp3->buf, mp3->buf_size);
return MAD_FLOW_CONTINUE;
}
static ssize_t scale(mad_fixed_t sample)
{
/* round */
sample += (1L << (MAD_F_FRACBITS - 16));
/* clip */
if (sample >= MAD_F_ONE) sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE) sample = -MAD_F_ONE;
/* quantize */
return sample >> (MAD_F_FRACBITS + 1 - 16);
}
static enum mad_flow decode_output(void *data, struct mad_header const *header, struct mad_pcm *pcm)
{
/* pcm->samplerate contains the sampling frequency */
uint nchannels = pcm->channels;
uint nsamples = pcm->length;
mad_fixed_t const *left_ch = pcm->samples[0];
mad_fixed_t const *right_ch = pcm->samples[1];
DecodeData * dd = (DecodeData*)data;
dd->decode_buf = realloc(dd->decode_buf, dd->decode_buf_size + nsamples * nchannels * 2);
dd->decode_buf_size += nsamples * nchannels * 2;
int16_t *output = (int16_t *)dd->decode_buf;
while (nsamples--)
{
*output++ = scale(*(left_ch++));
*output++ = scale(*(right_ch++));
}
//將解碼後的數據通過alsa寫入聲音設備
int out_count = output_decode_data(dd->decode_buf, dd->decode_buf_size);
if (out_count > 0)
{
dd->decode_buf_size -= out_count;
}
return MAD_FLOW_CONTINUE;
}
static enum mad_flow decode_error(void *data, struct mad_stream *ms, struct mad_frame *frame)
{
if (ms->error == MAD_ERROR_LOSTSYNC)
{
signed long tagsize;
tagsize = id3_tag_query(ms->this_frame,
ms->bufend - ms->this_frame);
if (tagsize > 0)
{
mad_stream_skip(ms, tagsize);
}
}
return MAD_FLOW_CONTINUE;
}
DecodeData* new_decode_data(const char* file)
{
DecodeData *decode_data = (DecodeData*)malloc(sizeof(DecodeData));
decode_data->fd = open(file, O_RDONLY);
if (decode_data->fd == -1)
{
free(decode_data);
perror("open:");
return 0;
}
decode_data->buf = (uint8_t *)calloc(MAX_BUFF_SIZE, 1);
decode_data->buf_size = MAX_BUFF_SIZE;
decode_data->decode_buf = 0;
decode_data->decode_buf_size = 0;
return decode_data;
}
void del_decode_data(DecodeData** pdd)
{
DecodeData* dd = *pdd;
free(dd->buf);
free(dd->decode_buf);
close(dd->fd);
free(dd);
*pdd = 0;
}
int main(int argc, char *argv[])
{
if (argc != 3 && argc != 2)
{
printf("usage: app [file] [device](etc->pluginhw:1,0)\n");
return -1;
}
const char *dev = argc == 3 ? argv[2] : "default"; //"pluginhw:1,0"
int ret = alsa_device_open(&pcm, dev);
if (ret == -1)
{
return -1;
}
DecodeData *decode_data = new_decode_data(argv[1]);
struct mad_decoder decoder;
mad_decoder_init(&decoder, decode_data,
decode_input, 0 /* header */, 0 /* filter */, decode_output,
decode_error, 0 /* message */);
mad_decoder_options(&decoder, MAD_OPTION_IGNORECRC);
/* 運行解碼器,直到返回 MAD_FLOW_STOP, MAD_FLOW_BREAK */
int result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
mad_decoder_finish(&decoder);
alsa_device_close(pcm);
del_decode_data(&decode_data);
return 0;
}
編譯命令:gcc -g [源文件名].c -lasound -lmad -lid3tag -o [程序名]
遇到的問題
1、解碼錯誤 (0x0101 lost synchronization)
對於這個錯誤的解釋可參考http://www.mars.org/pipermail/mad-dev/2004-January/000975.html
需要的兩個庫的下載地址以及編譯方法如下:
這個庫的configure腳本沒有提供編譯器選項。直接運行configure程序後,打開產生的Makefile,將CC=gcc改爲你要使用的編譯器的名字。
依賴於zlib,需要指定交叉編譯工具名稱,以及zlib庫的頭文件路徑(-I)、庫路徑(-L)
運行./configure --host=arm-xxx CPPFLAGS=-I(zlib頭文件路徑) LDFLAGS=-L(zlib庫路徑)
2、找不到“/dev/dsp"
對於無法正常播放聲音的系統,可採用手動建立dsp設備的方式:
sudo mknod /dev/dsp c 14 3 (其中的設備號可通過linux源碼目錄下/Documentation/devices.txt文件中查找/dev/dsp得到)
sudo chmod 666 /dev/dsp (設置普通用戶可用)
對於棄用/dev/dsp方式的系統來說,不能採用上述方式,可使用padsp程序,如:
padsp madplay xxx.mp3