PocketSphinx语音识别系统的编程
关于语音识别的基础知识和sphinx的知识,具体可以参考我的另外的博文:
语音识别的基础知识与CMUsphinx介绍:
http://blog.csdn.net/zouxy09/article/details/7941585
PocketSphinx语音识别系统的编译、安装和使用:
http://blog.csdn.net/zouxy09/article/details/7942784
PocketSphinx语音识别系统语言模型的训练和声学模型的改进:
http://blog.csdn.net/zouxy09/article/details/7949126
PocketSphinx语音识别系统声学模型的训练与使用
http://blog.csdn.net/zouxy09/article/details/7962382
本文主要实现PocketSphinx语音识别系统的编程使用,主要分两个方面,一个是编程解码语音文件(主要参考CMU sphinx的wiki:http://cmusphinx.sourceforge.net/wiki/),二是编程识别麦克风的语音(主要参考PocketSphinx源码包里的pocketsphinx.c文件)。对于后面加入我的人机交互系统的话,采用的是识别麦克风的语音的编程,具体使用时还需要对其进行精简。
一、编程解码语音文件
1、编程:
- #include <pocketsphinx.h>
- int main(int argc, char *argv[])
- {
- ps_decoder_t *ps;
- cmd_ln_t *config;
- FILE *fh;
- char const *hyp, *uttid;
- int16 buf[512];
- int rv;
- int32 score;
- //1、初始化:创建一个配置对象 cmd_ln_t *
- //cmd_ln_init函数第一个参数是我们需要更新的上一个配置,因为这里是初次创建,所以传入NULL;
- //第二个参数是一个定义参数的数组,如果使用的是标准配置的参数集的话可以通过调用ps_args()去获得。
- //第三个参数是是一个标志,它决定了参数的解释是否严格,如果为TRUE,那么遇到重复的或者未知的参
- //数,将会导致解释失败;
- //MODELDIR这个宏,指定了模型的路径,包括声学模型,语言模型和字典三个文件,是由gcc命令行传入,
- //我们通过pkg-config工具从PocketSphinx的配置中去获得这个modeldir变量
- config = cmd_ln_init(NULL, ps_args(), TRUE,
- "-hmm", MODELDIR "/hmm/en_US/hub4wsj_sc_8k",
- "-lm", MODELDIR "/lm/en/turtle.DMP",
- "-dict", MODELDIR "/lm/en/turtle.dic",
- NULL);
- if (config == NULL)
- return 1;
- //2、初始化解码器(语言识别就是一个解码过程,通俗的将就是将你说的话解码成对应的文字串)
- ps = ps_init(config);
- if (ps == NULL)
- return 1;
- //3、解码文件流
- //因为音频输入接口(麦克风)受到一些特定平台的影响,不利用我们演示,所以我们通过解码音频文件流
- //来演示PocketSphinx API的用法,goforward.raw是一个包含了一些诸如“go forward ten meters”等用来
- //控制机器人的短语(指令)的音频文件,其在test/data/goforward.raw。把它复制到当前目录
- fh = fopen("/dev/input/event14", "rb");
- if (fh == NULL) {
- perror("Failed to open goforward.raw");
- return 1;
- }
- //4、使用ps_decode_raw()进行解码
- rv = ps_decode_raw(ps, fh, NULL, -1);
- if (rv < 0)
- return 1;
- //5、得到解码的结果(概率最大的字串) hypothesis
- hyp = ps_get_hyp(ps, &score, &uttid);
- if (hyp == NULL)
- return 1;
- printf("Recognized: %s\n", hyp);
- //从内存中解码音频数据
- //现在我们将再次解码相同的文件,但是使用API从内存块中解码音频数据。在这种情况下,首先我们
- //需要使用ps_start_utt()开始说话:
- fseek(fh, 0, SEEK_SET);
- rv = ps_start_utt(ps, NULL);
- if (rv < 0)
- return 1;
- while (!feof(fh)) {
- rv = ps_start_utt(ps, NULL);
- if (rv < 0)
- return 1;
- printf("ready:\n");
- size_t nsamp;
- nsamp = fread(buf, 2, 512, fh);
- printf("read:\n");
- //我们将每次从文件中读取512大小的样本,使用ps_process_raw()把它们放到解码器中:
- rv = ps_process_raw(ps, buf, nsamp, FALSE, FALSE);
- printf("process:\n");
- }
- //我们需要使用ps_end_utt()去标记说话的结尾处:
- rv = ps_end_utt(ps);
- if (rv < 0)
- return 1;
- //以相同精确的方式运行来检索假设的字符串:
- hyp = ps_get_hyp(ps, &score, &uttid);
- if (hyp == NULL)
- return 1;
- printf("Recognized: %s\n", hyp);
- }
- //6、清理工作:使用ps_free()释放使用ps_init()返回的对象,不用释放配置对象。
- fclose(fh);
- ps_free(ps);
- return 0;
- }
2、编译:
编译方法:
gcc -o test_ps test_ps.c \
-DMODELDIR=\"`pkg-config --variable=modeldir pocketsphinx`\" \
`pkg-config --cflags --libs pocketsphinx sphinxbase`
//gcc的-D选项,指定宏定义,如-Dmacro=defn 相当于C语言中的#define macro=defn那么上面就表示在test_ps.c文件中,新加入一个宏定义:
#define MODELDIR=\"`pkg-config --variable=modeldir pocketsphinx`\"
\表示转义符,把“号转义。
这么做是为什么呢?因为程序中需要指定MODELDIR这个变量,但是因为不同的使用者,这个变量不一样,没办法指定死一个路径,所以只能放在编译时,让用户去根据自己的情况来指定。
pkg-config工具可以获得一个库的编译和连接等信息;
#pkg-config --cflags --libs pocketsphinx sphinxbase
显示:
-I/usr/local/include/sphinxbase -I/usr/local/include/pocketsphinx
-L/usr/local/lib -lpocketsphinx -lsphinxbase –lsphinxad
#pkg-config --variable=modeldir pocketsphinx
显示结果输出:/usr/local/share/pocketsphinx/model
二、编程解码麦克风的录音
1、编程
麦克风录音数据的获得主要是用sphinxbase封装了alsa的接口来实现。
- #include <stdio.h>
- #include <string.h>
- #include <sys/types.h>
- #include <sys/time.h>
- #include <signal.h>
- #include <setjmp.h>
- #include <sphinxbase/err.h>
- //generic live audio interface for recording and playback
- #include <sphinxbase/ad.h>
- #include <sphinxbase/cont_ad.h>
- #include "pocketsphinx.h"
- static ps_decoder_t *ps;
- static cmd_ln_t *config;
- static void print_word_times(int32 start)
- {
- ps_seg_t *iter = ps_seg_iter(ps, NULL);
- while (iter != NULL)
- {
- int32 sf, ef, pprob;
- float conf;
- ps_seg_frames (iter, &sf, &ef);
- pprob = ps_seg_prob (iter, NULL, NULL, NULL);
- conf = logmath_exp(ps_get_logmath(ps), pprob);
- printf ("%s %f %f %f\n", ps_seg_word (iter), (sf + start) / 100.0, (ef + start) / 100.0, conf);
- iter = ps_seg_next (iter);
- }
- }
- /* Sleep for specified msec */
- static void sleep_msec(int32 ms)
- {
- struct timeval tmo;
- tmo.tv_sec = 0;
- tmo.tv_usec = ms * 1000;
- select(0, NULL, NULL, NULL, &tmo);
- }
- /*
- * Main utterance processing loop:
- * for (;;) {
- * wait for start of next utterance;
- * decode utterance until silence of at least 1 sec observed;
- * print utterance result;
- * }
- */
- static void recognize_from_microphone()
- {
- ad_rec_t *ad;
- int16 adbuf[4096];
- int32 k, ts, rem;
- char const *hyp;
- char const *uttid;
- cont_ad_t *cont;
- char word[256];
- if ((ad = ad_open_dev(cmd_ln_str_r(config, "-adcdev"),
- (int)cmd_ln_float32_r(config, "-samprate"))) == NULL)
- E_FATAL("Failed top open audio device\n");
- /* Initialize continuous listening module */
- if ((cont = cont_ad_init(ad, ad_read)) == NULL)
- E_FATAL("Failed to initialize voice activity detection\n");
- if (ad_start_rec(ad) < 0)
- E_FATAL("Failed to start recording\n");
- if (cont_ad_calib(cont) < 0)
- E_FATAL("Failed to calibrate voice activity detection\n");
- for (;;) {
- /* Indicate listening for next utterance */
- printf("READY....\n");
- fflush(stdout);
- fflush(stderr);
- /* Wait data for next utterance */
- while ((k = cont_ad_read(cont, adbuf, 4096)) == 0)
- sleep_msec(100);
- if (k < 0)
- E_FATAL("Failed to read audio\n");
- /*
- * Non-zero amount of data received; start recognition of new utterance.
- * NULL argument to uttproc_begin_utt => automatic generation of utterance-id.
- */
- if (ps_start_utt(ps, NULL) < 0)
- E_FATAL("Failed to start utterance\n");
- ps_process_raw(ps, adbuf, k, FALSE, FALSE);
- printf("Listening...\n");
- fflush(stdout);
- /* Note timestamp for this first block of data */
- ts = cont->read_ts;
- /* Decode utterance until end (marked by a "long" silence, >1sec) */
- for (;;) {
- /* Read non-silence audio data, if any, from continuous listening module */
- if ((k = cont_ad_read(cont, adbuf, 4096)) < 0)
- E_FATAL("Failed to read audio\n");
- if (k == 0) {
- /*
- * No speech data available; check current timestamp with most recent
- * speech to see if more than 1 sec elapsed. If so, end of utterance.
- */
- if ((cont->read_ts - ts) > DEFAULT_SAMPLES_PER_SEC)
- break;
- }
- else {
- /* New speech data received; note current timestamp */
- ts = cont->read_ts;
- }
- /*
- * Decode whatever data was read above.
- */
- rem = ps_process_raw(ps, adbuf, k, FALSE, FALSE);
- /* If no work to be done, sleep a bit */
- if ((rem == 0) && (k == 0))
- sleep_msec(20);
- }
- /*
- * Utterance ended; flush any accumulated, unprocessed A/D data and stop
- * listening until current utterance completely decoded
- */
- ad_stop_rec(ad);
- while (ad_read(ad, adbuf, 4096) >= 0);
- cont_ad_reset(cont);
- printf("Stopped listening, please wait...\n");
- fflush(stdout);
- /* Finish decoding, obtain and print result */
- ps_end_utt(ps);
- hyp = ps_get_hyp(ps, NULL, &uttid);
- printf("%s: %s\n", uttid, hyp);
- fflush(stdout);
- /* Exit if the first word spoken was GOODBYE */
- if (hyp) {
- sscanf(hyp, "%s", word);
- if (strcmp(word, "goodbye") == 0)
- break;
- }
- /* Resume A/D recording for next utterance */
- if (ad_start_rec(ad) < 0)
- E_FATAL("Failed to start recording\n");
- }
- cont_ad_close(cont);
- ad_close(ad);
- }
- static jmp_buf jbuf;
- static void sighandler(int signo)
- {
- longjmp(jbuf, 1);
- }
- int main(int argc, char *argv[])
- {
- config = cmd_ln_init(NULL, ps_args(), TRUE,
- "-hmm", MODELDIR "/hmm/en_US/hub4wsj_sc_8k",
- "-lm", MODELDIR "/lm/en/turtle.DMP",
- "-dict", MODELDIR "/lm/en/turtle.dic",
- NULL);
- if (config == NULL)
- return 1;
- ps = ps_init(config);
- if (ps == NULL)
- return 1;
- signal(SIGINT, &sighandler);
- if (setjmp(jbuf) == 0)
- recognize_from_microphone();
- ps_free(ps);
- return 0;
- }
2、编译
和1.2一样。
至于说后面把PocketSphinx语音识别系统加入我的人机交互系统这个阶段,因为感觉这个系统本身的识别率不是很高,自己做了适应和重新训练声学和语言模型后,提升还是有限,暂时实用性还不是很强,所以暂时搁置下,看能不能通过其他方法去改进目前的状态。希望有牛人指导下。另外,由于开学了,需要上课,所以后续的进程可能会稍微减慢,不过依然期待各位多多交流!呵呵