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📄 audio.c

📁 Trolltech公司发布的图形界面操作系统。可在qt-embedded-2.3.10平台上编译为嵌入式图形界面操作系统。
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/* * Linux audio play and grab interface * Copyright (c) 2000, 2001 Fabrice Bellard. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA */#include "avformat.h"#include <stdlib.h>#include <stdio.h>#include <string.h>#include <sys/soundcard.h>#include <unistd.h>#include <fcntl.h>#include <sys/ioctl.h>#include <sys/mman.h>#include <sys/time.h>#define AUDIO_BLOCK_SIZE 4096typedef struct {    int fd;    int sample_rate;    int channels;    int frame_size; /* in bytes ! */    int codec_id;    int flip_left : 1;    uint8_t buffer[AUDIO_BLOCK_SIZE];    int buffer_ptr;} AudioData;static int audio_open(AudioData *s, int is_output, const char *audio_device){    int audio_fd;    int tmp, err;    char *flip = getenv("AUDIO_FLIP_LEFT");    /* open linux audio device */    if (!audio_device)        audio_device = "/dev/dsp";    if (is_output)        audio_fd = open(audio_device, O_WRONLY);    else        audio_fd = open(audio_device, O_RDONLY);    if (audio_fd < 0) {        perror(audio_device);        return -EIO;    }    if (flip && *flip == '1') {        s->flip_left = 1;    }    /* non blocking mode */    if (!is_output)        fcntl(audio_fd, F_SETFL, O_NONBLOCK);    s->frame_size = AUDIO_BLOCK_SIZE;#if 0    tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;    err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);    if (err < 0) {        perror("SNDCTL_DSP_SETFRAGMENT");    }#endif    /* select format : favour native format */    err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);    #ifdef WORDS_BIGENDIAN    if (tmp & AFMT_S16_BE) {        tmp = AFMT_S16_BE;    } else if (tmp & AFMT_S16_LE) {        tmp = AFMT_S16_LE;    } else {        tmp = 0;    }#else    if (tmp & AFMT_S16_LE) {        tmp = AFMT_S16_LE;    } else if (tmp & AFMT_S16_BE) {        tmp = AFMT_S16_BE;    } else {        tmp = 0;    }#endif    switch(tmp) {    case AFMT_S16_LE:        s->codec_id = CODEC_ID_PCM_S16LE;        break;    case AFMT_S16_BE:        s->codec_id = CODEC_ID_PCM_S16BE;        break;    default:        fprintf(stderr, "Soundcard does not support 16 bit sample format\n");        close(audio_fd);        return -EIO;    }    err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);    if (err < 0) {        perror("SNDCTL_DSP_SETFMT");        goto fail;    }        tmp = (s->channels == 2);    err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);    if (err < 0) {        perror("SNDCTL_DSP_STEREO");        goto fail;    }    if (tmp)        s->channels = 2;        tmp = s->sample_rate;    err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);    if (err < 0) {        perror("SNDCTL_DSP_SPEED");        goto fail;    }    s->sample_rate = tmp; /* store real sample rate */    s->fd = audio_fd;    return 0; fail:    close(audio_fd);    return -EIO;}static int audio_close(AudioData *s){    close(s->fd);    return 0;}/* sound output support */static int audio_write_header(AVFormatContext *s1){    AudioData *s = s1->priv_data;    AVStream *st;    int ret;    st = s1->streams[0];    s->sample_rate = st->codec.sample_rate;    s->channels = st->codec.channels;    ret = audio_open(s, 1, NULL);    if (ret < 0) {        return -EIO;    } else {        return 0;    }}static int audio_write_packet(AVFormatContext *s1, int stream_index,                              const uint8_t *buf, int size, int64_t pts){    AudioData *s = s1->priv_data;    int len, ret;    while (size > 0) {        len = AUDIO_BLOCK_SIZE - s->buffer_ptr;        if (len > size)            len = size;        memcpy(s->buffer + s->buffer_ptr, buf, len);        s->buffer_ptr += len;        if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {            for(;;) {                ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);                if (ret > 0)                    break;                if (ret < 0 && (errno != EAGAIN && errno != EINTR))                    return -EIO;            }            s->buffer_ptr = 0;        }        buf += len;        size -= len;    }    return 0;}static int audio_write_trailer(AVFormatContext *s1){    AudioData *s = s1->priv_data;    audio_close(s);    return 0;}/* grab support */static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap){    AudioData *s = s1->priv_data;    AVStream *st;    int ret;    if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)        return -1;    st = av_new_stream(s1, 0);    if (!st) {        return -ENOMEM;    }    s->sample_rate = ap->sample_rate;    s->channels = ap->channels;    ret = audio_open(s, 0, ap->device);    if (ret < 0) {        av_free(st);        return -EIO;    }    /* take real parameters */    st->codec.codec_type = CODEC_TYPE_AUDIO;    st->codec.codec_id = s->codec_id;    st->codec.sample_rate = s->sample_rate;    st->codec.channels = s->channels;    av_set_pts_info(s1, 48, 1, 1000000);  /* 48 bits pts in us */    return 0;}static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt){    AudioData *s = s1->priv_data;    int ret, bdelay;    int64_t cur_time;    struct audio_buf_info abufi;        if (av_new_packet(pkt, s->frame_size) < 0)        return -EIO;    for(;;) {        struct timeval tv;        fd_set fds;        tv.tv_sec = 0;        tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */        FD_ZERO(&fds);        FD_SET(s->fd, &fds);        /* This will block until data is available or we get a timeout */        (void) select(s->fd + 1, &fds, 0, 0, &tv);        ret = read(s->fd, pkt->data, pkt->size);        if (ret > 0)            break;        if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {            av_free_packet(pkt);            pkt->size = 0;            return 0;        }        if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {            av_free_packet(pkt);            return -EIO;        }    }    pkt->size = ret;    /* compute pts of the start of the packet */    cur_time = av_gettime();    bdelay = ret;    if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {        bdelay += abufi.bytes;    }    /* substract time represented by the number of bytes in the audio fifo */    cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);    /* convert to wanted units */    pkt->pts = cur_time & ((1LL << 48) - 1);    if (s->flip_left && s->channels == 2) {        int i;        short *p = (short *) pkt->data;        for (i = 0; i < ret; i += 4) {            *p = ~*p;            p += 2;        }    }    return 0;}static int audio_read_close(AVFormatContext *s1){    AudioData *s = s1->priv_data;    audio_close(s);    return 0;}static AVInputFormat audio_in_format = {    "audio_device",    "audio grab and output",    sizeof(AudioData),    NULL,    audio_read_header,    audio_read_packet,    audio_read_close,    .flags = AVFMT_NOFILE,};static AVOutputFormat audio_out_format = {    "audio_device",    "audio grab and output",    "",    "",    sizeof(AudioData),    /* XXX: we make the assumption that the soundcard accepts this format */    /* XXX: find better solution with "preinit" method, needed also in       other formats */#ifdef WORDS_BIGENDIAN    CODEC_ID_PCM_S16BE,#else    CODEC_ID_PCM_S16LE,#endif    CODEC_ID_NONE,    audio_write_header,    audio_write_packet,    audio_write_trailer,    .flags = AVFMT_NOFILE,};int audio_init(void){    av_register_input_format(&audio_in_format);    av_register_output_format(&audio_out_format);    return 0;}

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