view audio.c @ 2660:022174d849d5 libavformat

fix issue 225, instead of stoping when wrong atom size is found, limit atom size to what is left, assuming container atom has correct size.. cricket4.3g2 has incorrect moov atom size which indicates that file size should be 2 bytes bigger than it is and quicktime reads it correctly though.

/*
 * Linux audio play and grab interface
 * Copyright (c) 2000, 2001 Fabrice Bellard.
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
#include "avformat.h"

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#ifdef HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <sys/time.h>

#define AUDIO_BLOCK_SIZE 4096

typedef 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");

    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 AVERROR(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:
        av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
        close(audio_fd);
        return AVERROR(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 AVERROR(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, s1->filename);
    if (ret < 0) {
        return AVERROR(EIO);
    } else {
        return 0;
    }
}

static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
    AudioData *s = s1->priv_data;
    int len, ret;
    int size= pkt->size;
    uint8_t *buf= pkt->data;

    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 AVERROR(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->sample_rate <= 0 || ap->channels <= 0)
        return -1;

    st = av_new_stream(s1, 0);
    if (!st) {
        return AVERROR(ENOMEM);
    }
    s->sample_rate = ap->sample_rate;
    s->channels = ap->channels;

    ret = audio_open(s, 0, s1->filename);
    if (ret < 0) {
        av_free(st);
        return AVERROR(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(st, 64, 1, 1000000);  /* 64 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 AVERROR(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;
            pkt->pts = av_gettime();
            return 0;
        }
        if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
            av_free_packet(pkt);
            return AVERROR(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;

    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;
}

#ifdef CONFIG_OSS_DEMUXER
AVInputFormat oss_demuxer = {
    "oss",
    "audio grab and output",
    sizeof(AudioData),
    NULL,
    audio_read_header,
    audio_read_packet,
    audio_read_close,
    .flags = AVFMT_NOFILE,
};
#endif

#ifdef CONFIG_OSS_MUXER
AVOutputFormat oss_muxer = {
    "oss",
    "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,
};
#endif