chasing indexing error

1 files changed, 0 insertions, 328 deletions

diff --git a/ffmpeg/libavdevice/oss_audio.c b/ffmpeg/libavdevice/oss_audio.c
deleted file mode 100644
index 916908c..0000000
--- a/ffmpeg/libavdevice/oss_audio.c
+++ /dev/null
@@ -1,328 +0,0 @@
-/*
- * 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 "config.h"
-#include <stdlib.h>
-#include <stdio.h>
-#include <stdint.h>
-#include <string.h>
-#include <errno.h>
-#if HAVE_SOUNDCARD_H
-#include <soundcard.h>
-#else
-#include <sys/soundcard.h>
-#endif
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/ioctl.h>
-
-#include "libavutil/internal.h"
-#include "libavutil/log.h"
-#include "libavutil/opt.h"
-#include "libavutil/time.h"
-#include "libavcodec/avcodec.h"
-#include "avdevice.h"
-#include "libavformat/internal.h"
-
-#define AUDIO_BLOCK_SIZE 4096
-
-typedef struct {
-    AVClass *class;
-    int fd;
-    int sample_rate;
-    int channels;
-    int frame_size; /* in bytes ! */
-    enum AVCodecID codec_id;
-    unsigned int flip_left : 1;
-    uint8_t buffer[AUDIO_BLOCK_SIZE];
-    int buffer_ptr;
-} AudioData;
-
-static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
-{
-    AudioData *s = s1->priv_data;
-    int audio_fd;
-    int tmp, err;
-    char *flip = getenv("AUDIO_FLIP_LEFT");
-
-    if (is_output)
-        audio_fd = avpriv_open(audio_device, O_WRONLY);
-    else
-        audio_fd = avpriv_open(audio_device, O_RDONLY);
-    if (audio_fd < 0) {
-        av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
-        return AVERROR(EIO);
-    }
-
-    if (flip && *flip == '1') {
-        s->flip_left = 1;
-    }
-
-    /* non blocking mode */
-    if (!is_output) {
-        if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) {
-            av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, strerror(errno));
-        }
-    }
-
-    s->frame_size = AUDIO_BLOCK_SIZE;
-
-    /* select format : favour native format */
-    err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
-
-#if HAVE_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 = AV_CODEC_ID_PCM_S16LE;
-        break;
-    case AFMT_S16_BE:
-        s->codec_id = AV_CODEC_ID_PCM_S16BE;
-        break;
-    default:
-        av_log(s1, 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) {
-        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
-        goto fail;
-    }
-
-    tmp = (s->channels == 2);
-    err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
-    if (err < 0) {
-        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
-        goto fail;
-    }
-
-    tmp = s->sample_rate;
-    err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
-    if (err < 0) {
-        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
-        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(s1, 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 = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, 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)
-{
-    AudioData *s = s1->priv_data;
-    AVStream *st;
-    int ret;
-
-    st = avformat_new_stream(s1, NULL);
-    if (!st) {
-        return AVERROR(ENOMEM);
-    }
-
-    ret = audio_open(s1, 0, s1->filename);
-    if (ret < 0) {
-        return AVERROR(EIO);
-    }
-
-    /* take real parameters */
-    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = s->codec_id;
-    st->codec->sample_rate = s->sample_rate;
-    st->codec->channels = s->channels;
-
-    avpriv_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 ((ret=av_new_packet(pkt, s->frame_size)) < 0)
-        return ret;
-
-    ret = read(s->fd, pkt->data, pkt->size);
-    if (ret <= 0){
-        av_free_packet(pkt);
-        pkt->size = 0;
-        if (ret<0)  return AVERROR(errno);
-        else        return AVERROR_EOF;
-    }
-    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;
-    }
-    /* subtract 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;
-}
-
-#if CONFIG_OSS_INDEV
-static const AVOption options[] = {
-    { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
-    { "channels",    "", offsetof(AudioData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
-    { NULL },
-};
-
-static const AVClass oss_demuxer_class = {
-    .class_name     = "OSS demuxer",
-    .item_name      = av_default_item_name,
-    .option         = options,
-    .version        = LIBAVUTIL_VERSION_INT,
-};
-
-AVInputFormat ff_oss_demuxer = {
-    .name           = "oss",
-    .long_name      = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
-    .priv_data_size = sizeof(AudioData),
-    .read_header    = audio_read_header,
-    .read_packet    = audio_read_packet,
-    .read_close     = audio_read_close,
-    .flags          = AVFMT_NOFILE,
-    .priv_class     = &oss_demuxer_class,
-};
-#endif
-
-#if CONFIG_OSS_OUTDEV
-AVOutputFormat ff_oss_muxer = {
-    .name           = "oss",
-    .long_name      = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
-    .priv_data_size = 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 */
-    .audio_codec    = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
-    .video_codec    = AV_CODEC_ID_NONE,
-    .write_header   = audio_write_header,
-    .write_packet   = audio_write_packet,
-    .write_trailer  = audio_write_trailer,
-    .flags          = AVFMT_NOFILE,
-};
-#endif