chasing indexing error

1 files changed, 0 insertions, 157 deletions

diff --git a/ffmpeg/libavdevice/alsa-audio-dec.c b/ffmpeg/libavdevice/alsa-audio-dec.c
deleted file mode 100644
index 03154b0..0000000
--- a/ffmpeg/libavdevice/alsa-audio-dec.c
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * ALSA input and output
- * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
- * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
- *
- * 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
- */
-
-/**
- * @file
- * ALSA input and output: input
- * @author Luca Abeni ( lucabe72 email it )
- * @author Benoit Fouet ( benoit fouet free fr )
- * @author Nicolas George ( nicolas george normalesup org )
- *
- * This avdevice decoder allows to capture audio from an ALSA (Advanced
- * Linux Sound Architecture) device.
- *
- * The filename parameter is the name of an ALSA PCM device capable of
- * capture, for example "default" or "plughw:1"; see the ALSA documentation
- * for naming conventions. The empty string is equivalent to "default".
- *
- * The capture period is set to the lower value available for the device,
- * which gives a low latency suitable for real-time capture.
- *
- * The PTS are an Unix time in microsecond.
- *
- * Due to a bug in the ALSA library
- * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
- * decoder does not work with certain ALSA plugins, especially the dsnoop
- * plugin.
- */
-
-#include <alsa/asoundlib.h>
-#include "libavformat/internal.h"
-#include "libavutil/opt.h"
-#include "libavutil/mathematics.h"
-#include "libavutil/time.h"
-
-#include "avdevice.h"
-#include "alsa-audio.h"
-
-static av_cold int audio_read_header(AVFormatContext *s1)
-{
-    AlsaData *s = s1->priv_data;
-    AVStream *st;
-    int ret;
-    enum AVCodecID codec_id;
-
-    st = avformat_new_stream(s1, NULL);
-    if (!st) {
-        av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
-
-        return AVERROR(ENOMEM);
-    }
-    codec_id    = s1->audio_codec_id;
-
-    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
-        &codec_id);
-    if (ret < 0) {
-        return AVERROR(EIO);
-    }
-
-    /* take real parameters */
-    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id    = 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 */
-    /* microseconds instead of seconds, MHz instead of Hz */
-    s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
-                                      s->period_size, 1.5E-6);
-    if (!s->timefilter)
-        goto fail;
-
-    return 0;
-
-fail:
-    snd_pcm_close(s->h);
-    return AVERROR(EIO);
-}
-
-static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
-{
-    AlsaData *s  = s1->priv_data;
-    int res;
-    int64_t dts;
-    snd_pcm_sframes_t delay = 0;
-
-    if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
-        return AVERROR(EIO);
-    }
-
-    while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
-        if (res == -EAGAIN) {
-            av_free_packet(pkt);
-
-            return AVERROR(EAGAIN);
-        }
-        if (ff_alsa_xrun_recover(s1, res) < 0) {
-            av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
-                   snd_strerror(res));
-            av_free_packet(pkt);
-
-            return AVERROR(EIO);
-        }
-        ff_timefilter_reset(s->timefilter);
-    }
-
-    dts = av_gettime();
-    snd_pcm_delay(s->h, &delay);
-    dts -= av_rescale(delay + res, 1000000, s->sample_rate);
-    pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
-    s->last_period = res;
-
-    pkt->size = res * s->frame_size;
-
-    return 0;
-}
-
-static const AVOption options[] = {
-    { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
-    { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
-    { NULL },
-};
-
-static const AVClass alsa_demuxer_class = {
-    .class_name     = "ALSA demuxer",
-    .item_name      = av_default_item_name,
-    .option         = options,
-    .version        = LIBAVUTIL_VERSION_INT,
-};
-
-AVInputFormat ff_alsa_demuxer = {
-    .name           = "alsa",
-    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
-    .priv_data_size = sizeof(AlsaData),
-    .read_header    = audio_read_header,
-    .read_packet    = audio_read_packet,
-    .read_close     = ff_alsa_close,
-    .flags          = AVFMT_NOFILE,
-    .priv_class     = &alsa_demuxer_class,
-};