avdevice: Apply a more consistent file naming scheme

1 files changed, 0 insertions, 117 deletions

diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
deleted file mode 100644
index db4f5e426f..0000000000
--- a/libavdevice/alsa-audio-enc.c
+++ /dev/null
@@ -1,117 +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 Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * ALSA input and output: output
- * @author Luca Abeni ( lucabe72 email it )
- * @author Benoit Fouet ( benoit fouet free fr )
- *
- * This avdevice encoder allows to play audio to 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 playback period is set to the lower value available for the device,
- * which gives a low latency suitable for real-time playback.
- */
-
-#include <alsa/asoundlib.h>
-
-#include "libavutil/internal.h"
-
-#include "libavformat/avformat.h"
-
-#include "alsa-audio.h"
-
-static av_cold int audio_write_header(AVFormatContext *s1)
-{
-    AlsaData *s = s1->priv_data;
-    AVStream *st;
-    unsigned int sample_rate;
-    enum AVCodecID codec_id;
-    int res;
-
-    st = s1->streams[0];
-    sample_rate = st->codec->sample_rate;
-    codec_id    = st->codec->codec_id;
-    res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
-        st->codec->channels, &codec_id);
-    if (sample_rate != st->codec->sample_rate) {
-        av_log(s1, AV_LOG_ERROR,
-               "sample rate %d not available, nearest is %d\n",
-               st->codec->sample_rate, sample_rate);
-        goto fail;
-    }
-
-    return res;
-
-fail:
-    snd_pcm_close(s->h);
-    return AVERROR(EIO);
-}
-
-static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
-{
-    AlsaData *s = s1->priv_data;
-    int res;
-    int size     = pkt->size;
-    uint8_t *buf = pkt->data;
-
-    size /= s->frame_size;
-    if (s->reorder_func) {
-        if (size > s->reorder_buf_size)
-            if (ff_alsa_extend_reorder_buf(s, size))
-                return AVERROR(ENOMEM);
-        s->reorder_func(buf, s->reorder_buf, size);
-        buf = s->reorder_buf;
-    }
-    while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
-        if (res == -EAGAIN) {
-
-            return AVERROR(EAGAIN);
-        }
-
-        if (ff_alsa_xrun_recover(s1, res) < 0) {
-            av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
-                   snd_strerror(res));
-
-            return AVERROR(EIO);
-        }
-    }
-
-    return 0;
-}
-
-AVOutputFormat ff_alsa_muxer = {
-    .name           = "alsa",
-    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
-    .priv_data_size = sizeof(AlsaData),
-    .audio_codec    = DEFAULT_CODEC_ID,
-    .video_codec    = AV_CODEC_ID_NONE,
-    .write_header   = audio_write_header,
-    .write_packet   = audio_write_packet,
-    .write_trailer  = ff_alsa_close,
-    .flags          = AVFMT_NOFILE,
-};