avdevice: Apply a more consistent file naming scheme

1 files changed, 0 insertions, 96 deletions

diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h
deleted file mode 100644
index 26eaee6acf..0000000000
--- a/libavdevice/alsa-audio.h
+++ /dev/null
@@ -1,96 +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: definitions and structures
- * @author Luca Abeni ( lucabe72 email it )
- * @author Benoit Fouet ( benoit fouet free fr )
- */
-
-#ifndef AVDEVICE_ALSA_AUDIO_H
-#define AVDEVICE_ALSA_AUDIO_H
-
-#include <alsa/asoundlib.h>
-#include "config.h"
-#include "libavformat/avformat.h"
-#include "libavutil/log.h"
-
-/* XXX: we make the assumption that the soundcard accepts this format */
-/* XXX: find better solution with "preinit" method, needed also in
-        other formats */
-#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
-
-#define ALSA_BUFFER_SIZE_MAX 32768
-
-typedef struct AlsaData {
-    AVClass *class;
-    snd_pcm_t *h;
-    int frame_size;  ///< preferred size for reads and writes
-    int period_size; ///< bytes per sample * channels
-    int sample_rate; ///< sample rate set by user
-    int channels;    ///< number of channels set by user
-    void (*reorder_func)(const void *, void *, int);
-    void *reorder_buf;
-    int reorder_buf_size; ///< in frames
-} AlsaData;
-
-/**
- * Open an ALSA PCM.
- *
- * @param s media file handle
- * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
- * @param sample_rate in: requested sample rate;
- *                    out: actually selected sample rate
- * @param channels number of channels
- * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
- *                 out: actually selected AVCodecID, changed only if
- *                 AV_CODEC_ID_NONE was requested
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
-                 unsigned int *sample_rate,
-                 int channels, enum AVCodecID *codec_id);
-
-/**
- * Close the ALSA PCM.
- *
- * @param s1 media file handle
- *
- * @return 0
- */
-int ff_alsa_close(AVFormatContext *s1);
-
-/**
- * Try to recover from ALSA buffer underrun.
- *
- * @param s1 media file handle
- * @param err error code reported by the previous ALSA call
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
-
-int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
-
-#endif /* AVDEVICE_ALSA_AUDIO_H */