Rename directories, move files (step 1 of 2) (does not compile)

Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.

1 files changed, 0 insertions, 310 deletions

diff --git a/libmpcodecs/ad_spdif.c b/libmpcodecs/ad_spdif.c
deleted file mode 100644
index 877bc99317..0000000000
--- a/libmpcodecs/ad_spdif.c
+++ /dev/null
@@ -1,310 +0,0 @@
-/*
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <string.h>
-
-#include <libavformat/avformat.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/opt.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "ad_internal.h"
-
-static const ad_info_t info = {
-    "libavformat/spdifenc audio pass-through decoder.",
-    "spdif",
-    "Naoya OYAMA",
-    "Naoya OYAMA",
-    "For ALL hardware decoders"
-};
-
-LIBAD_EXTERN(spdif)
-
-#define FILENAME_SPDIFENC "spdif"
-#define OUTBUF_SIZE 65536
-struct spdifContext {
-    AVFormatContext *lavf_ctx;
-    int              iec61937_packet_size;
-    int              out_buffer_len;
-    int              out_buffer_size;
-    uint8_t         *out_buffer;
-    uint8_t          pb_buffer[OUTBUF_SIZE];
-};
-
-static int read_packet(void *p, uint8_t *buf, int buf_size)
-{
-    // spdifenc does not use read callback.
-    return 0;
-}
-
-static int write_packet(void *p, uint8_t *buf, int buf_size)
-{
-    int len;
-    struct spdifContext *ctx = p;
-
-    len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
-    memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
-    ctx->out_buffer_len += len;
-    return len;
-}
-
-static int64_t seek(void *p, int64_t offset, int whence)
-{
-    // spdifenc does not use seek callback.
-    return 0;
-}
-
-static int preinit(sh_audio_t *sh)
-{
-    sh->samplesize = 2;
-    return 1;
-}
-
-static int init(sh_audio_t *sh)
-{
-    int i, x, in_size, srate, bps, *dtshd_rate;
-    unsigned char *start;
-    double pts;
-    static const struct {
-        const char *name; enum CodecID id;
-    } fmt_id_type[] = {
-        { "aac" , CODEC_ID_AAC    },
-        { "ac3" , CODEC_ID_AC3    },
-        { "dca" , CODEC_ID_DTS    },
-        { "eac3", CODEC_ID_EAC3   },
-        { "mpa" , CODEC_ID_MP3    },
-        { "thd" , CODEC_ID_TRUEHD },
-        { NULL  , 0 }
-    };
-    AVFormatContext     *lavf_ctx  = NULL;
-    AVStream            *stream    = NULL;
-    const AVOption      *opt       = NULL;
-    struct spdifContext *spdif_ctx = NULL;
-
-    spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
-    if (!spdif_ctx)
-        goto fail;
-    spdif_ctx->lavf_ctx = avformat_alloc_context();
-    if (!spdif_ctx->lavf_ctx)
-        goto fail;
-
-    sh->context = spdif_ctx;
-    lavf_ctx    = spdif_ctx->lavf_ctx;
-
-    lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
-    if (!lavf_ctx->oformat)
-        goto fail;
-    lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
-    if (!lavf_ctx->priv_data)
-        goto fail;
-    lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
-                            read_packet, write_packet, seek);
-    if (!lavf_ctx->pb)
-        goto fail;
-    stream = avformat_new_stream(lavf_ctx, 0);
-    if (!stream)
-        goto fail;
-    lavf_ctx->duration   = AV_NOPTS_VALUE;
-    lavf_ctx->start_time = AV_NOPTS_VALUE;
-    for (i = 0; fmt_id_type[i].name; i++) {
-        if (!strcmp(sh->codec->dll, fmt_id_type[i].name)) {
-            lavf_ctx->streams[0]->codec->codec_id = fmt_id_type[i].id;
-            break;
-        }
-    }
-    lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
-    if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
-        mp_msg(MSGT_DECAUDIO,MSGL_INFO,
-               "This codec is not supported by spdifenc.\n");
-        goto fail;
-    }
-
-    // get sample_rate & bitrate from parser
-    bps = srate = 0;
-    x = ds_get_packet_pts(sh->ds, &start, &pts);
-    in_size = x;
-    if (x <= 0) {
-        pts = MP_NOPTS_VALUE;
-        x = 0;
-    }
-    ds_parse(sh->ds, &start, &x, pts, 0);
-    if (x == 0) { // not enough buffer
-        srate = 48000;    //fake value
-        bps   = 768000/8; //fake value
-    } else  if (sh->avctx) {
-        if (sh->avctx->sample_rate < 44100) {
-            mp_msg(MSGT_DECAUDIO,MSGL_INFO,
-                   "This stream sample_rate[%d Hz] may be broken. "
-                   "Force reset 48000Hz.\n",
-                   sh->avctx->sample_rate);
-            srate = 48000; //fake value
-        } else
-            srate = sh->avctx->sample_rate;
-        bps = sh->avctx->bit_rate/8;
-    }
-    sh->ds->buffer_pos -= in_size;
-
-    switch (lavf_ctx->streams[0]->codec->codec_id) {
-    case CODEC_ID_AAC:
-        spdif_ctx->iec61937_packet_size = 16384;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = srate;
-        sh->channels                    = 2;
-        sh->i_bps                       = bps;
-        break;
-    case CODEC_ID_AC3:
-        spdif_ctx->iec61937_packet_size = 6144;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = srate;
-        sh->channels                    = 2;
-        sh->i_bps                       = bps;
-        break;
-    case CODEC_ID_DTS: // FORCE USE DTS-HD
-        opt = av_opt_find(&lavf_ctx->oformat->priv_class,
-                          "dtshd_rate", NULL, 0, 0);
-        if (!opt)
-            goto fail;
-        dtshd_rate                      = (int*)(((uint8_t*)lavf_ctx->priv_data) +
-                                          opt->offset);
-        *dtshd_rate                     = 192000*4;
-        spdif_ctx->iec61937_packet_size = 32768;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = 192000; // DTS core require 48000
-        sh->channels                    = 2*4;
-        sh->i_bps                       = bps;
-        break;
-    case CODEC_ID_EAC3:
-        spdif_ctx->iec61937_packet_size = 24576;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = 192000;
-        sh->channels                    = 2;
-        sh->i_bps                       = bps;
-        break;
-    case CODEC_ID_MP3:
-        spdif_ctx->iec61937_packet_size = 4608;
-        sh->sample_format               = AF_FORMAT_MPEG2;
-        sh->samplerate                  = srate;
-        sh->channels                    = 2;
-        sh->i_bps                       = bps;
-        break;
-    case CODEC_ID_TRUEHD:
-        spdif_ctx->iec61937_packet_size = 61440;
-        sh->sample_format               = AF_FORMAT_IEC61937_LE;
-        sh->samplerate                  = 192000;
-        sh->channels                    = 8;
-        sh->i_bps                       = bps;
-        break;
-    default:
-        break;
-    }
-
-    return 1;
-
-fail:
-    uninit(sh);
-    return 0;
-}
-
-static int decode_audio(sh_audio_t *sh, unsigned char *buf,
-                        int minlen, int maxlen)
-{
-    struct spdifContext *spdif_ctx = sh->context;
-    AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
-    AVPacket            pkt;
-    double              pts;
-    int                 ret, in_size, consumed, x;
-    unsigned char       *start = NULL;
-
-    consumed = spdif_ctx->out_buffer_len  = 0;
-    spdif_ctx->out_buffer_size = maxlen;
-    spdif_ctx->out_buffer      = buf;
-    while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
-           && spdif_ctx->out_buffer_len < minlen) {
-        if (sh->ds->eof)
-            break;
-        x = ds_get_packet_pts(sh->ds, &start, &pts);
-        if (x <= 0) {
-            x = 0;
-            ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
-            if (x == 0)
-                continue; // END_NOT_FOUND
-            in_size = x;
-        } else {
-            in_size = x;
-            consumed = ds_parse(sh->ds, &start, &x, pts, 0);
-            if (x == 0) {
-                mp_msg(MSGT_DECAUDIO,MSGL_V,
-                       "start[%p] in_size[%d] consumed[%d] x[%d].\n",
-                       start, in_size, consumed, x);
-                continue; // END_NOT_FOUND
-            }
-            sh->ds->buffer_pos -= in_size - consumed;
-        }
-        av_init_packet(&pkt);
-        pkt.data = start;
-        pkt.size = x;
-        mp_msg(MSGT_DECAUDIO,MSGL_V,
-               "start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
-               start, pkt.size, in_size, consumed, x);
-        if (pts != MP_NOPTS_VALUE) {
-            sh->pts       = pts;
-            sh->pts_bytes = 0;
-        }
-        ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
-        if (ret < 0)
-            break;
-    }
-    sh->pts_bytes += spdif_ctx->out_buffer_len;
-    return spdif_ctx->out_buffer_len;
-}
-
-static int control(sh_audio_t *sh, int cmd, void* arg, ...)
-{
-    unsigned char *start;
-    double pts;
-
-    switch (cmd) {
-    case ADCTRL_RESYNC_STREAM:
-    case ADCTRL_SKIP_FRAME:
-        ds_get_packet_pts(sh->ds, &start, &pts);
-        return CONTROL_TRUE;
-    }
-    return CONTROL_UNKNOWN;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-    struct spdifContext *spdif_ctx = sh->context;
-    AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
-
-    if (lavf_ctx) {
-        if (lavf_ctx->oformat)
-            lavf_ctx->oformat->write_trailer(lavf_ctx);
-        av_freep(&lavf_ctx->pb);
-        if (lavf_ctx->streams) {
-            av_freep(&lavf_ctx->streams[0]->codec);
-            av_freep(&lavf_ctx->streams[0]->info);
-            av_freep(&lavf_ctx->streams[0]);
-        }
-        av_freep(&lavf_ctx->streams);
-        av_freep(&lavf_ctx->priv_data);
-    }
-    av_freep(&lavf_ctx);
-    av_freep(&spdif_ctx);
-}