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.

57 files changed, 0 insertions, 16306 deletions

diff --git a/libmpcodecs/ad.c b/libmpcodecs/ad.c
deleted file mode 100644
index 93cebed86d..0000000000
--- a/libmpcodecs/ad.c
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * audio decoder interface
- *
- * 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 <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "config.h"
-
-#include "stream/stream.h"
-#include "libmpdemux/demuxer.h"
-#include "libmpdemux/stheader.h"
-#include "ad.h"
-
-/* Missed vorbis, mad, dshow */
-
-extern const ad_functions_t mpcodecs_ad_mpg123;
-extern const ad_functions_t mpcodecs_ad_ffmpeg;
-extern const ad_functions_t mpcodecs_ad_pcm;
-extern const ad_functions_t mpcodecs_ad_dvdpcm;
-extern const ad_functions_t mpcodecs_ad_spdif;
-
-const ad_functions_t * const mpcodecs_ad_drivers[] =
-{
-#ifdef CONFIG_MPG123
-  &mpcodecs_ad_mpg123,
-#endif
-  &mpcodecs_ad_ffmpeg,
-  &mpcodecs_ad_pcm,
-  &mpcodecs_ad_dvdpcm,
-  &mpcodecs_ad_spdif,
-  NULL
-};
diff --git a/libmpcodecs/ad.h b/libmpcodecs/ad.h
deleted file mode 100644
index 5396085d04..0000000000
--- a/libmpcodecs/ad.h
+++ /dev/null
@@ -1,54 +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.
- */
-
-#ifndef MPLAYER_AD_H
-#define MPLAYER_AD_H
-
-#include "mpc_info.h"
-#include "libmpdemux/stheader.h"
-
-typedef struct mp_codec_info ad_info_t;
-
-/* interface of video decoder drivers */
-typedef struct ad_functions
-{
-    const ad_info_t *info;
-    int (*preinit)(sh_audio_t *sh);
-    int (*init)(sh_audio_t *sh);
-    void (*uninit)(sh_audio_t *sh);
-    int (*control)(sh_audio_t *sh,int cmd,void* arg, ...);
-    int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
-                        int maxlen);
-} ad_functions_t;
-
-// NULL terminated array of all drivers
-extern const ad_functions_t * const mpcodecs_ad_drivers[];
-
-// fallback if ADCTRL_RESYNC not implemented: sh_audio->a_in_buffer_len=0;
-#define ADCTRL_RESYNC_STREAM 1   // resync, called after seeking
-
-// fallback if ADCTRL_SKIP not implemented: ds_fill_buffer(sh_audio->ds);
-#define ADCTRL_SKIP_FRAME 2      // skip block/frame, called while seeking
-
-// fallback if ADCTRL_QUERY_FORMAT not implemented: sh_audio->sample_format
-#define ADCTRL_QUERY_FORMAT 3    // test for availabilty of a format
-
-// fallback: use hw mixer in libao
-#define ADCTRL_SET_VOLUME 4      // not used at the moment
-
-#endif /* MPLAYER_AD_H */
diff --git a/libmpcodecs/ad_dvdpcm.c b/libmpcodecs/ad_dvdpcm.c
deleted file mode 100644
index 41f6a1426d..0000000000
--- a/libmpcodecs/ad_dvdpcm.c
+++ /dev/null
@@ -1,162 +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 <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "ad_internal.h"
-
-static const ad_info_t info =
-{
-	"Uncompressed DVD/VOB LPCM audio decoder",
-	"dvdpcm",
-	"Nick Kurshev",
-	"A'rpi",
-	""
-};
-
-LIBAD_EXTERN(dvdpcm)
-
-static int init(sh_audio_t *sh)
-{
-/* DVD PCM Audio:*/
-    sh->i_bps = 0;
-    if(sh->codecdata_len==3){
-	// we have LPCM header:
-	unsigned char h=sh->codecdata[1];
-	sh->channels=1+(h&7);
-	switch((h>>4)&3){
-	case 0: sh->samplerate=48000;break;
-	case 1: sh->samplerate=96000;break;
-	case 2: sh->samplerate=44100;break;
-	case 3: sh->samplerate=32000;break;
-	}
-	switch ((h >> 6) & 3) {
-	  case 0:
-	    sh->sample_format = AF_FORMAT_S16_BE;
-	    sh->samplesize = 2;
-	    break;
-	  case 1:
-	    mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
-	    sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
-	  case 2:
-	    sh->sample_format = AF_FORMAT_S24_BE;
-	    sh->samplesize = 3;
-	    break;
-	  default:
-	    sh->sample_format = AF_FORMAT_S16_BE;
-	    sh->samplesize = 2;
-	}
-    } else {
-	// use defaults:
-	sh->channels=2;
-	sh->samplerate=48000;
-	sh->sample_format = AF_FORMAT_S16_BE;
-	sh->samplesize = 2;
-    }
-    if (!sh->i_bps)
-    sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
-    return 1;
-}
-
-static int preinit(sh_audio_t *sh)
-{
-  sh->audio_out_minsize=2048;
-  return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-}
-
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
-{
-  int skip;
-    switch(cmd)
-    {
-      case ADCTRL_SKIP_FRAME:
-	skip=sh->i_bps/16;
-	skip=skip&(~3);
-	demux_read_data(sh->ds,NULL,skip);
-	return CONTROL_TRUE;
-    }
-  return CONTROL_UNKNOWN;
-}
-
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
-{
-  int j,len;
-  if (sh_audio->samplesize == 3) {
-    if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
-      // 20 bit
-      // not sure if the "& 0xf0" and "<< 4" are the right way around
-      // can somebody clarify?
-      for (j = 0; j < minlen; j += 12) {
-        char tmp[10];
-        len = demux_read_data(sh_audio->ds, tmp, 10);
-        if (len < 10) break;
-        // first sample
-        buf[j + 0] = tmp[0];
-        buf[j + 1] = tmp[1];
-        buf[j + 2] = tmp[8] & 0xf0;
-        // second sample
-        buf[j + 3] = tmp[2];
-        buf[j + 4] = tmp[3];
-        buf[j + 5] = tmp[8] << 4;
-        // third sample
-        buf[j + 6] = tmp[4];
-        buf[j + 7] = tmp[5];
-        buf[j + 8] = tmp[9] & 0xf0;
-        // fourth sample
-        buf[j + 9] = tmp[6];
-        buf[j + 10] = tmp[7];
-        buf[j + 11] = tmp[9] << 4;
-      }
-      len = j;
-    } else {
-      // 24 bit
-      for (j = 0; j < minlen; j += 12) {
-        char tmp[12];
-        len = demux_read_data(sh_audio->ds, tmp, 12);
-        if (len < 12) break;
-        // first sample
-        buf[j + 0] = tmp[0];
-        buf[j + 1] = tmp[1];
-        buf[j + 2] = tmp[8];
-        // second sample
-        buf[j + 3] = tmp[2];
-        buf[j + 4] = tmp[3];
-        buf[j + 5] = tmp[9];
-        // third sample
-        buf[j + 6] = tmp[4];
-        buf[j + 7] = tmp[5];
-        buf[j + 8] = tmp[10];
-        // fourth sample
-        buf[j + 9] = tmp[6];
-        buf[j + 10] = tmp[7];
-        buf[j + 11] = tmp[11];
-      }
-      len = j;
-    }
-  } else
-  len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
-  return len;
-}
diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
deleted file mode 100644
index 2eacfadb8f..0000000000
--- a/libmpcodecs/ad_ffmpeg.c
+++ /dev/null
@@ -1,413 +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 <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <stdbool.h>
-#include <assert.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavutil/opt.h>
-
-#include "talloc.h"
-
-#include "config.h"
-#include "mp_msg.h"
-#include "options.h"
-
-#include "ad_internal.h"
-#include "libaf/reorder_ch.h"
-
-#include "mpbswap.h"
-
-static const ad_info_t info =
-{
-    "libavcodec audio decoders",
-    "ffmpeg",
-    "",
-    "",
-    "",
-    .print_name = "libavcodec",
-};
-
-LIBAD_EXTERN(ffmpeg)
-
-struct priv {
-    AVCodecContext *avctx;
-    AVFrame *avframe;
-    char *output;
-    char *output_packed; // used by deplanarize to store packed audio samples
-    int output_left;
-    int unitsize;
-    int previous_data_left;  // input demuxer packet data
-};
-
-static int preinit(sh_audio_t *sh)
-{
-    return 1;
-}
-
-/* Prefer playing audio with the samplerate given in container data
- * if available, but take number the number of channels and sample format
- * from the codec, since if the codec isn't using the correct values for
- * those everything breaks anyway.
- */
-static int setup_format(sh_audio_t *sh_audio,
-                        const AVCodecContext *lavc_context)
-{
-    int sample_format = sh_audio->sample_format;
-    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
-    case AV_SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
-    case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
-    case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
-    case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
-    default:
-        mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
-        sample_format = AF_FORMAT_UNKNOWN;
-    }
-
-    bool broken_srate        = false;
-    int samplerate           = lavc_context->sample_rate;
-    int container_samplerate = sh_audio->container_out_samplerate;
-    if (!container_samplerate && sh_audio->wf)
-        container_samplerate = sh_audio->wf->nSamplesPerSec;
-    if (lavc_context->codec_id == CODEC_ID_AAC
-        && samplerate == 2 * container_samplerate)
-        broken_srate = true;
-    else if (container_samplerate)
-        samplerate = container_samplerate;
-
-    if (lavc_context->channels != sh_audio->channels ||
-        samplerate != sh_audio->samplerate ||
-        sample_format != sh_audio->sample_format) {
-        sh_audio->channels = lavc_context->channels;
-        sh_audio->samplerate = samplerate;
-        sh_audio->sample_format = sample_format;
-        sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
-        if (broken_srate)
-            mp_msg(MSGT_DECAUDIO, MSGL_WARN,
-                   "Ignoring broken container sample rate for AAC with SBR\n");
-        return 1;
-    }
-    return 0;
-}
-
-static int init(sh_audio_t *sh_audio)
-{
-    struct MPOpts *opts = sh_audio->opts;
-    AVCodecContext *lavc_context;
-    AVCodec *lavc_codec;
-
-    if (sh_audio->codec->dll) {
-        lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
-        if (!lavc_codec) {
-            mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
-                    "Cannot find codec '%s' in libavcodec...\n",
-                    sh_audio->codec->dll);
-            return 0;
-        }
-    } else if (!sh_audio->libav_codec_id) {
-        mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. "
-                "Generic lavc decoder is not applicable.\n");
-        return 0;
-    } else {
-        lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id);
-        if (!lavc_codec) {
-            mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder "
-                   "for this codec\n");
-            return 0;
-        }
-    }
-
-    sh_audio->codecname = lavc_codec->long_name;
-    if (!sh_audio->codecname)
-        sh_audio->codecname = lavc_codec->name;
-
-    struct priv *ctx = talloc_zero(NULL, struct priv);
-    sh_audio->context = ctx;
-    lavc_context = avcodec_alloc_context3(lavc_codec);
-    ctx->avctx = lavc_context;
-    ctx->avframe = avcodec_alloc_frame();
-
-    // Always try to set - option only exists for AC3 at the moment
-    av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
-                      AV_OPT_SEARCH_CHILDREN);
-    lavc_context->sample_rate = sh_audio->samplerate;
-    lavc_context->bit_rate = sh_audio->i_bps * 8;
-    if (sh_audio->wf) {
-        lavc_context->channels = sh_audio->wf->nChannels;
-        lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
-        lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
-        lavc_context->block_align = sh_audio->wf->nBlockAlign;
-        lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
-    }
-    lavc_context->request_channels = opts->audio_output_channels;
-    lavc_context->codec_tag = sh_audio->format; //FOURCC
-    if (sh_audio->gsh->lavf_codec_tag)
-        lavc_context->codec_tag = sh_audio->gsh->lavf_codec_tag;
-    lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
-    lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
-
-    /* alloc extra data */
-    if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
-        lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
-        lavc_context->extradata_size = sh_audio->wf->cbSize;
-        memcpy(lavc_context->extradata, sh_audio->wf + 1,
-               lavc_context->extradata_size);
-    }
-
-    // for QDM2
-    if (sh_audio->codecdata_len && sh_audio->codecdata &&
-            !lavc_context->extradata) {
-        lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
-                                            FF_INPUT_BUFFER_PADDING_SIZE);
-        lavc_context->extradata_size = sh_audio->codecdata_len;
-        memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
-               lavc_context->extradata_size);
-    }
-
-    /* open it */
-    if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
-        mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
-        uninit(sh_audio);
-        return 0;
-    }
-    mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
-           lavc_codec->name);
-
-    if (sh_audio->format == 0x3343414D) {
-        // MACE 3:1
-        sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
-        sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
-    } else if (sh_audio->format == 0x3643414D) {
-        // MACE 6:1
-        sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
-        sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
-    }
-
-    // Decode at least 1 byte:  (to get header filled)
-    for (int tries = 0;;) {
-        int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
-                             sh_audio->a_buffer_size);
-        if (x > 0) {
-            sh_audio->a_buffer_len = x;
-            break;
-        }
-        if (++tries >= 5) {
-            mp_msg(MSGT_DECAUDIO, MSGL_ERR,
-                   "ad_ffmpeg: initial decode failed\n");
-            uninit(sh_audio);
-            return 0;
-        }
-    }
-
-    sh_audio->i_bps = lavc_context->bit_rate / 8;
-    if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
-        sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
-
-    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
-    case AV_SAMPLE_FMT_U8:
-    case AV_SAMPLE_FMT_S16:
-    case AV_SAMPLE_FMT_S32:
-    case AV_SAMPLE_FMT_FLT:
-        break;
-    default:
-        uninit(sh_audio);
-        return 0;
-    }
-    return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-    sh->codecname = NULL;
-    struct priv *ctx = sh->context;
-    if (!ctx)
-        return;
-    AVCodecContext *lavc_context = ctx->avctx;
-
-    if (lavc_context) {
-        if (avcodec_close(lavc_context) < 0)
-            mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
-        av_freep(&lavc_context->extradata);
-        av_freep(&lavc_context);
-    }
-    avcodec_free_frame(&ctx->avframe);
-    talloc_free(ctx);
-    sh->context = NULL;
-}
-
-static int control(sh_audio_t *sh, int cmd, void *arg, ...)
-{
-    struct priv *ctx = sh->context;
-    switch (cmd) {
-    case ADCTRL_RESYNC_STREAM:
-        avcodec_flush_buffers(ctx->avctx);
-        ds_clear_parser(sh->ds);
-        ctx->previous_data_left = 0;
-        ctx->output_left = 0;
-        return CONTROL_TRUE;
-    }
-    return CONTROL_UNKNOWN;
-}
-
-static av_always_inline void deplanarize(struct sh_audio *sh)
-{
-    struct priv *priv = sh->context;
-
-    size_t bps        = av_get_bytes_per_sample(priv->avctx->sample_fmt);
-    size_t nb_samples = priv->avframe->nb_samples;
-    size_t channels   = priv->avctx->channels;
-    size_t size       = bps * nb_samples * channels;
-
-    if (talloc_get_size(priv->output_packed) != size)
-        priv->output_packed =
-            talloc_realloc_size(priv, priv->output_packed, size);
-
-    size_t offset = 0;
-    unsigned char *output_ptr = priv->output_packed;
-    unsigned char **src = priv->avframe->data;
-
-    for (size_t s = 0; s < nb_samples; s++) {
-        for (size_t c = 0; c < channels; c++) {
-            memcpy(output_ptr, src[c] + offset, bps);
-            output_ptr += bps;
-        }
-        offset += bps;
-    }
-
-    priv->output = priv->output_packed;
-}
-
-static int decode_new_packet(struct sh_audio *sh)
-{