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, 162 insertions, 0 deletions

diff --git a/audio/decode/ad_dvdpcm.c b/audio/decode/ad_dvdpcm.c
new file mode 100644
index 0000000000..41f6a1426d
--- /dev/null
+++ b/audio/decode/ad_dvdpcm.c
@@ -0,0 +1,162 @@
+/*
+ * 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;
+}