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, 560 deletions

diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
deleted file mode 100644
index 9d4dde4837..0000000000
--- a/libao2/ao_oss.c
+++ /dev/null
@@ -1,560 +0,0 @@
-/*
- * OSS audio output driver
- *
- * 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 <sys/ioctl.h>
-#include <unistd.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <string.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "mixer.h"
-
-#ifdef HAVE_SYS_SOUNDCARD_H
-#include <sys/soundcard.h>
-#else
-#ifdef HAVE_SOUNDCARD_H
-#include <soundcard.h>
-#endif
-#endif
-
-#include "libaf/format.h"
-
-#include "audio_out.h"
-#include "audio_out_internal.h"
-
-static const ao_info_t info =
-{
-	"OSS/ioctl audio output",
-	"oss",
-	"A'rpi",
-	""
-};
-
-/* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
-
-LIBAO_EXTERN(oss)
-
-static int format2oss(int format)
-{
-    switch(format)
-    {
-    case AF_FORMAT_U8: return AFMT_U8;
-    case AF_FORMAT_S8: return AFMT_S8;
-    case AF_FORMAT_U16_LE: return AFMT_U16_LE;
-    case AF_FORMAT_U16_BE: return AFMT_U16_BE;
-    case AF_FORMAT_S16_LE: return AFMT_S16_LE;
-    case AF_FORMAT_S16_BE: return AFMT_S16_BE;
-#ifdef AFMT_S24_PACKED
-    case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
-#endif
-#ifdef AFMT_U32_LE
-    case AF_FORMAT_U32_LE: return AFMT_U32_LE;
-#endif
-#ifdef AFMT_U32_BE
-    case AF_FORMAT_U32_BE: return AFMT_U32_BE;
-#endif
-#ifdef AFMT_S32_LE
-    case AF_FORMAT_S32_LE: return AFMT_S32_LE;
-#endif
-#ifdef AFMT_S32_BE
-    case AF_FORMAT_S32_BE: return AFMT_S32_BE;
-#endif
-#ifdef AFMT_FLOAT
-    case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
-#endif
-    // SPECIALS
-    case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
-    case AF_FORMAT_A_LAW: return AFMT_A_LAW;
-    case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
-#ifdef AFMT_MPEG
-    case AF_FORMAT_MPEG2: return AFMT_MPEG;
-#endif
-#ifdef AFMT_AC3
-    case AF_FORMAT_AC3_NE: return AFMT_AC3;
-#endif
-    }
-    mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
-    return -1;
-}
-
-static int oss2format(int format)
-{
-    switch(format)
-    {
-    case AFMT_U8: return AF_FORMAT_U8;
-    case AFMT_S8: return AF_FORMAT_S8;
-    case AFMT_U16_LE: return AF_FORMAT_U16_LE;
-    case AFMT_U16_BE: return AF_FORMAT_U16_BE;
-    case AFMT_S16_LE: return AF_FORMAT_S16_LE;
-    case AFMT_S16_BE: return AF_FORMAT_S16_BE;
-#ifdef AFMT_S24_PACKED
-    case AFMT_S24_PACKED: return AF_FORMAT_S24_LE;
-#endif
-#ifdef AFMT_U32_LE
-    case AFMT_U32_LE: return AF_FORMAT_U32_LE;
-#endif
-#ifdef AFMT_U32_BE
-    case AFMT_U32_BE: return AF_FORMAT_U32_BE;
-#endif
-#ifdef AFMT_S32_LE
-    case AFMT_S32_LE: return AF_FORMAT_S32_LE;
-#endif
-#ifdef AFMT_S32_BE
-    case AFMT_S32_BE: return AF_FORMAT_S32_BE;
-#endif
-#ifdef AFMT_FLOAT
-    case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
-#endif
-    // SPECIALS
-    case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
-    case AFMT_A_LAW: return AF_FORMAT_A_LAW;
-    case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
-#ifdef AFMT_MPEG
-    case AFMT_MPEG: return AF_FORMAT_MPEG2;
-#endif
-#ifdef AFMT_AC3
-    case AFMT_AC3: return AF_FORMAT_AC3_NE;
-#endif
-    }
-    mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format);
-    return -1;
-}
-
-static char *dsp=PATH_DEV_DSP;
-static audio_buf_info zz;
-static int audio_fd=-1;
-static int prepause_space;
-
-static const char *oss_mixer_device = PATH_DEV_MIXER;
-static int oss_mixer_channel = SOUND_MIXER_PCM;
-
-#ifdef SNDCTL_DSP_GETPLAYVOL
-static int volume_oss4(ao_control_vol_t *vol, int cmd) {
-    int v;
-
-    if (audio_fd < 0)
-        return CONTROL_ERROR;
-
-    if (cmd == AOCONTROL_GET_VOLUME) {
-        if (ioctl(audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1)
-            return CONTROL_ERROR;
-        vol->right = (v & 0xff00) >> 8;
-        vol->left = v & 0x00ff;
-        return CONTROL_OK;
-    } else if (cmd == AOCONTROL_SET_VOLUME) {
-        v = ((int) vol->right << 8) | (int) vol->left;
-        if (ioctl(audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1)
-            return CONTROL_ERROR;
-        return CONTROL_OK;
-    } else
-        return CONTROL_UNKNOWN;
-}
-#endif
-
-// to set/get/query special features/parameters
-static int control(int cmd,void *arg){
-    switch(cmd){
-	case AOCONTROL_GET_VOLUME:
-	case AOCONTROL_SET_VOLUME:
-	{
-	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
-	    int fd, v, devs;
-
-#ifdef SNDCTL_DSP_GETPLAYVOL
-        // Try OSS4 first
-        if (volume_oss4(vol, cmd) == CONTROL_OK)
-            return CONTROL_OK;
-#endif
-
-	    if(AF_FORMAT_IS_AC3(ao_data.format))
-		return CONTROL_TRUE;
-
-	    if ((fd = open(oss_mixer_device, O_RDONLY)) != -1)
-	    {
-		ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
-		if (devs & (1 << oss_mixer_channel))
-		{
-		    if (cmd == AOCONTROL_GET_VOLUME)
-		    {
-		        ioctl(fd, MIXER_READ(oss_mixer_channel), &v);
-			vol->right = (v & 0xFF00) >> 8;
-			vol->left = v & 0x00FF;
-		    }
-		    else
-		    {
-		        v = ((int)vol->right << 8) | (int)vol->left;
-			ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
-		    }
-		}
-		else
-		{
-		    close(fd);
-		    return CONTROL_ERROR;
-		}
-		close(fd);
-		return CONTROL_OK;
-	    }
-	}
-	return CONTROL_ERROR;
-    }
-    return CONTROL_UNKNOWN;
-}
-
-// open & setup audio device
-// return: 1=success 0=fail
-static int init(int rate,int channels,int format,int flags){
-  char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
-  int oss_format;
-  char *mdev = mixer_device, *mchan = mixer_channel;
-
-  mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz  %d chans  %s\n",rate,channels,
-    af_fmt2str_short(format));
-
-  if (ao_subdevice) {
-    char *m,*c;
-    m = strchr(ao_subdevice,':');
-    if(m) {
-      c = strchr(m+1,':');
-      if(c) {
-        mchan = c+1;
-        c[0] = '\0';
-      }
-      mdev = m+1;
-      m[0] = '\0';
-    }
-    dsp = ao_subdevice;
-  }
-
-  if(mdev)
-    oss_mixer_device=mdev;
-  else
-    oss_mixer_device=PATH_DEV_MIXER;
-
-  if(mchan){
-    int fd, devs, i;
-
-    if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
-      mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n",
-        oss_mixer_device, strerror(errno));
-    }else{
-      ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
-      close(fd);
-
-      for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
-        if(!strcasecmp(mixer_channels[i], mchan)){
-          if(!(devs & (1 << i))){
-            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
-            i = SOUND_MIXER_NRDEVICES+1;
-            break;
-          }
-          oss_mixer_channel = i;
-          break;
-        }
-      }
-      if(i==SOUND_MIXER_NRDEVICES){
-        mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
-      }
-    }
-  } else
-    oss_mixer_channel = SOUND_MIXER_PCM;
-
-  mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
-  mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device);
-  mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]);
-
-#ifdef __linux__
-  audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
-#else
-  audio_fd=open(dsp, O_WRONLY);
-#endif
-  if(audio_fd<0){
-    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno));
-    return 0;
-  }
-
-#ifdef __linux__
-  /* Remove the non-blocking flag */
-  if(fcntl(audio_fd, F_SETFL, 0) < 0) {
-   mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't make file descriptor blocking: %s\n", strerror(errno));
-   return 0;
-  }
-#endif
-
-#if defined(FD_CLOEXEC) && defined(F_SETFD)
-  fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
-#endif
-
-  if(AF_FORMAT_IS_AC3(format)) {
-    ao_data.samplerate=rate;
-    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
-  }
-
-ac3_retry:
-  if (AF_FORMAT_IS_AC3(format))
-    format = AF_FORMAT_AC3_NE;
-  ao_data.format=format;
-  oss_format=format2oss(format);
-  if (oss_format == -1) {
-#if BYTE_ORDER == BIG_ENDIAN
-    oss_format=AFMT_S16_BE;
-#else
-    oss_format=AFMT_S16_LE;
-#endif
-    format=AF_FORMAT_S16_NE;
-  }
-  if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
-      oss_format != format2oss(format)) {
-    mp_tmsg(MSGT_AO,MSGL_WARN, "[AO OSS] Can't set audio device %s to %s output, trying %s...\n", dsp,
-            af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
-    format=AF_FORMAT_S16_NE;
-    goto ac3_retry;
-  }
-#if 0
-  if(oss_format!=format2oss(format))
-	mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-af format'\n",audio_out_format_name(format));
-#endif
-
-  ao_data.format = oss2format(oss_format);
-  if (ao_data.format == -1) return 0;
-
-  mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
-    af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
-
-  ao_data.channels = channels;
-  if(!AF_FORMAT_IS_AC3(format)) {
-    // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
-    if (ao_data.channels > 2) {
-      if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
-	   ao_data.channels != channels ) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", channels);
-	return 0;
-      }
-    }
-    else {
-      int c = ao_data.channels-1;
-      if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", ao_data.channels);
-	return 0;
-      }
-      ao_data.channels=c+1;
-    }
-    mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
-    // set rate
-    ao_data.samplerate=rate;
-    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
-    mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
-  }
-
-  if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
-      int r=0;
-      mp_tmsg(MSGT_AO,MSGL_WARN,"[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
-      if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
-          mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
-      } else {
-          ao_data.outburst=r;
-          mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
-      }
-  } else {
-      mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d  (%d bytes/frag)  free: %6d\n",
-          zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
-      if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
-      ao_data.outburst=zz.fragsize;
-  }
-
-  if(ao_data.buffersize==-1){
-    // Measuring buffer size:
-    void* data;
-    ao_data.buffersize=0;
-#ifdef HAVE_AUDIO_SELECT
-    data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
-    while(ao_data.buffersize<0x40000){
-      fd_set rfds;
-      struct timeval tv;
-      FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
-      tv.tv_sec=0; tv.tv_usec = 0;
-      if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
-      write(audio_fd,data,ao_data.outburst);
-      ao_data.buffersize+=ao_data.outburst;
-    }
-    free(data);
-    if(ao_data.buffersize==0){
-        mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\n   ***  Your audio driver DOES NOT support select()  ***\n Recompile mpv with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
-        return 0;
-    }
-#endif
-  }
-
-  ao_data.bps=ao_data.channels;
-  switch (ao_data.format & AF_FORMAT_BITS_MASK) {
-  case AF_FORMAT_8BIT:
-    break;
-  case AF_FORMAT_16BIT:
-    ao_data.bps*=2;
-    break;
-  case AF_FORMAT_24BIT:
-    ao_data.bps*=3;
-    break;
-  case AF_FORMAT_32BIT:
-    ao_data.bps*=4;
-    break;
-  }
-
-  ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
-  ao_data.bps*=ao_data.samplerate;
-
-    return 1;
-}
-
-// close audio device
-static void uninit(int immed){
-    if(audio_fd == -1) return;
-#ifdef SNDCTL_DSP_SYNC
-    // to get the buffer played
-    if (!immed)
-	ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
-#endif
-#ifdef SNDCTL_DSP_RESET
-    if (immed)
-	ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
-#endif
-    close(audio_fd);
-    audio_fd = -1;
-}
-
-// stop playing and empty buffers (for seeking/pause)
-static void reset(void){
-  int oss_format;
-    uninit(1);
-    audio_fd=open(dsp, O_WRONLY);
-    if(audio_fd < 0){
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
-	return;
-    }
-
-#if defined(FD_CLOEXEC) && defined(F_SETFD)
-  fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
-#endif
-
-  oss_format = format2oss(ao_data.format);
-  if(AF_FORMAT_IS_AC3(ao_data.format))
-    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
-  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
-  if(!AF_FORMAT_IS_AC3(ao_data.format)) {
-    if (ao_data.channels > 2)
-      ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
-    else {
-      int c = ao_data.channels-1;
-      ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
-    }
-    ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
-  }
-}
-
-// stop playing, keep buffers (for pause)
-static void audio_pause(void)
-{
-    prepause_space = get_space();
-    uninit(1);
-}
-
-// resume playing, after audio_pause()
-static void audio_resume(void)
-{
-    int fillcnt;
-    reset();
-    fillcnt = get_space() - prepause_space;
-    if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
-      void *silence = calloc(fillcnt, 1);
-      play(silence, fillcnt, 0);
-      free(silence);
-    }
-}
-
-
-// return: how many bytes can be played without blocking
-static int get_space(void){
-  int playsize=ao_data.outburst;
-
-#ifdef SNDCTL_DSP_GETOSPACE
-  if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
-      // calculate exact buffer space:
-      playsize = zz.fragments*zz.fragsize;
-      return playsize;
-  }
-#endif
-
-    // check buffer
-#ifdef HAVE_AUDIO_SELECT
-    {  fd_set rfds;
-       struct timeval tv;
-       FD_ZERO(&rfds);
-       FD_SET(audio_fd, &rfds);
-       tv.tv_sec = 0;
-       tv.tv_usec = 0;
-       if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
-    }
-#endif
-
-  return ao_data.outburst;
-}
-
-// plays 'len' bytes of 'data'
-// it should round it down to outburst*n
-// return: number of bytes played
-static int play(void* data,int len,int flags){
-    if(len==0)
-        return len;
-    if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
-        len/=ao_data.outburst;
-        len*=ao_data.outburst;
-    }
-    len=write(audio_fd,data,len);
-    return len;
-}
-
-static int audio_delay_method=2;
-
-// return: delay in seconds between first and last sample in buffer
-static float get_delay(void){
-  /* Calculate how many bytes/second is sent out */
-  if(audio_delay_method==2){
-#ifdef SNDCTL_DSP_GETODELAY
-      int r=0;
-      if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
-         return ((float)r)/(float)ao_data.bps;
-#endif
-      audio_delay_method=1; // fallback if not supported
-  }
-  if(audio_delay_method==1){
-      // SNDCTL_DSP_GETOSPACE
-      if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
-         return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
-      audio_delay_method=0; // fallback if not supported
-  }
-  return ((float)ao_data.buffersize)/(float)ao_data.bps;
-}