libmpcodecs: remove redundant audio and video decoders

Probably all of these are supported by libavcodec. Missing things can
be added back.

Also remove qtpalette.h. It was used by demux_mov.c, and should have
been deleted with commit 1fde09db6f4ce.

1 files changed, 0 insertions, 577 deletions

diff --git a/libmpcodecs/ad_hwac3.c b/libmpcodecs/ad_hwac3.c
deleted file mode 100644
index 44e7657107..0000000000
--- a/libmpcodecs/ad_hwac3.c
+++ /dev/null
@@ -1,577 +0,0 @@
-/*
- * DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9"
- * (see http://www.dtek.chalmers.se/~dvd/)
- * Reference: DOCS/tech/hwac3.txt !!!!!
- *
- * 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.
- */
-
-#define _GNU_SOURCE
-#define _XOPEN_SOURCE 600
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <unistd.h>
-#include <sys/types.h>
-
-#include <libavutil/intreadwrite.h>
-#include <libavutil/common.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "mpbswap.h"
-
-#include "ad_internal.h"
-
-
-static int isdts = -1;
-
-static const ad_info_t info =
-{
-  "AC3/DTS pass-through S/PDIF",
-  "hwac3",
-  "Nick Kurshev/Peter Schüller",
-  "???",
-  ""
-};
-
-LIBAD_EXTERN(hwac3)
-
-
-static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate);
-static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf);
-
-
-static int a52_syncinfo (uint8_t *buf, int *sample_rate, int *bit_rate)
-{
-    static const uint16_t rate[] = { 32,  40,  48,  56,  64,  80,  96, 112,
-                                    128, 160, 192, 224, 256, 320, 384, 448,
-                                    512, 576, 640};
-    int frmsizecod;
-    int bitrate;
-    int half;
-
-    if (buf[0] != 0x0b || buf[1] != 0x77)    /* syncword */
-        return 0;
-
-    if (buf[5] >= 0x60)                      /* bsid >= 12 */
-        return 0;
-    half = buf[5] >> 3;
-    half = FFMAX(half - 8, 0);
-
-    frmsizecod = buf[4] & 63;
-    if (frmsizecod >= 38)
-        return 0;
-    bitrate = rate[frmsizecod >> 1];
-    *bit_rate = (bitrate * 1000) >> half;
-
-    switch (buf[4] & 0xc0) {
-    case 0:
-        *sample_rate = 48000 >> half;
-        return 4 * bitrate;
-    case 0x40:
-        *sample_rate = 44100 >> half;
-        return 2 * (320 * bitrate / 147 + (frmsizecod & 1));
-    case 0x80:
-        *sample_rate = 32000 >> half;
-        return 6 * bitrate;
-    default:
-        return 0;
-    }
-}
-
-static int ac3dts_fillbuff(sh_audio_t *sh_audio)
-{
-  int length = 0;
-  int flags = 0;
-  int sample_rate = 0;
-  int bit_rate = 0;
-
-  sh_audio->a_in_buffer_len = 0;
-  /* sync frame:*/
-  while(1)
-  {
-    // Original code DTS has a 10 bytes header.
-    // Now max 12 bytes for 14 bits DTS header.
-    while(sh_audio->a_in_buffer_len < 12)
-    {
-      int c = demux_getc(sh_audio->ds);
-      if(c<0)
-        return -1; /* EOF*/
-      sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++] = c;
-    }
-
-    if (sh_audio->format == 0x2001)
-    {
-      length = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
-      if(length >= 12)
-      {
-        if(isdts != 1)
-        {
-          mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to DTS, %d bps, %d Hz\n", bit_rate, sample_rate);
-          isdts = 1;
-        }
-        break;
-      }
-    }
-    else
-    {
-      length = a52_syncinfo(sh_audio->a_in_buffer, &sample_rate, &bit_rate);
-      if(length >= 7 && length <= 3840)
-      {
-        if(isdts != 0)
-        {
-          mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to AC3, %d bps, %d Hz\n", bit_rate, sample_rate);
-          isdts = 0;
-        }
-        break; /* we're done.*/
-      }
-    }
-    /* bad file => resync*/
-    memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer + 1, 11);
-    --sh_audio->a_in_buffer_len;
-  }
-  mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "ac3dts: %s len=%d  flags=0x%X  %d Hz %d bit/s\n", isdts == 1 ? "DTS" : isdts == 0 ? "AC3" : "unknown", length, flags, sample_rate, bit_rate);
-
-  sh_audio->samplerate = sample_rate;
-  sh_audio->i_bps = bit_rate / 8;
-  demux_read_data(sh_audio->ds, sh_audio->a_in_buffer + 12, length - 12);
-  sh_audio->a_in_buffer_len = length;
-
-  return length;
-}
-
-
-static int preinit(sh_audio_t *sh)
-{
-  /* Dolby AC3 audio: */
-  sh->audio_out_minsize = 128 * 32 * 2 * 2; // DTS seems to need more than AC3
-  sh->audio_in_minsize = 8192;
-  sh->channels = 2;
-  sh->samplesize = 2;
-  sh->sample_format = AF_FORMAT_AC3_BE;
-  // HACK for DTS where useless swapping can't easily be removed
-  if (sh->format == 0x2001)
-    sh->sample_format = AF_FORMAT_AC3_NE;
-  return 1;
-}
-
-static int init(sh_audio_t *sh_audio)
-{
-  /* Dolby AC3 passthrough:*/
-  if(ac3dts_fillbuff(sh_audio) < 0)
-  {
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "AC3/DTS sync failed\n");
-    return 0;
-  }
-  return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-}
-
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
-{
-  switch(cmd)
-  {
-  case ADCTRL_RESYNC_STREAM:
-  case ADCTRL_SKIP_FRAME:
-      ac3dts_fillbuff(sh);
-      return CONTROL_TRUE;
-  }
-  return CONTROL_UNKNOWN;
-}
-
-
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
-{
-  int len = sh_audio->a_in_buffer_len;
-
-  if(len <= 0)
-    if((len = ac3dts_fillbuff(sh_audio)) <= 0)
-      return len; /*EOF*/
-  sh_audio->a_in_buffer_len = 0;
-
-  if(isdts == 1)
-  {
-    return decode_audio_dts(sh_audio->a_in_buffer, len, buf);
-  }
-  else if(isdts == 0)
-  {
-    AV_WB16(buf,     0xF872);   // iec 61937 syncword 1
-    AV_WB16(buf + 2, 0x4E1F);   // iec 61937 syncword 2
-    buf[4] = sh_audio->a_in_buffer[5] & 0x7; // bsmod
-    buf[5] = 0x01;              // data-type ac3
-    AV_WB16(buf + 6, len << 3); // number of bits in payload
-    memcpy(buf + 8, sh_audio->a_in_buffer, len);
-    memset(buf + 8 + len, 0, 6144 - 8 - len);
-
-    return 6144;
-  }
-  else
-    return -1;
-}
-
-
-static const int DTS_SAMPLEFREQS[16] =
-{
-  0,
-  8000,
-  16000,
-  32000,
-  64000,
-  128000,
-  11025,
-  22050,
-  44100,
-  88200,
-  176400,
-  12000,
-  24000,
-  48000,
-  96000,
-  192000
-};
-
-static const int DTS_BITRATES[30] =
-{
-  32000,
-  56000,
-  64000,
-  96000,
-  112000,
-  128000,
-  192000,
-  224000,
-  256000,
-  320000,
-  384000,
-  448000,
-  512000,
-  576000,
-  640000,
-  768000,
-  896000,
-  1024000,
-  1152000,
-  1280000,
-  1344000,
-  1408000,
-  1411200,
-  1472000,
-  1536000,
-  1920000,
-  2048000,
-  3072000,
-  3840000,
-  4096000
-};
-
-static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq)
-{
-  int ftype;
-  int surp;
-  int unknown_bit av_unused;
-  int fsize;
-  int amode av_unused;
-
-  int word_mode;
-  int le_mode;
-
-  unsigned int first4bytes = indata_ptr[0] << 24 | indata_ptr[1] << 16
-                             | indata_ptr[2] << 8 | indata_ptr[3];
-
-  switch(first4bytes)
-  {
-    /* 14 bits LE */
-    case 0xff1f00e8:
-      /* Also make sure frame type is 1. */
-      if ((indata_ptr[4]&0xf0) != 0xf0 || indata_ptr[5] != 0x07)
-        return -1;
-      word_mode = 0;
-      le_mode = 1;
-      break;
-    /* 14 bits BE */
-    case 0x1fffe800:
-      /* Also make sure frame type is 1. */
-      if (indata_ptr[4] != 0x07 || (indata_ptr[5]&0xf0) != 0xf0)
-        return -1;
-      word_mode = 0;
-      le_mode = 0;
-      break;
-    /* 16 bits LE */
-    case 0xfe7f0180:
-      word_mode = 1;
-      le_mode = 1;
-      break;
-    /* 16 bits BE */
-    case 0x7ffe8001:
-      word_mode = 1;
-      le_mode = 0;
-      break;
-    default:
-      return -1;
-  }
-
-  if(word_mode)
-  {
-    /* First bit after first 32 bits:
-       Frame type ( 1: Normal frame; 0: Termination frame ) */
-    ftype = indata_ptr[4+le_mode] >> 7;
-
-  if(ftype != 1)
-  {
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Termination frames not handled, REPORT BUG\n");
-    return -1;
-  }
-    /* Next 5 bits: Surplus Sample Count V SURP 5 bits */
-    surp = indata_ptr[4+le_mode] >> 2 & 0x1f;
-    /* Number of surplus samples */
-    surp = (surp + 1) % 32;
-
-    /* One unknown bit, crc? */
-    unknown_bit = indata_ptr[4+le_mode] >> 1 & 0x01;
-
-    /* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
-    *nblks = (indata_ptr[4+le_mode] & 0x01) << 6 | indata_ptr[5-le_mode] >> 2;
-    /* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
-       encoded in the current frame per channel. */
-    ++(*nblks);
-
-    /* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
-       (ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
-       FSIZE defines the byte size of the current audio frame. */
-    fsize = (indata_ptr[5-le_mode] & 0x03) << 12 | indata_ptr[6+le_mode] << 4
-            | indata_ptr[7-le_mode] >> 4;
-    ++fsize;
-
-    /* Audio Channel Arrangement ACC AMODE 6 bits */
-    amode = (indata_ptr[7-le_mode] & 0x0f) << 2 | indata_ptr[8+le_mode] >> 6;
-
-    /* Source Sampling rate ACC SFREQ 4 bits */
-    *sfreq = indata_ptr[8+le_mode] >> 2 & 0x0f;
-    /* Transmission Bit Rate ACC RATE 5 bits */
-    *rate = (indata_ptr[8+le_mode] & 0x03) << 3
-            | (indata_ptr[9-le_mode] >> 5 & 0x07);
-  }
-  else
-  {
-    /* in the case judgement, we assure this */
-    ftype = 1;
-    surp = 0;
-    /* 14 bits support, every 2 bytes, & 0x3fff, got used 14 bits */
-    /* Bits usage:
-       32 bits: Sync code (28 + 4)      1th and 2th word, 4 bits in 3th word
-       1  bits: Frame type              1 bits in 3th word
-       5  bits: SURP                    5 bits in 3th word
-       1  bits: crc?                    1 bits in 3th word
-       7  bits: NBLKS                   3 bits in 3th word, 4 bits in 4th word
-       14 bits: FSIZE                   10 bits in 4th word, 4 bits in 5th word
-                                        in 14 bits mode, FSIZE = FSIZE*8/14*2
-       6  bits: AMODE                   6 bits in 5th word
-       4  bits: SFREQ                   4 bits in 5th word
-       5  bits: RATE                    5 bits in 6th word
-       total bits: 75 bits    */
-
-    /* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
-    *nblks = (indata_ptr[5-le_mode] & 0x07) << 4
-             | (indata_ptr[6+le_mode] & 0x3f) >> 2;
-    /* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
-       encoded in the current frame per channel. */
-    ++(*nblks);
-
-    /* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
-       (ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
-       FSIZE defines the byte size of the current audio frame. */
-    fsize = (indata_ptr[6+le_mode] & 0x03) << 12 | indata_ptr[7-le_mode] << 4
-            | (indata_ptr[8+le_mode] & 0x3f) >> 2;
-    ++fsize;
-    fsize = fsize * 8 / 14 * 2;
-
-    /* Audio Channel Arrangement ACC AMODE 6 bits */
-    amode = (indata_ptr[8+le_mode] & 0x03) << 4
-            | (indata_ptr[9-le_mode] & 0xf0) >> 4;
-
-    /* Source Sampling rate ACC SFREQ 4 bits */
-    *sfreq = indata_ptr[9-le_mode] & 0x0f;
-    /* Transmission Bit Rate ACC RATE 5 bits */
-    *rate = (indata_ptr[10+le_mode] & 0x3f) >> 1;
-  }
-#if 0
-  if(*sfreq != 13)
-  {
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Only 48kHz supported, REPORT BUG\n");
-    return -1;
-  }
-#endif
-  if((fsize > 8192) || (fsize < 96))
-  {
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: fsize: %d invalid, REPORT BUG\n", fsize);
-    return -1;
-  }
-
-  if(*nblks != 8 &&
-    *nblks != 16 &&
-    *nblks != 32 &&
-    *nblks != 64 &&
-    *nblks != 128 &&
-    ftype == 1)
-  {
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: nblks %d not valid for normal frame, REPORT BUG\n", *nblks);
-    return -1;
-  }
-
-  return fsize;
-}
-
-static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate)
-{
-  int nblks;
-  int fsize;
-  int rate;
-  int sfreq;
-
-  fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
-  if(fsize >= 0)
-  {
-    if(rate >= 0 && rate <= 29)
-      *bit_rate = DTS_BITRATES[rate];
-    else
-      *bit_rate = 0;
-    if(sfreq >= 1 && sfreq <= 15)
-      *sample_rate = DTS_SAMPLEFREQS[sfreq];
-    else
-      *sample_rate = 0;
-  }
-  return fsize;
-}
-
-static int convert_14bits_to_16bits(const unsigned char *src,
-                                    unsigned char *dest,
-                                    int len,
-                                    int is_le)
-{
-  uint16_t *p = (uint16_t *)dest;
-  uint16_t buf = 0;
-  int spacebits = 16;
-  if (len <= 0) return 0;
-  while (len > 0) {
-    uint16_t v;
-    if (len == 1)
-      v = is_le ? src[0] : src[0] << 8;
-    else
-      v = is_le ? src[1] << 8 | src[0] : src[0] << 8 | src[1];
-    v <<= 2;
-    src += 2;
-    len -= 2;
-    buf |= v >> (16 - spacebits);
-    spacebits -= 14;
-    if (spacebits < 0) {
-      *p++ = buf;
-      spacebits += 16;
-      buf = v << (spacebits - 2);
-    }
-  }
-  *p++ = buf;
-  return (unsigned char *)p - dest;
-}
-
-static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf)
-{
-  int nblks;
-  int fsize;
-  int rate;
-  int sfreq;
-  int nr_samples;
-  int convert_16bits = 0;
-  uint16_t *buf16 = (uint16_t *)buf;
-
-  fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
-  if(fsize < 0)
-    return -1;
-  nr_samples = nblks * 32;
-
-  buf16[0] = 0xf872; /* iec 61937     */
-  buf16[1] = 0x4e1f; /*  syncword     */
-  switch(nr_samples)
-  {
-  case 512:
-    buf16[2] = 0x000b;      /* DTS-1 (512-sample bursts) */
-    break;
-  case 1024:
-    buf16[2] = 0x000c;      /* DTS-2 (1024-sample bursts) */
-    break;
-  case 2048:
-    buf16[2] = 0x000d;      /* DTS-3 (2048-sample bursts) */
-    break;
-  default:
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: %d-sample bursts not supported\n", nr_samples);
-    buf16[2] = 0x0000;
-    break;
-  }
-
-  if(fsize + 8 > nr_samples * 2 * 2)
-  {
-    // dts wav (14bits LE) match this condition, one way to passthrough
-    // is not add iec 61937 header, decoders will notice the dts header
-    // and identify the dts stream. Another way here is convert
-    // the stream from 14 bits to 16 bits.
-    if ((indata_ptr[0] == 0xff || indata_ptr[0] == 0x1f)
-        && fsize * 14 / 16 + 8 <= nr_samples * 2 * 2) {
-      // The input stream is 14 bits, we can shrink it to 16 bits
-      // to save space for add the 61937 header
-      fsize = convert_14bits_to_16bits(indata_ptr,
-                                       &buf[8],
-                                       fsize,
-                                       indata_ptr[0] == 0xff /* is LE */
-                                       );
-      mp_msg(MSGT_DECAUDIO, MSGL_DBG3, "DTS: shrink 14 bits stream to "
-             "16 bits %02x%02x%02x%02x => %02x%02x%02x%02x, new size %d.\n",
-             indata_ptr[0], indata_ptr[1], indata_ptr[2], indata_ptr[3],
-             buf[8], buf[9], buf[10], buf[11], fsize);
-      convert_16bits = 1;
-    }
-    else
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: more data than fits\n");
-  }
-
-  buf16[3] = fsize << 3;
-
-  if (!convert_16bits) {
-#ifdef BIG_ENDIAN
-  /* BE stream */
-  if (indata_ptr[0] == 0x1f || indata_ptr[0] == 0x7f)
-#else
-  /* LE stream */
-  if (indata_ptr[0] == 0xff || indata_ptr[0] == 0xfe)
-#endif
-  memcpy(&buf[8], indata_ptr, fsize);
-  else
-  {
-  swab(indata_ptr, &buf[8], fsize);
-  if (fsize & 1) {
-    buf[8+fsize-1] = 0;
-    buf[8+fsize] = indata_ptr[fsize-1];
-    fsize++;
-  }
-  }
-  }
-  memset(&buf[fsize + 8], 0, nr_samples * 2 * 2 - (fsize + 8));
-
-  return nr_samples * 2 * 2;
-}