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-rw-r--r--libmpcodecs/ad_msadpcm.c237
1 files changed, 0 insertions, 237 deletions
diff --git a/libmpcodecs/ad_msadpcm.c b/libmpcodecs/ad_msadpcm.c
deleted file mode 100644
index eee89de2ea..0000000000
--- a/libmpcodecs/ad_msadpcm.c
+++ /dev/null
@@ -1,237 +0,0 @@
-/*
- * MS ADPCM decoder
- *
- * This file is responsible for decoding Microsoft ADPCM data.
- * Details about the data format can be found here:
- * http://www.pcisys.net/~melanson/codecs/
- *
- * Copyright (c) 2002 Mike Melanson
- *
- * 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 <libavutil/common.h>
-#include <libavutil/intreadwrite.h>
-
-#include "config.h"
-#include "mpbswap.h"
-#include "ad_internal.h"
-
-static const ad_info_t info =
-{
- "MS ADPCM audio decoder",
- "msadpcm",
- "Nick Kurshev",
- "Mike Melanson",
- ""
-};
-
-LIBAD_EXTERN(msadpcm)
-
-static const int ms_adapt_table[] =
-{
- 230, 230, 230, 230, 307, 409, 512, 614,
- 768, 614, 512, 409, 307, 230, 230, 230
-};
-
-static const uint8_t ms_adapt_coeff1[] =
-{
- 64, 128, 0, 48, 60, 115, 98
-};
-
-static const int8_t ms_adapt_coeff2[] =
-{
- 0, -64, 0, 16, 0, -52, -58
-};
-
-#define MS_ADPCM_PREAMBLE_SIZE 6
-
-#define LE_16(x) ((int16_t)AV_RL16(x))
-
-// clamp a number between 0 and 88
-#define CLAMP_0_TO_88(x) x = av_clip(x, 0, 88);
-// clamp a number within a signed 16-bit range
-#define CLAMP_S16(x) x = av_clip_int16(x);
-// clamp a number above 16
-#define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
-// sign extend a 4-bit value
-#define SE_4BIT(x) if (x & 0x8) x -= 0x10;
-
-static int preinit(sh_audio_t *sh_audio)
-{
- sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
- sh_audio->ds->ss_div =
- (sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
- sh_audio->audio_in_minsize =
- sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
- return 1;
-}
-
-static int init(sh_audio_t *sh_audio)
-{
- sh_audio->channels=sh_audio->wf->nChannels;
- sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
- sh_audio->i_bps = sh_audio->wf->nBlockAlign *
- (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
- sh_audio->samplesize=2;
-
- return 1;
-}
-
-static void uninit(sh_audio_t *sh_audio)
-{
-}
-
-static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
-{
- if(cmd==ADCTRL_SKIP_FRAME){
- demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
- return CONTROL_TRUE;
- }
- return CONTROL_UNKNOWN;
-}
-
-static inline int check_coeff(uint8_t c) {
- if (c > 6) {
- mp_msg(MSGT_DECAUDIO, MSGL_WARN,
- "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
- c);
- c = 6;
- }
- return c;
-}
-
-static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
- int channels, int block_size)
-{
- int current_channel = 0;
- int coeff_idx;
- int idelta[2];
- int sample1[2];
- int sample2[2];
- int coeff1[2];
- int coeff2[2];
- int stream_ptr = 0;
- int out_ptr = 0;
- int upper_nibble = 1;
- int nibble;
- int snibble; // signed nibble
- int predictor;
-
- if (channels != 1) channels = 2;
- if (block_size < 7 * channels)
- return -1;
-
- // fetch the header information, in stereo if both channels are present
- coeff_idx = check_coeff(input[stream_ptr]);
- coeff1[0] = ms_adapt_coeff1[coeff_idx];
- coeff2[0] = ms_adapt_coeff2[coeff_idx];
- stream_ptr++;
- if (channels == 2)
- {
- coeff_idx = check_coeff(input[stream_ptr]);
- coeff1[1] = ms_adapt_coeff1[coeff_idx];
- coeff2[1] = ms_adapt_coeff2[coeff_idx];
- stream_ptr++;
- }
-
- idelta[0] = LE_16(&input[stream_ptr]);
- stream_ptr += 2;
- if (channels == 2)
- {
- idelta[1] = LE_16(&input[stream_ptr]);
- stream_ptr += 2;
- }
-
- sample1[0] = LE_16(&input[stream_ptr]);
- stream_ptr += 2;
- if (channels == 2)
- {
- sample1[1] = LE_16(&input[stream_ptr]);
- stream_ptr += 2;
- }
-
- sample2[0] = LE_16(&input[stream_ptr]);
- stream_ptr += 2;
- if (channels == 2)
- {
- sample2[1] = LE_16(&input[stream_ptr]);
- stream_ptr += 2;
- }
-
- if (channels == 1)
- {
- output[out_ptr++] = sample2[0];
- output[out_ptr++] = sample1[0];
- } else {
- output[out_ptr++] = sample2[0];
- output[out_ptr++] = sample2[1];
- output[out_ptr++] = sample1[0];
- output[out_ptr++] = sample1[1];
- }
-
- while (stream_ptr < block_size)
- {
- // get the next nibble
- if (upper_nibble)
- nibble = snibble = input[stream_ptr] >> 4;
- else
- nibble = snibble = input[stream_ptr++] & 0x0F;
- upper_nibble ^= 1;
- SE_4BIT(snibble);
-
- // should this really be a division and not a shift?
- // coefficients were originally scaled by for, which might have
- // been an optimization for 8-bit CPUs _if_ a shift is correct
- predictor = (
- ((sample1[current_channel] * coeff1[current_channel]) +
- (sample2[current_channel] * coeff2[current_channel])) / 64) +
- (snibble * idelta[current_channel]);
- CLAMP_S16(predictor);
- sample2[current_channel] = sample1[current_channel];
- sample1[current_channel] = predictor;
- output[out_ptr++] = predictor;
-
- // compute the next adaptive scale factor (a.k.a. the variable idelta)
- idelta[current_channel] =
- (ms_adapt_table[nibble] * idelta[current_channel]) / 256;
- CLAMP_ABOVE_16(idelta[current_channel]);
-
- // toggle the channel
- current_channel ^= channels - 1;
- }
-
- return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
-}
-
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
-{
- int res;
- if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
- sh_audio->ds->ss_mul) !=
- sh_audio->ds->ss_mul)
- return -1; /* EOF */
-
- res = ms_adpcm_decode_block(
- (unsigned short*)buf, sh_audio->a_in_buffer,
- sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
- return res < 0 ? res : 2 * res;
-}