Mercurial > mplayer.hg
view libmpcodecs/ad_msadpcm.c @ 30431:5fef3206196f
Document that -af-adv force=0 currently does the same as =1
author | reimar |
---|---|
date | Sun, 31 Jan 2010 01:42:11 +0000 |
parents | bbb6ebec87a0 |
children | cc27da5d7286 |
line wrap: on
line source
/* * 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 "config.h" #include "libavutil/common.h" #include "libavutil/intreadwrite.h" #include "mpbswap.h" #include "ad_internal.h" static 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; }