Mercurial > mplayer.hg
view libmpcodecs/ad_msadpcm.c @ 18402:b3c5e79aeab8
2nd part of review by Pierre Lombard + some more fixes by me.
Hopefully now this file doesn't like it was babelfished ;)
author | gpoirier |
---|---|
date | Mon, 08 May 2006 14:14:48 +0000 |
parents | 9d0b052c4f74 |
children | 1767c271d710 |
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/* MS ADPCM Decoder for MPlayer by Mike Melanson This file is responsible for decoding Microsoft ADPCM data. Details about the data format can be found here: http://www.pcisys.net/~melanson/codecs/ */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "bswap.h" #include "ad_internal.h" static ad_info_t info = { "MS ADPCM audio decoder", "msadpcm", "Nick Kurshev", "Mike Melanson", "" }; LIBAD_EXTERN(msadpcm) static int ms_adapt_table[] = { 230, 230, 230, 230, 307, 409, 512, 614, 768, 614, 512, 409, 307, 230, 230, 230 }; static int ms_adapt_coeff1[] = { 256, 512, 0, 192, 240, 460, 392 }; static int ms_adapt_coeff2[] = { 0, -256, 0, 64, 0, -208, -232 }; #define MS_ADPCM_PREAMBLE_SIZE 6 #define LE_16(x) ((x)[0]+(256*((x)[1]))) //#define LE_16(x) (le2me_16((x)[1]+(256*((x)[0])))) //#define LE_16(x) (le2me_16(*(unsigned short *)(x))) //#define LE_32(x) (le2me_32(*(unsigned int *)(x))) // useful macros // clamp a number between 0 and 88 #define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88; // clamp a number within a signed 16-bit range #define CLAMP_S16(x) if (x < -32768) x = -32768; \ else if (x > 32767) x = 32767; // clamp a number above 16 #define CLAMP_ABOVE_16(x) if (x < 16) x = 16; // sign extend a 16-bit value #define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; // 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 int ms_adpcm_decode_block(unsigned short *output, unsigned char *input, int channels, int block_size) { int current_channel = 0; 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; // fetch the header information, in stereo if both channels are present if (input[stream_ptr] > 6) mp_msg(MSGT_DECAUDIO, MSGL_WARN, "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n", input[stream_ptr]); coeff1[0] = ms_adapt_coeff1[input[stream_ptr]]; coeff2[0] = ms_adapt_coeff2[input[stream_ptr]]; stream_ptr++; if (channels == 2) { if (input[stream_ptr] > 6) mp_msg(MSGT_DECAUDIO, MSGL_WARN, "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n", input[stream_ptr]); coeff1[1] = ms_adapt_coeff1[input[stream_ptr]]; coeff2[1] = ms_adapt_coeff2[input[stream_ptr]]; stream_ptr++; } idelta[0] = LE_16(&input[stream_ptr]); stream_ptr += 2; SE_16BIT(idelta[0]); if (channels == 2) { idelta[1] = LE_16(&input[stream_ptr]); stream_ptr += 2; SE_16BIT(idelta[1]); } sample1[0] = LE_16(&input[stream_ptr]); stream_ptr += 2; SE_16BIT(sample1[0]); if (channels == 2) { sample1[1] = LE_16(&input[stream_ptr]); stream_ptr += 2; SE_16BIT(sample1[1]); } sample2[0] = LE_16(&input[stream_ptr]); stream_ptr += 2; SE_16BIT(sample2[0]); if (channels == 2) { sample2[1] = LE_16(&input[stream_ptr]); stream_ptr += 2; SE_16BIT(sample2[1]); } 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); predictor = ( ((sample1[current_channel] * coeff1[current_channel]) + (sample2[current_channel] * coeff2[current_channel])) / 256) + (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) { 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 */ return 2 * ms_adpcm_decode_block( (unsigned short*)buf, sh_audio->a_in_buffer, sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign); }