/***************************************************************************** * adpcm.c : adpcm variant audio decoder ***************************************************************************** * Copyright (C) 2001, 2002 VLC authors and VideoLAN * $Id$ * * Authors: Laurent Aimar * Rémi Denis-Courmont * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU Lesser General Public License as published by * the Free Software Foundation; either version 2.1 of the License, or * (at your option) any later version. * * This program 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 Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public License * along with this program; if not, write to the Free Software Foundation, * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble * * Documentation: http://www.pcisys.net/~melanson/codecs/adpcm.txt *****************************************************************************/ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include /***************************************************************************** * Module descriptor *****************************************************************************/ static int OpenDecoder( vlc_object_t * ); static void CloseDecoder( vlc_object_t * ); static int DecodeAudio( decoder_t *, block_t * ); static void Flush( decoder_t * ); vlc_module_begin () set_description( N_("ADPCM audio decoder") ) set_capability( "audio decoder", 50 ) set_category( CAT_INPUT ) set_subcategory( SUBCAT_INPUT_ACODEC ) set_callbacks( OpenDecoder, CloseDecoder ) vlc_module_end () /***************************************************************************** * Local prototypes *****************************************************************************/ enum adpcm_codec_e { ADPCM_IMA_QT, ADPCM_IMA_WAV, ADPCM_MS, ADPCM_DK3, ADPCM_DK4, ADPCM_EA }; struct decoder_sys_t { enum adpcm_codec_e codec; size_t i_block; size_t i_samplesperblock; date_t end_date; int16_t *prev; }; static void DecodeAdpcmMs ( decoder_t *, int16_t *, uint8_t * ); static void DecodeAdpcmImaWav( decoder_t *, int16_t *, uint8_t * ); static void DecodeAdpcmImaQT ( decoder_t *, int16_t *, uint8_t * ); static void DecodeAdpcmDk4 ( decoder_t *, int16_t *, uint8_t * ); static void DecodeAdpcmDk3 ( decoder_t *, int16_t *, uint8_t * ); static void DecodeAdpcmEA ( decoder_t *, int16_t *, uint8_t * ); static const int pi_channels_maps[6] = { 0, AOUT_CHAN_CENTER, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARLEFT, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARLEFT }; /* Various table from http://www.pcisys.net/~melanson/codecs/adpcm.txt */ static const int i_index_table[16] = { -1, -1, -1, -1, 2, 4, 6, 8, -1, -1, -1, -1, 2, 4, 6, 8 }; static const int i_step_table[89] = { 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 }; static const int i_adaptation_table[16] = { 230, 230, 230, 230, 307, 409, 512, 614, 768, 614, 512, 409, 307, 230, 230, 230 }; static const int i_adaptation_coeff1[7] = { 256, 512, 0, 192, 240, 460, 392 }; static const int i_adaptation_coeff2[7] = { 0, -256, 0, 64, 0, -208, -232 }; /***************************************************************************** * OpenDecoder: probe the decoder and return score *****************************************************************************/ static int OpenDecoder( vlc_object_t *p_this ) { decoder_t *p_dec = (decoder_t*)p_this; decoder_sys_t *p_sys; switch( p_dec->fmt_in.i_codec ) { case VLC_CODEC_ADPCM_IMA_QT: case VLC_CODEC_ADPCM_IMA_WAV: case VLC_CODEC_ADPCM_MS: case VLC_CODEC_ADPCM_DK4: case VLC_CODEC_ADPCM_DK3: case VLC_CODEC_ADPCM_XA_EA: break; default: return VLC_EGENERIC; } if( p_dec->fmt_in.audio.i_rate <= 0 ) { msg_Err( p_dec, "bad samplerate" ); return VLC_EGENERIC; } /* Allocate the memory needed to store the decoder's structure */ p_sys = malloc(sizeof(*p_sys)); if( unlikely(p_sys == NULL) ) return VLC_ENOMEM; p_sys->prev = NULL; p_sys->i_samplesperblock = 0; unsigned i_channels = p_dec->fmt_in.audio.i_channels; uint8_t i_max_channels = 5; switch( p_dec->fmt_in.i_codec ) { case VLC_CODEC_ADPCM_IMA_QT: /* IMA ADPCM */ p_sys->codec = ADPCM_IMA_QT; i_max_channels = 2; break; case VLC_CODEC_ADPCM_IMA_WAV: /* IMA ADPCM */ p_sys->codec = ADPCM_IMA_WAV; i_max_channels = 2; break; case VLC_CODEC_ADPCM_MS: /* MS ADPCM */ p_sys->codec = ADPCM_MS; i_max_channels = 2; break; case VLC_CODEC_ADPCM_DK4: /* Duck DK4 ADPCM */ p_sys->codec = ADPCM_DK4; i_max_channels = 2; break; case VLC_CODEC_ADPCM_DK3: /* Duck DK3 ADPCM */ p_sys->codec = ADPCM_DK3; i_max_channels = 2; break; case VLC_CODEC_ADPCM_XA_EA: /* EA ADPCM */ p_sys->codec = ADPCM_EA; p_sys->prev = calloc( 2 * p_dec->fmt_in.audio.i_channels, sizeof( int16_t ) ); if( unlikely(p_sys->prev == NULL) ) { free( p_sys ); return VLC_ENOMEM; } break; } if (i_channels > i_max_channels || i_channels == 0) { free(p_sys->prev); free(p_sys); msg_Err( p_dec, "Invalid number of channels %i", p_dec->fmt_in.audio.i_channels ); return VLC_EGENERIC; } if( p_dec->fmt_in.audio.i_blockalign <= 0 ) { p_sys->i_block = (p_sys->codec == ADPCM_IMA_QT) ? 34 * p_dec->fmt_in.audio.i_channels : 1024; msg_Warn( p_dec, "block size undefined, using %zu", p_sys->i_block ); } else { p_sys->i_block = p_dec->fmt_in.audio.i_blockalign; } /* calculate samples per block */ switch( p_sys->codec ) { case ADPCM_IMA_QT: p_sys->i_samplesperblock = 64; break; case ADPCM_IMA_WAV: if( p_sys->i_block >= 4 * i_channels ) { p_sys->i_samplesperblock = 2 * ( p_sys->i_block - 4 * i_channels ) / i_channels; } break; case ADPCM_MS: if( p_sys->i_block >= 7 * i_channels ) { p_sys->i_samplesperblock = 2 * (p_sys->i_block - 7 * i_channels) / i_channels + 2; } break; case ADPCM_DK4: if( p_sys->i_block >= 4 * i_channels ) { p_sys->i_samplesperblock = 2 * (p_sys->i_block - 4 * i_channels) / i_channels + 1; } break; case ADPCM_DK3: i_channels = 2; if( p_sys->i_block >= 16 ) p_sys->i_samplesperblock = ( 4 * ( p_sys->i_block - 16 ) + 2 )/ 3; break; case ADPCM_EA: if( p_sys->i_block >= i_channels ) { p_sys->i_samplesperblock = 2 * (p_sys->i_block - i_channels) / i_channels; } } msg_Dbg( p_dec, "format: samplerate:%d Hz channels:%d bits/sample:%d " "blockalign:%zu samplesperblock:%zu", p_dec->fmt_in.audio.i_rate, i_channels, p_dec->fmt_in.audio.i_bitspersample, p_sys->i_block, p_sys->i_samplesperblock ); if (p_sys->i_samplesperblock == 0) { free(p_sys->prev); free(p_sys); msg_Err( p_dec, "Error computing number of samples per block"); return VLC_EGENERIC; } p_dec->p_sys = p_sys; p_dec->fmt_out.i_codec = VLC_CODEC_S16N; p_dec->fmt_out.audio.i_rate = p_dec->fmt_in.audio.i_rate; p_dec->fmt_out.audio.i_channels = i_channels; p_dec->fmt_out.audio.i_physical_channels = pi_channels_maps[i_channels]; date_Init( &p_sys->end_date, p_dec->fmt_out.audio.i_rate, 1 ); date_Set( &p_sys->end_date, 0 ); p_dec->pf_decode = DecodeAudio; p_dec->pf_flush = Flush; return VLC_SUCCESS; } /***************************************************************************** * Flush: *****************************************************************************/ static void Flush( decoder_t *p_dec ) { decoder_sys_t *p_sys = p_dec->p_sys; date_Set( &p_sys->end_date, 0 ); } /***************************************************************************** * DecodeBlock: *****************************************************************************/ static block_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block ) { decoder_sys_t *p_sys = p_dec->p_sys; block_t *p_block; if( !*pp_block ) return NULL; p_block = *pp_block; if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) ) { Flush( p_dec ); if( p_block->i_flags & BLOCK_FLAG_CORRUPTED ) goto drop; } if( p_block->i_pts > VLC_TICK_INVALID && p_block->i_pts != date_Get( &p_sys->end_date ) ) { date_Set( &p_sys->end_date, p_block->i_pts ); } else if( !date_Get( &p_sys->end_date ) ) /* We've just started the stream, wait for the first PTS. */ goto drop; /* Don't re-use the same pts twice */ p_block->i_pts = VLC_TICK_INVALID; if( p_block->i_buffer >= p_sys->i_block ) { block_t *p_out; if( decoder_UpdateAudioFormat( p_dec ) ) goto drop; p_out = decoder_NewAudioBuffer( p_dec, p_sys->i_samplesperblock ); if( p_out == NULL ) goto drop; p_out->i_pts = date_Get( &p_sys->end_date ); p_out->i_length = date_Increment( &p_sys->end_date, p_sys->i_samplesperblock ) - p_out->i_pts; switch( p_sys->codec ) { case ADPCM_IMA_QT: DecodeAdpcmImaQT( p_dec, (int16_t*)p_out->p_buffer, p_block->p_buffer ); break; case ADPCM_IMA_WAV: DecodeAdpcmImaWav( p_dec, (int16_t*)p_out->p_buffer, p_block->p_buffer ); break; case ADPCM_MS: DecodeAdpcmMs( p_dec, (int16_t*)p_out->p_buffer, p_block->p_buffer ); break; case ADPCM_DK4: DecodeAdpcmDk4( p_dec, (int16_t*)p_out->p_buffer, p_block->p_buffer ); break; case ADPCM_DK3: DecodeAdpcmDk3( p_dec, (int16_t*)p_out->p_buffer, p_block->p_buffer ); break; case ADPCM_EA: DecodeAdpcmEA( p_dec, (int16_t*)p_out->p_buffer, p_block->p_buffer ); default: break; } p_block->p_buffer += p_sys->i_block; p_block->i_buffer -= p_sys->i_block; return p_out; } drop: block_Release( p_block ); *pp_block = NULL; return NULL; } static int DecodeAudio( decoder_t *p_dec, block_t *p_block ) { if( p_block == NULL ) /* No Drain */ return VLCDEC_SUCCESS; block_t **pp_block = &p_block, *p_out; while( ( p_out = DecodeBlock( p_dec, pp_block ) ) != NULL ) decoder_QueueAudio( p_dec, p_out ); return VLCDEC_SUCCESS; } /***************************************************************************** * CloseDecoder: *****************************************************************************/ static void CloseDecoder( vlc_object_t *p_this ) { decoder_t *p_dec = (decoder_t *)p_this; decoder_sys_t *p_sys = p_dec->p_sys; free( p_sys->prev ); free( p_sys ); } /***************************************************************************** * Local functions *****************************************************************************/ #define CLAMP( v, min, max ) \ if( (v) < (min) ) (v) = (min); \ if( (v) > (max) ) (v) = (max) #define GetByte( v ) \ (v) = *p_buffer; p_buffer++; #define GetWord( v ) \ (v) = *p_buffer; p_buffer++; \ (v) |= ( *p_buffer ) << 8; p_buffer++; \ if( (v)&0x8000 ) (v) -= 0x010000; /* * MS */ typedef struct adpcm_ms_channel_s { int i_idelta; int i_sample1, i_sample2; int i_coeff1, i_coeff2; } adpcm_ms_channel_t; static int AdpcmMsExpandNibble(adpcm_ms_channel_t *p_channel, int i_nibble ) { int i_predictor; int i_snibble; /* expand sign */ i_snibble = i_nibble - ( i_nibble&0x08 ? 0x10 : 0 ); i_predictor = ( p_channel->i_sample1 * p_channel->i_coeff1 + p_channel->i_sample2 * p_channel->i_coeff2 ) / 256 + i_snibble * p_channel->i_idelta; CLAMP( i_predictor, -32768, 32767 ); p_channel->i_sample2 = p_channel->i_sample1; p_channel->i_sample1 = i_predictor; p_channel->i_idelta = ( i_adaptation_table[i_nibble] * p_channel->i_idelta ) / 256; if( p_channel->i_idelta < 16 ) { p_channel->i_idelta = 16; } return( i_predictor ); } static void DecodeAdpcmMs( decoder_t *p_dec, int16_t *p_sample, uint8_t *p_buffer ) { decoder_sys_t *p_sys = p_dec->p_sys; adpcm_ms_channel_t channel[2]; int b_stereo; int i_block_predictor; size_t i_total_samples = p_sys->i_samplesperblock; if(i_total_samples < 2) return; b_stereo = p_dec->fmt_out.audio.i_channels == 2 ? 1 : 0; GetByte( i_block_predictor ); CLAMP( i_block_predictor, 0, 6 ); channel[0].i_coeff1 = i_adaptation_coeff1[i_block_predictor]; channel[0].i_coeff2 = i_adaptation_coeff2[i_block_predictor]; if( b_stereo ) { GetByte( i_block_predictor ); CLAMP( i_block_predictor, 0, 6 ); channel[1].i_coeff1 = i_adaptation_coeff1[i_block_predictor]; channel[1].i_coeff2 = i_adaptation_coeff2[i_block_predictor]; } GetWord( channel[0].i_idelta ); if( b_stereo ) { GetWord( channel[1].i_idelta ); } GetWord( channel[0].i_sample1 ); if( b_stereo ) { GetWord( channel[1].i_sample1 ); } GetWord( channel[0].i_sample2 ); if( b_stereo ) { GetWord( channel[1].i_sample2 ); } if( b_stereo ) { *p_sample++ = channel[0].i_sample2; *p_sample++ = channel[1].i_sample2; *p_sample++ = channel[0].i_sample1; *p_sample++ = channel[1].i_sample1; } else { *p_sample++ = channel[0].i_sample2; *p_sample++ = channel[0].i_sample1; } for( i_total_samples -= 2; i_total_samples >= 2; i_total_samples -= 2, p_buffer++ ) { *p_sample++ = AdpcmMsExpandNibble( &channel[0], (*p_buffer) >> 4); *p_sample++ = AdpcmMsExpandNibble( &channel[b_stereo ? 1 : 0], (*p_buffer)&0x0f); } } /* * IMA-WAV */ typedef struct adpcm_ima_wav_channel_s { int i_predictor; int i_step_index; } adpcm_ima_wav_channel_t; static int AdpcmImaWavExpandNibble(adpcm_ima_wav_channel_t *p_channel, int i_nibble ) { int i_diff; i_diff = i_step_table[p_channel->i_step_index] >> 3; if( i_nibble&0x04 ) i_diff += i_step_table[p_channel->i_step_index]; if( i_nibble&0x02 ) i_diff += i_step_table[p_channel->i_step_index]>>1; if( i_nibble&0x01 ) i_diff += i_step_table[p_channel->i_step_index]>>2; if( i_nibble&0x08 ) p_channel->i_predictor -= i_diff; else p_channel->i_predictor += i_diff; CLAMP( p_channel->i_predictor, -32768, 32767 ); p_channel->i_step_index += i_index_table[i_nibble]; CLAMP( p_channel->i_step_index, 0, 88 ); return( p_channel->i_predictor ); } static void DecodeAdpcmImaWav( decoder_t *p_dec, int16_t *p_sample, uint8_t *p_buffer ) { decoder_sys_t *p_sys = p_dec->p_sys; adpcm_ima_wav_channel_t channel[2]; int i_nibbles; int b_stereo; b_stereo = p_dec->fmt_out.audio.i_channels == 2 ? 1 : 0; GetWord( channel[0].i_predictor ); GetByte( channel[0].i_step_index ); CLAMP( channel[0].i_step_index, 0, 88 ); p_buffer++; if( b_stereo ) { GetWord( channel[1].i_predictor ); GetByte( channel[1].i_step_index ); CLAMP( channel[1].i_step_index, 0, 88 ); p_buffer++; } if( b_stereo ) { for( i_nibbles = 2 * (p_sys->i_block - 8); i_nibbles > 0; i_nibbles -= 16 ) { int i; for( i = 0; i < 4; i++ ) { p_sample[i * 4] = AdpcmImaWavExpandNibble(&channel[0],p_buffer[i]&0x0f); p_sample[i * 4 + 2] = AdpcmImaWavExpandNibble(&channel[0],p_buffer[i] >> 4); } p_buffer += 4; for( i = 0; i < 4; i++ ) { p_sample[i * 4 + 1] = AdpcmImaWavExpandNibble(&channel[1],p_buffer[i]&0x0f); p_sample[i * 4 + 3] = AdpcmImaWavExpandNibble(&channel[1],p_buffer[i] >> 4); } p_buffer += 4; p_sample += 16; } } else { for( i_nibbles = 2 * (p_sys->i_block - 4); i_nibbles > 0; i_nibbles -= 2, p_buffer++ ) { *p_sample++ =AdpcmImaWavExpandNibble( &channel[0], (*p_buffer)&0x0f ); *p_sample++ =AdpcmImaWavExpandNibble( &channel[0], (*p_buffer) >> 4 ); } } } /* * Ima4 in QT file */ static void DecodeAdpcmImaQT( decoder_t *p_dec, int16_t *p_sample, uint8_t *p_buffer ) { adpcm_ima_wav_channel_t channel[2]; int i_nibbles; int i_ch; int i_step; i_step = p_dec->fmt_out.audio.i_channels; for( i_ch = 0; i_ch < p_dec->fmt_out.audio.i_channels; i_ch++ ) { /* load preamble */ channel[i_ch].i_predictor = (int16_t)((( ( p_buffer[0] << 1 )|( p_buffer[1] >> 7 ) ))<<7); channel[i_ch].i_step_index = p_buffer[1]&0x7f; CLAMP( channel[i_ch].i_step_index, 0, 88 ); p_buffer += 2; for( i_nibbles = 0; i_nibbles < 64; i_nibbles +=2 ) { *p_sample = AdpcmImaWavExpandNibble( &channel[i_ch], (*p_buffer)&0x0f); p_sample += i_step; *p_sample = AdpcmImaWavExpandNibble( &channel[i_ch], (*p_buffer >> 4)&0x0f); p_sample += i_step; p_buffer++; } /* Next channel */ p_sample += 1 - 64 * i_step; } } /* * Dk4 */ static void DecodeAdpcmDk4( decoder_t *p_dec, int16_t *p_sample, uint8_t *p_buffer ) { decoder_sys_t *p_sys = p_dec->p_sys; adpcm_ima_wav_channel_t channel[2]; size_t i_nibbles; int b_stereo; b_stereo = p_dec->fmt_out.audio.i_channels == 2 ? 1 : 0; GetWord( channel[0].i_predictor ); GetByte( channel[0].i_step_index ); CLAMP( channel[0].i_step_index, 0, 88 ); p_buffer++; if( b_stereo ) { GetWord( channel[1].i_predictor ); GetByte( channel[1].i_step_index ); CLAMP( channel[1].i_step_index, 0, 88 ); p_buffer++; } /* first output predictor */ *p_sample++ = channel[0].i_predictor; if( b_stereo ) { *p_sample++ = channel[1].i_predictor; } for( i_nibbles = 0; i_nibbles < p_sys->i_block - 4 * (b_stereo ? 2:1 ); i_nibbles++ ) { *p_sample++ = AdpcmImaWavExpandNibble( &channel[0], (*p_buffer) >> 4); *p_sample++ = AdpcmImaWavExpandNibble( &channel[b_stereo ? 1 : 0], (*p_buffer)&0x0f); p_buffer++; } } /* * Dk3 */ static void DecodeAdpcmDk3( decoder_t *p_dec, int16_t *p_sample, uint8_t *p_buffer ) { decoder_sys_t *p_sys = p_dec->p_sys; uint8_t *p_end = &p_buffer[p_sys->i_block]; adpcm_ima_wav_channel_t sum; adpcm_ima_wav_channel_t diff; int i_diff_value; p_buffer += 10; GetWord( sum.i_predictor ); GetWord( diff.i_predictor ); GetByte( sum.i_step_index ); GetByte( diff.i_step_index ); i_diff_value = diff.i_predictor; /* we process 6 nibbles at once */ while( p_buffer + 1 <= p_end ) { /* first 3 nibbles */ AdpcmImaWavExpandNibble( &sum, (*p_buffer)&0x0f); AdpcmImaWavExpandNibble( &diff, (*p_buffer) >> 4 ); i_diff_value = ( i_diff_value + diff.i_predictor ) / 2; *p_sample++ = sum.i_predictor + i_diff_value; *p_sample++ = sum.i_predictor - i_diff_value; p_buffer++; AdpcmImaWavExpandNibble( &sum, (*p_buffer)&0x0f); *p_sample++ = sum.i_predictor + i_diff_value; *p_sample++ = sum.i_predictor - i_diff_value; /* now last 3 nibbles */ AdpcmImaWavExpandNibble( &sum, (*p_buffer)>>4); p_buffer++; if( p_buffer < p_end ) { AdpcmImaWavExpandNibble( &diff, (*p_buffer)&0x0f ); i_diff_value = ( i_diff_value + diff.i_predictor ) / 2; *p_sample++ = sum.i_predictor + i_diff_value; *p_sample++ = sum.i_predictor - i_diff_value; AdpcmImaWavExpandNibble( &sum, (*p_buffer)>>4); p_buffer++; *p_sample++ = sum.i_predictor + i_diff_value; *p_sample++ = sum.i_predictor - i_diff_value; } } } /* * EA ADPCM */ #define MAX_CHAN 5 static void DecodeAdpcmEA( decoder_t *p_dec, int16_t *p_sample, uint8_t *p_buffer ) { static const int16_t EATable[]= { 0x0000, 0x00F0, 0x01CC, 0x0188, 0x0000, 0x0000, 0xFF30, 0xFF24, 0x0000, 0x0001, 0x0003, 0x0004, 0x0007, 0x0008, 0x000A, 0x000B, 0x0000, 0xFFFF, 0xFFFD, 0xFFFC, }; decoder_sys_t *p_sys = p_dec->p_sys; int_fast32_t c1[MAX_CHAN], c2[MAX_CHAN]; int_fast8_t d[MAX_CHAN]; unsigned chans = p_dec->fmt_out.audio.i_channels; const uint8_t *p_end = &p_buffer[p_sys->i_block]; int16_t *prev = p_sys->prev; int16_t *cur = prev + chans; for (unsigned c = 0; c < chans; c++) { uint8_t input = p_buffer[c]; c1[c] = EATable[input >> 4]; c2[c] = EATable[(input >> 4) + 4]; d[c] = (input & 0xf) + 8; } for (p_buffer += chans; p_buffer < p_end; p_buffer += chans) { union { uint32_t u; int32_t i; } spl; for (unsigned c = 0; c < chans; c++) { spl.u = (p_buffer[c] & 0xf0u) << 24u; spl.i >>= d[c]; spl.i = (spl.i + cur[c] * c1[c] + prev[c] * c2[c] + 0x80) >> 8; CLAMP(spl.i, -32768, 32767); prev[c] = cur[c]; cur[c] = spl.i; *(p_sample++) = spl.i; } for (unsigned c = 0; c < chans; c++) { spl.u = (p_buffer[c] & 0x0fu) << 28u; spl.i >>= d[c]; spl.i = (spl.i + cur[c] * c1[c] + prev[c] * c2[c] + 0x80) >> 8; CLAMP(spl.i, -32768, 32767); prev[c] = cur[c]; cur[c] = spl.i; *(p_sample++) = spl.i; } } }