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Diffstat (limited to 'src/libdts/decoder.c')
| -rw-r--r-- | src/libdts/decoder.c | 851 |
1 files changed, 0 insertions, 851 deletions
diff --git a/src/libdts/decoder.c b/src/libdts/decoder.c deleted file mode 100644 index 6023366b4..000000000 --- a/src/libdts/decoder.c +++ /dev/null @@ -1,851 +0,0 @@ -/* - * Copyright (C) 2000-2003 the xine project - * - * This file is part of xine, a unix video player. - * - * xine 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. - * - * xine 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 this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * $Id: decoder.c,v 1.2 2003/12/07 15:34:30 f1rmb Exp $ - * - * 04-08-2003 DTS software decode (C) James Courtier-Dutton - * - */ - -#ifndef __sun -/* required for swab() */ -#define _XOPEN_SOURCE 500 -#endif - -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <fcntl.h> -#include <netinet/in.h> /* ntohs */ -#include <assert.h> - -#include "xine_internal.h" -#include "xineutils.h" -#include "audio_out.h" -#include "buffer.h" - -#include "dts_debug.h" -#include "decoder.h" -#include "decoder_internal.h" -#include "print_info.h" - -#ifdef ENABLE_DTS_PARSE - -typedef struct { - uint8_t *start; - uint32_t byte_position; - uint32_t bit_position; - uint8_t byte; -} getbits_state_t; - -static float AdjTable[] = { - 1.0000, - 1.1250, - 1.2500, - 1.4375 -}; - -#include "huffman_tables.h" - -static int32_t getbits_init(getbits_state_t *state, uint8_t *start) { - if ((state == NULL) || (start == NULL)) return -1; - state->start = start; - state->bit_position = 0; - state->byte_position = 0; - state->byte = start[0]; - return 0; -} -/* Non-optimized getbits. */ -/* This can easily be optimized for particular platforms. */ -static uint32_t getbits(getbits_state_t *state, uint32_t number_of_bits) { - uint32_t result=0; - uint8_t byte=0; - if (number_of_bits > 32) { - printf("Number of bits > 32 in getbits\n"); - abort(); - } - - if ((state->bit_position) > 0) { /* Last getbits left us in the middle of a byte. */ - if (number_of_bits > (8-state->bit_position)) { /* this getbits will span 2 or more bytes. */ - byte = state->byte; - byte = byte >> (state->bit_position); - result = byte; - number_of_bits -= (8-state->bit_position); - state->bit_position = 0; - state->byte_position++; - state->byte = state->start[state->byte_position]; - } else { - byte=state->byte; - state->byte = state->byte << number_of_bits; - byte = byte >> (8 - number_of_bits); - result = byte; - state->bit_position += number_of_bits; /* Here it is impossible for bit_position > 8 */ - if (state->bit_position == 8) { - state->bit_position = 0; - state->byte_position++; - state->byte = state->start[state->byte_position]; - } - number_of_bits = 0; - } - } - if ((state->bit_position) == 0) - while (number_of_bits > 7) { - result = (result << 8) + state->byte; - state->byte_position++; - state->byte = state->start[state->byte_position]; - number_of_bits -= 8; - } - if (number_of_bits > 0) { /* number_of_bits < 8 */ - byte = state->byte; - state->byte = state->byte << number_of_bits; - state->bit_position += number_of_bits; /* Here it is impossible for bit_position > 7 */ - if (state->bit_position > 7) printf ("bit_pos2 too large: %d\n",state->bit_position); - byte = byte >> (8 - number_of_bits); - result = (result << number_of_bits) + byte; - number_of_bits = 0; - } - - return result; -} - -static int32_t huff_lookup(getbits_state_t *state, int32_t HuffTable[][2] ) { - int32_t n=1; - int32_t bit; - - { - bit = getbits(state, 1); - n = HuffTable[n][bit]; - } while (n > 0); - /* printf("returning %d\n", n + HuffTable[0][0]); */ - return n + HuffTable[0][0]; -} - - -static int32_t qscales(int32_t nQSelect, getbits_state_t *state, int32_t *nScale) { -/* FIXME: IMPLEMENT */ -return 0; -} - -/* Used by dts.wav files, only 14 bits of the 16 possible are used in the CD. */ -static void squash14to16(uint8_t *buf_from, uint8_t *buf_to, uint32_t number_of_bytes) { - int32_t from; - int32_t to=0; - uint16_t sample1; - uint16_t sample2; - uint16_t sample3; - uint16_t sample4; - uint16_t sample16bit; - /* This should convert the 14bit sync word into a 16bit one. */ - printf("libdts: squashing %d bytes.\n", number_of_bytes); - for(from=0;from<number_of_bytes;from+=8) { - sample1 = buf_from[from+0] | buf_from[from+1] << 8; - sample1 = (sample1 & 0x1fff) | ((sample1 & 0x8000) >> 2); - sample2 = buf_from[from+2] | buf_from[from+3] << 8; - sample2 = (sample2 & 0x1fff) | ((sample2 & 0x8000) >> 2); - sample16bit = (sample1 << 2) | (sample2 >> 12); - buf_to[to++] = sample16bit >> 8; /* Add some swabbing in as well */ - buf_to[to++] = sample16bit & 0xff; - sample3 = buf_from[from+4] | buf_from[from+5] << 8; - sample3 = (sample3 & 0x1fff) | ((sample3 & 0x8000) >> 2); - sample16bit = ((sample2 & 0xfff) << 4) | (sample3 >> 10); - buf_to[to++] = sample16bit >> 8; /* Add some swabbing in as well */ - buf_to[to++] = sample16bit & 0xff; - sample4 = buf_from[from+6] | buf_from[from+7] << 8; - sample4 = (sample4 & 0x1fff) | ((sample4 & 0x8000) >> 2); - sample16bit = ((sample3 & 0x3ff) << 6) | (sample4 >> 8); - buf_to[to++] = sample16bit >> 8; /* Add some swabbing in as well */ - buf_to[to++] = sample16bit & 0xff; - buf_to[to++] = sample4 & 0xff; - } - -} - -#if 0 -/* FIXME: Make this re-entrant */ -static void InverseADPCM(void) { -/* - * NumADPCMCoeff =4, the number of ADPCM coefficients. - * raADPCMcoeff[] are the ADPCM coefficients extracted - * from the bit stream. - * raSample[NumADPCMCoeff], ..., raSample[-1] are the - * history from last subframe or subsubframe. It must - * updated each time before reverse ADPCM is run for a - * block of samples for each subband. - */ -for (m=0; m<nNumSample; m++) -for (n=0; n<NumADPCMCoeff; n++) -raSample[m] += raADPCMcoeff[n]*raSample[m-n-1]; -} -#endif - - -void dts_parse_data (dts_decoder_t *this, buf_element_t *buf) { - uint8_t *data_in = (uint8_t *)buf->content; - getbits_state_t state; - decoder_data_t decoder_data; - decoder_data.sync_type=0; - decoder_data.header_crc_check_bytes=0; - - int32_t n, ch, i; - printf("libdts: buf->size = %d\n", buf->size); - printf("libdts: parse1: "); - for(i=0;i<16;i++) { - printf("%02x ",data_in[i]); - } - printf("\n"); - - if ((data_in[0] == 0x7f) && - (data_in[1] == 0xfe) && - (data_in[2] == 0x80) && - (data_in[3] == 0x01)) { - decoder_data.sync_type=1; - } - if (data_in[0] == 0xff && - data_in[1] == 0x1f && - data_in[2] == 0x00 && - data_in[3] == 0xe8 && - data_in[4] == 0xf1 && /* DTS standard document was wrong here! */ - data_in[5] == 0x07 ) { /* DTS standard document was wrong here! */ - squash14to16(&data_in[0], &data_in[0], buf->size); - buf->size = buf->size - (buf->size / 8); /* size = size * 7 / 8; */ - decoder_data.sync_type=2; - } - if (decoder_data.sync_type == 0) { - printf("libdts: DTS Sync bad\n"); - return; - } - printf("libdts: DTS Sync OK. type=%d\n", decoder_data.sync_type); - printf("libdts: parse2: "); - for(i=0;i<16;i++) { - printf("%02x ",data_in[i]); - } - printf("\n"); - - getbits_init(&state, &data_in[4]); - - /* B.2 Unpack Frame Header Routine */ - /* Frame Type V FTYPE 1 bit */ - decoder_data.frame_type = getbits(&state, 1); /* 1: Normal Frame, 2:Termination Frame */ - /* Deficit Sample Count V SHORT 5 bits */ - decoder_data.deficit_sample_count = getbits(&state, 5); - /* CRC Present Flag V CPF 1 bit */ - decoder_data.crc_present_flag = getbits(&state, 1); - /* Number of PCM Sample Blocks V NBLKS 7 bits */ - decoder_data.number_of_pcm_blocks = getbits(&state, 7); - /* Primary Frame Byte Size V FSIZE 14 bits */ - decoder_data.primary_frame_byte_size = getbits(&state, 14); - /* Audio Channel Arrangement ACC AMODE 6 bits */ - decoder_data.audio_channel_arrangement = getbits(&state, 6); - /* Core Audio Sampling Frequency ACC SFREQ 4 bits */ - decoder_data.core_audio_sampling_frequency = getbits(&state, 4); - /* Transmission Bit Rate ACC RATE 5 bits */ - decoder_data.transmission_bit_rate = getbits(&state, 5); - /* Embedded Down Mix Enabled V MIX 1 bit */ - decoder_data.embedded_down_mix_enabled = getbits(&state, 1); - /* Embedded Dynamic Range Flag V DYNF 1 bit */ - decoder_data.embedded_dynamic_range_flag = getbits(&state, 1); - /* Embedded Time Stamp Flag V TIMEF 1 bit */ - decoder_data.embedded_time_stamp_flag = getbits(&state, 1); - /* Auxiliary Data Flag V AUXF 1 bit */ - decoder_data.auxiliary_data_flag = getbits(&state, 1); - /* HDCD NV HDCD 1 bits */ - decoder_data.hdcd = getbits(&state, 1); - /* Extension Audio Descriptor Flag ACC EXT_AUDIO_ID 3 bits */ - decoder_data.extension_audio_descriptor_flag = getbits(&state, 3); - /* Extended Coding Flag ACC EXT_AUDIO 1 bit */ - decoder_data.extended_coding_flag = getbits(&state, 1); - /* Audio Sync Word Insertion Flag ACC ASPF 1 bit */ - decoder_data.audio_sync_word_insertion_flag = getbits(&state, 1); - /* Low Frequency Effects Flag V LFF 2 bits */ - decoder_data.low_frequency_effects_flag = getbits(&state, 2); - /* Predictor History Flag Switch V HFLAG 1 bit */ - decoder_data.predictor_history_flag_switch = getbits(&state, 1); - /* Header CRC Check Bytes V HCRC 16 bits */ - if (decoder_data.crc_present_flag == 1) - decoder_data.header_crc_check_bytes = getbits(&state, 16); - /* Multirate Interpolator Switch NV FILTS 1 bit */ - decoder_data.multirate_interpolator_switch = getbits(&state, 1); - /* Encoder Software Revision ACC/NV VERNUM 4 bits */ - decoder_data.encoder_software_revision = getbits(&state, 4); - /* Copy History NV CHIST 2 bits */ - decoder_data.copy_history = getbits(&state, 2); - /* Source PCM Resolution ACC/NV PCMR 3 bits */ - decoder_data.source_pcm_resolution = getbits(&state, 3); - /* Front Sum/Difference Flag V SUMF 1 bit */ - decoder_data.front_sum_difference_flag = getbits(&state, 1); - /* Surrounds Sum/Difference Flag V SUMS 1 bit */ - decoder_data.surrounds_sum_difference_flag = getbits(&state, 1); - /* Dialog Normalisation Parameter/Unspecified V DIALNORM/UNSPEC 4 bits */ - switch (decoder_data.encoder_software_revision) { - case 6: - decoder_data.dialog_normalisation_unspecified = 0; - decoder_data.dialog_normalisation_parameter = getbits(&state, 4); - decoder_data.dialog_normalisation_gain = - (16+decoder_data.dialog_normalisation_parameter); - break; - case 7: - decoder_data.dialog_normalisation_unspecified = 0; - decoder_data.dialog_normalisation_parameter = getbits(&state, 4); - decoder_data.dialog_normalisation_gain = - (decoder_data.dialog_normalisation_parameter); - break; - default: - decoder_data.dialog_normalisation_unspecified = getbits(&state, 4); - decoder_data.dialog_normalisation_gain = decoder_data.dialog_normalisation_parameter = 0; - break; - } - - /* B.3 Audio Decoding */ - /* B.3.1 Primary Audio Coding Header */ - - /* Number of Subframes V SUBFS 4 bits */ - decoder_data.number_of_subframes = getbits(&state, 4) + 1 ; - /* Number of Primary Audio Channels V PCHS 3 bits */ - decoder_data.number_of_primary_audio_channels = getbits(&state, 3) + 1 ; - /* Subband Activity Count V SUBS 5 bits per channel */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - decoder_data.subband_activity_count[ch] = getbits(&state, 5) + 2 ; - } - /* High Frequency VQ Start Subband V VQSUB 5 bits per channel */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - decoder_data.high_frequency_VQ_start_subband[ch] = getbits(&state, 5) + 1 ; - } - /* Joint Intensity Coding Index V JOINX 3 bits per channel */ - for (n=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - decoder_data.joint_intensity_coding_index[ch] = getbits(&state, 3) ; - } - /* Transient Mode Code Book V THUFF 2 bits per channel */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - decoder_data.transient_mode_code_book[ch] = getbits(&state, 2) ; - } - /* Scale Factor Code Book V SHUFF 3 bits per channel */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - decoder_data.scales_factor_code_book[ch] = getbits(&state, 3) ; - } - /* Bit Allocation Quantizer Select BHUFF V 3 bits per channel */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - decoder_data.bit_allocation_quantizer_select[ch] = getbits(&state, 3) ; - } - /* Quantization Index Codebook Select V SEL variable bits */ - /* ABITS=1: */ - n=0; - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) - decoder_data.quantization_index_codebook_select[ch][n] = getbits(&state, 1); - /* ABITS = 2 to 5: */ - for (n=1; n<5; n++) - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) - decoder_data.quantization_index_codebook_select[ch][n] = getbits(&state, 2); - /* ABITS = 6 to 10: */ - for (n=5; n<10; n++) - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) - decoder_data.quantization_index_codebook_select[ch][n] = getbits(&state, 3); - /* ABITS = 11 to 26: */ - for (n=10; n<26; n++) - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) - decoder_data.quantization_index_codebook_select[ch][n] = 0; /* Not transmitted, set to zero. */ - - /* Scale Factor Adjustment Index V ADJ 2 bits per occasion */ - /* ABITS = 1: */ - n = 0; - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - int32_t adj; - if ( decoder_data.quantization_index_codebook_select[ch][n] == 0 ) { /* Transmitted only if quantization_index_codebook_select=0 (Huffman code used) */ - /* Extract ADJ index */ - adj = getbits(&state, 2); - /* Look up ADJ table */ - decoder_data.scale_factor_adjustment_index[ch][n] = AdjTable[adj]; - } - } - /* ABITS = 2 to 5: */ - for (n=1; n<5; n++){ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++){ - int32_t adj; - if ( decoder_data.quantization_index_codebook_select[ch][n] < 3 ) { /* Transmitted only when quantization_index_codebook_select<3 */ - /* Extract ADJ index */ - adj = getbits(&state, 2); - /* Look up ADJ table */ - decoder_data.scale_factor_adjustment_index[ch][n] = AdjTable[adj]; - } - } - } - /* ABITS = 6 to 10: */ - for (n=5; n<10; n++){ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++){ - int32_t adj; - if ( decoder_data.quantization_index_codebook_select[ch][n] < 7 ) { /* Transmitted only when quantization_index_codebook_select<7 */ - /* Extract ADJ index */ - adj = getbits(&state, 2); - /* Look up ADJ table */ - decoder_data.scale_factor_adjustment_index[ch][n] = AdjTable[adj]; - } - } - } - - if (decoder_data.crc_present_flag == 1) { /* Present only if CPF=1. */ - decoder_data.audio_header_crc_check_word = getbits(&state, 16); - } - -/* B.3.2 Unpack Subframes */ -/* B.3.2.1 Primary Audio Coding Side Information */ - -/* Subsubframe Count V SSC 2 bit */ - decoder_data.subsubframe_count = getbits(&state, 2) + 1; -/* Partial Subsubframe Sample Count V PSC 3 bit */ - decoder_data.partial_subsubframe_sample_count = getbits(&state, 3); -/* Prediction Mode V PMODE 1 bit per subband */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - for (n=0; n<decoder_data.subband_activity_count[ch]; n++) { - decoder_data.prediction_mode[ch][n] = getbits(&state, 1); - } - } - -/* Prediction Coefficients VQ Address V PVQ 12 bits per occurrence */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - for (n=0; n<decoder_data.subband_activity_count[ch]; n++) { - decoder_data.PVQIndex[ch][n] = 0; - if ( decoder_data.prediction_mode[ch][n]>0 ) { /* Transmitted only when ADPCM active */ - /* Extract the VQindex */ - decoder_data.nVQIndex = getbits(&state,12); - /* Look up the VQ table for prediction coefficients. */ - /* FIXME: How to implement LookUp? */ - decoder_data.PVQIndex[ch][n] = decoder_data.nVQIndex; - /* FIXME: We don't have the ADPCMCoeff table. */ - /* ADPCMCoeffVQ.LookUp(nVQIndex, PVQ[ch][n]);*/ /* 4 coefficients FIXME: Need to work out what this does. */ - } - } - } - - - /* Bit Allocation Index V ABITS variable bits */ - /* FIXME: No getbits here InverseQ does the getbits */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - /* Bit Allocation Quantizer Select tells which codebook was used */ - decoder_data.nQSelect = decoder_data.bit_allocation_quantizer_select[ch]; - /* Use this codebook to decode the bit stream for bit_allocation_index[ch][n] */ - for (n=0; n<decoder_data.high_frequency_VQ_start_subband[ch]; n++) { - /* Not for VQ encoded subbands. */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - /* This basically selects a huffman table number nQSelect, */ - /* and uses it to read a variable amount of bits and does a huffman search to find the value. */ - /* FIXME: Need to implement InverseQ, so we can uncomment this line */ - if (decoder_data.nQSelect == 6) { - decoder_data.bit_allocation_index[ch][n] = getbits(&state,5); - } else { - printf("bit_alloc parse failed, (nQSelect != 6) not implemented yet."); - abort(); - } - - /*QABITS.ppQ[nQSelect]->InverseQ(&state, bit_allocation_index[ch][n]); */ - } - } - - /* Transition Mode V TMODE variable bits */ - - /* Always assume no transition unless told */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++){ - for (n=0; n<decoder_data.subband_activity_count[ch]; n++) { - decoder_data.transition_mode[ch][n] = 0; - } - /* Decode transition_mode[ch][n] */ - if ( decoder_data.subsubframe_count>1 ) { - /* Transient possible only if more than one subsubframe. */ - for (ch=0; ch<decoder_data.number_of_primary_audio_channels; ch++) { - /* transition_mode[ch][n] is encoded by a codebook indexed by transient_mode_code_book[ch] */ - decoder_data.nQSelect = decoder_data.transient_mode_code_book[ch]; - for (n=0; n<decoder_data.high_frequency_VQ_start_subband[ch]; n++) { - /* No VQ encoded subbands */ - if ( decoder_data.bit_allocation_index[ch][n] >0 ) { - /* Present only if bits allocated */ - /* Use codebook nQSelect to decode transition_mode from the bit stream */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - if (decoder_data.nQSelect == 0) { - decoder_data.transition_mode[ch][n] = huff_lookup(&state, HuffA4); - } else { - printf("transition mod parse failed, (nQSelect != 0) not implemented yet."); - abort(); - } - - /* QTMODE.ppQ[nQSelect]->InverseQ(&state,transition_mode[ch][n]); */ - } else { - decoder_data.transition_mode[ch][n] = 0; - } - } - } - } - } - -/* WORKING ON THIS BIT */ - - -#if 0 - /* Scale Factors V SCALES variable bits */ - for (ch=0; ch<number_of_primary_audio_channels; ch++) { - /* Clear scale_factors */ - for (n=0; n<subband_activity_count[ch]; n++) { - scale_factors[ch][n][0] = 0; - scale_factors[ch][n][1] = 0; - } - /* scales_factor_code_book indicates which codebook was used to encode scale_factors */ - nQSelect = scales_factor_code_book[ch]; - /* Select the root square table (scale_factors were nonlinearly */ - /* quantized). */ - /* Assume nQSelect != 6 */ - /* So RMS is always 6 bit. */ - if ( nQSelect == 6 ) { - /* pScaleTable = &RMS7Bit;*/ /* 7-bit root square table */ - } else { - /* pScaleTable = &RMS6Bit;*/ /* 6-bit root square table */ - } - /* - * Clear accumulation (if Huffman code was used, the difference - * of scale_factors was encoded). - */ - nScaleSum = 0; - /* - * Extract scale_factors for Subbands up to high_frequency_VQ_start_subband[ch] - */ - for (n=0; n<high_frequency_VQ_start_subband[ch]; n++) { - if ( bit_allocation_index[ch][n] >0 ) { /* Not present if no bit allocated */ - /* - * First scale factor - */ - /* Use the (Huffman) code indicated by nQSelect to decode */ - /* the quantization index of scale_factors from the bit stream */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - qscales(nQSelect, &state, &nScale); - /* QSCALES.ppQ[nQSelect]->InverseQ(InputFrame, nScale); */ - /* Take care of difference encoding */ - if ( nQSelect < 5 ) { /* Huffman encoded, nScale is the difference */ - nScaleSum += nScale; /* of the quantization indexes of scale_factors. */ - } else { /* Otherwise, nScale is the quantization */ - nScaleSum = nScale; /* level of scale_factors. */ - } - /* Look up scale_factors from the root square table */ - /* FIXME: How to implement LookUp? */ - pScaleTable->LookUp(nScaleSum, scale_factors[ch][n][0]) - /* - * Two scale factors transmitted if there is a transient - */ - if (transition_mode[ch][n]>0) { - /* Use the (Huffman) code indicated by nQSelect to decode */ - /* the quantization index of scale_factors from the bit stream */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - QSCALES.ppQ[nQSelect]->InverseQ(InputFrame, nScale); - /* Take care of difference encoding */ - if ( nQSelect < 5 ) /* Huffman encoded, nScale is the difference */ - nScaleSum += nScale; /* of the quantization indexes of scale_factors. */ - else /* Otherwise, nScale is the quantization */ - nScaleSum = nScale; /* level of scale_factors. */ - /* Look up scale_factors from the root square table */ - /* FIXME: How to implement LookUp? */ - pScaleTable->LookUp(nScaleSum, scale_factors[ch][n][1]); - } - } - } - /* - * High frequency VQ subbands - */ - for (n=high_frequency_VQ_start_subband[ch]; n<subband_activity_count[ch]; n++) { - /* Use the code book indicated by nQSelect to decode */ - /* the quantization index of scale_factors from the bit stream */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - QSCALES.ppQ[nQSelect]->InverseQ(InputFrame, nScale); - /* Take care of difference encoding */ - if ( nQSelect < 5 ) /* Huffman encoded, nScale is the difference */ - nScaleSum += nScale; /* of the quantization indexes of scale_factors. */ - else /* Otherwise, nScale is the quantization */ - nScaleSum = nScale; /* level of scale_factors. */ - /* Look up scale_factors from the root square table */ - /* FIXME: How to implement LookUp? */ - pScaleTable->LookUp(nScaleSum, scale_factors[ch][n][0]) - } - } - -/* #if 0 */ -/* FIXME: ALL CODE BELOW HERE does not compile yet. */ - - - /* Joint Subband Scale Factor Codebook Select V JOIN SHUFF 3 bits per channel */ - for (ch=0; ch<number_of_primary_audio_channels; ch++) - if (joint_intensity_coding_index[ch]>0 ) /* Transmitted only if joint subband coding enabled. */ - joint_subband_scale_factor_codebook_select[ch] = getbits(&state,3); - - /* Scale Factors for Joint Subband Coding V JOIN SCALES variable bits */ - int nSourceCh; - for (ch=0; ch<number_of_primary_audio_channels; ch++) { - if (joint_intensity_coding_index[ch]>0 ) { /* Only if joint subband coding enabled. */ - nSourceCh = joint_intensity_coding_index[ch]-1; /* Get source channel. joint_intensity_coding_index counts */ - /* channels as 1,2,3,4,5, so minus 1. */ - nQSelect = joint_subband_scale_factor_codebook_select[ch]; /* Select code book. */ - for (n=subband_activity_count[ch]; n<subband_activity_count[nSourceCh]; n++) { - /* Use the code book indicated by nQSelect to decode */ - /* the quantization index of scale_factors_for_joint_subband_coding */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - QSCALES.ppQ[nQSelect]->InverseQ(InputFrame, nJScale); - /* Bias by 64 */ - nJScale = nJScale + 64; - /* Look up scale_factors_for_joint_subband_coding from the joint scale table */ - /* FIXME: How to implement LookUp? */ - JScaleTbl.LookUp(nJScale, scale_factors_for_joint_subband_coding[ch][n]); - } - } - } - - /* Stereo Down-Mix Coefficients NV DOWN 7 bits per coefficient */ - if ( (MIX!=0) && (number_of_primary_audio_channels>2) ) { - /* Extract down mix indexes */ - for (ch=0; ch<number_of_primary_audio_channels; ch++) { /* Each primary channel */ - stereo_down_mix_coefficients[ch][0] = getbits(&state,7); - stereo_down_mix_coefficients[ch][1] = getbits(&state,7); - } - } - /* Look up down mix coefficients */ - for (n=0; n<subband_activity_count; n++) { /* Each active subbands */ - LeftChannel = 0; - RightChannel = 0; - for (ch=0; ch<number_of_primary_audio_channels; ch++) { /* Each primary channels */ - LeftChannel += stereo_down_mix_coefficients[ch][0]*Sample[Ch]; - RightChannel += stereo_down_mix_coefficients[ch][1]*Sample[Ch]; - } - } - /* Down mixing may also be performed on the PCM samples after the filterbank reconstruction. */ - - /* Dynamic Range Coefficient NV RANGE 8 bits */ - if ( embedded_dynamic_range_flag != 0 ) { - nIndex = getbits(&state,8); - /* FIXME: How to implement LookUp? */ - RANGEtbl.LookUp(nIndex,dynamic_range_coefficient); - /* The following range adjustment is to be performed */ - /* after QMF reconstruction */ - for (ch=0; ch<number_of_primary_audio_channels; ch++) - for (n=0; n<nNumSamples; n++) - AudioCh[ch].ReconstructedSamples[n] *= dynamic_range_coefficient; - } - - /* Side Information CRC Check Word V SICRC 16 bits */ - if ( CPF==1 ) /* Present only if CPF=1. */ - SICRC = getbits(&state,16); - - /* B.3.3 Primary Audio Data Arrays */ - - /* VQ Encoded High Frequency Subbands NV HFREQ 10 bits per applicable subbands */ - for (ch=0; ch<number_of_primary_audio_channels; ch++) { - for (n=high_frequency_VQ_start_subband[ch]; n<subband_activity_count[ch]; n++) { - /* Extract the VQ address from the bit stream */ - nVQIndex = getbits(&state,10); - /* Look up the VQ code book for 32 subband samples. */ - /* FIXME: How to implement LookUp? */ - HFreqVQ.LookUp(nVQIndex, VQ_encoded_high_frequency_subbands[ch][n]) - /* Scale and take the samples */ - Scale = (real)scale_factors[ch][n][0]; /* Get the scale factor */ - for (m=0; m<subsubframe_count*8; m++, nSample++) { - aPrmCh[ch].aSubband[n].raSample[m] = rScale*VQ_encoded_high_frequency_subbands[ch][n][m]; - } - } - } - - /* Low Frequency Effect Data V LFE 8 bits per sample */ - if ( low_frequency_effects_flag>0 ) { /* Present only if flagged by low_frequency_effects_flag */ - /* extract low_frequency_effect_data samples from the bit stream */ - for (n=0; n<2*low_frequency_effects_flag*subsubframe_count; n++) { - low_frequency_effect_data[n] = (signed int)(signed char)getbits(&state,8); - /* Use char to get sign extension because it */ - /* is 8-bit 2's compliment. */ - /* Extract scale factor index from the bit stream */ - } - LFEscaleIndex = getbits(&state,8); - /* Look up the 7-bit root square quantization table */ - /* FIXME: How to implement LookUp? */ - pLFE_RMS->LookUp(LFEscaleIndex,nScale); - /* Account for the quantizer step size which is 0.035 */ - rScale = nScale*0.035; - /* Get the actual low_frequency_effect_data samples */ - for (n=0; n<2*low_frequency_effects_flag*subsubframe_count; n++) { - LFECh.rLFE[k] = low_frequency_effect_data[n]*rScale; - } - /* Interpolation low_frequency_effect_data samples */ - LFECh.InterpolationFIR(low_frequency_effects_flag); /* low_frequency_effects_flag indicates which */ - /* interpolation filter to use */ - } - - /* Audio Data V AUDIO variable bits */ - /* - * Select quantization step size table - */ - if ( RATE == 0x1f ) { - pStepSizeTable = &StepSizeLossLess; /* Lossless quantization */ - } else { - pStepSizeTable = &StepSizeLossy; /* Lossy */ - } - /* - * Unpack the subband samples - */ - for (nSubSubFrame=0; nSubSubFrame<subsubframe_count; nSubSubFrame++) { - for (ch=0; ch<number_of_primary_audio_channels; ch++) { - for (n=0; n<high_frequency_VQ_start_subband[ch]; n++) { /* Not high frequency VQ subbands */ - /* - * Select the mid-tread linear quantizer - */ - nABITS = bit_allocation_index[ch][n]; /* Select the mid-tread quantizer */ - pCQGroup = &pCQGroupAUDIO[nABITS-1];/* Select the group of */ - /* code books corresponding to the */ - /* the mid-tread linear quantizer. */ - nNumQ = pCQGroupAUDIO[nABITS-1].nNumQ-1;/* Number of code */ - /* books in this group */ - /* - * Determine quantization index code book and its type - */ - /* Select quantization index code book */ - nSEL = quantization_index_codebook_select[ch][nABITS-1]; - /* Determine its type */ - nQType = 1; /* Assume Huffman type by default */ - if ( nSEL==nNumQ ) { /* Not Huffman type */ - if ( nABITS<=7 ) { - nQType = 3; /* Block code */ - } else { - nQType = 2; /* No further encoding */ - } - } - if ( nABITS==0 ) { /* No bits allocated */ - nQType = 0; - } - /* - * Extract bits from the bit stream - * This retrieves 8 AUDIO values - */ - switch ( nQType ) { - case 0: /* No bits allocated */ - for (m=0; m<8; m++) - AUDIO[m] = 0; - break; - case 1: /* Huffman code */ - for (m=0; m<8; m++) - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - pCQGroup->ppQ[nSEL]->InverseQ(InputFrame,AUDIO[m]); - break; - case 2: /* No further encoding */ - for (m=0; m<8; m++) { - /* Extract quantization index from the bit stream */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - pCQGroup->ppQ[nSEL]->InverseQ(InputFrame, nCode) - /* Take care of 2's compliment */ - AUDIO[m] = pCQGroup->ppQ[nSEL]->SignExtension(nCode); - } - break; - case 3: /* Block code */ - /* Block code is just 1 value with 4 samples derived from it. - * with each sample a digit from the number (using a base derived from nABITS via a table) - * E.g. nABITS = 10, base = 5 (Base value taken from table.) - * 1st sample = (value % 5) - (int(5/2); (Values between -2 and +2 ) - * 2st sample = ((value / 5) % 5) - (int(5/2); - * 3rd sample = ((value / 25) % 5) - (int(5/2); - * 4th sample = ((value / 125) % 5) - (int(5/2); - * - */ - pCBQ = &pCBlockQ[nABITS-1]; /* Select block code book */ - m = 0; - for (nBlock=0; nBlock<2; nBlock++) { - /* Extract the block code index from the bit stream */ - /* FIXME: What is Inverse Quantization(InverseQ) ? */ - pCQGroup->ppQ[nSEL]->InverseQ(InputFrame, nCode) - /* Look up 4 samples from the block code book */ - /* FIXME: How to implement LookUp? */ - pCBQ->LookUp(nCode,&AUDIO[m]) - m += 4; - } - break; - default: /* Undefined */ - printf("ERROR: Unknown AUDIO quantization index code book."); - } - /* - * Account for quantization step size and scale factor - */ - /* Look up quantization step size */ - nABITS = bit_allocation_index[ch][n]; - /* FIXME: How to implement LookUp? */ - pStepSizeTable->LookUp(nABITS, rStepSize); - /* Identify transient location */ - nTmode = transition_mode[ch][n]; - if ( nTmode == 0 ) /* No transient */ - nTmode = subsubframe_count; - /* Determine proper scale factor */ - if (nSubSubFrame<nTmode) /* Pre-transient */ - rScale = rStepSize * scale_factors[ch][n][0]; /* Use first scale factor */ - else /* After-transient */ - rScale = rStepSize * scale_factors[ch][n][1]; /* Use second scale factor */ - /* Adjustmemt of scale factor */ - rScale *= scale_factor_adjustment_index[ch][quantization_index_codebook_select[ch][nABITS-1]]; /* scale_factor_adjustment_index[ ][ ] are assumed 1 */ - /* unless changed by bit */ - /* stream when quantization_index_codebook_select indicates */ - /* Huffman code. */ - /* Scale the samples */ - nSample = 8*nSubSubFrame; /* Set sample index */ - for (m=0; m<8; m++, nSample++) - aPrmCh[ch].aSubband[n].aSample[nSample] = rScale*AUDIO[m]; - /* - * Inverse ADPCM - */ - if ( PMODE[ch][n] != 0 ) /* Only when prediction mode is on. */ - aPrmCh[ch].aSubband[n].InverseADPCM(); - /* - * Check for DSYNC - */ - if ( (nSubSubFrame==(subsubframe_count-1)) || (ASPF==1) ) { - DSYNC = getbits(&state,16); - if ( DSYNC != 0xffff ) - printf("DSYNC error at end of subsubframe #%d", nSubSubFrame); - } - } - } -/* B.3.4 Unpack Optional Information */ -/* TODO ^^^ */ - -#endif -/* CODE BELOW here does compile */ - - printf("getbits status: byte_pos = %d, bit_pos = %d\n", - state.byte_position, - state.bit_position); -#if 0 - for(n=0;n<2016;n++) { - if((n % 32) == 0) printf("\n"); - printf("%02X ",state.start[state.byte_position+n]); - } - printf("\n"); -#endif - -#if 0 - if ((extension_audio_descriptor_flag == 0) - || (extension_audio_descriptor_flag == 3)) { - printf("libdts:trying extension...\n"); - channel_extension_sync_word = getbits(&state, 32); - extension_primary_frame_byte_size = getbits(&state, 10); - extension_channel_arrangement = getbits(&state, 4); - } -#endif - -#if 0 - extension_sync_word_SYNC96 = getbits(&state, 32); - extension_frame_byte_data_size_FSIZE96 = getbits(&state, 12); - revision_number = getbits(&state, 4); -#endif -dts_print_decoded_data(&decoder_data); -} - -#endif |