mirror of
https://github.com/FFmpeg/FFmpeg.git
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atrac: Convert to the new bitstream reader
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This commit is contained in:
parent
41679be1a2
commit
0272119202
@ -33,8 +33,9 @@
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#include <stdio.h>
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#include "libavutil/float_dsp.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "bitstream.h"
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#include "fft.h"
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#include "internal.h"
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#include "sinewin.h"
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@ -164,30 +165,31 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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* Parse the block size mode byte
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*/
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static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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static int at1_parse_bsm(BitstreamContext *bc,
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int log2_block_cnt[AT1_QMF_BANDS])
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{
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int log2_block_count_tmp, i;
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for (i = 0; i < 2; i++) {
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/* low and mid band */
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log2_block_count_tmp = get_bits(gb, 2);
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log2_block_count_tmp = bitstream_read(bc, 2);
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if (log2_block_count_tmp & 1)
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return AVERROR_INVALIDDATA;
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log2_block_cnt[i] = 2 - log2_block_count_tmp;
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}
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/* high band */
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log2_block_count_tmp = get_bits(gb, 2);
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log2_block_count_tmp = bitstream_read(bc, 2);
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if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
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return AVERROR_INVALIDDATA;
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log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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skip_bits(gb, 2);
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bitstream_skip(bc, 2);
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return 0;
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}
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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static int at1_unpack_dequant(BitstreamContext *bc, AT1SUCtx *su,
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float spec[AT1_SU_SAMPLES])
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{
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int bits_used, band_num, bfu_num, i;
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@ -195,22 +197,22 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
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/* parse the info byte (2nd byte) telling how much BFUs were coded */
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su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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su->num_bfus = bfu_amount_tab1[bitstream_read(bc, 3)];
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/* calc number of consumed bits:
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num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
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+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */
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bits_used = su->num_bfus * 10 + 32 +
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bfu_amount_tab2[get_bits(gb, 2)] +
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(bfu_amount_tab3[get_bits(gb, 3)] << 1);
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bfu_amount_tab2[bitstream_read(bc, 2)] +
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(bfu_amount_tab3[bitstream_read(bc, 3)] << 1);
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/* get word length index (idwl) for each BFU */
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for (i = 0; i < su->num_bfus; i++)
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idwls[i] = get_bits(gb, 4);
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idwls[i] = bitstream_read(bc, 4);
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/* get scalefactor index (idsf) for each BFU */
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for (i = 0; i < su->num_bfus; i++)
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idsfs[i] = get_bits(gb, 6);
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idsfs[i] = bitstream_read(bc, 6);
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/* zero idwl/idsf for empty BFUs */
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for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
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@ -240,7 +242,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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/* read in a quantized spec and convert it to
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* signed int and then inverse quantization
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*/
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spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
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spec[pos+i] = bitstream_read_signed(bc, word_len) * scale_factor * max_quant;
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}
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} else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
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memset(&spec[pos], 0, num_specs * sizeof(float));
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@ -277,7 +279,7 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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int buf_size = avpkt->size;
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AT1Ctx *q = avctx->priv_data;
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int ch, ret;
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GetBitContext gb;
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BitstreamContext bc;
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if (buf_size < 212 * avctx->channels) {
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@ -295,14 +297,14 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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for (ch = 0; ch < avctx->channels; ch++) {
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AT1SUCtx* su = &q->SUs[ch];
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init_get_bits(&gb, &buf[212 * ch], 212 * 8);
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bitstream_init(&bc, &buf[212 * ch], 212 * 8);
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/* parse block_size_mode, 1st byte */
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ret = at1_parse_bsm(&gb, su->log2_block_count);
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ret = at1_parse_bsm(&bc, su->log2_block_count);
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if (ret < 0)
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return ret;
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ret = at1_unpack_dequant(&gb, su, q->spec);
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ret = at1_unpack_dequant(&bc, su, q->spec);
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if (ret < 0)
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return ret;
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@ -38,10 +38,11 @@
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#include "libavutil/attributes.h"
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#include "libavutil/float_dsp.h"
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#include "avcodec.h"
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#include "bitstream.h"
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#include "bytestream.h"
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#include "fft.h"
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#include "get_bits.h"
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#include "internal.h"
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#include "atrac.h"
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@ -80,7 +81,7 @@ typedef struct ChannelUnit {
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} ChannelUnit;
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typedef struct ATRAC3Context {
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GetBitContext gb;
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BitstreamContext bc;
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//@{
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/** stream data */
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int coding_mode;
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@ -203,7 +204,7 @@ static av_cold int atrac3_decode_close(AVCodecContext *avctx)
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* @param mantissas mantissa output table
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* @param num_codes number of values to get
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*/
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static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
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static void read_quant_spectral_coeffs(BitstreamContext *bc, int selector,
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int coding_flag, int *mantissas,
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int num_codes)
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{
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@ -219,7 +220,7 @@ static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
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if (selector > 1) {
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for (i = 0; i < num_codes; i++) {
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if (num_bits)
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code = get_sbits(gb, num_bits);
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code = bitstream_read_signed(bc, num_bits);
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else
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code = 0;
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mantissas[i] = code;
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@ -227,7 +228,7 @@ static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
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} else {
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for (i = 0; i < num_codes; i++) {
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if (num_bits)
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code = get_bits(gb, num_bits); // num_bits is always 4 in this case
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code = bitstream_read(bc, num_bits); // num_bits is always 4 in this case
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else
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code = 0;
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mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
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@ -238,8 +239,8 @@ static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
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/* variable length coding (VLC) */
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if (selector != 1) {
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for (i = 0; i < num_codes; i++) {
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huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
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spectral_coeff_tab[selector-1].bits, 3);
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huff_symb = bitstream_read_vlc(bc, spectral_coeff_tab[selector-1].table,
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spectral_coeff_tab[selector-1].bits, 3);
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huff_symb += 1;
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code = huff_symb >> 1;
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if (huff_symb & 1)
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@ -248,8 +249,8 @@ static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
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}
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} else {
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for (i = 0; i < num_codes; i++) {
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huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
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spectral_coeff_tab[selector - 1].bits, 3);
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huff_symb = bitstream_read_vlc(bc, spectral_coeff_tab[selector - 1].table,
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spectral_coeff_tab[selector - 1].bits, 3);
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mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
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mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
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}
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@ -262,24 +263,24 @@ static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
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*
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* @return subband count, fix for broken specification/files
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*/
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static int decode_spectrum(GetBitContext *gb, float *output)
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static int decode_spectrum(BitstreamContext *bc, float *output)
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{
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int num_subbands, coding_mode, i, j, first, last, subband_size;
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int subband_vlc_index[32], sf_index[32];
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int mantissas[128];
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float scale_factor;
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num_subbands = get_bits(gb, 5); // number of coded subbands
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coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
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num_subbands = bitstream_read(bc, 5); // number of coded subbands
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coding_mode = bitstream_read_bit(bc); // coding Mode: 0 - VLC/ 1 - CLC
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/* get the VLC selector table for the subbands, 0 means not coded */
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for (i = 0; i <= num_subbands; i++)
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subband_vlc_index[i] = get_bits(gb, 3);
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subband_vlc_index[i] = bitstream_read(bc, 3);
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/* read the scale factor indexes from the stream */
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for (i = 0; i <= num_subbands; i++) {
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if (subband_vlc_index[i] != 0)
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sf_index[i] = get_bits(gb, 6);
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sf_index[i] = bitstream_read(bc, 6);
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}
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for (i = 0; i <= num_subbands; i++) {
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@ -292,7 +293,7 @@ static int decode_spectrum(GetBitContext *gb, float *output)
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/* decode spectral coefficients for this subband */
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/* TODO: This can be done faster is several blocks share the
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* same VLC selector (subband_vlc_index) */
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read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
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read_quant_spectral_coeffs(bc, subband_vlc_index[i], coding_mode,
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mantissas, subband_size);
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/* decode the scale factor for this subband */
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@ -320,7 +321,7 @@ static int decode_spectrum(GetBitContext *gb, float *output)
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* @param components tonal components
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* @param num_bands number of coded bands
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*/
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static int decode_tonal_components(GetBitContext *gb,
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static int decode_tonal_components(BitstreamContext *bc,
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TonalComponent *components, int num_bands)
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{
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int i, b, c, m;
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@ -328,13 +329,13 @@ static int decode_tonal_components(GetBitContext *gb,
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int band_flags[4], mantissa[8];
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int component_count = 0;
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nb_components = get_bits(gb, 5);
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nb_components = bitstream_read(bc, 5);
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/* no tonal components */
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if (nb_components == 0)
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return 0;
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coding_mode_selector = get_bits(gb, 2);
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coding_mode_selector = bitstream_read(bc, 2);
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if (coding_mode_selector == 2)
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return AVERROR_INVALIDDATA;
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@ -344,16 +345,16 @@ static int decode_tonal_components(GetBitContext *gb,
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int coded_values_per_component, quant_step_index;
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for (b = 0; b <= num_bands; b++)
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band_flags[b] = get_bits1(gb);
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band_flags[b] = bitstream_read_bit(bc);
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coded_values_per_component = get_bits(gb, 3);
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coded_values_per_component = bitstream_read(bc, 3);
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quant_step_index = get_bits(gb, 3);
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quant_step_index = bitstream_read(bc, 3);
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if (quant_step_index <= 1)
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return AVERROR_INVALIDDATA;
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if (coding_mode_selector == 3)
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coding_mode = get_bits1(gb);
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coding_mode = bitstream_read_bit(bc);
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for (b = 0; b < (num_bands + 1) * 4; b++) {
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int coded_components;
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@ -361,18 +362,18 @@ static int decode_tonal_components(GetBitContext *gb,
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if (band_flags[b >> 2] == 0)
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continue;
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coded_components = get_bits(gb, 3);
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coded_components = bitstream_read(bc, 3);
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for (c = 0; c < coded_components; c++) {
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TonalComponent *cmp = &components[component_count];
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int sf_index, coded_values, max_coded_values;
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float scale_factor;
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sf_index = get_bits(gb, 6);
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sf_index = bitstream_read(bc, 6);
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if (component_count >= 64)
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return AVERROR_INVALIDDATA;
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cmp->pos = b * 64 + get_bits(gb, 6);
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cmp->pos = b * 64 + bitstream_read(bc, 6);
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max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
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coded_values = coded_values_per_component + 1;
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@ -381,7 +382,7 @@ static int decode_tonal_components(GetBitContext *gb,
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scale_factor = ff_atrac_sf_table[sf_index] *
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inv_max_quant[quant_step_index];
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read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
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read_quant_spectral_coeffs(bc, quant_step_index, coding_mode,
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mantissa, coded_values);
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cmp->num_coefs = coded_values;
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@ -404,7 +405,7 @@ static int decode_tonal_components(GetBitContext *gb,
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* @param block the gainblock for the current band
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* @param num_bands amount of coded bands
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*/
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static int decode_gain_control(GetBitContext *gb, GainBlock *block,
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static int decode_gain_control(BitstreamContext *bc, GainBlock *block,
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int num_bands)
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{
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int i, j;
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@ -413,13 +414,13 @@ static int decode_gain_control(GetBitContext *gb, GainBlock *block,
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AtracGainInfo *gain = block->g_block;
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for (i = 0; i <= num_bands; i++) {
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gain[i].num_points = get_bits(gb, 3);
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gain[i].num_points = bitstream_read(bc, 3);
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level = gain[i].lev_code;
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loc = gain[i].loc_code;
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for (j = 0; j < gain[i].num_points; j++) {
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level[j] = get_bits(gb, 4);
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loc[j] = get_bits(gb, 5);
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level[j] = bitstream_read(bc, 4);
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loc[j] = bitstream_read(bc, 5);
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if (j && loc[j] <= loc[j - 1])
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return AVERROR_INVALIDDATA;
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}
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@ -567,7 +568,7 @@ static void channel_weighting(float *su1, float *su2, int *p3)
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* @param channel_num channel number
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* @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
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*/
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static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
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static int decode_channel_sound_unit(ATRAC3Context *q, BitstreamContext *bc,
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ChannelUnit *snd, float *output,
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int channel_num, int coding_mode)
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{
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@ -576,30 +577,30 @@ static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
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GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
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if (coding_mode == JOINT_STEREO && channel_num == 1) {
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if (get_bits(gb, 2) != 3) {
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if (bitstream_read(bc, 2) != 3) {
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av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
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return AVERROR_INVALIDDATA;
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}
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} else {
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if (get_bits(gb, 6) != 0x28) {
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if (bitstream_read(bc, 6) != 0x28) {
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av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
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return AVERROR_INVALIDDATA;
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}
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}
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/* number of coded QMF bands */
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snd->bands_coded = get_bits(gb, 2);
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snd->bands_coded = bitstream_read(bc, 2);
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ret = decode_gain_control(gb, gain2, snd->bands_coded);
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ret = decode_gain_control(bc, gain2, snd->bands_coded);
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if (ret)
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return ret;
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snd->num_components = decode_tonal_components(gb, snd->components,
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snd->num_components = decode_tonal_components(bc, snd->components,
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snd->bands_coded);
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if (snd->num_components < 0)
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return snd->num_components;
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num_subbands = decode_spectrum(gb, snd->spectrum);
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num_subbands = decode_spectrum(bc, snd->spectrum);
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/* Merge the decoded spectrum and tonal components. */
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last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
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@ -644,9 +645,9 @@ static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
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if (q->coding_mode == JOINT_STEREO) {
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/* channel coupling mode */
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/* decode Sound Unit 1 */
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init_get_bits(&q->gb, databuf, avctx->block_align * 8);
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bitstream_init(&q->bc, databuf, avctx->block_align * 8);
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ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
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ret = decode_channel_sound_unit(q, &q->bc, q->units, out_samples[0], 0,
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JOINT_STEREO);
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if (ret != 0)
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return ret;
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@ -673,22 +674,22 @@ static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
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/* set the bitstream reader at the start of the second Sound Unit*/
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init_get_bits(&q->gb, ptr1, (avctx->block_align - i) * 8);
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bitstream_init(&q->bc, ptr1, (avctx->block_align - i) * 8);
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/* Fill the Weighting coeffs delay buffer */
|
||||
memmove(q->weighting_delay, &q->weighting_delay[2],
|
||||
4 * sizeof(*q->weighting_delay));
|
||||
q->weighting_delay[4] = get_bits1(&q->gb);
|
||||
q->weighting_delay[5] = get_bits(&q->gb, 3);
|
||||
q->weighting_delay[4] = bitstream_read_bit(&q->bc);
|
||||
q->weighting_delay[5] = bitstream_read(&q->bc, 3);
|
||||
|
||||
for (i = 0; i < 4; i++) {
|
||||
q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
|
||||
q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
|
||||
q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
|
||||
q->matrix_coeff_index_next[i] = bitstream_read(&q->bc, 2);
|
||||
}
|
||||
|
||||
/* Decode Sound Unit 2. */
|
||||
ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
|
||||
ret = decode_channel_sound_unit(q, &q->bc, &q->units[1],
|
||||
out_samples[1], 1, JOINT_STEREO);
|
||||
if (ret != 0)
|
||||
return ret;
|
||||
@ -704,11 +705,11 @@ static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
|
||||
/* Decode the channel sound units. */
|
||||
for (i = 0; i < avctx->channels; i++) {
|
||||
/* Set the bitstream reader at the start of a channel sound unit. */
|
||||
init_get_bits(&q->gb,
|
||||
databuf + i * avctx->block_align / avctx->channels,
|
||||
avctx->block_align * 8 / avctx->channels);
|
||||
bitstream_init(&q->bc,
|
||||
databuf + i * avctx->block_align / avctx->channels,
|
||||
avctx->block_align * 8 / avctx->channels);
|
||||
|
||||
ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
|
||||
ret = decode_channel_sound_unit(q, &q->bc, &q->units[i],
|
||||
out_samples[i], i, q->coding_mode);
|
||||
if (ret != 0)
|
||||
return ret;
|
||||
|
Loading…
Reference in New Issue
Block a user