#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "rdft.h"
+#include "vlc.h"
#include "qdm2data.h"
#include "qdm2_tablegen.h"
BitstreamContext bc;
int i, j, k, n, ch, run, level, diff;
- bitstream_init(&bc, node->packet->data, node->packet->size * 8);
+ bitstream_init8(&bc, node->packet->data, node->packet->size);
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
BitstreamContext bc;
if (node) {
- bitstream_init(&bc, node->packet->data, node->packet->size * 8);
+ bitstream_init8(&bc, node->packet->data, node->packet->size);
init_tone_level_dequantization(q, &bc);
fill_tone_level_array(q, 1);
} else {
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
- bitstream_init(&bc, q->compressed_data, q->compressed_size * 8);
+ bitstream_init8(&bc, q->compressed_data, q->compressed_size);
qdm2_decode_sub_packet_header(&bc, &header);
if (header.type < 2 || header.type >= 8) {
q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
packet_bytes = (q->compressed_size - bitstream_tell(&bc) / 8);
- bitstream_init(&bc, header.data, header.size * 8);
+ bitstream_init8(&bc, header.data, header.size);
if (header.type == 2 || header.type == 4 || header.type == 5) {
int csum = 257 * bitstream_read(&bc, 8);
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
/* seek to next block */
- bitstream_init(&bc, header.data, header.size * 8);
+ bitstream_init8(&bc, header.data, header.size);
bitstream_skip(&bc, next_index * 8);
if (next_index >= header.size)
return;
/* decode FFT tones */
- bitstream_init(&bc, packet->data, packet->size * 8);
+ bitstream_init8(&bc, packet->data, packet->size);
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
unknown_flag = 1;