/*
* Bink Audio decoder
- * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
+ * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
*
* This file is part of FFmpeg.
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
-#include "fft.h"
+#include "dct.h"
+#include "rdft.h"
#include "fmtconvert.h"
+#include "libavutil/intfloat_readwrite.h"
extern const uint16_t ff_wma_critical_freqs[25];
GetBitContext gb;
DSPContext dsp;
FmtConvertContext fmt_conv;
+ int version_b; ///< Bink version 'b'
int first;
int channels;
int frame_len; ///< transform size (samples)
int num_bands;
unsigned int *bands;
float root;
- DECLARE_ALIGNED(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
+ DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
union {
return -1;
}
+ s->version_b = avctx->codec_tag == MKTAG('B','I','K','b');
+
if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
// audio is already interleaved for the RDFT format variant
sample_rate *= avctx->channels;
s->channels = 1;
- frame_len_bits += av_log2(avctx->channels);
+ if (!s->version_b)
+ frame_len_bits += av_log2(avctx->channels);
} else {
s->channels = avctx->channels;
}
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
- coeffs[0] = get_float(gb) * s->root;
- coeffs[1] = get_float(gb) * s->root;
+ if (s->version_b) {
+ coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
+ coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
+ } else {
+ coeffs[0] = get_float(gb) * s->root;
+ coeffs[1] = get_float(gb) * s->root;
+ }
for (i = 0; i < s->num_bands; i++) {
/* constant is result of 0.066399999/log10(M_E) */
// parse coefficients
i = 2;
while (i < s->frame_len) {
- if (get_bits1(gb)) {
+ if (s->version_b) {
+ j = i + 16;
+ } else if (get_bits1(gb)) {
j = i + rle_length_tab[get_bits(gb, 4)] * 8;
} else {
j = i + 8;
if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
coeffs[0] /= 0.5;
- ff_dct_calc (&s->trans.dct, coeffs);
+ s->trans.dct.dct_calc(&s->trans.dct, coeffs);
s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
}
else if (CONFIG_BINKAUDIO_RDFT_DECODER)
- ff_rdft_calc(&s->trans.rdft, coeffs);
+ s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
}
s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,