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@@ -1,6 +1,6 @@
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/*
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* Bink Audio decoder
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- * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
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+ * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
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* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
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*
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* This file is part of FFmpeg.
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@@ -34,6 +34,7 @@
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#include "dsputil.h"
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#include "fft.h"
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#include "fmtconvert.h"
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+#include "libavutil/intfloat_readwrite.h"
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extern const uint16_t ff_wma_critical_freqs[25];
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@@ -44,6 +45,7 @@ typedef struct {
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GetBitContext gb;
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DSPContext dsp;
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FmtConvertContext fmt_conv;
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+ int version_b; ///< Bink version 'b'
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int first;
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int channels;
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int frame_len; ///< transform size (samples)
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@@ -87,11 +89,14 @@ static av_cold int decode_init(AVCodecContext *avctx)
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return -1;
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}
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+ s->version_b = avctx->codec_tag == MKTAG('B','I','K','b');
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+
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if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
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// audio is already interleaved for the RDFT format variant
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sample_rate *= avctx->channels;
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s->channels = 1;
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- frame_len_bits += av_log2(avctx->channels);
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+ if (!s->version_b)
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+ frame_len_bits += av_log2(avctx->channels);
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} else {
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s->channels = avctx->channels;
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}
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@@ -162,8 +167,13 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
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for (ch = 0; ch < s->channels; ch++) {
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FFTSample *coeffs = s->coeffs_ptr[ch];
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- coeffs[0] = get_float(gb) * s->root;
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- coeffs[1] = get_float(gb) * s->root;
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+ if (s->version_b) {
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+ coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
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+ coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
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+ } else {
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+ coeffs[0] = get_float(gb) * s->root;
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+ coeffs[1] = get_float(gb) * s->root;
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+ }
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for (i = 0; i < s->num_bands; i++) {
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/* constant is result of 0.066399999/log10(M_E) */
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@@ -177,7 +187,9 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
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// parse coefficients
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i = 2;
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while (i < s->frame_len) {
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- if (get_bits1(gb)) {
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+ if (s->version_b) {
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+ j = i + 16;
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+ } else if (get_bits1(gb)) {
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j = i + rle_length_tab[get_bits(gb, 4)] * 8;
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} else {
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j = i + 8;
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