/*
* 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.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/binkaudio.c
+ * @file
* Bink Audio decoder
*
* Technical details here:
#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];
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
typedef struct {
- AVCodecContext *avctx;
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 {
int i;
int frame_len_bits;
- s->avctx = avctx;
dsputil_init(&s->dsp, avctx);
+ ff_fmt_convert_init(&s->fmt_conv, avctx);
/* determine frame length */
if (avctx->sample_rate < 22050) {
} else {
frame_len_bits = 11;
}
- s->frame_len = 1 << frame_len_bits;
- if (s->channels > MAX_CHANNELS) {
- av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
return -1;
}
+ s->version_b = avctx->extradata && avctx->extradata[3] == 'b';
+
if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
// audio is already interleaved for the RDFT format variant
sample_rate *= avctx->channels;
- s->frame_len *= avctx->channels;
s->channels = 1;
- if (avctx->channels == 2)
- frame_len_bits++;
+ if (!s->version_b)
+ frame_len_bits += av_log2(avctx->channels);
} else {
s->channels = avctx->channels;
}
+ s->frame_len = 1 << frame_len_bits;
s->overlap_len = s->frame_len / 16;
s->block_size = (s->frame_len - s->overlap_len) * s->channels;
sample_rate_half = (sample_rate + 1) / 2;
return AVERROR(ENOMEM);
/* populate bands data */
- s->bands[0] = 1;
+ s->bands[0] = 2;
for (i = 1; i < s->num_bands; i++)
- s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half;
- s->bands[s->num_bands] = s->frame_len / 2;
+ s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
+ s->bands[s->num_bands] = s->frame_len;
s->first = 1;
- avctx->sample_fmt = SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
for (i = 0; i < s->channels; i++)
s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
- ff_dct_init(&s->trans.dct, frame_len_bits, 1);
+ ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
else
return -1;
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
- q = 0.0f;
- 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) */
quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
}
- // find band (k)
- for (k = 0; s->bands[k] < 1; k++) {
- q = quant[k];
- }
+ k = 0;
+ q = quant[0];
// 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 (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
- while (s->bands[k] * 2 < i)
+ while (s->bands[k] < i)
q = quant[k++];
} else {
while (i < j) {
- if (s->bands[k] * 2 == i)
+ if (s->bands[k] == i)
q = quant[k++];
coeff = get_bits(gb, width);
if (coeff) {
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);
}
- if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- for (i = 0; i < s->channels; i++)
- for (j = 0; j < s->frame_len; j++)
- s->coeffs_ptr[i][j] = 385.0 + s->coeffs_ptr[i][j]*(1.0/32767.0);
- }
- s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
+ s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
+ s->frame_len, s->channels);
if (!s->first) {
int count = s->overlap_len * s->channels;
return buf_size;
}
-AVCodec binkaudio_rdft_decoder = {
- "binkaudio_rdft",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_BINKAUDIO_RDFT,
- sizeof(BinkAudioContext),
- decode_init,
- NULL,
- decode_end,
- decode_frame,
+AVCodec ff_binkaudio_rdft_decoder = {
+ .name = "binkaudio_rdft",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_BINKAUDIO_RDFT,
+ .priv_data_size = sizeof(BinkAudioContext),
+ .init = decode_init,
+ .close = decode_end,
+ .decode = decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
};
-AVCodec binkaudio_dct_decoder = {
- "binkaudio_dct",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_BINKAUDIO_DCT,
- sizeof(BinkAudioContext),
- decode_init,
- NULL,
- decode_end,
- decode_frame,
+AVCodec ff_binkaudio_dct_decoder = {
+ .name = "binkaudio_dct",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_BINKAUDIO_DCT,
+ .priv_data_size = sizeof(BinkAudioContext),
+ .init = decode_init,
+ .close = decode_end,
+ .decode = decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
};
projects / ffmpeg / blobdiff