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ffmpeg/libavcodec/binkaudio.c

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/*
* Bink Audio decoder
* Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
*
* This file is part of FFmpeg.
*
* FFmpeg 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,
* 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
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Bink Audio decoder
*
* Technical details here:
* http://wiki.multimedia.cx/index.php?title=Bink_Audio
*/
#include "avcodec.h"
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.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 MAX_CHANNELS 2
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
typedef struct {
GetBitContext gb;
DSPContext dsp;
FmtConvertContext fmt_conv;
int version_b; ///< Bink version 'b'
int first;
int channels;
int frame_len; ///< transform size (samples)
int overlap_len; ///< overlap size (samples)
int block_size;
int num_bands;
unsigned int *bands;
float root;
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 {
RDFTContext rdft;
DCTContext dct;
} trans;
} BinkAudioContext;
static av_cold int decode_init(AVCodecContext *avctx)
{
BinkAudioContext *s = avctx->priv_data;
int sample_rate = avctx->sample_rate;
int sample_rate_half;
int i;
int frame_len_bits;
dsputil_init(&s->dsp, avctx);
ff_fmt_convert_init(&s->fmt_conv, avctx);
/* determine frame length */
if (avctx->sample_rate < 22050) {
frame_len_bits = 9;
} else if (avctx->sample_rate < 44100) {
frame_len_bits = 10;
} else {
frame_len_bits = 11;
}
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->channels = 1;
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;
s->root = 2.0 / sqrt(s->frame_len);
/* calculate number of bands */
for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
break;
s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
if (!s->bands)
return AVERROR(ENOMEM);
/* populate bands data */
s->bands[0] = 2;
for (i = 1; i < s->num_bands; i++)
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 = 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, DCT_III);
else
return -1;
return 0;
}
static float get_float(GetBitContext *gb)
{
int power = get_bits(gb, 5);
float f = ldexpf(get_bits_long(gb, 23), power - 23);
if (get_bits1(gb))
f = -f;
return f;
}
static const uint8_t rle_length_tab[16] = {
2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
};
/**
* Decode Bink Audio block
* @param[out] out Output buffer (must contain s->block_size elements)
*/
static void decode_block(BinkAudioContext *s, short *out, int use_dct)
{
int ch, i, j, k;
float q, quant[25];
int width, coeff;
GetBitContext *gb = &s->gb;
if (use_dct)
skip_bits(gb, 2);
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
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) */
int value = get_bits(gb, 8);
quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
}
k = 0;
q = quant[0];
// parse coefficients
i = 2;
while (i < s->frame_len) {
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;
}
j = FFMIN(j, s->frame_len);
width = get_bits(gb, 4);
if (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
while (s->bands[k] < i)
q = quant[k++];
} else {
while (i < j) {
if (s->bands[k] == i)
q = quant[k++];
coeff = get_bits(gb, width);
if (coeff) {
if (get_bits1(gb))
coeffs[i] = -q * coeff;
else
coeffs[i] = q * coeff;
} else {
coeffs[i] = 0.0f;
}
i++;
}
}
}
if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
coeffs[0] /= 0.5;
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)
s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
}
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;
int shift = av_log2(count);
for (i = 0; i < count; i++) {
out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
}
}
memcpy(s->previous, out + s->block_size,
s->overlap_len * s->channels * sizeof(*out));
s->first = 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
BinkAudioContext * s = avctx->priv_data;
av_freep(&s->bands);
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_end(&s->trans.rdft);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
ff_dct_end(&s->trans.dct);
return 0;
}
static void get_bits_align32(GetBitContext *s)
{
int n = (-get_bits_count(s)) & 31;
if (n) skip_bits(s, n);
}
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
short *samples = data;
short *samples_end = (short*)((uint8_t*)data + *data_size);
int reported_size;
GetBitContext *gb = &s->gb;
init_get_bits(gb, buf, buf_size * 8);
reported_size = get_bits_long(gb, 32);
while (get_bits_count(gb) / 8 < buf_size &&
samples + s->block_size <= samples_end) {
decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
samples += s->block_size;
get_bits_align32(gb);
}
*data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
return buf_size;
}
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 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)")
};
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