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/*
* NellyMoser audio decoder
* Copyright (c) 2007 a840bda5870ba11f19698ff6eb9581dfb0f95fa5,
* 539459aeb7d425140b62a3ec7dbf6dc8e408a306, and
* 520e17cd55896441042b14df2566a6eb610ed444
* Copyright (c) 2007 Loic Minier <lool at dooz.org>
* Benjamin Larsson
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*/
/**
* @file
* The 3 alphanumeric copyright notices are md5summed they are from the original
* implementors. The original code is available from http://code.google.com/p/nelly2pcm/
*/
#include "nellymoser.h"
#include "libavutil/lfg.h"
#include "libavutil/random_seed.h"
#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
typedef struct NellyMoserDecodeContext {
AVCodecContext* avctx;
float *float_buf;
DECLARE_ALIGNED(16, float, state)[NELLY_BUF_LEN];
AVLFG random_state;
GetBitContext gb;
float scale_bias;
DSPContext dsp;
FFTContext imdct_ctx;
FmtConvertContext fmt_conv;
DECLARE_ALIGNED(32, float, imdct_out)[NELLY_BUF_LEN * 2];
} NellyMoserDecodeContext;
static void nelly_decode_block(NellyMoserDecodeContext *s,
const unsigned char block[NELLY_BLOCK_LEN],
float audio[NELLY_SAMPLES])
{
int i,j;
float buf[NELLY_FILL_LEN], pows[NELLY_FILL_LEN];
float *aptr, *bptr, *pptr, val, pval;
int bits[NELLY_BUF_LEN];
unsigned char v;
init_get_bits(&s->gb, block, NELLY_BLOCK_LEN * 8);
bptr = buf;
pptr = pows;
val = ff_nelly_init_table[get_bits(&s->gb, 6)];
for (i=0 ; i<NELLY_BANDS ; i++) {
if (i > 0)
val += ff_nelly_delta_table[get_bits(&s->gb, 5)];
pval = -pow(2, val/2048) * s->scale_bias;
for (j = 0; j < ff_nelly_band_sizes_table[i]; j++) {
*bptr++ = val;
*pptr++ = pval;
}
}
ff_nelly_get_sample_bits(buf, bits);
for (i = 0; i < 2; i++) {
aptr = audio + i * NELLY_BUF_LEN;
init_get_bits(&s->gb, block, NELLY_BLOCK_LEN * 8);
skip_bits_long(&s->gb, NELLY_HEADER_BITS + i*NELLY_DETAIL_BITS);
for (j = 0; j < NELLY_FILL_LEN; j++) {
if (bits[j] <= 0) {
aptr[j] = M_SQRT1_2*pows[j];
if (av_lfg_get(&s->random_state) & 1)
aptr[j] *= -1.0;
} else {
v = get_bits(&s->gb, bits[j]);
aptr[j] = ff_nelly_dequantization_table[(1<<bits[j])-1+v]*pows[j];
}
}
memset(&aptr[NELLY_FILL_LEN], 0,
(NELLY_BUF_LEN - NELLY_FILL_LEN) * sizeof(float));
s->imdct_ctx.imdct_calc(&s->imdct_ctx, s->imdct_out, aptr);
/* XXX: overlapping and windowing should be part of a more
generic imdct function */
s->dsp.vector_fmul_reverse(s->state, s->state, ff_sine_128, NELLY_BUF_LEN);
s->dsp.vector_fmul_add(aptr, s->imdct_out, ff_sine_128, s->state, NELLY_BUF_LEN);
memcpy(s->state, s->imdct_out + NELLY_BUF_LEN, sizeof(float)*NELLY_BUF_LEN);
}
}
static av_cold int decode_init(AVCodecContext * avctx) {
NellyMoserDecodeContext *s = avctx->priv_data;
s->avctx = avctx;
av_lfg_init(&s->random_state, 0);
ff_mdct_init(&s->imdct_ctx, 8, 1, 1.0);
dsputil_init(&s->dsp, avctx);
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
s->scale_bias = 1.0/(32768*8);
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
} else {
s->scale_bias = 1.0/(1*8);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
ff_fmt_convert_init(&s->fmt_conv, avctx);
s->float_buf = av_mallocz(NELLY_SAMPLES * sizeof(*s->float_buf));
if (!s->float_buf) {
av_log(avctx, AV_LOG_ERROR, "error allocating float buffer\n");
return AVERROR(ENOMEM);
}
}
/* Generate overlap window */
if (!ff_sine_128[127])
ff_init_ff_sine_windows(7);
avctx->channel_layout = AV_CH_LAYOUT_MONO;
return 0;
}
static int decode_tag(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
NellyMoserDecodeContext *s = avctx->priv_data;
int blocks, i, block_size;
int16_t *samples_s16 = data;
float *samples_flt = data;
block_size = NELLY_SAMPLES * av_get_bytes_per_sample(avctx->sample_fmt);
blocks = buf_size / NELLY_BLOCK_LEN;
if (blocks <= 0) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
if (*data_size < blocks * block_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
if (buf_size % NELLY_BLOCK_LEN) {
av_log(avctx, AV_LOG_WARNING, "Leftover bytes: %d.\n",
buf_size % NELLY_BLOCK_LEN);
}
/* Normal numbers of blocks for sample rates:
* 8000 Hz - 1
* 11025 Hz - 2
* 16000 Hz - 3
* 22050 Hz - 4
* 44100 Hz - 8
*/
for (i=0 ; i<blocks ; i++) {
if (avctx->sample_fmt == SAMPLE_FMT_FLT) {
nelly_decode_block(s, buf, samples_flt);
samples_flt += NELLY_SAMPLES;
} else {
nelly_decode_block(s, buf, s->float_buf);
s->fmt_conv.float_to_int16(samples_s16, s->float_buf, NELLY_SAMPLES);
samples_s16 += NELLY_SAMPLES;
}
buf += NELLY_BLOCK_LEN;
}
*data_size = blocks * block_size;
return buf_size;
}
static av_cold int decode_end(AVCodecContext * avctx) {
NellyMoserDecodeContext *s = avctx->priv_data;
av_freep(&s->float_buf);
ff_mdct_end(&s->imdct_ctx);
return 0;
}
AVCodec ff_nellymoser_decoder = {
.name = "nellymoser",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_NELLYMOSER,
.priv_data_size = sizeof(NellyMoserDecodeContext),
.init = decode_init,
.close = decode_end,
.decode = decode_tag,
.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};