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FFmpeg/libavcodec/nellymoserdec.c
Clément Bœsch ae753dbd0d Merge commit 'b668662939de3a02454cfc9ba3e6d10b87527a40'
* commit 'b668662939de3a02454cfc9ba3e6d10b87527a40':
  get_bits: Move BITSTREAM_READER_LE definition before all relevant #includes

The merge commit also includes changes for libavcodec/interplayacm.c and
libavcodec/truemotion2rt.c

Merged-by: Clément Bœsch <clement@stupeflix.com>
2016-06-29 11:35:10 +02:00

212 lines
6.8 KiB
C

/*
* 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 "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/lfg.h"
#include "libavutil/random_seed.h"
#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "fft.h"
#include "get_bits.h"
#include "internal.h"
#include "nellymoser.h"
#include "sinewin.h"
typedef struct NellyMoserDecodeContext {
AVCodecContext* avctx;
AVLFG random_state;
GetBitContext gb;
float scale_bias;
AVFloatDSPContext *fdsp;
FFTContext imdct_ctx;
DECLARE_ALIGNED(32, float, imdct_buf)[2][NELLY_BUF_LEN];
float *imdct_out;
float *imdct_prev;
} 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 = -exp2(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_half(&s->imdct_ctx, s->imdct_out, aptr);
s->fdsp->vector_fmul_window(aptr, s->imdct_prev + NELLY_BUF_LEN / 2,
s->imdct_out, ff_sine_128,
NELLY_BUF_LEN / 2);
FFSWAP(float *, s->imdct_out, s->imdct_prev);
}
}
static av_cold int decode_init(AVCodecContext * avctx) {
NellyMoserDecodeContext *s = avctx->priv_data;
s->avctx = avctx;
s->imdct_out = s->imdct_buf[0];
s->imdct_prev = s->imdct_buf[1];
av_lfg_init(&s->random_state, 0);
ff_mdct_init(&s->imdct_ctx, 8, 1, 1.0);
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!s->fdsp)
return AVERROR(ENOMEM);
s->scale_bias = 1.0/(32768*8);
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
/* Generate overlap window */
if (!ff_sine_128[127])
ff_init_ff_sine_windows(7);
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
return 0;
}
static int decode_tag(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
const uint8_t *side=av_packet_get_side_data(avpkt, 'F', NULL);
int buf_size = avpkt->size;
NellyMoserDecodeContext *s = avctx->priv_data;
int blocks, i, ret;
float *samples_flt;
blocks = buf_size / NELLY_BLOCK_LEN;
if (blocks <= 0) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
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
*/
if(side && blocks>1 && avctx->sample_rate%11025==0 && (1<<((side[0]>>2)&3)) == blocks)
avctx->sample_rate= 11025*(blocks/2);
/* get output buffer */
frame->nb_samples = NELLY_SAMPLES * blocks;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
samples_flt = (float *)frame->data[0];
for (i=0 ; i<blocks ; i++) {
nelly_decode_block(s, buf, samples_flt);
samples_flt += NELLY_SAMPLES;
buf += NELLY_BLOCK_LEN;
}
*got_frame_ptr = 1;
return buf_size;
}
static av_cold int decode_end(AVCodecContext * avctx) {
NellyMoserDecodeContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct_ctx);
av_freep(&s->fdsp);
return 0;
}
AVCodec ff_nellymoser_decoder = {
.name = "nellymoser",
.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_NELLYMOSER,
.priv_data_size = sizeof(NellyMoserDecodeContext),
.init = decode_init,
.close = decode_end,
.decode = decode_tag,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_PARAM_CHANGE,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};