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FFmpeg/libavcodec/ra288.c
Andreas Rheinhardt 20f9727018 avcodec/codec_internal: Add FFCodec, hide internal part of AVCodec 
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.

This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-03-21 01:33:09 +01:00

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/*
* RealAudio 2.0 (28.8K)
* Copyright (c) 2003 The FFmpeg project
*
* 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
*/
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"
#include "libavutil/mem_internal.h"
#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "celp_filters.h"
#include "codec_internal.h"
#include "get_bits.h"
#include "internal.h"
#include "lpc.h"
#include "ra288.h"
#define MAX_BACKWARD_FILTER_ORDER 36
#define MAX_BACKWARD_FILTER_LEN 40
#define MAX_BACKWARD_FILTER_NONREC 35
#define RA288_BLOCK_SIZE 5
#define RA288_BLOCKS_PER_FRAME 32
typedef struct RA288Context {
void (*vector_fmul)(float *dst, const float *src0, const float *src1,
int len);
DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
/** speech data history (spec: SB).
* Its first 70 coefficients are updated only at backward filtering.
*/
float sp_hist[111];
/// speech part of the gain autocorrelation (spec: REXP)
float sp_rec[37];
/** log-gain history (spec: SBLG).
* Its first 28 coefficients are updated only at backward filtering.
*/
float gain_hist[38];
/// recursive part of the gain autocorrelation (spec: REXPLG)
float gain_rec[11];
} RA288Context;
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
RA288Context *ractx = avctx->priv_data;
AVFloatDSPContext *fdsp;
av_channel_layout_uninit(&avctx->ch_layout);
avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->block_align != 38) {
av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
return AVERROR_PATCHWELCOME;
}
fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!fdsp)
return AVERROR(ENOMEM);
ractx->vector_fmul = fdsp->vector_fmul;
av_free(fdsp);
return 0;
}
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
}
static void decode(RA288Context *ractx, float gain, int cb_coef)
{
int i;
double sumsum;
float sum, buffer[5];
float *block = ractx->sp_hist + 70 + 36; // current block
float *gain_block = ractx->gain_hist + 28;
memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
/* block 46 of G.728 spec */
sum = 32.0;
for (i=0; i < 10; i++)
sum -= gain_block[9-i] * ractx->gain_lpc[i];
/* block 47 of G.728 spec */
sum = av_clipf(sum, 0, 60);
/* block 48 of G.728 spec */
/* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
for (i=0; i < 5; i++)
buffer[i] = codetable[cb_coef][i] * sumsum;
sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
sum = FFMAX(sum, 5.0 / (1<<24));
/* shift and store */
memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}
/**
* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
* @param order filter order
* @param n input length
* @param non_rec number of non-recursive samples
* @param out filter output
* @param hist pointer to the input history of the filter
* @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(RA288Context *ractx,
int order, int n, int non_rec, float *out,
float *hist, float *out2, const float *window)
{
int i;
float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
MAX_BACKWARD_FILTER_LEN +
MAX_BACKWARD_FILTER_NONREC, 16)]);
av_assert2(order>=0);
ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
for (i=0; i <= order; i++) {
out2[i] = out2[i] * 0.5625 + buffer1[i];
out [i] = out2[i] + buffer2[i];
}
/* Multiply by the white noise correcting factor (WNCF). */
*out *= 257.0 / 256.0;
}
/**
* Backward synthesis filter, find the LPC coefficients from past speech data.
*/
static void backward_filter(RA288Context *ractx,
float *hist, float *rec, const float *window,
float *lpc, const float *tab,
int order, int n, int non_rec, int move_size)
{
float temp[MAX_BACKWARD_FILTER_ORDER+1];
do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
memmove(hist, hist + n, move_size*sizeof(*hist));
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
float *out;
int i, ret;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
"Error! Input buffer is too small [%d<%d]\n",
buf_size, avctx->block_align);
return AVERROR_INVALIDDATA;
}
ret = init_get_bits8(&gb, buf, avctx->block_align);
if (ret < 0)
return ret;
/* get output buffer */
frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
out = (float *)frame->data[0];
for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
float gain = amptable[get_bits(&gb, 3)];
int cb_coef = get_bits(&gb, 6 + (i&1));
decode(ractx, gain, cb_coef);
memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
out += RA288_BLOCK_SIZE;
if ((i & 7) == 3) {
backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
}
}
*got_frame_ptr = 1;
return avctx->block_align;
}
const FFCodec ff_ra_288_decoder = {
.p.name = "real_288",
.p.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_RA_288,
.priv_data_size = sizeof(RA288Context),
.init = ra288_decode_init,
.decode = ra288_decode_frame,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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