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aaf26cffba
And move compute_ref_coefs() to its only user: lpc.c There is no overlap between the users of compute_lpc_coefs() and lpc proper. Reviewed-by: Lynne <dev@lynne.ee> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
250 lines
8.0 KiB
C
250 lines
8.0 KiB
C
/*
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* RealAudio 2.0 (28.8K)
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* Copyright (c) 2003 The FFmpeg project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/internal.h"
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#include "libavutil/mem_internal.h"
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#define BITSTREAM_READER_LE
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#include "avcodec.h"
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#include "celp_filters.h"
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#include "codec_internal.h"
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#include "decode.h"
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#include "get_bits.h"
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#include "lpc_functions.h"
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#include "ra288.h"
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#define MAX_BACKWARD_FILTER_ORDER 36
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#define MAX_BACKWARD_FILTER_LEN 40
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#define MAX_BACKWARD_FILTER_NONREC 35
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#define RA288_BLOCK_SIZE 5
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#define RA288_BLOCKS_PER_FRAME 32
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typedef struct RA288Context {
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void (*vector_fmul)(float *dst, const float *src0, const float *src1,
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int len);
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DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
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DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
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/** speech data history (spec: SB).
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* Its first 70 coefficients are updated only at backward filtering.
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*/
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float sp_hist[111];
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/// speech part of the gain autocorrelation (spec: REXP)
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float sp_rec[37];
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/** log-gain history (spec: SBLG).
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* Its first 28 coefficients are updated only at backward filtering.
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*/
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float gain_hist[38];
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/// recursive part of the gain autocorrelation (spec: REXPLG)
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float gain_rec[11];
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} RA288Context;
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static av_cold int ra288_decode_init(AVCodecContext *avctx)
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{
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RA288Context *ractx = avctx->priv_data;
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AVFloatDSPContext *fdsp;
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av_channel_layout_uninit(&avctx->ch_layout);
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avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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if (avctx->block_align != 38) {
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av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
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return AVERROR_PATCHWELCOME;
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}
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fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
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if (!fdsp)
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return AVERROR(ENOMEM);
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ractx->vector_fmul = fdsp->vector_fmul;
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av_free(fdsp);
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return 0;
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}
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static void convolve(float *tgt, const float *src, int len, int n)
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{
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for (; n >= 0; n--)
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tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
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}
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static void decode(RA288Context *ractx, float gain, int cb_coef)
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{
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int i;
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double sumsum;
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float sum, buffer[5];
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float *block = ractx->sp_hist + 70 + 36; // current block
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float *gain_block = ractx->gain_hist + 28;
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memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
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/* block 46 of G.728 spec */
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sum = 32.0;
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for (i=0; i < 10; i++)
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sum -= gain_block[9-i] * ractx->gain_lpc[i];
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/* block 47 of G.728 spec */
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sum = av_clipf(sum, 0, 60);
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/* block 48 of G.728 spec */
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/* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
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sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
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for (i=0; i < 5; i++)
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buffer[i] = codetable[cb_coef][i] * sumsum;
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sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
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sum = FFMAX(sum, 5.0 / (1<<24));
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/* shift and store */
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memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
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gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
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ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
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}
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/**
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* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
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*
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* @param order filter order
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* @param n input length
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* @param non_rec number of non-recursive samples
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* @param out filter output
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* @param hist pointer to the input history of the filter
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* @param out pointer to the non-recursive part of the output
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* @param out2 pointer to the recursive part of the output
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* @param window pointer to the windowing function table
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*/
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static void do_hybrid_window(RA288Context *ractx,
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int order, int n, int non_rec, float *out,
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float *hist, float *out2, const float *window)
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{
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int i;
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float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
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float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
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LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
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MAX_BACKWARD_FILTER_LEN +
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MAX_BACKWARD_FILTER_NONREC, 16)]);
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av_assert2(order>=0);
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ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
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convolve(buffer1, work + order , n , order);
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convolve(buffer2, work + order + n, non_rec, order);
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for (i=0; i <= order; i++) {
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out2[i] = out2[i] * 0.5625 + buffer1[i];
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out [i] = out2[i] + buffer2[i];
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}
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/* Multiply by the white noise correcting factor (WNCF). */
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*out *= 257.0 / 256.0;
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}
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/**
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* Backward synthesis filter, find the LPC coefficients from past speech data.
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*/
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static void backward_filter(RA288Context *ractx,
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float *hist, float *rec, const float *window,
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float *lpc, const float *tab,
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int order, int n, int non_rec, int move_size)
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{
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float temp[MAX_BACKWARD_FILTER_ORDER+1];
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do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
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if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
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ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
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memmove(hist, hist + n, move_size*sizeof(*hist));
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}
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static int ra288_decode_frame(AVCodecContext * avctx, AVFrame *frame,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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float *out;
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int i, ret;
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RA288Context *ractx = avctx->priv_data;
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GetBitContext gb;
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if (buf_size < avctx->block_align) {
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av_log(avctx, AV_LOG_ERROR,
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"Error! Input buffer is too small [%d<%d]\n",
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buf_size, avctx->block_align);
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return AVERROR_INVALIDDATA;
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}
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ret = init_get_bits8(&gb, buf, avctx->block_align);
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if (ret < 0)
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return ret;
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/* get output buffer */
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frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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out = (float *)frame->data[0];
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for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
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float gain = amptable[get_bits(&gb, 3)];
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int cb_coef = get_bits(&gb, 6 + (i&1));
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decode(ractx, gain, cb_coef);
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memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
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out += RA288_BLOCK_SIZE;
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if ((i & 7) == 3) {
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backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
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ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
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backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
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ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
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}
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}
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*got_frame_ptr = 1;
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return avctx->block_align;
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}
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const FFCodec ff_ra_288_decoder = {
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.p.name = "real_288",
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CODEC_LONG_NAME("RealAudio 2.0 (28.8K)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_RA_288,
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.priv_data_size = sizeof(RA288Context),
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.init = ra288_decode_init,
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FF_CODEC_DECODE_CB(ra288_decode_frame),
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.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
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};
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