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Diffstat (limited to 'ffmpeg/libavcodec/ra288.c')
-rw-r--r--ffmpeg/libavcodec/ra288.c239
1 files changed, 0 insertions, 239 deletions
diff --git a/ffmpeg/libavcodec/ra288.c b/ffmpeg/libavcodec/ra288.c
deleted file mode 100644
index c1b9b6b..0000000
--- a/ffmpeg/libavcodec/ra288.c
+++ /dev/null
@@ -1,239 +0,0 @@
-/*
- * 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 "avcodec.h"
-#include "internal.h"
-#define BITSTREAM_READER_LE
-#include "get_bits.h"
-#include "ra288.h"
-#include "lpc.h"
-#include "celp_filters.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 {
- AVFloatDSPContext fdsp;
- 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;
-
- avctx->channels = 1;
- avctx->channel_layout = AV_CH_LAYOUT_MONO;
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
-
- if (avctx->block_align <= 0) {
- av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
- return AVERROR_PATCHWELCOME;
- }
-
- avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
-
- 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->fdsp.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->fdsp.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;
- }
-
- /* 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];
-
- init_get_bits8(&gb, buf, avctx->block_align);
-
- 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;
-}
-
-AVCodec ff_ra_288_decoder = {
- .name = "real_288",
- .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_RA_288,
- .priv_data_size = sizeof(RA288Context),
- .init = ra288_decode_init,
- .decode = ra288_decode_frame,
- .capabilities = CODEC_CAP_DR1,
-};
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