/*
* RealAudio 2.0 (28.8K)
- * Copyright (c) 2003 the ffmpeg project
+ * Copyright (c) 2003 The FFmpeg project
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; 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"
+
+#define BITSTREAM_READER_LE
#include "avcodec.h"
-#define ALT_BITSTREAM_READER_LE
#include "bitstream.h"
+#include "celp_filters.h"
+#include "internal.h"
+#include "lpc.h"
#include "ra288.h"
-typedef struct {
- float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
- float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
+#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 {
+ 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)
- float sp_hist[111]; ///< Speech data history (spec: SB)
+ /** 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) */
+ /// speech part of the gain autocorrelation (spec: REXP)
float sp_rec[37];
- float gain_hist[38]; ///< Log-gain history (spec: SBLG)
+ /** 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) */
+ /// recursive part of the gain autocorrelation (spec: REXPLG)
float gain_rec[11];
-
- float sp_block[41]; ///< Speech data of four blocks (spec: STTMP)
- float gain_block[10]; ///< Gain data of four blocks (spec: GSTATE)
} RA288Context;
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
- avctx->sample_fmt = SAMPLE_FMT_S16;
- return 0;
-}
+ RA288Context *ractx = avctx->priv_data;
-static inline float scalar_product_float(const float * v1, const float * v2,
- int size)
-{
- float res = 0.;
+ avctx->channels = 1;
+ avctx->channel_layout = AV_CH_LAYOUT_MONO;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- while (size--)
- res += *v1++ * *v2++;
+ avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
- return res;
+ return 0;
}
-static void colmult(float *tgt, const float *m1, const float *m2, int n)
+static void convolve(float *tgt, const float *src, int len, int n)
{
- while (n--)
- *tgt++ = *m1++ * *m2++;
+ 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, j;
+ int i;
double sumsum;
float sum, buffer[5];
- float *block = ractx->sp_block + 36; // Current block
-
- memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block));
+ float *block = ractx->sp_hist + 70 + 36; // current block
+ float *gain_block = ractx->gain_hist + 28;
- for (i=0; i < 5; i++) {
- block[i] = 0.;
- for (j=0; j < 36; j++)
- block[i] -= block[i-1-j]*ractx->sp_lpc[j];
- }
+ memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
/* block 46 of G.728 spec */
- sum = 32.;
+ sum = 32.0;
for (i=0; i < 10; i++)
- sum -= ractx->gain_block[9-i] * ractx->gain_lpc[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 */
- sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */
+ /* 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 = scalar_product_float(buffer, buffer, 5) / 5;
+ sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0);
sum = FFMAX(sum, 1);
/* shift and store */
- memmove(ractx->gain_block, ractx->gain_block + 1,
- 9 * sizeof(*ractx->gain_block));
+ memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
- ractx->gain_block[9] = 10 * log10(sum) - 32;
+ gain_block[9] = 10 * log10(sum) - 32;
- for (i=1; i < 5; i++)
- for (j=i-1; j >= 0; j--)
- buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j];
-
- /* output */
- for (i=0; i < 5; i++)
- block[i] = av_clipf(block[i] + buffer[i], -4095, 4095);
+ ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}
/**
- * Converts autocorrelation coefficients to LPC coefficients using the
- * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
+ * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
- * @return 0 if success, -1 if fail
- */
-static int eval_lpc_coeffs(const float *in, float *tgt, int n)
-{
- int i, j;
- double f0, f1, f2;
-
- if (in[n] == 0)
- return -1;
-
- if ((f0 = *in) <= 0)
- return -1;
-
- in--; // To avoid a -1 subtraction in the inner loop
-
- for (i=1; i <= n; i++) {
- f1 = in[i+1];
-
- for (j=0; j < i - 1; j++)
- f1 += in[i-j]*tgt[j];
-
- tgt[i-1] = f2 = -f1/f0;
- for (j=0; j < i >> 1; j++) {
- float temp = tgt[j] + tgt[i-j-2]*f2;
- tgt[i-j-2] += tgt[j]*f2;
- tgt[j] = temp;
- }
- if ((f0 += f1*f2) < 0)
- return -1;
- }
-
- return 0;
-}
-
-static void convolve(float *tgt, const float *src, int len, int n)
-{
- for (; n >= 0; n--)
- tgt[n] = scalar_product_float(src, src - n, len);
-
-}
-
-/**
- * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
- *
- * @param order the order of the filter
- * @param n the length of the input
- * @param non_rec the number of non-recursive samples
- * @param out the filter output
- * @param in pointer to the input of the filter
- * @param hist pointer to the input history of the filter. It is updated by
- * this function.
+ * @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(int order, int n, int non_rec, const float *in,
- float *out, float *hist, float *out2,
- const float *window)
+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[order + 1];
- float buffer2[order + 1];
- float work[order + n + non_rec];
+ 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)]);
- /* update history */
- memmove(hist, hist + n, (order + non_rec)*sizeof(*hist));
- memcpy (hist + order + non_rec, in, n * sizeof(*hist));
-
- colmult(work, window, hist, order + n + non_rec);
+ 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);
out [i] = out2[i] + buffer2[i];
}
- /* Multiply by the white noise correcting factor (WNCF) */
- *out *= 257./256.;
+ /* Multiply by the white noise correcting factor (WNCF). */
+ *out *= 257.0 / 256.0;
}
/**
- * Backward synthesis filter. Find the LPC coefficients from past speech data.
+ * Backward synthesis filter, find the LPC coefficients from past speech data.
*/
-static void backward_filter(RA288Context *ractx)
+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 temp1[37]; // RTMP in the spec
- float temp2[11]; // GPTPMP in the spec
-
- do_hybrid_window(36, 40, 35, ractx->sp_block+1, temp1, ractx->sp_hist,
- ractx->sp_rec, syn_window);
+ float temp[MAX_BACKWARD_FILTER_ORDER+1];
- if (!eval_lpc_coeffs(temp1, ractx->sp_lpc, 36))
- colmult(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
+ do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
- do_hybrid_window(10, 8, 20, ractx->gain_block+2, temp2, ractx->gain_hist,
- ractx->gain_rec, gain_window);
+ if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
+ ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
- if (!eval_lpc_coeffs(temp2, ractx->gain_lpc, 10))
- colmult(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
+ memmove(hist, hist + n, move_size*sizeof(*hist));
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
- int *data_size, const uint8_t * buf,
- int buf_size)
+ int *got_frame_ptr, AVPacket *avpkt)
{
- int16_t *out = data;
- int i, j;
+ 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;
+ BitstreamContext bc;
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 0;
+ return AVERROR_INVALIDDATA;
}
- init_get_bits(&gb, buf, avctx->block_align * 8);
+ /* get output buffer */
+ frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ out = (float *)frame->data[0];
- for (i=0; i < 32; i++) {
- float gain = amptable[get_bits(&gb, 3)];
- int cb_coef = get_bits(&gb, 6 + (i&1));
+ bitstream_init(&bc, buf, avctx->block_align * 8);
+
+ for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
+ float gain = amptable[bitstream_read(&bc, 3)];
+ int cb_coef = bitstream_read(&bc, 6 + (i & 1));
decode(ractx, gain, cb_coef);
- for (j=0; j < 5; j++)
- *(out++) = 8 * ractx->sp_block[36 + j];
+ 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);
- if ((i & 7) == 3)
- backward_filter(ractx);
+ backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
+ ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
+ }
}
- *data_size = (char *)out - (char *)data;
+ *got_frame_ptr = 1;
+
return avctx->block_align;
}
-AVCodec ra_288_decoder =
-{
- "real_288",
- CODEC_TYPE_AUDIO,
- CODEC_ID_RA_288,
- sizeof(RA288Context),
- ra288_decode_init,
- NULL,
- NULL,
- ra288_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
+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 = AV_CODEC_CAP_DR1,
};
projects / ffmpeg / blobdiff