* RealAudio 2.0 (28.8K)
* 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 "avcodec.h"
#define ALT_BITSTREAM_READER_LE
-#include "bitstream.h"
+#include "get_bits.h"
#include "ra288.h"
#include "lpc.h"
+#include "celp_math.h"
+#include "celp_filters.h"
+#include "dsputil.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 {
- float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
- float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
+ DSPContext dsp;
+ DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A)
+ DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB)
/** speech data history (spec: SB).
* Its first 70 coefficients are updated only at backward filtering.
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ RA288Context *ractx = avctx->priv_data;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ dsputil_init(&ractx->dsp, avctx);
return 0;
}
-static inline float scalar_product_float(const float * v1, const float * v2,
- int size)
+static void convolve(float *tgt, const float *src, int len, int n)
{
- float res = 0.;
-
- while (size--)
- res += *v1++ * *v2++;
-
- return res;
-}
+ for (; n >= 0; n--)
+ tgt[n] = ff_dot_productf(src, src - n, len);
-static void apply_window(float *tgt, const float *m1, const float *m2, int n)
-{
- while (n--)
- *tgt++ = *m1++ * *m2++;
}
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_hist + 70 + 36; // current block
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 * (1./2048.);
+ buffer[i] = codetable[cb_coef][i] * sumsum;
- sum = scalar_product_float(buffer, buffer, 5) / 5;
+ sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
sum = FFMAX(sum, 1);
gain_block[9] = 10 * log10(sum) - 32;
- for (i=0; i < 5; i++) {
- block[i] = buffer[i];
- for (j=0; j < 36; j++)
- block[i] -= block[i-1-j]*ractx->sp_lpc[j];
- }
-
- /* output */
- for (i=0; i < 5; i++)
- block[i] = av_clipf(block[i], -4095, 4095);
-}
-
-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);
-
+ ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
}
/**
* @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, float *out,
+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_16(float, work)[FFALIGN(MAX_BACKWARD_FILTER_ORDER +
+ MAX_BACKWARD_FILTER_LEN +
+ MAX_BACKWARD_FILTER_NONREC, 8)];
- apply_window(work, window, hist, order + n + non_rec);
+ ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8));
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
/**
* 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
+ float temp[MAX_BACKWARD_FILTER_ORDER+1];
- do_hybrid_window(36, 40, 35, temp1, ractx->sp_hist,
- ractx->sp_rec, syn_window);
+ do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
- if (!compute_lpc_coefs(temp1, 36, ractx->sp_lpc, 0, 1, 1))
- apply_window(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
+ if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
+ ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8));
- do_hybrid_window(10, 8, 20, temp2, ractx->gain_hist,
- ractx->gain_rec, gain_window);
-
- if (!compute_lpc_coefs(temp2, 10, ractx->gain_lpc, 0, 1, 1))
- apply_window(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
-
- memmove(ractx->gain_hist, ractx->gain_hist + 8,
- 28*sizeof(*ractx->gain_hist));
-
- memmove(ractx->sp_hist , ractx->sp_hist + 40,
- 70*sizeof(*ractx->sp_hist ));
+ 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 *data_size, AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
float *out = data;
- int i, j;
+ int i, out_size;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
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;
}
- if (*data_size < 32*5*4)
- return -1;
+ out_size = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ if (*data_size < out_size) {
+ av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+ return AVERROR(EINVAL);
+ }
init_get_bits(&gb, buf, avctx->block_align * 8);
- for (i=0; i < 32; i++) {
+ 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);
- for (j=0; j < 5; j++)
- *(out++) = (1/4096.) * ractx->sp_hist[70 + 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;
+ *data_size = out_size;
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",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_RA_288,
+ .priv_data_size = sizeof(RA288Context),
+ .init = ra288_decode_init,
+ .decode = ra288_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};
projects / ffmpeg / blobdiff