* 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"
-#define ALT_BITSTREAM_READER_LE
+#include "internal.h"
+#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "ra288.h"
#include "lpc.h"
-#include "celp_math.h"
#include "celp_filters.h"
#define MAX_BACKWARD_FILTER_ORDER 36
#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)
+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)
/** 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 = AV_SAMPLE_FMT_FLT;
- return 0;
-}
+ RA288Context *ractx = avctx->priv_data;
-static void apply_window(float *tgt, const float *m1, const float *m2, int n)
-{
- while (n--)
- *tgt++ = *m1++ * *m2++;
+ avctx->channels = 1;
+ avctx->channel_layout = AV_CH_LAYOUT_MONO;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+
+ 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] = ff_dot_productf(src, src - n, len);
+ tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
}
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 -= gain_block[9-i] * ractx->gain_lpc[i];
for (i=0; i < 5; i++)
buffer[i] = codetable[cb_coef][i] * sumsum;
- sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
+ sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0);
sum = FFMAX(sum, 1);
* @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[MAX_BACKWARD_FILTER_ORDER + 1];
float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
- float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];
+ LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
+ MAX_BACKWARD_FILTER_LEN +
+ MAX_BACKWARD_FILTER_NONREC, 16)]);
- apply_window(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);
}
/* Multiply by the white noise correcting factor (WNCF). */
- *out *= 257./256.;
+ *out *= 257.0 / 256.0;
}
/**
* Backward synthesis filter, find the LPC coefficients from past speech data.
*/
-static void backward_filter(float *hist, float *rec, const float *window,
+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(order, n, non_rec, temp, hist, rec, window);
+ do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
- apply_window(lpc, lpc, tab, order);
+ 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 *data_size, AVPacket *avpkt)
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
- float *out = data;
- int i, j, out_size;
+ float *out;
+ int i, ret;
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;
}
- 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);
+ /* 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];
init_get_bits(&gb, buf, avctx->block_align * 8);
decode(ractx, gain, cb_coef);
- for (j=0; j < RA288_BLOCK_SIZE; j++)
- *(out++) = 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->sp_hist, ractx->sp_rec, syn_window,
+ backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
- backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
+ backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
}
}
- *data_size = out_size;
+ *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 = CODEC_ID_RA_288,
+ .id = AV_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)"),
+ .capabilities = CODEC_CAP_DR1,
};
projects / ffmpeg / blobdiff