#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)
+ AVFrame frame;
+ 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)
{
+ RA288Context *ractx = avctx->priv_data;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- return 0;
-}
+ dsputil_init(&ractx->dsp, avctx);
-static void apply_window(float *tgt, const float *m1, const float *m2, int n)
-{
- while (n--)
- *tgt++ = *m1++ * *m2++;
+ avcodec_get_frame_defaults(&ractx->frame);
+ avctx->coded_frame = &ractx->frame;
+
+ return 0;
}
static void convolve(float *tgt, const float *src, int len, int n)
* @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_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(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->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8));
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)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
- float *out = data;
- int i, j;
+ 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;
}
- if (*data_size < 32*5*4)
- return -1;
+ /* get output buffer */
+ ractx->frame.nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
+ if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ out = (float *)ractx->frame.data[0];
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++) = 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 = (char *)out - (char *)data;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = ractx->frame;
+
return avctx->block_align;
}
.priv_data_size = sizeof(RA288Context),
.init = ra288_decode_init,
.decode = ra288_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};
projects / ffmpeg / blobdiff