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Add a shift parameter to celp_lp_synthesis_filter for reuse by the G.723.1 decoder

This commit is contained in:
Mohamed Naufal Basheer 2011-03-17 23:56:47 +01:00 committed by Michael Niedermayer
parent ecf31a68fd
commit bcc67dffa0
5 changed files with 10 additions and 6 deletions
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4
libavcodec/celp_filters.c
View File

@ -58,7 +58,7 @@ void ff_celp_circ_addf(float *out, const float *in,
int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
const int16_t *in, int buffer_length,
int filter_length, int stop_on_overflow,
int rounder)
int shift, int rounder)
{
int i,n;
@ -67,7 +67,7 @@ int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
for (i = 1; i <= filter_length; i++)
sum -= filter_coeffs[i-1] * out[n-i];
sum = (sum >> 12) + in[n];
sum = ((sum >> 12) + in[n]) >> shift;
if (sum + 0x8000 > 0xFFFFU) {
if (stop_on_overflow)

3
libavcodec/celp_filters.h
View File

@ -63,6 +63,7 @@ void ff_celp_circ_addf(float *out, const float *in,
* @param filter_length filter length (10 for 10th order LP filter)
* @param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
* @param shift the result is shifted right by this value
* @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
*
* @return 1 if overflow occurred, 0 - otherwise
@ -75,7 +76,7 @@ void ff_celp_circ_addf(float *out, const float *in,
int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
const int16_t *in, int buffer_length,
int filter_length, int stop_on_overflow,
int rounder);
int shift, int rounder);
/**
* LP synthesis filter.

3
libavcodec/g729dec.c
View File

@ -604,6 +604,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
SUBFRAME_SIZE,
10,
1,
0,
0x800))
/* Overflow occured, downscale excitation signal... */
for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
@ -625,6 +626,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
SUBFRAME_SIZE,
10,
0,
0,
0x800);
} else {
ff_celp_lp_synthesis_filter(
@ -634,6 +636,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
SUBFRAME_SIZE,
10,
0,
0,
0x800);
}
/* Save data (without postfilter) for use in next subframe. */

4
libavcodec/g729postfilter.c
View File

@ -433,7 +433,7 @@ static int16_t get_tilt_comp(DSPContext *dsp, int16_t *lp_gn,
lp_gn[10] = 4096; //1.0 in (3.12)
/* Apply 1/A(z/FORMANT_PP_FACTOR_DEN) filter to hf. */
ff_celp_lp_synthesis_filter(lp_gn + 11, lp_gd + 1, lp_gn + 11, 22, 10, 0, 0x800);
ff_celp_lp_synthesis_filter(lp_gn + 11, lp_gd + 1, lp_gn + 11, 22, 10, 0, 0, 0x800);
/* Now lp_gn (starting with 10) contains impulse response
of A(z/FORMANT_PP_FACTOR_NUM)/A(z/FORMANT_PP_FACTOR_DEN) filter. */
@ -554,7 +554,7 @@ void ff_g729_postfilter(DSPContext *dsp, int16_t* ht_prev_data, int* voicing,
/* Apply second half of short-term postfilter: 1/A(z/FORMANT_PP_FACTOR_DEN) */
ff_celp_lp_synthesis_filter(pos_filter_data + 10, lp_gd + 1,
residual_filt_buf + 10,
subframe_size, 10, 0, 0x800);
subframe_size, 10, 0, 0, 0x800);
memcpy(pos_filter_data, pos_filter_data + subframe_size, 10 * sizeof(int16_t));
*ht_prev_data = apply_tilt_comp(speech, pos_filter_data + 10, tilt_comp_coeff,

2
libavcodec/ra144.c
View File

@ -1715,6 +1715,6 @@ void ff_subblock_synthesis(RA144Context *ractx, const uint16_t *lpc_coefs,
10*sizeof(*ractx->curr_sblock));
if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
block, BLOCKSIZE, 10, 1, 0xfff))
block, BLOCKSIZE, 10, 1, 0, 0xfff))
memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
}