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Merge remote-tracking branch 'qatar/master'
* qatar/master: (22 commits) g723.1: do not pass large structs by value g723.1: do not bounce intermediate values via memory g723.1: declare a variable in the block it is used g723.1: avoid saving/restoring excitation g723.1: avoid unnecessary memcpy() in residual_interp() g723.1: make postfilter write directly to output buffer g723.1: drop unnecessary variable buf_ptr in formant_postfilter() g723.1: make scale_vector() output to a separate buffer g723.1: make autocorr_max() work on an arbitrary buffer g723.1: do not needlessly use int64_t g723.1: use saturating addition functions g723.1: optimise scale_vector() g723.1: remove useless uses of MUL64() g723.1: remove unnecessary argument 'shift' from dot_product() g723.1: deobfuscate "(x << 4) - x" to "15 * x" celp: optimise ff_celp_lp_synthesis_filter() libavutil: add saturating addition functions cllc: Implement ARGB support cllc: Add support for QRGB cllc: Rename some funcs to represent what they actually do ... Conflicts: LICENSE libavcodec/g723_1.c libavcodec/x86/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
commit
d8c3170c9f
36
LICENSE
36
LICENSE
@ -1,5 +1,4 @@
|
||||
FFmpeg:
|
||||
-------
|
||||
|
||||
Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
|
||||
or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
|
||||
@ -51,18 +50,29 @@ for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
|
||||
COPYING.GPLv3 to learn the exact legal terms that apply in this case.
|
||||
|
||||
|
||||
external libraries:
|
||||
-------------------
|
||||
external libraries
|
||||
==================
|
||||
|
||||
Some external libraries, e.g. libx264, are under GPL and can be used in
|
||||
conjunction with FFmpeg. They require --enable-gpl to be passed to configure
|
||||
as well.
|
||||
FFmpeg can be combined with a number of external libraries, which sometimes
|
||||
affect the licensing of binaries resulting from the combination.
|
||||
|
||||
The OpenCORE external libraries are under the Apache License 2.0. That license
|
||||
is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
|
||||
those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
|
||||
version needs to be upgraded by passing --enable-version3 to configure.
|
||||
compatible libraries
|
||||
--------------------
|
||||
|
||||
The nonfree external libraries libfaac and libaacplus can be hooked up in FFmpeg.
|
||||
You need to pass --enable-nonfree to configure to enable it. Employ this option
|
||||
with care as FFmpeg then becomes nonfree and unredistributable.
|
||||
The libcdio, libx264, libxavs and libxvid libraries are under GPL. When
|
||||
combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
|
||||
passing --enable-gpl to configure.
|
||||
|
||||
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
|
||||
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
|
||||
version 3 of those licenses. So to combine these libraries with FFmpeg, the
|
||||
license version needs to be upgraded by passing --enable-version3 to configure.
|
||||
|
||||
incompatible libraries
|
||||
----------------------
|
||||
|
||||
The Fraunhofer AAC library, FAAC and aacplus are under licenses incompatible
|
||||
with all (L)GPL versions. Thus, unfortunately, since both licenses cannot be
|
||||
satisfied simultaneously, binaries resulting from the combination of FFmpeg
|
||||
with these libraries are nonfree und unredistributable. If you wish to enable
|
||||
any of these libraries nonetheless, pass --enable-nonfree to configure.
|
||||
|
@ -63,17 +63,16 @@ int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
|
||||
int i,n;
|
||||
|
||||
for (n = 0; n < buffer_length; n++) {
|
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int sum = rounder;
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||||
int sum = -rounder, sum1;
|
||||
for (i = 1; i <= filter_length; i++)
|
||||
sum -= filter_coeffs[i-1] * out[n-i];
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||||
sum += filter_coeffs[i-1] * out[n-i];
|
||||
|
||||
sum = ((sum >> 12) + in[n]) >> shift;
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sum1 = ((-sum >> 12) + in[n]) >> shift;
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sum = av_clip_int16(sum1);
|
||||
|
||||
if (stop_on_overflow && sum != sum1)
|
||||
return 1;
|
||||
|
||||
if (sum + 0x8000 > 0xFFFFU) {
|
||||
if (stop_on_overflow)
|
||||
return 1;
|
||||
sum = (sum >> 31) ^ 32767;
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||||
}
|
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out[n] = sum;
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}
|
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|
||||
|
@ -65,7 +65,7 @@ typedef struct g723_1_context {
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int pf_gain; ///< formant postfilter
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///< gain scaling unit memory
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int postfilter;
|
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int16_t audio[FRAME_LEN + LPC_ORDER];
|
||||
int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX];
|
||||
int16_t prev_data[HALF_FRAME_LEN];
|
||||
int16_t prev_weight_sig[PITCH_MAX];
|
||||
|
||||
@ -245,32 +245,27 @@ static int normalize_bits(int num, int width)
|
||||
|
||||
#define normalize_bits_int16(num) normalize_bits(num, 15)
|
||||
#define normalize_bits_int32(num) normalize_bits(num, 31)
|
||||
#define dot_product(a,b,c,d) (ff_dot_product(a,b,c)<<(d))
|
||||
|
||||
/**
|
||||
* Scale vector contents based on the largest of their absolutes.
|
||||
*/
|
||||
static int scale_vector(int16_t *vector, int length)
|
||||
static int scale_vector(int16_t *dst, const int16_t *vector, int length)
|
||||
{
|
||||
int bits, scale, max = 0;
|
||||
int bits, max = 0;
|
||||
int i;
|
||||
|
||||
const int16_t shift_table[16] = {
|
||||
0x0001, 0x0002, 0x0004, 0x0008, 0x0010, 0x0020, 0x0040, 0x0080,
|
||||
0x0100, 0x0200, 0x0400, 0x0800, 0x1000, 0x2000, 0x4000, 0x7fff
|
||||
};
|
||||
|
||||
for (i = 0; i < length; i++)
|
||||
max = FFMAX(max, FFABS(vector[i]));
|
||||
max |= FFABS(vector[i]);
|
||||
|
||||
max = FFMIN(max, 0x7FFF);
|
||||
bits = normalize_bits(max, 15);
|
||||
scale = shift_table[bits];
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||||
|
||||
for (i = 0; i < length; i++) {
|
||||
av_assert2(av_clipl_int32(vector[i] * (int64_t)scale << 1) == vector[i] * (int64_t)scale << 1);
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||||
vector[i] = (vector[i] * scale) >> 3;
|
||||
}
|
||||
if (bits == 15)
|
||||
for (i = 0; i < length; i++)
|
||||
dst[i] = vector[i] * 0x7fff >> 3;
|
||||
else
|
||||
for (i = 0; i < length; i++)
|
||||
dst[i] = vector[i] << bits >> 3;
|
||||
|
||||
return bits - 3;
|
||||
}
|
||||
@ -369,11 +364,11 @@ static void lsp2lpc(int16_t *lpc)
|
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for (j = 0; j < LPC_ORDER; j++) {
|
||||
int index = lpc[j] >> 7;
|
||||
int offset = lpc[j] & 0x7f;
|
||||
int64_t temp1 = cos_tab[index] << 16;
|
||||
int temp1 = cos_tab[index] << 16;
|
||||
int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
|
||||
((offset << 8) + 0x80) << 1;
|
||||
|
||||
lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 16);
|
||||
lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
|
||||
}
|
||||
|
||||
/*
|
||||
@ -473,7 +468,7 @@ static void gen_dirac_train(int16_t *buf, int pitch_lag)
|
||||
* @param pitch_lag closed loop pitch lag
|
||||
* @param index current subframe index
|
||||
*/
|
||||
static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm,
|
||||
static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
|
||||
enum Rate cur_rate, int pitch_lag, int index)
|
||||
{
|
||||
int temp, i, j;
|
||||
@ -481,34 +476,34 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm,
|
||||
memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
|
||||
|
||||
if (cur_rate == RATE_6300) {
|
||||
if (subfrm.pulse_pos >= max_pos[index])
|
||||
if (subfrm->pulse_pos >= max_pos[index])
|
||||
return;
|
||||
|
||||
/* Decode amplitudes and positions */
|
||||
j = PULSE_MAX - pulses[index];
|
||||
temp = subfrm.pulse_pos;
|
||||
temp = subfrm->pulse_pos;
|
||||
for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
|
||||
temp -= combinatorial_table[j][i];
|
||||
if (temp >= 0)
|
||||
continue;
|
||||
temp += combinatorial_table[j++][i];
|
||||
if (subfrm.pulse_sign & (1 << (PULSE_MAX - j))) {
|
||||
vector[subfrm.grid_index + GRID_SIZE * i] =
|
||||
-fixed_cb_gain[subfrm.amp_index];
|
||||
if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
|
||||
vector[subfrm->grid_index + GRID_SIZE * i] =
|
||||
-fixed_cb_gain[subfrm->amp_index];
|
||||
} else {
|
||||
vector[subfrm.grid_index + GRID_SIZE * i] =
|
||||
fixed_cb_gain[subfrm.amp_index];
|
||||
vector[subfrm->grid_index + GRID_SIZE * i] =
|
||||
fixed_cb_gain[subfrm->amp_index];
|
||||
}
|
||||
if (j == PULSE_MAX)
|
||||
break;
|
||||
}
|
||||
if (subfrm.dirac_train == 1)
|
||||
if (subfrm->dirac_train == 1)
|
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gen_dirac_train(vector, pitch_lag);
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} else { /* 5300 bps */
|
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int cb_gain = fixed_cb_gain[subfrm.amp_index];
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int cb_shift = subfrm.grid_index;
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int cb_sign = subfrm.pulse_sign;
|
||||
int cb_pos = subfrm.pulse_pos;
|
||||
int cb_gain = fixed_cb_gain[subfrm->amp_index];
|
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int cb_shift = subfrm->grid_index;
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int cb_sign = subfrm->pulse_sign;
|
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int cb_pos = subfrm->pulse_pos;
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int offset, beta, lag;
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|
||||
for (i = 0; i < 8; i += 2) {
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@ -519,9 +514,9 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm,
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}
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|
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/* Enhance harmonic components */
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lag = pitch_contrib[subfrm.ad_cb_gain << 1] + pitch_lag +
|
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subfrm.ad_cb_lag - 1;
|
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beta = pitch_contrib[(subfrm.ad_cb_gain << 1) + 1];
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lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
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subfrm->ad_cb_lag - 1;
|
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beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
|
||||
|
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if (lag < SUBFRAME_LEN - 2) {
|
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for (i = lag; i < SUBFRAME_LEN; i++)
|
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@ -546,19 +541,25 @@ static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
|
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residual[i] = prev_excitation[offset + (i - 2) % lag];
|
||||
}
|
||||
|
||||
static int dot_product(const int16_t *a, const int16_t *b, int length)
|
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{
|
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int sum = ff_dot_product(a,b,length);
|
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return av_sat_add32(sum, sum);
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}
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|
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/**
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* Generate adaptive codebook excitation.
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*/
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static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
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int pitch_lag, G723_1_Subframe subfrm,
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int pitch_lag, G723_1_Subframe *subfrm,
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enum Rate cur_rate)
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{
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int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
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const int16_t *cb_ptr;
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int lag = pitch_lag + subfrm.ad_cb_lag - 1;
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int lag = pitch_lag + subfrm->ad_cb_lag - 1;
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|
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int i;
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int64_t sum;
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int sum;
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|
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get_residual(residual, prev_excitation, lag);
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@ -569,28 +570,27 @@ static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
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cb_ptr = adaptive_cb_gain170;
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/* Calculate adaptive vector */
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cb_ptr += subfrm.ad_cb_gain * 20;
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cb_ptr += subfrm->ad_cb_gain * 20;
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for (i = 0; i < SUBFRAME_LEN; i++) {
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sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
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vector[i] = av_clipl_int32((sum << 2) + (1 << 15)) >> 16;
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vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
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}
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}
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|
||||
/**
|
||||
* Estimate maximum auto-correlation around pitch lag.
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*
|
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* @param p the context
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* @param buf buffer with offset applied
|
||||
* @param offset offset of the excitation vector
|
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* @param ccr_max pointer to the maximum auto-correlation
|
||||
* @param pitch_lag decoded pitch lag
|
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* @param length length of autocorrelation
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* @param dir forward lag(1) / backward lag(-1)
|
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*/
|
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static int autocorr_max(G723_1_Context *p, int offset, int *ccr_max,
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||||
static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
|
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int pitch_lag, int length, int dir)
|
||||
{
|
||||
int limit, ccr, lag = 0;
|
||||
int16_t *buf = p->excitation + offset;
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int i;
|
||||
|
||||
pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
|
||||
@ -600,7 +600,7 @@ static int autocorr_max(G723_1_Context *p, int offset, int *ccr_max,
|
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limit = pitch_lag + 3;
|
||||
|
||||
for (i = pitch_lag - 3; i <= limit; i++) {
|
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ccr = ff_dot_product(buf, buf + dir * i, length)<<1;
|
||||
ccr = dot_product(buf, buf + dir * i, length);
|
||||
|
||||
if (ccr > *ccr_max) {
|
||||
*ccr_max = ccr;
|
||||
@ -624,7 +624,7 @@ static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
|
||||
int tgt_eng, int ccr, int res_eng)
|
||||
{
|
||||
int pf_residual; /* square of postfiltered residual */
|
||||
int64_t temp1, temp2;
|
||||
int temp1, temp2;
|
||||
|
||||
ppf->index = lag;
|
||||
|
||||
@ -641,7 +641,7 @@ static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
|
||||
/* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
|
||||
temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
|
||||
temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
|
||||
pf_residual = av_clipl_int32(temp1 + temp2 + (1 << 15)) >> 16;
|
||||
pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
|
||||
|
||||
if (tgt_eng >= pf_residual << 1) {
|
||||
temp1 = 0x7fff;
|
||||
@ -674,7 +674,7 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
|
||||
|
||||
int16_t scale;
|
||||
int i;
|
||||
int64_t temp1, temp2;
|
||||
int temp1, temp2;
|
||||
|
||||
/*
|
||||
* 0 - target energy
|
||||
@ -684,10 +684,10 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
|
||||
* 4 - backward residual energy
|
||||
*/
|
||||
int energy[5] = {0, 0, 0, 0, 0};
|
||||
int16_t *buf = p->excitation + offset;
|
||||
int fwd_lag = autocorr_max(p, offset, &energy[1], pitch_lag,
|
||||
int16_t *buf = p->audio + LPC_ORDER + offset;
|
||||
int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
|
||||
SUBFRAME_LEN, 1);
|
||||
int back_lag = autocorr_max(p, offset, &energy[3], pitch_lag,
|
||||
int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
|
||||
SUBFRAME_LEN, -1);
|
||||
|
||||
ppf->index = 0;
|
||||
@ -699,17 +699,15 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
|
||||
return;
|
||||
|
||||
/* Compute target energy */
|
||||
energy[0] = ff_dot_product(buf, buf, SUBFRAME_LEN)<<1;
|
||||
energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
|
||||
|
||||
/* Compute forward residual energy */
|
||||
if (fwd_lag)
|
||||
energy[2] = ff_dot_product(buf + fwd_lag, buf + fwd_lag,
|
||||
SUBFRAME_LEN)<<1;
|
||||
energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
|
||||
|
||||
/* Compute backward residual energy */
|
||||
if (back_lag)
|
||||
energy[4] = ff_dot_product(buf - back_lag, buf - back_lag,
|
||||
SUBFRAME_LEN)<<1;
|
||||
energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
|
||||
|
||||
/* Normalize and shorten */
|
||||
temp1 = 0;
|
||||
@ -758,28 +756,28 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag,
|
||||
int *exc_eng, int *scale)
|
||||
{
|
||||
int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
|
||||
int16_t *buf = p->excitation + offset;
|
||||
int16_t *buf = p->audio + LPC_ORDER;
|
||||
|
||||
int index, ccr, tgt_eng, best_eng, temp;
|
||||
|
||||
*scale = scale_vector(p->excitation, FRAME_LEN + PITCH_MAX);
|
||||
*scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
|
||||
buf += offset;
|
||||
|
||||
/* Compute maximum backward cross-correlation */
|
||||
ccr = 0;
|
||||
index = autocorr_max(p, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
|
||||
ccr = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16;
|
||||
index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
|
||||
ccr = av_sat_add32(ccr, 1 << 15) >> 16;
|
||||
|
||||
/* Compute target energy */
|
||||
tgt_eng = ff_dot_product(buf, buf, SUBFRAME_LEN * 2)<<1;
|
||||
*exc_eng = av_clipl_int32(tgt_eng + (1 << 15)) >> 16;
|
||||
tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2);
|
||||
*exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
|
||||
|
||||
if (ccr <= 0)
|
||||
return 0;
|
||||
|
||||
/* Compute best energy */
|
||||
best_eng = ff_dot_product(buf - index, buf - index,
|
||||
SUBFRAME_LEN * 2)<<1;
|
||||
best_eng = av_clipl_int32((int64_t)best_eng + (1 << 15)) >> 16;
|
||||
best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
|
||||
best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
|
||||
|
||||
temp = best_eng * *exc_eng >> 3;
|
||||
|
||||
@ -806,10 +804,9 @@ static void residual_interp(int16_t *buf, int16_t *out, int lag,
|
||||
int16_t *vector_ptr = buf + PITCH_MAX;
|
||||
/* Attenuate */
|
||||
for (i = 0; i < lag; i++)
|
||||
vector_ptr[i - lag] = vector_ptr[i - lag] * 3 >> 2;
|
||||
av_memcpy_backptr((uint8_t*)vector_ptr, lag * sizeof(*vector_ptr),
|
||||
FRAME_LEN * sizeof(*vector_ptr));
|
||||
memcpy(out, vector_ptr, FRAME_LEN * sizeof(*vector_ptr));
|
||||
out[i] = vector_ptr[i - lag] * 3 >> 2;
|
||||
av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
|
||||
(FRAME_LEN - lag) * sizeof(*out));
|
||||
} else { /* Unvoiced */
|
||||
for (i = 0; i < FRAME_LEN; i++) {
|
||||
*rseed = *rseed * 521 + 259;
|
||||
@ -861,9 +858,9 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
|
||||
num = energy;
|
||||
denom = 0;
|
||||
for (i = 0; i < SUBFRAME_LEN; i++) {
|
||||
int64_t temp = buf[i] >> 2;
|
||||
temp = av_clipl_int32(MUL64(temp, temp) << 1);
|
||||
denom = av_clipl_int32(denom + temp);
|
||||
int temp = buf[i] >> 2;
|
||||
temp *= temp;
|
||||
denom = av_sat_dadd32(denom, temp);
|
||||
}
|
||||
|
||||
if (num && denom) {
|
||||
@ -882,7 +879,7 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
|
||||
}
|
||||
|
||||
for (i = 0; i < SUBFRAME_LEN; i++) {
|
||||
p->pf_gain = ((p->pf_gain << 4) - p->pf_gain + gain + (1 << 3)) >> 4;
|
||||
p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
|
||||
buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
|
||||
(1 << 10)) >> 11);
|
||||
}
|
||||
@ -893,11 +890,13 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
|
||||
*
|
||||
* @param p the context
|
||||
* @param lpc quantized lpc coefficients
|
||||
* @param buf output buffer
|
||||
* @param buf input buffer
|
||||
* @param dst output buffer
|
||||
*/
|
||||
static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
|
||||
static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
|
||||
int16_t *buf, int16_t *dst)
|
||||
{
|
||||
int16_t filter_coef[2][LPC_ORDER], *buf_ptr;
|
||||
int16_t filter_coef[2][LPC_ORDER];
|
||||
int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
|
||||
int i, j, k;
|
||||
|
||||
@ -919,23 +918,19 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
|
||||
memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
|
||||
memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
|
||||
|
||||
buf_ptr = buf + LPC_ORDER;
|
||||
buf += LPC_ORDER;
|
||||
signal_ptr = filter_signal + LPC_ORDER;
|
||||
for (i = 0; i < SUBFRAMES; i++) {
|
||||
int16_t temp_vector[SUBFRAME_LEN];
|
||||
int temp;
|
||||
int auto_corr[2];
|
||||
int scale, energy;
|
||||
|
||||
/* Normalize */
|
||||
memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(*temp_vector));
|
||||
scale = scale_vector(temp_vector, SUBFRAME_LEN);
|
||||
scale = scale_vector(dst, buf, SUBFRAME_LEN);
|
||||
|
||||
/* Compute auto correlation coefficients */
|
||||
auto_corr[0] = ff_dot_product(temp_vector, temp_vector + 1,
|
||||
SUBFRAME_LEN - 1)<<1;
|
||||
auto_corr[1] = ff_dot_product(temp_vector, temp_vector,
|
||||
SUBFRAME_LEN)<<1;
|
||||
auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
|
||||
auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN);
|
||||
|
||||
/* Compute reflection coefficient */
|
||||
temp = auto_corr[1] >> 16;
|
||||
@ -947,9 +942,8 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
|
||||
|
||||
/* Compensation filter */
|
||||
for (j = 0; j < SUBFRAME_LEN; j++) {
|
||||
buf_ptr[j] = av_clipl_int32((int64_t)signal_ptr[j] +
|
||||
((signal_ptr[j - 1] >> 16) *
|
||||
temp << 1)) >> 16;
|
||||
dst[j] = av_sat_dadd32(signal_ptr[j],
|
||||
(signal_ptr[j - 1] >> 16) * temp) >> 16;
|
||||
}
|
||||
|
||||
/* Compute normalized signal energy */
|
||||
@ -959,10 +953,11 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
|
||||
} else
|
||||
energy = auto_corr[1] >> temp;
|
||||
|
||||
gain_scale(p, buf_ptr, energy);
|
||||
gain_scale(p, dst, energy);
|
||||
|
||||
buf_ptr += SUBFRAME_LEN;
|
||||
buf += SUBFRAME_LEN;
|
||||
signal_ptr += SUBFRAME_LEN;
|
||||
dst += SUBFRAME_LEN;
|
||||
}
|
||||
}
|
||||
|
||||
@ -978,9 +973,9 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
|
||||
int16_t cur_lsp[LPC_ORDER];
|
||||
int16_t lpc[SUBFRAMES * LPC_ORDER];
|
||||
int16_t acb_vector[SUBFRAME_LEN];
|
||||
int16_t *vector_ptr;
|
||||
int16_t *out;
|
||||
int bad_frame = 0, i, j, ret;
|
||||
int16_t *audio = p->audio;
|
||||
|
||||
if (buf_size < frame_size[dec_mode]) {
|
||||
if (buf_size)
|
||||
@ -1022,48 +1017,38 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
|
||||
/* Generate the excitation for the frame */
|
||||
memcpy(p->excitation, p->prev_excitation,
|
||||
PITCH_MAX * sizeof(*p->excitation));
|
||||
vector_ptr = p->excitation + PITCH_MAX;
|
||||
if (!p->erased_frames) {
|
||||
int16_t *vector_ptr = p->excitation + PITCH_MAX;
|
||||
|
||||
/* Update interpolation gain memory */
|
||||
p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
|
||||
p->subframe[3].amp_index) >> 1];
|
||||
for (i = 0; i < SUBFRAMES; i++) {
|
||||
gen_fcb_excitation(vector_ptr, p->subframe[i], p->cur_rate,
|
||||
gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
|
||||
p->pitch_lag[i >> 1], i);
|
||||
gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
|
||||
p->pitch_lag[i >> 1], p->subframe[i],
|
||||
p->pitch_lag[i >> 1], &p->subframe[i],
|
||||
p->cur_rate);
|
||||
/* Get the total excitation */
|
||||
for (j = 0; j < SUBFRAME_LEN; j++) {
|
||||
vector_ptr[j] = av_clip_int16(vector_ptr[j] << 1);
|
||||
vector_ptr[j] = av_clip_int16(vector_ptr[j] +
|
||||
acb_vector[j]);
|
||||
int v = av_clip_int16(vector_ptr[j] << 1);
|
||||
vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
|
||||
}
|
||||
vector_ptr += SUBFRAME_LEN;
|
||||
}
|
||||
|
||||
vector_ptr = p->excitation + PITCH_MAX;
|
||||
|
||||
/* Save the excitation */
|
||||
memcpy(p->audio + LPC_ORDER, vector_ptr, FRAME_LEN * sizeof(*p->audio));
|
||||
|
||||
p->interp_index = comp_interp_index(p, p->pitch_lag[1],
|
||||
&p->sid_gain, &p->cur_gain);
|
||||
|
||||
/* Peform pitch postfiltering */
|
||||
if (p->postfilter) {
|
||||
i = PITCH_MAX;
|
||||
for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
||||
comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
|
||||
ppf + j, p->cur_rate);
|
||||
}
|
||||
|
||||
/* Restore the original excitation */
|
||||
memcpy(p->excitation, p->prev_excitation,
|
||||
PITCH_MAX * sizeof(*p->excitation));
|
||||
memcpy(vector_ptr, p->audio + LPC_ORDER, FRAME_LEN * sizeof(*vector_ptr));
|
||||
|
||||
/* Peform pitch postfiltering */
|
||||
if (p->postfilter)
|
||||
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
||||
ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
|
||||
vector_ptr + i,
|
||||
@ -1071,24 +1056,35 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
|
||||
ppf[j].sc_gain,
|
||||
ppf[j].opt_gain,
|
||||
1 << 14, 15, SUBFRAME_LEN);
|
||||
} else {
|
||||
audio = vector_ptr - LPC_ORDER;
|
||||
}
|
||||
|
||||
/* Save the excitation for the next frame */
|
||||
memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
|
||||
PITCH_MAX * sizeof(*p->excitation));
|
||||
} else {
|
||||
p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
|
||||
if (p->erased_frames == 3) {
|
||||
/* Mute output */
|
||||
memset(p->excitation, 0,
|
||||
(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
|
||||
memset(p->prev_excitation, 0,
|
||||
PITCH_MAX * sizeof(*p->excitation));
|
||||
memset(p->frame.data[0], 0,
|
||||
(FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
|
||||
} else {
|
||||
int16_t *buf = p->audio + LPC_ORDER;
|
||||
|
||||
/* Regenerate frame */
|
||||
residual_interp(p->excitation, p->audio + LPC_ORDER, p->interp_index,
|
||||
residual_interp(p->excitation, buf, p->interp_index,
|
||||
p->interp_gain, &p->random_seed);
|
||||
|
||||
/* Save the excitation for the next frame */
|
||||
memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
|
||||
PITCH_MAX * sizeof(*p->excitation));
|
||||
}
|
||||
}
|
||||
/* Save the excitation for the next frame */
|
||||
memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
|
||||
PITCH_MAX * sizeof(*p->excitation));
|
||||
} else {
|
||||
memset(out, 0, FRAME_LEN * 2);
|
||||
av_log(avctx, AV_LOG_WARNING,
|
||||
@ -1104,13 +1100,12 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
|
||||
memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
|
||||
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
|
||||
ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
|
||||
p->audio + i, SUBFRAME_LEN, LPC_ORDER,
|
||||
audio + i, SUBFRAME_LEN, LPC_ORDER,
|
||||
0, 1, 1 << 12);
|
||||
memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
|
||||
|
||||
if (p->postfilter) {
|
||||
formant_postfilter(p, lpc, p->audio);
|
||||
memcpy(p->frame.data[0], p->audio + LPC_ORDER, FRAME_LEN * 2);
|
||||
formant_postfilter(p, lpc, p->audio, out);
|
||||
} else { // if output is not postfiltered it should be scaled by 2
|
||||
for (i = 0; i < FRAME_LEN; i++)
|
||||
out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
|
||||
@ -1214,14 +1209,14 @@ static void comp_autocorr(int16_t *buf, int16_t *autocorr)
|
||||
int16_t vector[LPC_FRAME];
|
||||
|
||||
memcpy(vector, buf, LPC_FRAME * sizeof(int16_t));
|
||||
scale_vector(vector, LPC_FRAME);
|
||||
scale_vector(vector, vector, LPC_FRAME);
|
||||
|
||||
/* Apply the Hamming window */
|
||||
for (i = 0; i < LPC_FRAME; i++)
|
||||
vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
|
||||
|
||||
/* Compute the first autocorrelation coefficient */
|
||||
temp = dot_product(vector, vector, LPC_FRAME, 0);
|
||||
temp = ff_dot_product(vector, vector, LPC_FRAME);
|
||||
|
||||
/* Apply a white noise correlation factor of (1025/1024) */
|
||||
temp += temp >> 10;
|
||||
@ -1236,7 +1231,7 @@ static void comp_autocorr(int16_t *buf, int16_t *autocorr)
|
||||
memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
|
||||
} else {
|
||||
for (i = 1; i <= LPC_ORDER; i++) {
|
||||
temp = dot_product(vector, vector + i, LPC_FRAME - i, 0);
|
||||
temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
|
||||
temp = MULL2((temp << scale), binomial_window[i - 1]);
|
||||
autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
|
||||
}
|
||||
@ -1416,8 +1411,8 @@ static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
|
||||
temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
|
||||
(1 << 14)) >> 15;\
|
||||
}\
|
||||
error = dot_product(lsp + (offset), temp, size, 1) << 1;\
|
||||
error -= dot_product(lsp_band##num[i], temp, size, 1);\
|
||||
error = dot_product(lsp + (offset), temp, size) << 1;\
|
||||
error -= dot_product(lsp_band##num[i], temp, size);\
|
||||
if (error > max) {\
|
||||
max = error;\
|
||||
lsp_index[num] = i;\
|
||||
@ -1522,7 +1517,7 @@ static int estimate_pitch(int16_t *buf, int start)
|
||||
|
||||
int i;
|
||||
|
||||
orig_eng = dot_product(buf + offset, buf + offset, HALF_FRAME_LEN, 0);
|
||||
orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
|
||||
|
||||
for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
|
||||
offset--;
|
||||
@ -1530,7 +1525,7 @@ static int estimate_pitch(int16_t *buf, int start)
|
||||
/* Update energy and compute correlation */
|
||||
orig_eng += buf[offset] * buf[offset] -
|
||||
buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
|
||||
ccr = dot_product(buf + start, buf + offset, HALF_FRAME_LEN, 0);
|
||||
ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
|
||||
if (ccr <= 0)
|
||||
continue;
|
||||
|
||||
@ -1591,13 +1586,13 @@ static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
|
||||
|
||||
for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
|
||||
/* Compute residual energy */
|
||||
energy[i << 1] = dot_product(buf - j, buf - j, SUBFRAME_LEN, 0);
|
||||
energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
|
||||
/* Compute correlation */
|
||||
energy[(i << 1) + 1] = dot_product(buf, buf - j, SUBFRAME_LEN, 0);
|
||||
energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
|
||||
}
|
||||
|
||||
/* Compute target energy */
|
||||
energy[14] = dot_product(buf, buf, SUBFRAME_LEN, 0);
|
||||
energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
|
||||
|
||||
/* Normalize */
|
||||
max = 0;
|
||||
@ -1778,19 +1773,19 @@ static void acb_search(G723_1_Context *p, int16_t *residual,
|
||||
|
||||
/* Compute crosscorrelation with the signal */
|
||||
for (j = 0; j < PITCH_ORDER; j++) {
|
||||
temp = dot_product(buf, flt_buf[j], SUBFRAME_LEN, 0);
|
||||
temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
|
||||
ccr_buf[count++] = av_clipl_int32(temp << 1);
|
||||
}
|
||||
|
||||
/* Compute energies */
|
||||
for (j = 0; j < PITCH_ORDER; j++) {
|
||||
ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
|
||||
SUBFRAME_LEN, 1);
|
||||
SUBFRAME_LEN);
|
||||
}
|
||||
|
||||
for (j = 1; j < PITCH_ORDER; j++) {
|
||||
for (k = 0; k < j; k++) {
|
||||
temp = dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN, 0);
|
||||
temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
|
||||
ccr_buf[count++] = av_clipl_int32(temp<<2);
|
||||
}
|
||||
}
|
||||
@ -1893,20 +1888,20 @@ static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
|
||||
temp_corr[i] = impulse_r[i] >> 1;
|
||||
|
||||
/* Compute impulse response autocorrelation */
|
||||
temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN, 1);
|
||||
temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
|
||||
|
||||
scale = normalize_bits_int32(temp);
|
||||
impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
|
||||
|
||||
for (i = 1; i < SUBFRAME_LEN; i++) {
|
||||
temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i, 1);
|
||||
temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
|
||||
impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
|
||||
}
|
||||
|
||||
/* Compute crosscorrelation of impulse response with residual signal */
|
||||
scale -= 4;
|
||||
for (i = 0; i < SUBFRAME_LEN; i++){
|
||||
temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i, 1);
|
||||
temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
|
||||
if (scale < 0)
|
||||
ccr1[i] = temp >> -scale;
|
||||
else
|
||||
@ -2185,7 +2180,7 @@ static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
||||
memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
|
||||
memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
|
||||
|
||||
scale_vector(vector, FRAME_LEN + PITCH_MAX);
|
||||
scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
|
||||
|
||||
p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
|
||||
p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
|
||||
@ -2237,14 +2232,14 @@ static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
||||
|
||||
acb_search(p, residual, impulse_resp, in, i);
|
||||
gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
|
||||
p->subframe[i], p->cur_rate);
|
||||
&p->subframe[i], p->cur_rate);
|
||||
sub_acb_contrib(residual, impulse_resp, in);
|
||||
|
||||
fcb_search(p, impulse_resp, in, i);
|
||||
|
||||
/* Reconstruct the excitation */
|
||||
gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
|
||||
p->subframe[i], RATE_6300);
|
||||
&p->subframe[i], RATE_6300);
|
||||
|
||||
memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
|
||||
sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
|
||||
|
@ -41,7 +41,7 @@ YASM-OBJS-$(CONFIG_AAC_DECODER) += x86/sbrdsp.o
|
||||
YASM-OBJS-$(CONFIG_AC3DSP) += x86/ac3dsp.o
|
||||
YASM-OBJS-$(CONFIG_DCT) += x86/dct32_sse.o
|
||||
YASM-OBJS-$(CONFIG_DIRAC_DECODER) += x86/diracdsp_mmx.o x86/diracdsp_yasm.o
|
||||
YASM-OBJS-$(CONFIG_ENCODERS) += x86/dsputilenc_yasm.o
|
||||
YASM-OBJS-$(CONFIG_ENCODERS) += x86/dsputilenc.o
|
||||
YASM-OBJS-$(CONFIG_FFT) += x86/fft_mmx.o \
|
||||
$(YASM-OBJS-FFT-yes)
|
||||
|
||||
@ -65,11 +65,11 @@ YASM-OBJS-$(CONFIG_RV30_DECODER) += x86/rv34dsp.o
|
||||
YASM-OBJS-$(CONFIG_RV40_DECODER) += x86/rv34dsp.o \
|
||||
x86/rv40dsp.o
|
||||
YASM-OBJS-$(CONFIG_V210_DECODER) += x86/v210.o
|
||||
YASM-OBJS-$(CONFIG_VC1_DECODER) += x86/vc1dsp_yasm.o
|
||||
YASM-OBJS-$(CONFIG_VC1_DECODER) += x86/vc1dsp.o
|
||||
YASM-OBJS-$(CONFIG_VP3DSP) += x86/vp3dsp.o
|
||||
YASM-OBJS-$(CONFIG_VP6_DECODER) += x86/vp56dsp.o
|
||||
YASM-OBJS-$(CONFIG_VP8_DECODER) += x86/vp8dsp.o
|
||||
|
||||
YASM-OBJS += x86/dsputil_yasm.o \
|
||||
YASM-OBJS += x86/dsputil.o \
|
||||
x86/deinterlace.o \
|
||||
x86/fmtconvert.o \
|
||||
|
@ -83,6 +83,21 @@ static av_always_inline av_const unsigned av_clip_uintp2_arm(int a, int p)
|
||||
return x;
|
||||
}
|
||||
|
||||
#define av_sat_add32 av_sat_add32_arm
|
||||
static av_always_inline int av_sat_add32_arm(int a, int b)
|
||||
{
|
||||
int r;
|
||||
__asm__ ("qadd %0, %1, %2" : "=r"(r) : "r"(a), "r"(b));
|
||||
return r;
|
||||
}
|
||||
|
||||
#define av_sat_dadd32 av_sat_dadd32_arm
|
||||
static av_always_inline int av_sat_dadd32_arm(int a, int b)
|
||||
{
|
||||
int r;
|
||||
__asm__ ("qdadd %0, %1, %2" : "=r"(r) : "r"(a), "r"(b));
|
||||
return r;
|
||||
}
|
||||
|
||||
#else /* HAVE_ARMV6 */
|
||||
|
||||
|
@ -186,6 +186,30 @@ static av_always_inline av_const unsigned av_clip_uintp2_c(int a, int p)
|
||||
else return a;
|
||||
}
|
||||
|
||||
/**
|
||||
* Add two signed 32-bit values with saturation.
|
||||
*
|
||||
* @param a one value
|
||||
* @param b another value
|
||||
* @return sum with signed saturation
|
||||
*/
|
||||
static av_always_inline int av_sat_add32_c(int a, int b)
|
||||
{
|
||||
return av_clipl_int32((int64_t)a + b);
|
||||
}
|
||||
|
||||
/**
|
||||
* Add a doubled value to another value with saturation at both stages.
|
||||
*
|
||||
* @param a first value
|
||||
* @param b value doubled and added to a
|
||||
* @return sum with signed saturation
|
||||
*/
|
||||
static av_always_inline int av_sat_dadd32_c(int a, int b)
|
||||
{
|
||||
return av_sat_add32(a, av_sat_add32(b, b));
|
||||
}
|
||||
|
||||
/**
|
||||
* Clip a float value into the amin-amax range.
|
||||
* @param a value to clip
|
||||
@ -392,6 +416,12 @@ static av_always_inline av_const int av_popcount64_c(uint64_t x)
|
||||
#ifndef av_clip_uintp2
|
||||
# define av_clip_uintp2 av_clip_uintp2_c
|
||||
#endif
|
||||
#ifndef av_sat_add32
|
||||
# define av_sat_add32 av_sat_add32_c
|
||||
#endif
|
||||
#ifndef av_sat_dadd32
|
||||
# define av_sat_dadd32 av_sat_dadd32_c
|
||||
#endif
|
||||
#ifndef av_clipf
|
||||
# define av_clipf av_clipf_c
|
||||
#endif
|
||||
|
Loading…
Reference in New Issue
Block a user