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-rw-r--r--LICENSE36
-rw-r--r--libavcodec/celp_filters.c15
-rw-r--r--libavcodec/g723_1.c255
-rw-r--r--libavcodec/x86/Makefile6
-rw-r--r--libavcodec/x86/dsputil.asm (renamed from libavcodec/x86/dsputil_yasm.asm)0
-rw-r--r--libavcodec/x86/dsputilenc.asm (renamed from libavcodec/x86/dsputilenc_yasm.asm)0
-rw-r--r--libavcodec/x86/vc1dsp.asm (renamed from libavcodec/x86/vc1dsp_yasm.asm)0
-rw-r--r--libavutil/arm/intmath.h15
-rw-r--r--libavutil/common.h30
9 files changed, 203 insertions, 154 deletions
diff --git a/LICENSE b/LICENSE
index 1607742b0c..7dff535a72 100644
--- a/LICENSE
+++ b/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.
diff --git a/libavcodec/celp_filters.c b/libavcodec/celp_filters.c
index 8047a78452..cf2198d325 100644
--- a/libavcodec/celp_filters.c
+++ b/libavcodec/celp_filters.c
@@ -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++) {
- int sum = rounder;
+ int sum = -rounder, sum1;
for (i = 1; i <= filter_length; i++)
- sum -= filter_coeffs[i-1] * out[n-i];
+ sum += filter_coeffs[i-1] * out[n-i];
- sum = ((sum >> 12) + in[n]) >> shift;
+ sum1 = ((-sum >> 12) + in[n]) >> shift;
+ 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;
- }
out[n] = sum;
}
diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
index c4016f8112..c1e7d2bb49 100644
--- a/libavcodec/g723_1.c
+++ b/libavcodec/g723_1.c
@@ -65,7 +65,7 @@ typedef struct g723_1_context {
int pf_gain; ///< formant postfilter
///< gain scaling unit memory
int postfilter;
- 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];
- for (i = 0; i < length; i++) {
- av_assert2(av_clipl_int32(vector[i] * (int64_t)scale << 1) == vector[i] * (int64_t)scale << 1);
- 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)
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)
gen_dirac_train(vector, pitch_lag);
} else { /* 5300 bps */
- int cb_gain = fixed_cb_gain[subfrm.amp_index];
- int cb_shift = subfrm.grid_index;
- int cb_sign = subfrm.pulse_sign;
- int cb_pos = subfrm.pulse_pos;
+ int cb_gain = fixed_cb_gain[subfrm->amp_index];
+ int cb_shift = subfrm->grid_index;
+ int cb_sign = subfrm->pulse_sign;
+ int cb_pos = subfrm->pulse_pos;
int offset, beta, lag;
for (i = 0; i < 8; i += 2) {
@@ -519,9 +514,9 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm,
}
/* Enhance harmonic components */
- lag = pitch_contrib[subfrm.ad_cb_gain << 1] + pitch_lag +
- subfrm.ad_cb_lag - 1;
- beta = pitch_contrib[(subfrm.ad_cb_gain << 1) + 1];
+ lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
+ subfrm->ad_cb_lag - 1;
+ beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
if (lag < SUBFRAME_LEN - 2) {
for (i = lag; i < SUBFRAME_LEN; i++)
@@ -546,19 +541,25 @@ static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
residual[i] = prev_excitation[offset + (i - 2) % lag];
}
+static int dot_product(const int16_t *a, const int16_t *b, int length)
+{
+ int sum = ff_dot_product(a,b,length);
+ return av_sat_add32(sum, sum);
+}
+
/**
* Generate adaptive codebook excitation.
*/
static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
- int pitch_lag, G723_1_Subframe subfrm,
+ int pitch_lag, G723_1_Subframe *subfrm,
enum Rate cur_rate)
{
int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
const int16_t *cb_ptr;
- int lag = pitch_lag + subfrm.ad_cb_lag - 1;
+ int lag = pitch_lag + subfrm->ad_cb_lag - 1;
int i;
- int64_t sum;
+ int sum;
get_residual(residual, prev_excitation, lag);
@@ -569,28 +570,27 @@ static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
cb_ptr = adaptive_cb_gain170;
/* Calculate adaptive vector */
- cb_ptr += subfrm.ad_cb_gain * 20;
+ cb_ptr += subfrm->ad_cb_gain * 20;
for (i = 0; i < SUBFRAME_LEN; i++) {
sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
- vector[i] = av_clipl_int32((sum << 2) + (1 << 15)) >> 16;
+ vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
}
}
/**
* Estimate maximum auto-correlation around pitch lag.
*
- * @param p the context
+ * @param buf buffer with offset applied
* @param offset offset of the excitation vector
* @param ccr_max pointer to the maximum auto-correlation
* @param pitch_lag decoded pitch lag
* @param length length of autocorrelation
* @param dir forward lag(1) / backward lag(-1)
*/
-static int autocorr_max(G723_1_Context *p, int offset, int *ccr_max,
+static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
int pitch_lag, int length, int dir)
{
int limit, ccr, lag = 0;
- int16_t *buf = p->excitation + offset;
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,
limit = pitch_lag + 3;
for (i = pitch_lag - 3; i <= limit; i++) {
- 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));
diff --git a/libavcodec/x86/Makefile b/libavcodec/x86/Makefile
index 3e998efafc..e01454a1b7 100644
--- a/libavcodec/x86/Makefile
+++ b/libavcodec/x86/Makefile
@@ -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 \
diff --git a/libavcodec/x86/dsputil_yasm.asm b/libavcodec/x86/dsputil.asm
index 19884a36a8..19884a36a8 100644
--- a/libavcodec/x86/dsputil_yasm.asm
+++ b/libavcodec/x86/dsputil.asm
diff --git a/libavcodec/x86/dsputilenc_yasm.asm b/libavcodec/x86/dsputilenc.asm
index 2d805c7da8..2d805c7da8 100644
--- a/libavcodec/x86/dsputilenc_yasm.asm
+++ b/libavcodec/x86/dsputilenc.asm
diff --git a/libavcodec/x86/vc1dsp_yasm.asm b/libavcodec/x86/vc1dsp.asm
index 590aa509a7..590aa509a7 100644
--- a/libavcodec/x86/vc1dsp_yasm.asm
+++ b/libavcodec/x86/vc1dsp.asm
diff --git a/libavutil/arm/intmath.h b/libavutil/arm/intmath.h
index 4af4bf32d1..cf6293525f 100644
--- a/libavutil/arm/intmath.h
+++ b/libavutil/arm/intmath.h
@@ -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 */
diff --git a/libavutil/common.h b/libavutil/common.h
index a11a3251e9..3e3baab3a1 100644
--- a/libavutil/common.h
+++ b/libavutil/common.h
@@ -187,6 +187,30 @@ static av_always_inline av_const unsigned av_clip_uintp2_c(int a, int p)
}
/**
+ * 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
* @param amin minimum value of the clip range
@@ -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
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-05 22:43:15 +0000