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 similarity index 100% rename from libavcodec/x86/dsputil_yasm.asm rename to libavcodec/x86/dsputil.asm diff --git a/libavcodec/x86/dsputilenc_yasm.asm b/libavcodec/x86/dsputilenc.asm similarity index 100% rename from libavcodec/x86/dsputilenc_yasm.asm rename to libavcodec/x86/dsputilenc.asm diff --git a/libavcodec/x86/vc1dsp_yasm.asm b/libavcodec/x86/vc1dsp.asm similarity index 100% rename from libavcodec/x86/vc1dsp_yasm.asm rename to 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 @@ -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