From 41ae2b03a5cf87f9673f82efbc9cea53df70f150 Mon Sep 17 00:00:00 2001 From: Lynne Date: Sat, 16 Mar 2024 02:43:33 +0100 Subject: [PATCH] aacdec: move spectrum decode and dequantization to a new file --- libavcodec/aac/aacdec.c | 4 + libavcodec/aac/aacdec_dsp_template.c | 2 +- libavcodec/aac/aacdec_fixed.c | 5 + libavcodec/aac/aacdec_fixed_dequant.h | 174 +++++++++++++ libavcodec/aac/aacdec_float.c | 74 ++++++ libavcodec/aac/aacdec_proc_template.c | 354 ++++++++++++++++++++++++++ libavcodec/aacdec.c | 68 ----- libavcodec/aacdec.h | 15 +- libavcodec/aacdec_fixed.c | 130 ---------- libavcodec/aacdec_template.c | 331 +----------------------- 10 files changed, 626 insertions(+), 531 deletions(-) create mode 100644 libavcodec/aac/aacdec_fixed_dequant.h create mode 100644 libavcodec/aac/aacdec_proc_template.c diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index 26612f4a14..d31c64d08d 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -48,6 +48,9 @@ extern const AACDecDSP aac_dsp; extern const AACDecDSP aac_dsp_fixed; +extern const AACDecProc aac_proc; +extern const AACDecProc aac_proc_fixed; + av_cold int ff_aac_decode_close(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; @@ -119,6 +122,7 @@ av_cold int ff_aac_decode_init_common(AVCodecContext *avctx) return ret; ac->dsp = is_fixed ? aac_dsp_fixed : aac_dsp; + ac->proc = is_fixed ? aac_proc_fixed : aac_proc; ac->dsp.init_tables(); diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c index 56c51c3e07..a04d358883 100644 --- a/libavcodec/aac/aacdec_dsp_template.c +++ b/libavcodec/aac/aacdec_dsp_template.c @@ -151,7 +151,7 @@ static void AAC_RENAME(apply_intensity_stereo)(AACDecContext *ac, scale = c * sce1->AAC_RENAME(sf)[idx]; for (group = 0; group < ics->group_len[g]; group++) #if USE_FIXED - ac->subband_scale(coef1 + group * 128 + offsets[i], + subband_scale(coef1 + group * 128 + offsets[i], coef0 + group * 128 + offsets[i], scale, 23, diff --git a/libavcodec/aac/aacdec_fixed.c b/libavcodec/aac/aacdec_fixed.c index 9dd8f34f55..a2ddc5aac2 100644 --- a/libavcodec/aac/aacdec_fixed.c +++ b/libavcodec/aac/aacdec_fixed.c @@ -38,6 +38,7 @@ #include "libavcodec/aactab.h" #include "libavcodec/sinewin_fixed_tablegen.h" #include "libavcodec/kbdwin.h" +#include "libavcodec/cbrt_data.h" DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_long_1024))[1024]; DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_short_128))[128]; @@ -61,4 +62,8 @@ static void init_tables_fixed(void) ff_thread_once(&init_fixed_once, init_tables_fixed_fn); } +/** Dequantization-related */ +#include "aacdec_fixed_dequant.h" + #include "aacdec_dsp_template.c" +#include "aacdec_proc_template.c" diff --git a/libavcodec/aac/aacdec_fixed_dequant.h b/libavcodec/aac/aacdec_fixed_dequant.h new file mode 100644 index 0000000000..5fb84fbed0 --- /dev/null +++ b/libavcodec/aac/aacdec_fixed_dequant.h @@ -0,0 +1,174 @@ +/* + * AAC decoder + * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) + * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) + * Copyright (c) 2008-2013 Alex Converse + * + * AAC LATM decoder + * Copyright (c) 2008-2010 Paul Kendall + * Copyright (c) 2010 Janne Grunau + * + * AAC decoder fixed-point implementation + * Copyright (c) 2013 + * MIPS Technologies, Inc., California. + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H +#define AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H + +#include "aacdec_tab.h" + +static void inline vector_pow43(int *coefs, int len) +{ + int i, coef; + + for (i=0; i> 2); + + if (s > 31) { + for (i=0; i 0) { + round = 1 << (s-1); + for (i=0; i> 32); + dst[i] = ((int)(out+round) >> s) * ssign; + } + } else if (s > -32) { + s = s + 32; + round = 1U << (s-1); + for (i=0; i> s); + dst[i] = out * (unsigned)ssign; + } + } else { + av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n"); + } +} + +static void noise_scale(int *coefs, int scale, int band_energy, int len) +{ + int s = -scale; + unsigned int round; + int i, out, c = exp2tab[s & 3]; + int nlz = 0; + + av_assert0(s >= 0); + while (band_energy > 0x7fff) { + band_energy >>= 1; + nlz++; + } + c /= band_energy; + s = 21 + nlz - (s >> 2); + + if (s > 31) { + for (i=0; i= 0) { + round = s ? 1 << (s-1) : 0; + for (i=0; i> 32); + coefs[i] = -((int)(out+round) >> s); + } + } + else { + s = s + 32; + if (s > 0) { + round = 1 << (s-1); + for (i=0; i> s); + coefs[i] = -out; + } + } else { + for (i=0; i> 4 & 15) - 4; + + return dst + 2; +} + +static inline int *DEC_SQUAD(int *dst, unsigned idx) +{ + dst[0] = (idx & 3) - 1; + dst[1] = (idx >> 2 & 3) - 1; + dst[2] = (idx >> 4 & 3) - 1; + dst[3] = (idx >> 6 & 3) - 1; + + return dst + 4; +} + +static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign) +{ + dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE)); + dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2)); + + return dst + 2; +} + +static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign) +{ + unsigned nz = idx >> 12; + + dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2)); + sign <<= nz & 1; + nz >>= 1; + dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2)); + sign <<= nz & 1; + nz >>= 1; + dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2)); + sign <<= nz & 1; + nz >>= 1; + dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2)); + + return dst + 4; +} + +#endif /* AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H */ diff --git a/libavcodec/aac/aacdec_float.c b/libavcodec/aac/aacdec_float.c index 6a5e8483b0..355980d169 100644 --- a/libavcodec/aac/aacdec_float.c +++ b/libavcodec/aac/aacdec_float.c @@ -38,6 +38,7 @@ #include "libavcodec/aactab.h" #include "libavcodec/sinewin.h" #include "libavcodec/kbdwin.h" +#include "libavcodec/cbrt_data.h" DECLARE_ALIGNED(32, static float, sine_120)[120]; DECLARE_ALIGNED(32, static float, sine_960)[960]; @@ -63,4 +64,77 @@ static void init_tables(void) ff_thread_once(&init_float_once, init_tables_float_fn); } +/** Dequantization-related **/ +#include "aacdec_tab.h" +#include "libavutil/intfloat.h" + +#ifndef VMUL2 +static inline float *VMUL2(float *dst, const float *v, unsigned idx, + const float *scale) +{ + float s = *scale; + *dst++ = v[idx & 15] * s; + *dst++ = v[idx>>4 & 15] * s; + return dst; +} +#endif + +#ifndef VMUL4 +static inline float *VMUL4(float *dst, const float *v, unsigned idx, + const float *scale) +{ + float s = *scale; + *dst++ = v[idx & 3] * s; + *dst++ = v[idx>>2 & 3] * s; + *dst++ = v[idx>>4 & 3] * s; + *dst++ = v[idx>>6 & 3] * s; + return dst; +} +#endif + +#ifndef VMUL2S +static inline float *VMUL2S(float *dst, const float *v, unsigned idx, + unsigned sign, const float *scale) +{ + union av_intfloat32 s0, s1; + + s0.f = s1.f = *scale; + s0.i ^= sign >> 1 << 31; + s1.i ^= sign << 31; + + *dst++ = v[idx & 15] * s0.f; + *dst++ = v[idx>>4 & 15] * s1.f; + + return dst; +} +#endif + +#ifndef VMUL4S +static inline float *VMUL4S(float *dst, const float *v, unsigned idx, + unsigned sign, const float *scale) +{ + unsigned nz = idx >> 12; + union av_intfloat32 s = { .f = *scale }; + union av_intfloat32 t; + + t.i = s.i ^ (sign & 1U<<31); + *dst++ = v[idx & 3] * t.f; + + sign <<= nz & 1; nz >>= 1; + t.i = s.i ^ (sign & 1U<<31); + *dst++ = v[idx>>2 & 3] * t.f; + + sign <<= nz & 1; nz >>= 1; + t.i = s.i ^ (sign & 1U<<31); + *dst++ = v[idx>>4 & 3] * t.f; + + sign <<= nz & 1; + t.i = s.i ^ (sign & 1U<<31); + *dst++ = v[idx>>6 & 3] * t.f; + + return dst; +} +#endif + #include "aacdec_dsp_template.c" +#include "aacdec_proc_template.c" diff --git a/libavcodec/aac/aacdec_proc_template.c b/libavcodec/aac/aacdec_proc_template.c new file mode 100644 index 0000000000..c3d607b4d3 --- /dev/null +++ b/libavcodec/aac/aacdec_proc_template.c @@ -0,0 +1,354 @@ +/* + * AAC decoder + * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) + * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) + * Copyright (c) 2008-2013 Alex Converse + * + * AAC LATM decoder + * Copyright (c) 2008-2010 Paul Kendall + * Copyright (c) 2010 Janne Grunau + * + * AAC decoder fixed-point implementation + * Copyright (c) 2013 + * MIPS Technologies, Inc., California. + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * linear congruential pseudorandom number generator + * + * @param previous_val pointer to the current state of the generator + * + * @return Returns a 32-bit pseudorandom integer + */ +static av_always_inline int lcg_random(unsigned previous_val) +{ + union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 }; + return v.s; +} + +/** + * Decode spectral data; reference: table 4.50. + * Dequantize and scale spectral data; reference: 4.6.3.3. + * + * @param coef array of dequantized, scaled spectral data + * @param sf array of scalefactors or intensity stereo positions + * @param pulse_present set if pulses are present + * @param pulse pointer to pulse data struct + * @param band_type array of the used band type + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int AAC_RENAME(decode_spectrum_and_dequant)(AACDecContext *ac, + GetBitContext *gb, + const Pulse *pulse, + SingleChannelElement *sce) +{ + int i, k, g, idx = 0; + INTFLOAT *coef = sce->AAC_RENAME(coeffs); + IndividualChannelStream *ics = &sce->ics; + const int c = 1024 / ics->num_windows; + const uint16_t *offsets = ics->swb_offset; + const INTFLOAT *sf = sce->AAC_RENAME(sf); + const enum BandType *band_type = sce->band_type; + INTFLOAT *coef_base = coef; + + for (g = 0; g < ics->num_windows; g++) + memset(coef + g * 128 + offsets[ics->max_sfb], 0, + sizeof(INTFLOAT) * (c - offsets[ics->max_sfb])); + + for (g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (i = 0; i < ics->max_sfb; i++, idx++) { + const unsigned cbt_m1 = band_type[idx] - 1; + INTFLOAT *cfo = coef + offsets[i]; + int off_len = offsets[i + 1] - offsets[i]; + int group; + + if (cbt_m1 >= INTENSITY_BT2 - 1) { + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { + memset(cfo, 0, off_len * sizeof(*cfo)); + } + } else if (cbt_m1 == NOISE_BT - 1) { + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { + INTFLOAT band_energy; +#if USE_FIXED + for (k = 0; k < off_len; k++) { + ac->random_state = lcg_random(ac->random_state); + cfo[k] = ac->random_state >> 3; + } + + band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len); + band_energy = fixed_sqrt(band_energy, 31); + noise_scale(cfo, sf[idx], band_energy, off_len); +#else + float scale; + + for (k = 0; k < off_len; k++) { + ac->random_state = lcg_random(ac->random_state); + cfo[k] = ac->random_state; + } + + band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len); + scale = sf[idx] / sqrtf(band_energy); + ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len); +#endif /* USE_FIXED */ + } + } else { +#if !USE_FIXED + const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; +#endif /* !USE_FIXED */ + const VLCElem *vlc_tab = ff_vlc_spectral[cbt_m1]; + OPEN_READER(re, gb); + + switch (cbt_m1 >> 1) { + case 0: + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { + INTFLOAT *cf = cfo; + int len = off_len; + + do { + int code; + unsigned cb_idx; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + cb_idx = code; +#if USE_FIXED + cf = DEC_SQUAD(cf, cb_idx); +#else + cf = VMUL4(cf, vq, cb_idx, sf + idx); +#endif /* USE_FIXED */ + } while (len -= 4); + } + break; + + case 1: + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { + INTFLOAT *cf = cfo; + int len = off_len; + + do { + int code; + unsigned nnz; + unsigned cb_idx; + uint32_t bits; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + cb_idx = code; + nnz = cb_idx >> 8 & 15; + bits = nnz ? GET_CACHE(re, gb) : 0; + LAST_SKIP_BITS(re, gb, nnz); +#if USE_FIXED + cf = DEC_UQUAD(cf, cb_idx, bits); +#else + cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); +#endif /* USE_FIXED */ + } while (len -= 4); + } + break; + + case 2: + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { + INTFLOAT *cf = cfo; + int len = off_len; + + do { + int code; + unsigned cb_idx; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + cb_idx = code; +#if USE_FIXED + cf = DEC_SPAIR(cf, cb_idx); +#else + cf = VMUL2(cf, vq, cb_idx, sf + idx); +#endif /* USE_FIXED */ + } while (len -= 2); + } + break; + + case 3: + case 4: + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { + INTFLOAT *cf = cfo; + int len = off_len; + + do { + int code; + unsigned nnz; + unsigned cb_idx; + unsigned sign; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + cb_idx = code; + nnz = cb_idx >> 8 & 15; + sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0; + LAST_SKIP_BITS(re, gb, nnz); +#if USE_FIXED + cf = DEC_UPAIR(cf, cb_idx, sign); +#else + cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); +#endif /* USE_FIXED */ + } while (len -= 2); + } + break; + + default: + for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { +#if USE_FIXED + int *icf = cfo; + int v; +#else + float *cf = cfo; + uint32_t *icf = (uint32_t *) cf; +#endif /* USE_FIXED */ + int len = off_len; + + do { + int code; + unsigned nzt, nnz; + unsigned cb_idx; + uint32_t bits; + int j; + + UPDATE_CACHE(re, gb); + GET_VLC(code, re, gb, vlc_tab, 8, 2); + cb_idx = code; + + if (cb_idx == 0x0000) { + *icf++ = 0; + *icf++ = 0; + continue; + } + + nnz = cb_idx >> 12; + nzt = cb_idx >> 8; + bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); + LAST_SKIP_BITS(re, gb, nnz); + + for (j = 0; j < 2; j++) { + if (nzt & 1< 8) { + av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); + return AVERROR_INVALIDDATA; + } + + SKIP_BITS(re, gb, b + 1); + b += 4; + n = (1 << b) + SHOW_UBITS(re, gb, b); + LAST_SKIP_BITS(re, gb, b); +#if USE_FIXED + v = n; + if (bits & 1U<<31) + v = -v; + *icf++ = v; +#else + *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31); +#endif /* USE_FIXED */ + bits <<= 1; + } else { +#if USE_FIXED + v = cb_idx & 15; + if (bits & 1U<<31) + v = -v; + *icf++ = v; +#else + unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; + *icf++ = (bits & 1U<<31) | v; +#endif /* USE_FIXED */ + bits <<= !!v; + } + cb_idx >>= 4; + } + } while (len -= 2); +#if !USE_FIXED + ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len); +#endif /* !USE_FIXED */ + } + } + + CLOSE_READER(re, gb); + } + } + coef += g_len << 7; + } + + if (pulse) { + idx = 0; + for (i = 0; i < pulse->num_pulse; i++) { + INTFLOAT co = coef_base[ pulse->pos[i] ]; + while (offsets[idx + 1] <= pulse->pos[i]) + idx++; + if (band_type[idx] != NOISE_BT && sf[idx]) { + INTFLOAT ico = -pulse->amp[i]; +#if USE_FIXED + if (co) { + ico = co + (co > 0 ? -ico : ico); + } + coef_base[ pulse->pos[i] ] = ico; +#else + if (co) { + co /= sf[idx]; + ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); + } + coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; +#endif /* USE_FIXED */ + } + } + } +#if USE_FIXED + coef = coef_base; + idx = 0; + for (g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (i = 0; i < ics->max_sfb; i++, idx++) { + const unsigned cbt_m1 = band_type[idx] - 1; + int *cfo = coef + offsets[i]; + int off_len = offsets[i + 1] - offsets[i]; + int group; + + if (cbt_m1 < NOISE_BT - 1) { + for (group = 0; group < (int)g_len; group++, cfo+=128) { + vector_pow43(cfo, off_len); + subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx); + } + } + } + coef += g_len << 7; + } +#endif /* USE_FIXED */ + return 0; +} + +const AACDecProc AAC_RENAME(aac_proc) = { + .decode_spectrum_and_dequant = AAC_RENAME(decode_spectrum_and_dequant), +}; diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 67cdda8cde..9642c45015 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -78,74 +78,6 @@ static av_always_inline void reset_predict_state(PredictorState *ps) ps->var1 = 1.0f; } -#ifndef VMUL2 -static inline float *VMUL2(float *dst, const float *v, unsigned idx, - const float *scale) -{ - float s = *scale; - *dst++ = v[idx & 15] * s; - *dst++ = v[idx>>4 & 15] * s; - return dst; -} -#endif - -#ifndef VMUL4 -static inline float *VMUL4(float *dst, const float *v, unsigned idx, - const float *scale) -{ - float s = *scale; - *dst++ = v[idx & 3] * s; - *dst++ = v[idx>>2 & 3] * s; - *dst++ = v[idx>>4 & 3] * s; - *dst++ = v[idx>>6 & 3] * s; - return dst; -} -#endif - -#ifndef VMUL2S -static inline float *VMUL2S(float *dst, const float *v, unsigned idx, - unsigned sign, const float *scale) -{ - union av_intfloat32 s0, s1; - - s0.f = s1.f = *scale; - s0.i ^= sign >> 1 << 31; - s1.i ^= sign << 31; - - *dst++ = v[idx & 15] * s0.f; - *dst++ = v[idx>>4 & 15] * s1.f; - - return dst; -} -#endif - -#ifndef VMUL4S -static inline float *VMUL4S(float *dst, const float *v, unsigned idx, - unsigned sign, const float *scale) -{ - unsigned nz = idx >> 12; - union av_intfloat32 s = { .f = *scale }; - union av_intfloat32 t; - - t.i = s.i ^ (sign & 1U<<31); - *dst++ = v[idx & 3] * t.f; - - sign <<= nz & 1; nz >>= 1; - t.i = s.i ^ (sign & 1U<<31); - *dst++ = v[idx>>2 & 3] * t.f; - - sign <<= nz & 1; nz >>= 1; - t.i = s.i ^ (sign & 1U<<31); - *dst++ = v[idx>>4 & 3] * t.f; - - sign <<= nz & 1; - t.i = s.i ^ (sign & 1U<<31); - *dst++ = v[idx>>6 & 3] * t.f; - - return dst; -} -#endif - static av_always_inline float flt16_round(float pf) { union av_intfloat32 tmp; diff --git a/libavcodec/aacdec.h b/libavcodec/aacdec.h index 87462adb02..811beb77f2 100644 --- a/libavcodec/aacdec.h +++ b/libavcodec/aacdec.h @@ -200,6 +200,16 @@ typedef struct DynamicRangeControl { */ } DynamicRangeControl; +/** + * Decode-specific primitives + */ +typedef struct AACDecProc { + int (*decode_spectrum_and_dequant)(AACDecContext *ac, + GetBitContext *gb, + const Pulse *pulse, + SingleChannelElement *sce); +} AACDecProc; + /** * DSP-specific primitives */ @@ -232,6 +242,7 @@ struct AACDecContext { struct AVCodecContext *avctx; AACDecDSP dsp; + AACDecProc proc; struct AVFrame *frame; @@ -309,10 +320,6 @@ struct AACDecContext { int warned_he_aac_mono; int is_fixed; - - /* aacdec functions pointers */ - void (*vector_pow43)(int *coefs, int len); - void (*subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context); }; #if defined(USE_FIXED) && USE_FIXED diff --git a/libavcodec/aacdec_fixed.c b/libavcodec/aacdec_fixed.c index 7633f4adb0..efc666a6ce 100644 --- a/libavcodec/aacdec_fixed.c +++ b/libavcodec/aacdec_fixed.c @@ -147,136 +147,6 @@ static av_always_inline void reset_predict_state(PredictorState *ps) static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75 -static inline int *DEC_SPAIR(int *dst, unsigned idx) -{ - dst[0] = (idx & 15) - 4; - dst[1] = (idx >> 4 & 15) - 4; - - return dst + 2; -} - -static inline int *DEC_SQUAD(int *dst, unsigned idx) -{ - dst[0] = (idx & 3) - 1; - dst[1] = (idx >> 2 & 3) - 1; - dst[2] = (idx >> 4 & 3) - 1; - dst[3] = (idx >> 6 & 3) - 1; - - return dst + 4; -} - -static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign) -{ - dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE)); - dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2)); - - return dst + 2; -} - -static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign) -{ - unsigned nz = idx >> 12; - - dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2)); - sign <<= nz & 1; - nz >>= 1; - dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2)); - sign <<= nz & 1; - nz >>= 1; - dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2)); - sign <<= nz & 1; - nz >>= 1; - dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2)); - - return dst + 4; -} - -static void vector_pow43(int *coefs, int len) -{ - int i, coef; - - for (i=0; i> 2); - - if (s > 31) { - for (i=0; i 0) { - round = 1 << (s-1); - for (i=0; i> 32); - dst[i] = ((int)(out+round) >> s) * ssign; - } - } else if (s > -32) { - s = s + 32; - round = 1U << (s-1); - for (i=0; i> s); - dst[i] = out * (unsigned)ssign; - } - } else { - av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n"); - } -} - -static void noise_scale(int *coefs, int scale, int band_energy, int len) -{ - int s = -scale; - unsigned int round; - int i, out, c = exp2tab[s & 3]; - int nlz = 0; - - av_assert0(s >= 0); - while (band_energy > 0x7fff) { - band_energy >>= 1; - nlz++; - } - c /= band_energy; - s = 21 + nlz - (s >> 2); - - if (s > 31) { - for (i=0; i= 0) { - round = s ? 1 << (s-1) : 0; - for (i=0; i> 32); - coefs[i] = -((int)(out+round) >> s); - } - } - else { - s = s + 32; - if (s > 0) { - round = 1 << (s-1); - for (i=0; i> s); - coefs[i] = -out; - } - } else { - for (i=0; iAAC_RENAME(coeffs); - IndividualChannelStream *ics = &sce->ics; - const int c = 1024 / ics->num_windows; - const uint16_t *offsets = ics->swb_offset; - const INTFLOAT *sf = sce->AAC_RENAME(sf); - const enum BandType *band_type = sce->band_type; - INTFLOAT *coef_base = coef; - - for (g = 0; g < ics->num_windows; g++) - memset(coef + g * 128 + offsets[ics->max_sfb], 0, - sizeof(INTFLOAT) * (c - offsets[ics->max_sfb])); - - for (g = 0; g < ics->num_window_groups; g++) { - unsigned g_len = ics->group_len[g]; - - for (i = 0; i < ics->max_sfb; i++, idx++) { - const unsigned cbt_m1 = band_type[idx] - 1; - INTFLOAT *cfo = coef + offsets[i]; - int off_len = offsets[i + 1] - offsets[i]; - int group; - - if (cbt_m1 >= INTENSITY_BT2 - 1) { - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { - memset(cfo, 0, off_len * sizeof(*cfo)); - } - } else if (cbt_m1 == NOISE_BT - 1) { - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { - INTFLOAT band_energy; -#if USE_FIXED - for (k = 0; k < off_len; k++) { - ac->random_state = lcg_random(ac->random_state); - cfo[k] = ac->random_state >> 3; - } - - band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len); - band_energy = fixed_sqrt(band_energy, 31); - noise_scale(cfo, sf[idx], band_energy, off_len); -#else - float scale; - - for (k = 0; k < off_len; k++) { - ac->random_state = lcg_random(ac->random_state); - cfo[k] = ac->random_state; - } - - band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len); - scale = sf[idx] / sqrtf(band_energy); - ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len); -#endif /* USE_FIXED */ - } - } else { -#if !USE_FIXED - const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; -#endif /* !USE_FIXED */ - const VLCElem *vlc_tab = ff_vlc_spectral[cbt_m1]; - OPEN_READER(re, gb); - - switch (cbt_m1 >> 1) { - case 0: - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { - INTFLOAT *cf = cfo; - int len = off_len; - - do { - int code; - unsigned cb_idx; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = code; -#if USE_FIXED - cf = DEC_SQUAD(cf, cb_idx); -#else - cf = VMUL4(cf, vq, cb_idx, sf + idx); -#endif /* USE_FIXED */ - } while (len -= 4); - } - break; - - case 1: - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { - INTFLOAT *cf = cfo; - int len = off_len; - - do { - int code; - unsigned nnz; - unsigned cb_idx; - uint32_t bits; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = code; - nnz = cb_idx >> 8 & 15; - bits = nnz ? GET_CACHE(re, gb) : 0; - LAST_SKIP_BITS(re, gb, nnz); -#if USE_FIXED - cf = DEC_UQUAD(cf, cb_idx, bits); -#else - cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); -#endif /* USE_FIXED */ - } while (len -= 4); - } - break; - - case 2: - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { - INTFLOAT *cf = cfo; - int len = off_len; - - do { - int code; - unsigned cb_idx; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = code; -#if USE_FIXED - cf = DEC_SPAIR(cf, cb_idx); -#else - cf = VMUL2(cf, vq, cb_idx, sf + idx); -#endif /* USE_FIXED */ - } while (len -= 2); - } - break; - - case 3: - case 4: - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { - INTFLOAT *cf = cfo; - int len = off_len; - - do { - int code; - unsigned nnz; - unsigned cb_idx; - unsigned sign; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = code; - nnz = cb_idx >> 8 & 15; - sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0; - LAST_SKIP_BITS(re, gb, nnz); -#if USE_FIXED - cf = DEC_UPAIR(cf, cb_idx, sign); -#else - cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); -#endif /* USE_FIXED */ - } while (len -= 2); - } - break; - - default: - for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) { -#if USE_FIXED - int *icf = cfo; - int v; -#else - float *cf = cfo; - uint32_t *icf = (uint32_t *) cf; -#endif /* USE_FIXED */ - int len = off_len; - - do { - int code; - unsigned nzt, nnz; - unsigned cb_idx; - uint32_t bits; - int j; - - UPDATE_CACHE(re, gb); - GET_VLC(code, re, gb, vlc_tab, 8, 2); - cb_idx = code; - - if (cb_idx == 0x0000) { - *icf++ = 0; - *icf++ = 0; - continue; - } - - nnz = cb_idx >> 12; - nzt = cb_idx >> 8; - bits = SHOW_UBITS(re, gb, nnz) << (32-nnz); - LAST_SKIP_BITS(re, gb, nnz); - - for (j = 0; j < 2; j++) { - if (nzt & 1< 8) { - av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); - return AVERROR_INVALIDDATA; - } - - SKIP_BITS(re, gb, b + 1); - b += 4; - n = (1 << b) + SHOW_UBITS(re, gb, b); - LAST_SKIP_BITS(re, gb, b); -#if USE_FIXED - v = n; - if (bits & 1U<<31) - v = -v; - *icf++ = v; -#else - *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31); -#endif /* USE_FIXED */ - bits <<= 1; - } else { -#if USE_FIXED - v = cb_idx & 15; - if (bits & 1U<<31) - v = -v; - *icf++ = v; -#else - unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; - *icf++ = (bits & 1U<<31) | v; -#endif /* USE_FIXED */ - bits <<= !!v; - } - cb_idx >>= 4; - } - } while (len -= 2); -#if !USE_FIXED - ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len); -#endif /* !USE_FIXED */ - } - } - - CLOSE_READER(re, gb); - } - } - coef += g_len << 7; - } - - if (pulse) { - idx = 0; - for (i = 0; i < pulse->num_pulse; i++) { - INTFLOAT co = coef_base[ pulse->pos[i] ]; - while (offsets[idx + 1] <= pulse->pos[i]) - idx++; - if (band_type[idx] != NOISE_BT && sf[idx]) { - INTFLOAT ico = -pulse->amp[i]; -#if USE_FIXED - if (co) { - ico = co + (co > 0 ? -ico : ico); - } - coef_base[ pulse->pos[i] ] = ico; -#else - if (co) { - co /= sf[idx]; - ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); - } - coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; -#endif /* USE_FIXED */ - } - } - } -#if USE_FIXED - coef = coef_base; - idx = 0; - for (g = 0; g < ics->num_window_groups; g++) { - unsigned g_len = ics->group_len[g]; - - for (i = 0; i < ics->max_sfb; i++, idx++) { - const unsigned cbt_m1 = band_type[idx] - 1; - int *cfo = coef + offsets[i]; - int off_len = offsets[i + 1] - offsets[i]; - int group; - - if (cbt_m1 < NOISE_BT - 1) { - for (group = 0; group < (int)g_len; group++, cfo+=128) { - ac->vector_pow43(cfo, off_len); - ac->subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx); - } - } - } - coef += g_len << 7; - } -#endif /* USE_FIXED */ - return 0; -} - /** * Apply AAC-Main style frequency domain prediction. */ @@ -2047,9 +1727,9 @@ static int decode_ics(AACDecContext *ac, SingleChannelElement *sce, } } - ret = decode_spectrum_and_dequant(ac, gb, - pulse_present ? &pulse : NULL, - sce); + ret = ac->proc.decode_spectrum_and_dequant(ac, gb, + pulse_present ? &pulse : NULL, + sce); if (ret < 0) goto fail; @@ -2892,11 +2572,6 @@ static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, static void aacdec_init(AACDecContext *c) { -#if USE_FIXED - c->vector_pow43 = vector_pow43; - c->subband_scale = subband_scale; -#endif - #if !USE_FIXED #if ARCH_MIPS ff_aacdec_init_mips(c);