Sync already committed code with that in SoC and commit more OKed hunks of code

Originally committed as revision 14674 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/aac.c

@@ -82,6 +82,7 @@
 
 #include "aac.h"
 #include "aactab.h"
+#include "aacdectab.h"
 #include "mpeg4audio.h"
 
 #include <assert.h>
@@ -91,6 +92,7 @@
 
 #ifndef CONFIG_HARDCODED_TABLES
     static float ff_aac_ivquant_tab[IVQUANT_SIZE];
+    static float ff_aac_pow2sf_tab[316];
 #endif /* CONFIG_HARDCODED_TABLES */
 
 static VLC vlc_scalefactors;
@@ -104,27 +106,29 @@ static VLC vlc_spectral[11];
     num_assoc_data  = get_bits(gb, 3);
     num_cc          = get_bits(gb, 4);
 
-    newpcs->mono_mixdown_tag   = get_bits1(gb) ? get_bits(gb, 4) : -1;
-    newpcs->stereo_mixdown_tag = get_bits1(gb) ? get_bits(gb, 4) : -1;
+    if (get_bits1(gb))
+        skip_bits(gb, 4); // mono_mixdown_tag
+    if (get_bits1(gb))
+        skip_bits(gb, 4); // stereo_mixdown_tag
 
-    if (get_bits1(gb)) {
-        newpcs->mixdown_coeff_index = get_bits(gb, 2);
-        newpcs->pseudo_surround     = get_bits1(gb);
-    }
+    if (get_bits1(gb))
+        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
 
-    program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_FRONT, gb, num_front);
-    program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
-    program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_BACK,  gb, num_back );
-    program_config_element_parse_tags(NULL,                     newpcs->che_type[ID_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
+    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
+    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
+    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
+    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
 
     skip_bits_long(gb, 4 * num_assoc_data);
 
-    program_config_element_parse_tags(newpcs->che_type[ID_CCE], newpcs->che_type[ID_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
+    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
 
     align_get_bits(gb);
 
     /* comment field, first byte is length */
     skip_bits_long(gb, 8 * get_bits(gb, 8));
+    return 0;
+}
 
 static av_cold int aac_decode_init(AVCodecContext * avccontext) {
     AACContext * ac = avccontext->priv_data;
@@ -132,6 +136,10 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
 
     ac->avccontext = avccontext;
 
+    if (avccontext->extradata_size <= 0 ||
+        decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
+        return -1;
+
     avccontext->sample_rate = ac->m4ac.sample_rate;
     avccontext->frame_size  = 1024;
 
@@ -166,6 +174,8 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
 #ifndef CONFIG_HARDCODED_TABLES
     for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++)
         ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] =  cbrt(fabs(i)) * i;
+    for (i = 0; i < 316; i++)
+        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
 #endif /* CONFIG_HARDCODED_TABLES */
 
     INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]),
@@ -200,6 +210,112 @@ static inline float ivquant(int a) {
         return cbrtf(fabsf(a)) * a;
 }
 
+        int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
+    int g, idx = 0;
+    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
+    for (g = 0; g < ics->num_window_groups; g++) {
+        int k = 0;
+        while (k < ics->max_sfb) {
+            uint8_t sect_len = k;
+            int sect_len_incr;
+            int sect_band_type = get_bits(gb, 4);
+            if (sect_band_type == 12) {
+                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
+                return -1;
+            }
+            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
+                sect_len += sect_len_incr;
+            sect_len += sect_len_incr;
+            if (sect_len > ics->max_sfb) {
+                av_log(ac->avccontext, AV_LOG_ERROR,
+                    "Number of bands (%d) exceeds limit (%d).\n",
+                    sect_len, ics->max_sfb);
+                return -1;
+            }
+
+ *
+ * @param   mix_gain            channel gain (Not used by AAC bitstream.)
+ * @param   global_gain         first scalefactor value as scalefactors are differentially coded
+ * @param   band_type           array of the used band type
+ * @param   band_type_run_end   array of the last scalefactor band of a band type run
+ * @param   sf                  array of scalefactors or intensity stereo positions
+ *
+ * @return  Returns error status. 0 - OK, !0 - error
+ */
+static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
+        float mix_gain, unsigned int global_gain, IndividualChannelStream * ics,
+        enum BandType band_type[120], int band_type_run_end[120]) {
+    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
+    int g, i, idx = 0;
+    int offset[3] = { global_gain, global_gain - 90, 100 };
+    int noise_flag = 1;
+    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
+    ics->intensity_present = 0;
+    for (g = 0; g < ics->num_window_groups; g++) {
+        for (i = 0; i < ics->max_sfb;) {
+            int run_end = band_type_run_end[idx];
+            if (band_type[idx] == ZERO_BT) {
+                for(; i < run_end; i++, idx++)
+                    sf[idx] = 0.;
+            }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
+                ics->intensity_present = 1;
+                for(; i < run_end; i++, idx++) {
+                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+                    if(offset[2] > 255U) {
+                        av_log(ac->avccontext, AV_LOG_ERROR,
+                            "%s (%d) out of range.\n", sf_str[2], offset[2]);
+                        return -1;
+                    }
+                    sf[idx]  = ff_aac_pow2sf_tab[-offset[2] + 300];
+                    sf[idx] *= mix_gain;
+                }
+            }else if(band_type[idx] == NOISE_BT) {
+                for(; i < run_end; i++, idx++) {
+                    if(noise_flag-- > 0)
+                        offset[1] += get_bits(gb, 9) - 256;
+                    else
+                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+                    if(offset[1] > 255U) {
+                        av_log(ac->avccontext, AV_LOG_ERROR,
+                            "%s (%d) out of range.\n", sf_str[1], offset[1]);
+                        return -1;
+                    }
+                    sf[idx]  = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
+                    sf[idx] *= mix_gain;
+                }
+            }else {
+                for(; i < run_end; i++, idx++) {
+                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
+                    if(offset[0] > 255U) {
+                        av_log(ac->avccontext, AV_LOG_ERROR,
+                            "%s (%d) out of range.\n", sf_str[0], offset[0]);
+                        return -1;
+                    }
+                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
+                    sf[idx] *= mix_gain;
+                }
+            }
+        }
+    }
+    return 0;
+}
+
+/**
+ * Decode pulse data; reference: table 4.7.
+ */
+static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
+    int i;
+    pulse->num_pulse = get_bits(gb, 2) + 1;
+    pulse->start = get_bits(gb, 6);
+    for (i = 0; i < pulse->num_pulse; i++) {
+        pulse->offset[i] = get_bits(gb, 5);
+        pulse->amp   [i] = get_bits(gb, 4);
+    }
+}
+
+/**
+ * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
+ *
  * @param   pulse   pointer to pulse data struct
  * @param   icoef   array of quantized spectral data
  */
@@ -213,18 +329,97 @@ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualCha
     }
 }
 
+/**
+ * Parse Spectral Band Replication extension data; reference: table 4.55.
+ *
+ * @param   crc flag indicating the presence of CRC checksum
+ * @param   cnt length of TYPE_FIL syntactic element in bytes
+ * @return  Returns number of bytes consumed from the TYPE_FIL element.
+ */
+static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
+    // TODO : sbr_extension implementation
+    av_log(ac->avccontext, AV_LOG_DEBUG, "aac: SBR not yet supported.\n");
+    skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
+    return cnt;
+}
+
+    int crc_flag = 0;
+    int res = cnt;
+    switch (get_bits(gb, 4)) { // extension type
+        case EXT_SBR_DATA_CRC:
+            crc_flag++;
+        case EXT_SBR_DATA:
+            res = decode_sbr_extension(ac, gb, crc_flag, cnt);
+            break;
+        case EXT_DYNAMIC_RANGE:
+            res = decode_dynamic_range(&ac->che_drc, gb, cnt);
+            break;
+        case EXT_FILL:
+        case EXT_FILL_DATA:
+        case EXT_DATA_ELEMENT:
+        default:
+            skip_bits_long(gb, 8*cnt - 4);
+            break;
+    };
+    return res;
+}
+
+/**
+ * Apply dependent channel coupling (applied before IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
+    IndividualChannelStream * ics = &cc->ch[0].ics;
+    const uint16_t * offsets = ics->swb_offset;
+    float * dest = sce->coeffs;
+    const float * src = cc->ch[0].coeffs;
+    int g, i, group, k, idx = 0;
+    if(ac->m4ac.object_type == AOT_AAC_LTP) {
+        av_log(ac->avccontext, AV_LOG_ERROR,
+               "Dependent coupling is not supported together with LTP\n");
+        return;
+    }
+    for (g = 0; g < ics->num_window_groups; g++) {
+        for (i = 0; i < ics->max_sfb; i++, idx++) {
+            if (cc->ch[0].band_type[idx] != ZERO_BT) {
+                float gain = cc->coup.gain[index][idx] * sce->mixing_gain;
+                for (group = 0; group < ics->group_len[g]; group++) {
+                    for (k = offsets[i]; k < offsets[i+1]; k++) {
+                        // XXX dsputil-ize
+                        dest[group*128+k] += gain * src[group*128+k];
+                    }
+                }
+            }
+        }
+        dest += ics->group_len[g]*128;
+        src  += ics->group_len[g]*128;
+    }
+}
+
+/**
+ * Apply independent channel coupling (applied after IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
+    int i;
+    float gain = cc->coup.gain[index][0] * sce->mixing_gain;
+    for (i = 0; i < 1024; i++)
+        sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);
+}
+
 static av_cold int aac_decode_close(AVCodecContext * avccontext) {
     AACContext * ac = avccontext->priv_data;
     int i, j;
 
-    for (i = 0; i < MAX_TAGID; i++) {
+    for (i = 0; i < MAX_ELEM_ID; i++) {
         for(j = 0; j < 4; j++)
             av_freep(&ac->che[j][i]);
     }
 
     ff_mdct_end(&ac->mdct);
     ff_mdct_end(&ac->mdct_small);
-    av_freep(&ac->interleaved_output);
     return 0 ;
 }
 
@@ -238,4 +433,5 @@ AVCodec aac_decoder = {
     aac_decode_close,
     aac_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
 };

libavcodec/aac.h

@@ -42,8 +42,49 @@
         ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
         size);
 
+#define MAX_CHANNELS 64
+
 #define IVQUANT_SIZE 1024
 
+enum AudioObjectType {
+    AOT_NULL,
+                               // Support?                Name
+    AOT_AAC_MAIN,              ///< Y                       Main
+    AOT_AAC_LC,                ///< Y                       Low Complexity
+    AOT_AAC_SSR,               ///< N (code in SoC repo)    Scalable Sample Rate
+    AOT_AAC_LTP,               ///< N (code in SoC repo)    Long Term Prediction
+    AOT_SBR,                   ///< N (in progress)         Spectral Band Replication
+    AOT_AAC_SCALABLE,          ///< N                       Scalable
+    AOT_TWINVQ,                ///< N                       Twin Vector Quantizer
+    AOT_CELP,                  ///< N                       Code Excited Linear Prediction
+    AOT_HVXC,                  ///< N                       Harmonic Vector eXcitation Coding
+    AOT_TTSI             = 12, ///< N                       Text-To-Speech Interface
+    AOT_MAINSYNTH,             ///< N                       Main Synthesis
+    AOT_WAVESYNTH,             ///< N                       Wavetable Synthesis
+    AOT_MIDI,                  ///< N                       General MIDI
+    AOT_SAFX,                  ///< N                       Algorithmic Synthesis and Audio Effects
+    AOT_ER_AAC_LC,             ///< N                       Error Resilient Low Complexity
+    AOT_ER_AAC_LTP       = 19, ///< N                       Error Resilient Long Term Prediction
+    AOT_ER_AAC_SCALABLE,       ///< N                       Error Resilient Scalable
+    AOT_ER_TWINVQ,             ///< N                       Error Resilient Twin Vector Quantizer
+    AOT_ER_BSAC,               ///< N                       Error Resilient Bit-Sliced Arithmetic Coding
+    AOT_ER_AAC_LD,             ///< N                       Error Resilient Low Delay
+    AOT_ER_CELP,               ///< N                       Error Resilient Code Excited Linear Prediction
+    AOT_ER_HVXC,               ///< N                       Error Resilient Harmonic Vector eXcitation Coding
+    AOT_ER_HILN,               ///< N                       Error Resilient Harmonic and Individual Lines plus Noise
+    AOT_ER_PARAM,              ///< N                       Error Resilient Parametric
+    AOT_SSC,                   ///< N                       SinuSoidal Coding
+};
+
+enum ExtensionPayloadID {
+    EXT_FILL,
+    EXT_FILL_DATA,
+    EXT_DATA_ELEMENT,
+    EXT_DYNAMIC_RANGE = 0xb,
+    EXT_SBR_DATA      = 0xd,
+    EXT_SBR_DATA_CRC  = 0xe,
+};
+
 enum WindowSequence {
     ONLY_LONG_SEQUENCE,
     LONG_START_SEQUENCE,
@@ -51,7 +92,18 @@ enum WindowSequence {
     LONG_STOP_SEQUENCE,
 };
 
-enum ChannelType {
+enum BandType {
+    ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
+    FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
+    ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
+    NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
+    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
+    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
+};
+
+#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
+
+enum ChannelPosition {
     AAC_CHANNEL_FRONT = 1,
     AAC_CHANNEL_SIDE  = 2,
     AAC_CHANNEL_BACK  = 3,
@@ -59,12 +111,33 @@ enum ChannelType {
     AAC_CHANNEL_CC    = 5,
 };
 
+typedef struct {
+    int num_pulse;
+    int start;
+    int offset[4];
+    int amp[4];
+} Pulse;
+
+/**
+ * coupling parameters
+ */
+typedef struct {
+
 /**
  * main AAC context
  */
 typedef struct {
     AVCodecContext * avccontext;
 
+    MPEG4AudioConfig m4ac;
+
+    int is_saved;                 ///< Set if elements have stored overlap from previous frame.
+    DynamicRangeControl che_drc;
+
+    enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
+                                                   *   first index as the first 4 raw data block types
+                                                   */
+
     /**
      * @defgroup tables   Computed / set up during initialization.
      * @{
@@ -75,9 +148,10 @@ typedef struct {
     /** @} */
 
     /**
-     * @defgroup output   Members used for output interleaving and down-mixing.
+     * @defgroup output   Members used for output interleaving.
      * @{
      */
+    float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
     float add_bias;                                   ///< offset for dsp.float_to_int16
     float sf_scale;                                   ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
     int sf_offset;                                    ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16

libavcodec/aacdectab.h

+/*
+ * AAC decoder data
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * 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
+ */
+
+/**
+ * @file aacdectab.h
+ * AAC decoder data
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+#ifndef FFMPEG_AACDECTAB_H
+#define FFMPEG_AACDECTAB_H
+
+#include "aac.h"
+
+#include <stdint.h>
+
+/* @name swb_offsets
+ * Sample offset into the window indicating the beginning of a scalefactor
+ * window band
+ *
+ * scalefactor window band - term for scalefactor bands within a window,
+ * given in Table 4.110 to Table 4.128.
+ *
+ * scalefactor band - a set of spectral coefficients which are scaled by one
+ * scalefactor. In case of EIGHT_SHORT_SEQUENCE and grouping a scalefactor band
+ * may contain several scalefactor window bands of corresponding frequency. For
+ * all other window_sequences scalefactor bands and scalefactor window bands are
+ * identical.
+ * @{
+ */
+
+static const uint16_t swb_offset_1024_96[] = {
+      0,   4,   8,  12,  16,  20,  24,  28,
+     32,  36,  40,  44,  48,  52,  56,  64,
+     72,  80,  88,  96, 108, 120, 132, 144,
+    156, 172, 188, 212, 240, 276, 320, 384,
+    448, 512, 576, 640, 704, 768, 832, 896,
+    960, 1024
+};
+
+static const uint16_t swb_offset_128_96[] = {
+    0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 128
+};
+
+static const uint16_t swb_offset_1024_64[] = {
+      0,   4,   8,  12,  16,  20,  24,  28,
+     32,  36,  40,  44,  48,  52,  56,  64,
+     72,  80,  88, 100, 112, 124, 140, 156,
+    172, 192, 216, 240, 268, 304, 344, 384,
+    424, 464, 504, 544, 584, 624, 664, 704,
+    744, 784, 824, 864, 904, 944, 984, 1024
+};
+
+static const uint16_t swb_offset_1024_48[] = {
+      0,   4,   8,  12,  16,  20,  24,  28,
+     32,  36,  40,  48,  56,  64,  72,  80,
+     88,  96, 108, 120, 132, 144, 160, 176,
+    196, 216, 240, 264, 292, 320, 352, 384,
+    416, 448, 480, 512, 544, 576, 608, 640,
+    672, 704, 736, 768, 800, 832, 864, 896,
+    928, 1024
+};
+
+static const uint16_t swb_offset_128_48[] = {
+     0,   4,   8,  12,  16,  20,  28,  36,
+    44,  56,  68,  80,  96, 112, 128
+};
+
+static const uint16_t swb_offset_1024_32[] = {
+      0,   4,   8,  12,  16,  20,  24,  28,
+     32,  36,  40,  48,  56,  64,  72,  80,
+     88,  96, 108, 120, 132, 144, 160, 176,
+    196, 216, 240, 264, 292, 320, 352, 384,
+    416, 448, 480, 512, 544, 576, 608, 640,
+    672, 704, 736, 768, 800, 832, 864, 896,
+    928, 960, 992, 1024
+};
+
+static const uint16_t swb_offset_1024_24[] = {
+      0,   4,   8,  12,  16,  20,  24,  28,
+     32,  36,  40,  44,  52,  60,  68,  76,
+     84,  92, 100, 108, 116, 124, 136, 148,
+    160, 172, 188, 204, 220, 240, 260, 284,
+    308, 336, 364, 396, 432, 468, 508, 552,
+    600, 652, 704, 768, 832, 896, 960, 1024
+};
+
+static const uint16_t swb_offset_128_24[] = {
+     0,   4,   8,  12,  16,  20,  24,  28,
+    36,  44,  52,  64,  76,  92, 108, 128
+};
+
+static const uint16_t swb_offset_1024_16[] = {
+      0,   8,  16,  24,  32,  40,  48,  56,
+     64,  72,  80,  88, 100, 112, 124, 136,
+    148, 160, 172, 184, 196, 212, 228, 244,
+    260, 280, 300, 320, 344, 368, 396, 424,
+    456, 492, 532, 572, 616, 664, 716, 772,
+    832, 896, 960, 1024
+};
+
+static const uint16_t swb_offset_128_16[] = {
+     0,   4,   8,  12,  16,  20,  24,  28,
+    32,  40,  48,  60,  72,  88, 108, 128
+};
+
+static const uint16_t swb_offset_1024_8[] = {
+      0,  12,  24,  36,  48,  60,  72,  84,
+     96, 108, 120, 132, 144, 156, 172, 188,
+    204, 220, 236, 252, 268, 288, 308, 328,
+    348, 372, 396, 420, 448, 476, 508, 544,
+    580, 620, 664, 712, 764, 820, 880, 944,
+    1024
+};
+
+static const uint16_t swb_offset_128_8[] = {
+     0,   4,   8,  12,  16,  20,  24,  28,
+    36,  44,  52,  60,  72,  88, 108, 128
+};
+
+static const uint16_t *swb_offset_1024[] = {
+    swb_offset_1024_96, swb_offset_1024_96, swb_offset_1024_64,
+    swb_offset_1024_48, swb_offset_1024_48, swb_offset_1024_32,
+    swb_offset_1024_24, swb_offset_1024_24, swb_offset_1024_16,
+    swb_offset_1024_16, swb_offset_1024_16, swb_offset_1024_8
+};
+
+static const uint16_t *swb_offset_128[] = {
+    /* The last entry on the following row is swb_offset_128_64 but is a
+       duplicate of swb_offset_128_96. */
+    swb_offset_128_96, swb_offset_128_96, swb_offset_128_96,
+    swb_offset_128_48, swb_offset_128_48, swb_offset_128_48,
+    swb_offset_128_24, swb_offset_128_24, swb_offset_128_16,
+    swb_offset_128_16, swb_offset_128_16, swb_offset_128_8
+};
+
+// @}
+
+#endif /* FFMPEG_AACDECTAB_H */

libavcodec/aactab.c

@@ -27,6 +27,7 @@
  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
  */
 
+#include "libavutil/mem.h"
 #include "aac.h"
 
 #include <stdint.h>
@@ -795,4 +796,86 @@ const float ff_aac_ivquant_tab[IVQUANT_SIZE] = {
      4064.0312908,  4074.6805676,  4085.3368071,  4096.0000000,
 };
 
+const float ff_aac_pow2sf_tab[316] = {
+    8.88178420e-16, 1.05622810e-15, 1.25607397e-15, 1.49373210e-15,
+    1.77635684e-15, 2.11245619e-15, 2.51214793e-15, 2.98746420e-15,
+    3.55271368e-15, 4.22491238e-15, 5.02429587e-15, 5.97492839e-15,
+    7.10542736e-15, 8.44982477e-15, 1.00485917e-14, 1.19498568e-14,
+    1.42108547e-14, 1.68996495e-14, 2.00971835e-14, 2.38997136e-14,
+    2.84217094e-14, 3.37992991e-14, 4.01943669e-14, 4.77994272e-14,
+    5.68434189e-14, 6.75985982e-14, 8.03887339e-14, 9.55988543e-14,
+    1.13686838e-13, 1.35197196e-13, 1.60777468e-13, 1.91197709e-13,
+    2.27373675e-13, 2.70394393e-13, 3.21554936e-13, 3.82395417e-13,
+    4.54747351e-13, 5.40788785e-13, 6.43109871e-13, 7.64790834e-13,
+    9.09494702e-13, 1.08157757e-12, 1.28621974e-12, 1.52958167e-12,
+    1.81898940e-12, 2.16315514e-12, 2.57243948e-12, 3.05916334e-12,
+    3.63797881e-12, 4.32631028e-12, 5.14487897e-12, 6.11832668e-12,
+    7.27595761e-12, 8.65262056e-12, 1.02897579e-11, 1.22366534e-11,
+    1.45519152e-11, 1.73052411e-11, 2.05795159e-11, 2.44733067e-11,
+    2.91038305e-11, 3.46104823e-11, 4.11590317e-11, 4.89466134e-11,
+    5.82076609e-11, 6.92209645e-11, 8.23180635e-11, 9.78932268e-11,
+    1.16415322e-10, 1.38441929e-10, 1.64636127e-10, 1.95786454e-10,
+    2.32830644e-10, 2.76883858e-10, 3.29272254e-10, 3.91572907e-10,
+    4.65661287e-10, 5.53767716e-10, 6.58544508e-10, 7.83145814e-10,
+    9.31322575e-10, 1.10753543e-09, 1.31708902e-09, 1.56629163e-09,
+    1.86264515e-09, 2.21507086e-09, 2.63417803e-09, 3.13258326e-09,
+    3.72529030e-09, 4.43014173e-09, 5.26835606e-09, 6.26516652e-09,
+    7.45058060e-09, 8.86028346e-09, 1.05367121e-08, 1.25303330e-08,
+    1.49011612e-08, 1.77205669e-08, 2.10734243e-08, 2.50606661e-08,
+    2.98023224e-08, 3.54411338e-08, 4.21468485e-08, 5.01213321e-08,
+    5.96046448e-08, 7.08822677e-08, 8.42936970e-08, 1.00242664e-07,
+    1.19209290e-07, 1.41764535e-07, 1.68587394e-07, 2.00485328e-07,
+    2.38418579e-07, 2.83529071e-07, 3.37174788e-07, 4.00970657e-07,
+    4.76837158e-07, 5.67058141e-07, 6.74349576e-07, 8.01941314e-07,
+    9.53674316e-07, 1.13411628e-06, 1.34869915e-06, 1.60388263e-06,
+    1.90734863e-06, 2.26823256e-06, 2.69739830e-06, 3.20776526e-06,
+    3.81469727e-06, 4.53646513e-06, 5.39479661e-06, 6.41553051e-06,
+    7.62939453e-06, 9.07293026e-06, 1.07895932e-05, 1.28310610e-05,
+    1.52587891e-05, 1.81458605e-05, 2.15791864e-05, 2.56621220e-05,
+    3.05175781e-05, 3.62917210e-05, 4.31583729e-05, 5.13242441e-05,
+    6.10351562e-05, 7.25834421e-05, 8.63167458e-05, 1.02648488e-04,
+    1.22070312e-04, 1.45166884e-04, 1.72633492e-04, 2.05296976e-04,
+    2.44140625e-04, 2.90333768e-04, 3.45266983e-04, 4.10593953e-04,
+    4.88281250e-04, 5.80667537e-04, 6.90533966e-04, 8.21187906e-04,
+    9.76562500e-04, 1.16133507e-03, 1.38106793e-03, 1.64237581e-03,
+    1.95312500e-03, 2.32267015e-03, 2.76213586e-03, 3.28475162e-03,
+    3.90625000e-03, 4.64534029e-03, 5.52427173e-03, 6.56950324e-03,
+    7.81250000e-03, 9.29068059e-03, 1.10485435e-02, 1.31390065e-02,
+    1.56250000e-02, 1.85813612e-02, 2.20970869e-02, 2.62780130e-02,
+    3.12500000e-02, 3.71627223e-02, 4.41941738e-02, 5.25560260e-02,
+    6.25000000e-02, 7.43254447e-02, 8.83883476e-02, 1.05112052e-01,
+    1.25000000e-01, 1.48650889e-01, 1.76776695e-01, 2.10224104e-01,
+    2.50000000e-01, 2.97301779e-01, 3.53553391e-01, 4.20448208e-01,
+    5.00000000e-01, 5.94603558e-01, 7.07106781e-01, 8.40896415e-01,
+    1.00000000e+00, 1.18920712e+00, 1.41421356e+00, 1.68179283e+00,
+    2.00000000e+00, 2.37841423e+00, 2.82842712e+00, 3.36358566e+00,
+    4.00000000e+00, 4.75682846e+00, 5.65685425e+00, 6.72717132e+00,
+    8.00000000e+00, 9.51365692e+00, 1.13137085e+01, 1.34543426e+01,
+    1.60000000e+01, 1.90273138e+01, 2.26274170e+01, 2.69086853e+01,
+    3.20000000e+01, 3.80546277e+01, 4.52548340e+01, 5.38173706e+01,
+    6.40000000e+01, 7.61092554e+01, 9.05096680e+01, 1.07634741e+02,
+    1.28000000e+02, 1.52218511e+02, 1.81019336e+02, 2.15269482e+02,
+    2.56000000e+02, 3.04437021e+02, 3.62038672e+02, 4.30538965e+02,
+    5.12000000e+02, 6.08874043e+02, 7.24077344e+02, 8.61077929e+02,
+    1.02400000e+03, 1.21774809e+03, 1.44815469e+03, 1.72215586e+03,
+    2.04800000e+03, 2.43549617e+03, 2.89630938e+03, 3.44431172e+03,
+    4.09600000e+03, 4.87099234e+03, 5.79261875e+03, 6.88862343e+03,
+    8.19200000e+03, 9.74198469e+03, 1.15852375e+04, 1.37772469e+04,
+    1.63840000e+04, 1.94839694e+04, 2.31704750e+04, 2.75544937e+04,
+    3.27680000e+04, 3.89679387e+04, 4.63409500e+04, 5.51089875e+04,
+    6.55360000e+04, 7.79358775e+04, 9.26819000e+04, 1.10217975e+05,
+    1.31072000e+05, 1.55871755e+05, 1.85363800e+05, 2.20435950e+05,
+    2.62144000e+05, 3.11743510e+05, 3.70727600e+05, 4.40871900e+05,
+    5.24288000e+05, 6.23487020e+05, 7.41455200e+05, 8.81743800e+05,
+    1.04857600e+06, 1.24697404e+06, 1.48291040e+06, 1.76348760e+06,
+    2.09715200e+06, 2.49394808e+06, 2.96582080e+06, 3.52697520e+06,
+    4.19430400e+06, 4.98789616e+06, 5.93164160e+06, 7.05395040e+06,
+    8.38860800e+06, 9.97579232e+06, 1.18632832e+07, 1.41079008e+07,
+    1.67772160e+07, 1.99515846e+07, 2.37265664e+07, 2.82158016e+07,
+    3.35544320e+07, 3.99031693e+07, 4.74531328e+07, 5.64316032e+07,
+    6.71088640e+07, 7.98063385e+07, 9.49062656e+07, 1.12863206e+08,
+    1.34217728e+08, 1.59612677e+08, 1.89812531e+08, 2.25726413e+08,
+    2.68435456e+08, 3.19225354e+08, 3.79625062e+08, 4.51452825e+08,
+};
+
 #endif /* CONFIG_HARDCODED_TABLES */

libavcodec/aactab.h

@@ -30,6 +30,7 @@
 #ifndef FFMPEG_AACTAB_H
 #define FFMPEG_AACTAB_H
 
+#include "libavutil/mem.h"
 #include "aac.h"
 
 #include <stdint.h>
@@ -45,6 +46,7 @@ extern const int8_t *ff_aac_codebook_vectors[];
 
 #ifdef CONFIG_HARDCODED_TABLES
 extern const float ff_aac_ivquant_tab[IVQUANT_SIZE];
+extern const float ff_aac_pow2sf_tab[316];
 #endif /* CONFIG_HARDCODED_TABLES */
 
 #endif /* FFMPEG_AACTAB_H */