Backport AAC-HE v2 from trunk

This patch has seen testing for a couple of weeks in ubuntu maverick and debian/experimental w/o negative feedback so far.



Originally committed as revision 24576 to svn://svn.ffmpeg.org/ffmpeg/branches/0.6

libavcodec/Makefile

@@ -43,7 +43,7 @@ OBJS-$(CONFIG_VAAPI)                   += vaapi.o
 OBJS-$(CONFIG_VDPAU)                   += vdpau.o
 
 # decoders/encoders/hardware accelerators
-OBJS-$(CONFIG_AAC_DECODER)             += aac.o aactab.o aacsbr.o
+OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o
 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                           aacpsy.o aactab.o      \
                                           psymodel.o iirfilter.o \

libavcodec/aac.h

@@ -38,12 +38,6 @@
 
 #include <stdint.h>
 
-#define AAC_INIT_VLC_STATIC(num, size) \
-    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
-         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
-        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 MAX_ELEM_ID 16
 
@@ -241,7 +235,7 @@ typedef struct {
  * main AAC context
  */
 typedef struct {
-    AVCodecContext * avccontext;
+    AVCodecContext *avctx;
 
     MPEG4AudioConfig m4ac;
 
@@ -255,8 +249,9 @@ typedef struct {
     enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
                                                    *   first index as the first 4 raw data block types
                                                    */
-    ChannelElement * che[4][MAX_ELEM_ID];
-    ChannelElement * tag_che_map[4][MAX_ELEM_ID];
+    ChannelElement          *che[4][MAX_ELEM_ID];
+    ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
+    uint8_t tags_seen_this_frame[4][MAX_ELEM_ID];
     int tags_mapped;
     /** @} */
 

libavcodec/aac_tablegen.c

@@ -0,0 +1,39 @@
+/*
+ * Generate a header file for hardcoded AAC tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#include "aac_tablegen.h"
+#include "tableprint.h"
+
+int main(void)
+{
+    ff_aac_tableinit();
+
+    write_fileheader();
+
+    printf("const float ff_aac_pow2sf_tab[428] = {\n");
+    write_float_array(ff_aac_pow2sf_tab, 428);
+    printf("};\n");
+
+    return 0;
+}

libavcodec/aac_tablegen.h

@@ -0,0 +1,42 @@
+/*
+ * Header file for hardcoded AAC tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#ifndef AAC_TABLEGEN_H
+#define AAC_TABLEGEN_H
+
+#include "aac_tablegen_decl.h"
+
+#if CONFIG_HARDCODED_TABLES
+#include "libavcodec/aac_tables.h"
+#else
+#include "../libavutil/mathematics.h"
+float ff_aac_pow2sf_tab[428];
+
+void ff_aac_tableinit(void)
+{
+    int i;
+    for (i = 0; i < 428; i++)
+        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
+}
+#endif /* CONFIG_HARDCODED_TABLES */
+
+#endif /* AAC_TABLEGEN_H */

libavcodec/aac_tablegen_decl.h

@@ -0,0 +1,34 @@
+/*
+ * Header file for hardcoded AAC tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#ifndef AAC_TABLEGEN_INIT_H
+#define AAC_TABLEGEN_INIT_H
+
+#if CONFIG_HARDCODED_TABLES
+#define ff_aac_tableinit()
+extern const float ff_aac_pow2sf_tab[428];
+#else
+void ff_aac_tableinit(void);
+extern       float ff_aac_pow2sf_tab[428];
+#endif /* CONFIG_HARDCODED_TABLES */
+
+#endif /* AAC_TABLEGEN_INIT_H */

libavcodec/aac.c → libavcodec/aacdec.c

@@ -67,7 +67,7 @@
  * Y (not in this code) Layer-2
  * Y (not in this code) Layer-3
  * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
- * N (planned)          Parametric Stereo
+ * Y                    Parametric Stereo
  * N                    Direct Stream Transfer
  *
  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
@@ -113,6 +113,22 @@ static const char overread_err[] = "Input buffer exhausted before END element fo
 
 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
 {
+    /* Some buggy encoders appear to set all elem_ids to zero and rely on
+    channels always occurring in the same order. This is expressly forbidden
+    by the spec but we will try to work around it.
+    */
+    int err_printed = 0;
+    while (ac->tags_seen_this_frame[type][elem_id] && elem_id < MAX_ELEM_ID) {
+        if (ac->output_configured < OC_LOCKED && !err_printed) {
+            av_log(ac->avctx, AV_LOG_WARNING, "Duplicate channel tag found, attempting to remap.\n");
+            err_printed = 1;
+        }
+        elem_id++;
+    }
+    if (elem_id == MAX_ELEM_ID)
+        return NULL;
+    ac->tags_seen_this_frame[type][elem_id] = 1;
+
     if (ac->tag_che_map[type][elem_id]) {
         return ac->tag_che_map[type][elem_id];
     }
@@ -127,8 +143,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
         }
     case 6:
         /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
-           instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
-           encountered such a stream, transfer the LFE[0] element to SCE[1] */
+           instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
+           encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
         if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
             ac->tags_mapped++;
             return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
@@ -184,7 +200,8 @@ static av_cold int che_configure(AACContext *ac,
         ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
         if (type != TYPE_CCE) {
             ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
-            if (type == TYPE_CPE) {
+            if (type == TYPE_CPE ||
+                (type == TYPE_SCE && ac->m4ac.ps == 1)) {
                 ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
             }
         }
@@ -209,9 +226,10 @@ static av_cold int output_configure(AACContext *ac,
                             enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
                             int channel_config, enum OCStatus oc_type)
 {
-    AVCodecContext *avctx = ac->avccontext;
+    AVCodecContext *avctx = ac->avctx;
     int i, type, channels = 0, ret;
 
+    if (new_che_pos != che_pos)
     memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
 
     if (channel_config) {
@@ -292,7 +310,7 @@ static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_EL
 
     sampling_index = get_bits(gb, 4);
     if (ac->m4ac.sampling_index != sampling_index)
-        av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
+        av_log(ac->avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
 
     num_front       = get_bits(gb, 4);
     num_side        = get_bits(gb, 4);
@@ -323,7 +341,7 @@ static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_EL
     /* comment field, first byte is length */
     comment_len = get_bits(gb, 8) * 8;
     if (get_bits_left(gb) < comment_len) {
-        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
+        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
         return -1;
     }
     skip_bits_long(gb, comment_len);
@@ -343,7 +361,7 @@ static av_cold int set_default_channel_config(AACContext *ac,
                                       int channel_config)
 {
     if (channel_config < 1 || channel_config > 7) {
-        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
+        av_log(ac->avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
                channel_config);
         return -1;
     }
@@ -388,7 +406,7 @@ static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
     int extension_flag, ret;
 
     if (get_bits1(gb)) { // frameLengthFlag
-        av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
+        av_log_missing_feature(ac->avctx, "960/120 MDCT window is", 1);
         return -1;
     }
 
@@ -452,9 +470,11 @@ static int decode_audio_specific_config(AACContext *ac, void *data,
     if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
         return -1;
     if (ac->m4ac.sampling_index > 12) {
-        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+        av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
         return -1;
     }
+    if (ac->m4ac.sbr == 1 && ac->m4ac.ps == -1)
+        ac->m4ac.ps = 1;
 
     skip_bits_long(&gb, i);
 
@@ -465,7 +485,7 @@ static int decode_audio_specific_config(AACContext *ac, void *data,
             return -1;
         break;
     default:
-        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
+        av_log(ac->avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
                ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
         return -1;
     }
@@ -508,20 +528,25 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
         reset_predict_state(&ps[i]);
 }
 
-static av_cold int aac_decode_init(AVCodecContext *avccontext)
+#define AAC_INIT_VLC_STATIC(num, size) \
+    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
+         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
+        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
+        size);
+
+static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
-    AACContext *ac = avccontext->priv_data;
-    int i;
+    AACContext *ac = avctx->priv_data;
 
-    ac->avccontext = avccontext;
-    ac->m4ac.sample_rate = avccontext->sample_rate;
+    ac->avctx = avctx;
+    ac->m4ac.sample_rate = avctx->sample_rate;
 
-    if (avccontext->extradata_size > 0) {
-        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
+    if (avctx->extradata_size > 0) {
+        if (decode_audio_specific_config(ac, avctx->extradata, avctx->extradata_size))
             return -1;
     }
 
-    avccontext->sample_fmt = SAMPLE_FMT_S16;
+    avctx->sample_fmt = SAMPLE_FMT_S16;
 
     AAC_INIT_VLC_STATIC( 0, 304);
     AAC_INIT_VLC_STATIC( 1, 270);
@@ -537,7 +562,7 @@ static av_cold int aac_decode_init(AVCodecContext *avccontext)
 
     ff_aac_sbr_init();
 
-    dsputil_init(&ac->dsp, avccontext);
+    dsputil_init(&ac->dsp, avctx);
 
     ac->random_state = 0x1f2e3d4c;
 
@@ -555,10 +580,7 @@ static av_cold int aac_decode_init(AVCodecContext *avccontext)
         ac->sf_offset = 60;
     }
 
-#if !CONFIG_HARDCODED_TABLES
-    for (i = 0; i < 428; i++)
-        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
-#endif /* CONFIG_HARDCODED_TABLES */
+    ff_aac_tableinit();
 
     INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
                     ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
@@ -591,7 +613,7 @@ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
         align_get_bits(gb);
 
     if (get_bits_left(gb) < 8 * count) {
-        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
+        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
         return -1;
     }
     skip_bits_long(gb, 8 * count);
@@ -605,7 +627,7 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
     if (get_bits1(gb)) {
         ics->predictor_reset_group = get_bits(gb, 5);
         if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
             return -1;
         }
     }
@@ -624,7 +646,7 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                            GetBitContext *gb, int common_window)
 {
     if (get_bits1(gb)) {
-        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
+        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
         memset(ics, 0, sizeof(IndividualChannelStream));
         return -1;
     }
@@ -665,11 +687,11 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                     return -1;
                 }
             } else if (ac->m4ac.object_type == AOT_AAC_LC) {
-                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
+                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
                 memset(ics, 0, sizeof(IndividualChannelStream));
                 return -1;
             } else {
-                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
+                av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1);
                 memset(ics, 0, sizeof(IndividualChannelStream));
                 return -1;
             }
@@ -677,7 +699,7 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
     }
 
     if (ics->max_sfb > ics->num_swb) {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
                ics->max_sfb, ics->num_swb);
         memset(ics, 0, sizeof(IndividualChannelStream));
@@ -708,18 +730,18 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
             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");
+                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
                 return -1;
             }
             while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
                 sect_end += sect_len_incr;
             sect_end += sect_len_incr;
             if (get_bits_left(gb) < 0) {
-                av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
+                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
                 return -1;
             }
             if (sect_end > ics->max_sfb) {
-                av_log(ac->avccontext, AV_LOG_ERROR,
+                av_log(ac->avctx, AV_LOG_ERROR,
                        "Number of bands (%d) exceeds limit (%d).\n",
                        sect_end, ics->max_sfb);
                 return -1;
@@ -764,7 +786,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                 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,
+                        av_log(ac->avctx, AV_LOG_ERROR,
                                "%s (%d) out of range.\n", sf_str[2], offset[2]);
                         return -1;
                     }
@@ -777,7 +799,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                     else
                         offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                     if (offset[1] > 255U) {
-                        av_log(ac->avccontext, AV_LOG_ERROR,
+                        av_log(ac->avctx, AV_LOG_ERROR,
                                "%s (%d) out of range.\n", sf_str[1], offset[1]);
                         return -1;
                     }
@@ -787,7 +809,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                 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,
+                        av_log(ac->avctx, AV_LOG_ERROR,
                                "%s (%d) out of range.\n", sf_str[0], offset[0]);
                         return -1;
                     }
@@ -844,7 +866,7 @@ static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
                 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
 
                 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
-                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
+                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
                            tns->order[w][filt], tns_max_order);
                     tns->order[w][filt] = 0;
                     return -1;
@@ -1163,7 +1185,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
                                     b = 31 - av_log2(~b);
 
                                     if (b > 8) {
-                                        av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
+                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
                                         return -1;
                                     }
 
@@ -1216,7 +1238,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
     return 0;
 
 err_cb_overflow:
-    av_log(ac->avccontext, AV_LOG_ERROR,
+    av_log(ac->avctx, AV_LOG_ERROR,
            "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
            band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
     return -1;
@@ -1337,18 +1359,18 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
     if (!scale_flag) {
         if ((pulse_present = get_bits1(gb))) {
             if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
+                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
                 return -1;
             }
             if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
-                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
+                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
                 return -1;
             }
         }
         if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
             return -1;
         if (get_bits1(gb)) {
-            av_log_missing_feature(ac->avccontext, "SSR", 1);
+            av_log_missing_feature(ac->avctx, "SSR", 1);
             return -1;
         }
     }
@@ -1448,7 +1470,7 @@ static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
         cpe->ch[1].ics.use_kb_window[1] = i;
         ms_present = get_bits(gb, 2);
         if (ms_present == 3) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
             return -1;
         } else if (ms_present)
             decode_mid_side_stereo(cpe, gb, ms_present);
@@ -1635,16 +1657,20 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
         crc_flag++;
     case EXT_SBR_DATA:
         if (!che) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
             return res;
         } else if (!ac->m4ac.sbr) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
             skip_bits_long(gb, 8 * cnt - 4);
             return res;
         } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
             skip_bits_long(gb, 8 * cnt - 4);
             return res;
+        } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
+            ac->m4ac.sbr = 1;
+            ac->m4ac.ps = 1;
+            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
         } else {
             ac->m4ac.sbr = 1;
         }
@@ -1728,7 +1754,7 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float
     // imdct
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
-            av_log(ac->avccontext, AV_LOG_WARNING,
+            av_log(ac->avctx, AV_LOG_WARNING,
                    "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
                    "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
         for (i = 0; i < 1024; i += 128)
@@ -1794,7 +1820,7 @@ static void apply_dependent_coupling(AACContext *ac,
     const float *src = cce->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,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Dependent coupling is not supported together with LTP\n");
         return;
     }
@@ -1924,58 +1950,65 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
         } else if (ac->output_configured != OC_LOCKED) {
             ac->output_configured = OC_NONE;
         }
-        if (ac->output_configured != OC_LOCKED)
+        if (ac->output_configured != OC_LOCKED) {
             ac->m4ac.sbr = -1;
+            ac->m4ac.ps  = -1;
+        }
         ac->m4ac.sample_rate     = hdr_info.sample_rate;
         ac->m4ac.sampling_index  = hdr_info.sampling_index;
         ac->m4ac.object_type     = hdr_info.object_type;
-        if (!ac->avccontext->sample_rate)
-            ac->avccontext->sample_rate = hdr_info.sample_rate;
+        if (!ac->avctx->sample_rate)
+            ac->avctx->sample_rate = hdr_info.sample_rate;
         if (hdr_info.num_aac_frames == 1) {
             if (!hdr_info.crc_absent)
                 skip_bits(gb, 16);
         } else {
-            av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
+            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
             return -1;
         }
     }
     return size;
 }
 
-static int aac_decode_frame(AVCodecContext *avccontext, void *data,
+static int aac_decode_frame(AVCodecContext *avctx, void *data,
                             int *data_size, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
-    AACContext *ac = avccontext->priv_data;
+    AACContext *ac = avctx->priv_data;
     ChannelElement *che = NULL, *che_prev = NULL;
     GetBitContext gb;
     enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
     int err, elem_id, data_size_tmp;
     int buf_consumed;
-    int samples = 1024, multiplier;
+    int samples = 0, multiplier;
     int buf_offset;
 
     init_get_bits(&gb, buf, buf_size * 8);
 
     if (show_bits(&gb, 12) == 0xfff) {
         if (parse_adts_frame_header(ac, &gb) < 0) {
-            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
             return -1;
         }
         if (ac->m4ac.sampling_index > 12) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
             return -1;
         }
     }
 
+    memset(ac->tags_seen_this_frame, 0, sizeof(ac->tags_seen_this_frame));
     // parse
     while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
         elem_id = get_bits(&gb, 4);
 
-        if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
-            return -1;
+        if (elem_type < TYPE_DSE) {
+            if (!(che=get_che(ac, elem_type, elem_id))) {
+                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
+                       elem_type, elem_id);
+                return -1;
+            }
+            samples = 1024;
         }
 
         switch (elem_type) {
@@ -2006,7 +2039,7 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
             if ((err = decode_pce(ac, new_che_pos, &gb)))
                 break;
             if (ac->output_configured > OC_TRIAL_PCE)
-                av_log(avccontext, AV_LOG_ERROR,
+                av_log(avctx, AV_LOG_ERROR,
                        "Not evaluating a further program_config_element as this construct is dubious at best.\n");
             else
                 err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
@@ -2017,7 +2050,7 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
             if (elem_id == 15)
                 elem_id += get_bits(&gb, 8) - 1;
             if (get_bits_left(&gb) < 8 * elem_id) {
-                    av_log(avccontext, AV_LOG_ERROR, overread_err);
+                    av_log(avctx, AV_LOG_ERROR, overread_err);
                     return -1;
             }
             while (elem_id > 0)
@@ -2037,7 +2070,7 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
             return err;
 
         if (get_bits_left(&gb) < 3) {
-            av_log(avccontext, AV_LOG_ERROR, overread_err);
+            av_log(avctx, AV_LOG_ERROR, overread_err);
             return -1;
         }
     }
@@ -2047,20 +2080,21 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
     multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
     samples <<= multiplier;
     if (ac->output_configured < OC_LOCKED) {
-        avccontext->sample_rate = ac->m4ac.sample_rate << multiplier;
-        avccontext->frame_size = samples;
+        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
+        avctx->frame_size = samples;
     }
 
-    data_size_tmp = samples * avccontext->channels * sizeof(int16_t);
+    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
     if (*data_size < data_size_tmp) {
-        av_log(avccontext, AV_LOG_ERROR,
+        av_log(avctx, AV_LOG_ERROR,
                "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
                *data_size, data_size_tmp);
         return -1;
     }
     *data_size = data_size_tmp;
 
-    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avccontext->channels);
+    if (samples)
+        ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
 
     if (ac->output_configured)
         ac->output_configured = OC_LOCKED;
@@ -2073,9 +2107,9 @@ static int aac_decode_frame(AVCodecContext *avccontext, void *data,
     return buf_size > buf_offset ? buf_consumed : buf_size;
 }
 
-static av_cold int aac_decode_close(AVCodecContext *avccontext)
+static av_cold int aac_decode_close(AVCodecContext *avctx)
 {
-    AACContext *ac = avccontext->priv_data;
+    AACContext *ac = avctx->priv_data;
     int i, type;
 
     for (i = 0; i < MAX_ELEM_ID; i++) {

libavcodec/aacenc.c

@@ -201,13 +201,11 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     lengths[1] = ff_aac_num_swb_128[i];
     ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
     s->psypp = ff_psy_preprocess_init(avctx);
-    s->coder = &ff_aac_coders[0];
+    s->coder = &ff_aac_coders[2];
 
     s->lambda = avctx->global_quality ? avctx->global_quality : 120;
-#if !CONFIG_HARDCODED_TABLES
-    for (i = 0; i < 428; i++)
-        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
-#endif /* CONFIG_HARDCODED_TABLES */
+
+    ff_aac_tableinit();
 
     if (avctx->channels > 5)
         av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
@@ -234,25 +232,21 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
                 s->output[i] = sce->saved[i];
         }
         if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
-            j = channel;
-            for (i = 0; i < 1024; i++, j += avctx->channels) {
+            for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) {
                 s->output[i+1024]         = audio[j] * lwindow[1024 - i - 1];
                 sce->saved[i] = audio[j] * lwindow[i];
             }
         } else {
-            j = channel;
-            for (i = 0; i < 448; i++, j += avctx->channels)
+            for (i = 0, j = channel; i < 448; i++, j += avctx->channels)
                 s->output[i+1024]         = audio[j];
-            for (i = 448; i < 576; i++, j += avctx->channels)
+            for (; i < 576; i++, j += avctx->channels)
                 s->output[i+1024]         = audio[j] * swindow[576 - i - 1];
             memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
-            j = channel;
-            for (i = 0; i < 1024; i++, j += avctx->channels)
+            for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
                 sce->saved[i] = audio[j];
         }
         ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
     } else {
-        j = channel;
         for (k = 0; k < 1024; k += 128) {
             for (i = 448 + k; i < 448 + k + 256; i++)
                 s->output[i - 448 - k] = (i < 1024)
@@ -262,8 +256,7 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
             s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
             ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
         }
-        j = channel;
-        for (i = 0; i < 1024; i++, j += avctx->channels)
+        for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
             sce->saved[i] = audio[j];
     }
 }
@@ -562,6 +555,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
             cpe      = &s->cpe[i];
             for (j = 0; j < chans; j++) {
                 s->cur_channel = start_ch + j;
+                ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
             }
             cpe->common_window = 0;
@@ -592,7 +586,6 @@ static int aac_encode_frame(AVCodecContext *avctx,
             }
             for (j = 0; j < chans; j++) {
                 s->cur_channel = start_ch + j;
-                ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
                 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
             }
             start_ch += chans;

libavcodec/aacenc.h

@@ -64,7 +64,7 @@ typedef struct AACEncContext {
     int cur_channel;
     int last_frame;
     float lambda;
-    DECLARE_ALIGNED(16, int,   qcoefs)[96][2];   ///< quantized coefficients
+    DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
     DECLARE_ALIGNED(16, float, scoefs)[1024];    ///< scaled coefficients
 } AACEncContext;
 

libavcodec/aacps.h

@@ -0,0 +1,82 @@
+/*
+ * MPEG-4 Parametric Stereo definitions and declarations
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#ifndef AVCODEC_PS_H
+#define AVCODEC_PS_H
+
+#include <stdint.h>
+
+#include "avcodec.h"
+#include "get_bits.h"
+
+#define PS_MAX_NUM_ENV 5
+#define PS_MAX_NR_IIDICC 34
+#define PS_MAX_NR_IPDOPD 17
+#define PS_MAX_SSB 91
+#define PS_MAX_AP_BANDS 50
+#define PS_QMF_TIME_SLOTS 32
+#define PS_MAX_DELAY 14
+#define PS_AP_LINKS 3
+#define PS_MAX_AP_DELAY 5
+
+typedef struct {
+    int    start;
+    int    enable_iid;
+    int    iid_quant;
+    int    nr_iid_par;
+    int    nr_ipdopd_par;
+    int    enable_icc;
+    int    icc_mode;
+    int    nr_icc_par;
+    int    enable_ext;
+    int    frame_class;
+    int    num_env_old;
+    int    num_env;
+    int    enable_ipdopd;
+    int    border_position[PS_MAX_NUM_ENV+1];
+    int8_t iid_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Intensity Difference Parameters
+    int8_t icc_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-Channel Coherence Parameters
+    /* ipd/opd is iid/icc sized so that the same functions can handle both */
+    int8_t ipd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Inter-channel Phase Difference Parameters
+    int8_t opd_par[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC]; //<Overall Phase Difference Parameters
+    int    is34bands;
+    int    is34bands_old;
+
+    float  in_buf[5][44][2];
+    float  delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    float  ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    float  peak_decay_nrg[34];
+    float  power_smooth[34];
+    float  peak_decay_diff_smooth[34];
+    float  H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    int8_t opd_hist[PS_MAX_NR_IIDICC];
+    int8_t ipd_hist[PS_MAX_NR_IIDICC];
+} PSContext;
+
+void ff_ps_init(void);
+void ff_ps_ctx_init(PSContext *ps);
+int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
+int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
+
+#endif /* AVCODEC_PS_H */

libavcodec/aacps_tablegen.c

@@ -0,0 +1,93 @@
+/*
+ * Generate a header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#include "aacps_tablegen.h"
+#include "tableprint.h"
+
+void write_float_3d_array (const void *p, int b, int c, int d)
+{
+    int i;
+    const float *f = p;
+    for (i = 0; i < b; i++) {
+        printf("{\n");
+        write_float_2d_array(f, c, d);
+        printf("},\n");
+        f += c * d;
+    }
+}
+
+void write_float_4d_array (const void *p, int a, int b, int c, int d)
+{
+    int i;
+    const float *f = p;
+    for (i = 0; i < a; i++) {
+        printf("{\n");
+        write_float_3d_array(f, b, c, d);
+        printf("},\n");
+        f += b * c * d;
+    }
+}
+
+int main(void)
+{
+    ps_tableinit();
+
+    write_fileheader();
+
+    printf("static const float pd_re_smooth[8*8*8] = {\n");
+    write_float_array(pd_re_smooth, 8*8*8);
+    printf("};\n");
+    printf("static const float pd_im_smooth[8*8*8] = {\n");
+    write_float_array(pd_im_smooth, 8*8*8);
+    printf("};\n");
+
+    printf("static const float HA[46][8][4] = {\n");
+    write_float_3d_array(HA, 46, 8, 4);
+    printf("};\n");
+    printf("static const float HB[46][8][4] = {\n");
+    write_float_3d_array(HB, 46, 8, 4);
+    printf("};\n");
+
+    printf("static const float f20_0_8[8][7][2] = {\n");
+    write_float_3d_array(f20_0_8, 8, 7, 2);
+    printf("};\n");
+    printf("static const float f34_0_12[12][7][2] = {\n");
+    write_float_3d_array(f34_0_12, 12, 7, 2);
+    printf("};\n");
+    printf("static const float f34_1_8[8][7][2] = {\n");
+    write_float_3d_array(f34_1_8, 8, 7, 2);
+    printf("};\n");
+    printf("static const float f34_2_4[4][7][2] = {\n");
+    write_float_3d_array(f34_2_4, 4, 7, 2);
+    printf("};\n");
+
+    printf("static const float Q_fract_allpass[2][50][3][2] = {\n");
+    write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
+    printf("};\n");
+    printf("static const float phi_fract[2][50][2] = {\n");
+    write_float_3d_array(phi_fract, 2, 50, 2);
+    printf("};\n");
+
+    return 0;
+}

libavcodec/aacps_tablegen.h

@@ -0,0 +1,212 @@
+/*
+ * Header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#ifndef AACPS_TABLEGEN_H
+#define AACPS_TABLEGEN_H
+
+#include <stdint.h>
+
+#if CONFIG_HARDCODED_TABLES
+#define ps_tableinit()
+#include "libavcodec/aacps_tables.h"
+#else
+#include "../libavutil/common.h"
+#include "../libavutil/mathematics.h"
+#define NR_ALLPASS_BANDS20 30
+#define NR_ALLPASS_BANDS34 50
+#define PS_AP_LINKS 3
+static float pd_re_smooth[8*8*8];
+static float pd_im_smooth[8*8*8];
+static float HA[46][8][4];
+static float HB[46][8][4];
+static float f20_0_8 [ 8][7][2];
+static float f34_0_12[12][7][2];
+static float f34_1_8 [ 8][7][2];
+static float f34_2_4 [ 4][7][2];
+static float Q_fract_allpass[2][50][3][2];
+static float phi_fract[2][50][2];
+
+static const float g0_Q8[] = {
+    0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
+    0.09885108575264f, 0.11793710567217f, 0.125f
+};
+
+static const float g0_Q12[] = {
+    0.04081179924692f, 0.03812810994926f, 0.05144908135699f, 0.06399831151592f,
+    0.07428313801106f, 0.08100347892914f, 0.08333333333333f
+};
+
+static const float g1_Q8[] = {
+    0.01565675600122f, 0.03752716391991f, 0.05417891378782f, 0.08417044116767f,
+    0.10307344158036f, 0.12222452249753f, 0.125f
+};
+
+static const float g2_Q4[] = {
+    -0.05908211155639f, -0.04871498374946f, 0.0f,   0.07778723915851f,
+     0.16486303567403f,  0.23279856662996f, 0.25f
+};
+
+static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
+{
+    int q, n;
+    for (q = 0; q < bands; q++) {
+        for (n = 0; n < 7; n++) {
+            double theta = 2 * M_PI * (q + 0.5) * (n - 6) / bands;
+            filter[q][n][0] = proto[n] *  cos(theta);
+            filter[q][n][1] = proto[n] * -sin(theta);
+        }
+    }
+}
+
+static void ps_tableinit(void)
+{
+    static const float ipdopd_sin[] = { 0, M_SQRT1_2, 1,  M_SQRT1_2,  0, -M_SQRT1_2, -1, -M_SQRT1_2 };
+    static const float ipdopd_cos[] = { 1, M_SQRT1_2, 0, -M_SQRT1_2, -1, -M_SQRT1_2,  0,  M_SQRT1_2 };
+    int pd0, pd1, pd2;
+
+    static const float iid_par_dequant[] = {
+        //iid_par_dequant_default
+        0.05623413251903, 0.12589254117942, 0.19952623149689, 0.31622776601684,
+        0.44668359215096, 0.63095734448019, 0.79432823472428, 1,
+        1.25892541179417, 1.58489319246111, 2.23872113856834, 3.16227766016838,
+        5.01187233627272, 7.94328234724282, 17.7827941003892,
+        //iid_par_dequant_fine
+        0.00316227766017, 0.00562341325190, 0.01,             0.01778279410039,
+        0.03162277660168, 0.05623413251903, 0.07943282347243, 0.11220184543020,
+        0.15848931924611, 0.22387211385683, 0.31622776601684, 0.39810717055350,
+        0.50118723362727, 0.63095734448019, 0.79432823472428, 1,
+        1.25892541179417, 1.58489319246111, 1.99526231496888, 2.51188643150958,
+        3.16227766016838, 4.46683592150963, 6.30957344480193, 8.91250938133745,
+        12.5892541179417, 17.7827941003892, 31.6227766016838, 56.2341325190349,
+        100,              177.827941003892, 316.227766016837,
+    };
+    static const float icc_invq[] = {
+        1, 0.937,      0.84118,    0.60092,    0.36764,   0,      -0.589,    -1
+    };
+    static const float acos_icc_invq[] = {
+        0, 0.35685527, 0.57133466, 0.92614472, 1.1943263, M_PI/2, 2.2006171, M_PI
+    };
+    int iid, icc;
+
+    int k, m;
+    static const int8_t f_center_20[] = {
+        -3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
+    };
+    static const int8_t f_center_34[] = {
+         2,  6, 10, 14, 18, 22, 26, 30,
+        34,-10, -6, -2, 51, 57, 15, 21,
+        27, 33, 39, 45, 54, 66, 78, 42,
+       102, 66, 78, 90,102,114,126, 90,
+    };
+    static const float fractional_delay_links[] = { 0.43f, 0.75f, 0.347f };
+    const float fractional_delay_gain = 0.39f;
+
+    for (pd0 = 0; pd0 < 8; pd0++) {
+        float pd0_re = ipdopd_cos[pd0];
+        float pd0_im = ipdopd_sin[pd0];
+        for (pd1 = 0; pd1 < 8; pd1++) {
+            float pd1_re = ipdopd_cos[pd1];
+            float pd1_im = ipdopd_sin[pd1];
+            for (pd2 = 0; pd2 < 8; pd2++) {
+                float pd2_re = ipdopd_cos[pd2];
+                float pd2_im = ipdopd_sin[pd2];
+                float re_smooth = 0.25f * pd0_re + 0.5f * pd1_re + pd2_re;
+                float im_smooth = 0.25f * pd0_im + 0.5f * pd1_im + pd2_im;
+                float pd_mag = 1 / sqrt(im_smooth * im_smooth + re_smooth * re_smooth);
+                pd_re_smooth[pd0*64+pd1*8+pd2] = re_smooth * pd_mag;
+                pd_im_smooth[pd0*64+pd1*8+pd2] = im_smooth * pd_mag;
+            }
+        }
+    }
+
+    for (iid = 0; iid < 46; iid++) {
+        float c = iid_par_dequant[iid]; //<Linear Inter-channel Intensity Difference
+        float c1 = (float)M_SQRT2 / sqrtf(1.0f + c*c);
+        float c2 = c * c1;
+        for (icc = 0; icc < 8; icc++) {
+            /*if (PS_BASELINE || ps->icc_mode < 3)*/ {
+                float alpha = 0.5f * acos_icc_invq[icc];
+                float beta  = alpha * (c1 - c2) * (float)M_SQRT1_2;
+                HA[iid][icc][0] = c2 * cosf(beta + alpha);
+                HA[iid][icc][1] = c1 * cosf(beta - alpha);
+                HA[iid][icc][2] = c2 * sinf(beta + alpha);
+                HA[iid][icc][3] = c1 * sinf(beta - alpha);
+            } /* else */ {
+                float alpha, gamma, mu, rho;
+                float alpha_c, alpha_s, gamma_c, gamma_s;
+                rho = FFMAX(icc_invq[icc], 0.05f);
+                alpha = 0.5f * atan2f(2.0f * c * rho, c*c - 1.0f);
+                mu = c + 1.0f / c;
+                mu = sqrtf(1 + (4 * rho * rho - 4)/(mu * mu));
+                gamma = atanf(sqrtf((1.0f - mu)/(1.0f + mu)));
+                if (alpha < 0) alpha += M_PI/2;
+                alpha_c = cosf(alpha);
+                alpha_s = sinf(alpha);
+                gamma_c = cosf(gamma);
+                gamma_s = sinf(gamma);
+                HB[iid][icc][0] =  M_SQRT2 * alpha_c * gamma_c;
+                HB[iid][icc][1] =  M_SQRT2 * alpha_s * gamma_c;
+                HB[iid][icc][2] = -M_SQRT2 * alpha_s * gamma_s;
+                HB[iid][icc][3] =  M_SQRT2 * alpha_c * gamma_s;
+            }
+        }
+    }
+
+    for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
+        double f_center, theta;
+        if (k < FF_ARRAY_ELEMS(f_center_20))
+            f_center = f_center_20[k] * 0.125;
+        else
+            f_center = k - 6.5f;
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = -M_PI * fractional_delay_links[m] * f_center;
+            Q_fract_allpass[0][k][m][0] = cos(theta);
+            Q_fract_allpass[0][k][m][1] = sin(theta);
+        }
+        theta = -M_PI*fractional_delay_gain*f_center;
+        phi_fract[0][k][0] = cos(theta);
+        phi_fract[0][k][1] = sin(theta);
+    }
+    for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
+        double f_center, theta;
+        if (k < FF_ARRAY_ELEMS(f_center_34))
+            f_center = f_center_34[k] / 24.;
+        else
+            f_center = k - 26.5f;
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = -M_PI * fractional_delay_links[m] * f_center;
+            Q_fract_allpass[1][k][m][0] = cos(theta);
+            Q_fract_allpass[1][k][m][1] = sin(theta);
+        }
+        theta = -M_PI*fractional_delay_gain*f_center;
+        phi_fract[1][k][0] = cos(theta);
+        phi_fract[1][k][1] = sin(theta);
+    }
+
+    make_filters_from_proto(f20_0_8,  g0_Q8,   8);
+    make_filters_from_proto(f34_0_12, g0_Q12, 12);
+    make_filters_from_proto(f34_1_8,  g1_Q8,   8);
+    make_filters_from_proto(f34_2_4,  g2_Q4,   4);
+}
+#endif /* CONFIG_HARDCODED_TABLES */
+
+#endif /* AACPS_TABLEGEN_H */

libavcodec/aacpsdata.c

@@ -0,0 +1,163 @@
+/*
+ * MPEG-4 Parametric Stereo data tables
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+static const uint8_t huff_iid_df1_bits[] = {
+    18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14,
+    13, 12, 12, 11, 10, 10,  8,  7,  6,  5,  4,  3,  1,  3,  4,  5,  6,  7,
+     8,  9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18,
+    18, 18, 18, 18, 18, 18, 18,
+};
+
+static const uint32_t huff_iid_df1_codes[] = {
+    0x01FEB4, 0x01FEB5, 0x01FD76, 0x01FD77, 0x01FD74, 0x01FD75, 0x01FE8A,
+    0x01FE8B, 0x01FE88, 0x00FE80, 0x01FEB6, 0x00FE82, 0x00FEB8, 0x007F42,
+    0x007FAE, 0x003FAF, 0x001FD1, 0x001FE9, 0x000FE9, 0x0007EA, 0x0007FB,
+    0x0003FB, 0x0001FB, 0x0001FF, 0x00007C, 0x00003C, 0x00001C, 0x00000C,
+    0x000000, 0x000001, 0x000001, 0x000002, 0x000001, 0x00000D, 0x00001D,
+    0x00003D, 0x00007D, 0x0000FC, 0x0001FC, 0x0003FC, 0x0003F4, 0x0007EB,
+    0x000FEA, 0x001FEA, 0x001FD6, 0x003FD0, 0x007FAF, 0x007F43, 0x00FEB9,
+    0x00FE83, 0x01FEB7, 0x00FE81, 0x01FE89, 0x01FE8E, 0x01FE8F, 0x01FE8C,
+    0x01FE8D, 0x01FEB2, 0x01FEB3, 0x01FEB0, 0x01FEB1,
+};
+
+static const uint8_t huff_iid_dt1_bits[] = {
+    16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13,
+    13, 13, 12, 12, 11, 10,  9,  9,  7,  6,  5,  3,  1,  2,  5,  6,  7,  8,
+     9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16,
+    16, 16, 16, 16, 16, 16, 16,
+};
+
+static const uint16_t huff_iid_dt1_codes[] = {
+    0x004ED4, 0x004ED5, 0x004ECE, 0x004ECF, 0x004ECC, 0x004ED6, 0x004ED8,
+    0x004F46, 0x004F60, 0x002718, 0x002719, 0x002764, 0x002765, 0x00276D,
+    0x0027B1, 0x0013B7, 0x0013D6, 0x0009C7, 0x0009E9, 0x0009ED, 0x0004EE,
+    0x0004F7, 0x000278, 0x000139, 0x00009A, 0x00009F, 0x000020, 0x000011,
+    0x00000A, 0x000003, 0x000001, 0x000000, 0x00000B, 0x000012, 0x000021,
+    0x00004C, 0x00009B, 0x00013A, 0x000279, 0x000270, 0x0004EF, 0x0004E2,
+    0x0009EA, 0x0009D8, 0x0013D7, 0x0013D0, 0x0027B2, 0x0027A2, 0x00271A,
+    0x00271B, 0x004F66, 0x004F67, 0x004F61, 0x004F47, 0x004ED9, 0x004ED7,
+    0x004ECD, 0x004ED2, 0x004ED3, 0x004ED0, 0x004ED1,
+};
+
+static const uint8_t huff_iid_df0_bits[] = {
+    17, 17, 17, 17, 16, 15, 13, 10,  9,  7,  6,  5,  4,  3,  1,  3,  4,  5,
+     6,  6,  8, 11, 13, 14, 14, 15, 17, 18, 18,
+};
+
+static const uint32_t huff_iid_df0_codes[] = {
+    0x01FFFB, 0x01FFFC, 0x01FFFD, 0x01FFFA, 0x00FFFC, 0x007FFC, 0x001FFD,
+    0x0003FE, 0x0001FE, 0x00007E, 0x00003C, 0x00001D, 0x00000D, 0x000005,
+    0x000000, 0x000004, 0x00000C, 0x00001C, 0x00003D, 0x00003E, 0x0000FE,
+    0x0007FE, 0x001FFC, 0x003FFC, 0x003FFD, 0x007FFD, 0x01FFFE, 0x03FFFE,
+    0x03FFFF,
+};
+
+static const uint8_t huff_iid_dt0_bits[] = {
+    19, 19, 19, 20, 20, 20, 17, 15, 12, 10,  8,  6,  4,  2,  1,  3,  5,  7,
+     9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20,
+};
+
+static const uint32_t huff_iid_dt0_codes[] = {
+    0x07FFF9, 0x07FFFA, 0x07FFFB, 0x0FFFF8, 0x0FFFF9, 0x0FFFFA, 0x01FFFD,
+    0x007FFE, 0x000FFE, 0x0003FE, 0x0000FE, 0x00003E, 0x00000E, 0x000002,
+    0x000000, 0x000006, 0x00001E, 0x00007E, 0x0001FE, 0x0007FE, 0x001FFE,
+    0x003FFE, 0x01FFFC, 0x07FFF8, 0x0FFFFB, 0x0FFFFC, 0x0FFFFD, 0x0FFFFE,
+    0x0FFFFF,
+};
+
+static const uint8_t huff_icc_df_bits[] = {
+    14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13,
+};
+
+static const uint16_t huff_icc_df_codes[] = {
+    0x3FFF, 0x3FFE, 0x0FFE, 0x03FE, 0x007E, 0x001E, 0x0006, 0x0000,
+    0x0002, 0x000E, 0x003E, 0x00FE, 0x01FE, 0x07FE, 0x1FFE,
+};
+
+static const uint8_t huff_icc_dt_bits[] = {
+    14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14,
+};
+
+static const uint16_t huff_icc_dt_codes[] = {
+    0x3FFE, 0x1FFE, 0x07FE, 0x01FE, 0x007E, 0x001E, 0x0006, 0x0000,
+    0x0002, 0x000E, 0x003E, 0x00FE, 0x03FE, 0x0FFE, 0x3FFF,
+};
+
+static const uint8_t huff_ipd_df_bits[] = {
+    1, 3, 4, 4, 4, 4, 4, 4,
+};
+
+static const uint8_t huff_ipd_df_codes[] = {
+    0x01, 0x00, 0x06, 0x04, 0x02, 0x03, 0x05, 0x07,
+};
+
+static const uint8_t huff_ipd_dt_bits[] = {
+    1, 3, 4, 5, 5, 4, 4, 3,
+};
+
+static const uint8_t huff_ipd_dt_codes[] = {
+    0x01, 0x02, 0x02, 0x03, 0x02, 0x00, 0x03, 0x03,
+};
+
+static const uint8_t huff_opd_df_bits[] = {
+    1, 3, 4, 4, 5, 5, 4, 3,
+};
+
+static const uint8_t huff_opd_df_codes[] = {
+    0x01, 0x01, 0x06, 0x04, 0x0F, 0x0E, 0x05, 0x00,
+};
+
+static const uint8_t huff_opd_dt_bits[] = {
+    1, 3, 4, 5, 5, 4, 4, 3,
+};
+
+static const uint8_t huff_opd_dt_codes[] = {
+    0x01, 0x02, 0x01, 0x07, 0x06, 0x00, 0x02, 0x03,
+};
+
+static const int8_t huff_offset[] = {
+    30, 30,
+    14, 14,
+    7, 7,
+    0, 0,
+    0, 0,
+};
+
+///Table 8.48
+static const int8_t k_to_i_20[] = {
+     1,  0,  0,  1,  2,  3,  4,  5,  6,  7,  8,  9, 10, 11, 12, 13, 14, 14, 15,
+    15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18,
+    18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19,
+    19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19
+};
+///Table 8.49
+static const int8_t k_to_i_34[] = {
+     0,  1,  2,  3,  4,  5,  6,  6,  7,  2,  1,  0, 10, 10,  4,  5,  6,  7,  8,
+     9, 10, 11, 12,  9, 14, 11, 12, 13, 14, 15, 16, 13, 16, 17, 18, 19, 20, 21,
+    22, 22, 23, 23, 24, 24, 25, 25, 26, 26, 27, 27, 27, 28, 28, 28, 29, 29, 29,
+    30, 30, 30, 31, 31, 31, 31, 32, 32, 32, 32, 33, 33, 33, 33, 33, 33, 33, 33,
+    33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
+};
+
+static const float g1_Q2[] = {
+    0.0f,  0.01899487526049f, 0.0f, -0.07293139167538f,
+    0.0f,  0.30596630545168f, 0.5f
+};

libavcodec/aacsbr.c

@@ -31,6 +31,7 @@
 #include "aacsbr.h"
 #include "aacsbrdata.h"
 #include "fft.h"
+#include "aacps.h"
 
 #include <stdint.h>
 #include <float.h>
@@ -71,9 +72,6 @@ enum {
 static VLC vlc_sbr[10];
 static const int8_t vlc_sbr_lav[10] =
     { 60, 60, 24, 24, 31, 31, 12, 12, 31, 12 };
-static DECLARE_ALIGNED(16, float, analysis_cos_pre)[64];
-static DECLARE_ALIGNED(16, float, analysis_sin_pre)[64];
-static DECLARE_ALIGNED(16, float, analysis_cossin_post)[32][2];
 static const DECLARE_ALIGNED(16, float, zero64)[64];
 
 #define SBR_INIT_VLC_STATIC(num, size) \
@@ -87,7 +85,7 @@ static const DECLARE_ALIGNED(16, float, zero64)[64];
 
 av_cold void ff_aac_sbr_init(void)
 {
-    int n, k;
+    int n;
     static const struct {
         const void *sbr_codes, *sbr_bits;
         const unsigned int table_size, elem_size;
@@ -116,16 +114,6 @@ av_cold void ff_aac_sbr_init(void)
     SBR_INIT_VLC_STATIC(8, 592);
     SBR_INIT_VLC_STATIC(9, 512);
 
-    for (n = 0; n < 64; n++) {
-        float pre = M_PI * n / 64;
-        analysis_cos_pre[n] = cosf(pre);
-        analysis_sin_pre[n] = sinf(pre);
-    }
-    for (k = 0; k < 32; k++) {
-        float post = M_PI * (k + 0.5) / 128;
-        analysis_cossin_post[k][0] =  4.0 * cosf(post);
-        analysis_cossin_post[k][1] = -4.0 * sinf(post);
-    }
     for (n = 1; n < 320; n++)
         sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
     sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
@@ -133,6 +121,8 @@ av_cold void ff_aac_sbr_init(void)
 
     for (n = 0; n < 320; n++)
         sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
+
+    ff_ps_init();
 }
 
 av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
@@ -142,13 +132,14 @@ av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
     sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     ff_mdct_init(&sbr->mdct, 7, 1, 1.0/64);
-    ff_rdft_init(&sbr->rdft, 6, IDFT_R2C);
+    ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0);
+    ff_ps_ctx_init(&sbr->ps);
 }
 
 av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
 {
     ff_mdct_end(&sbr->mdct);
-    ff_rdft_end(&sbr->rdft);
+    ff_mdct_end(&sbr->mdct_ana);
 }
 
 static int qsort_comparison_function_int16(const void *a, const void *b)
@@ -293,15 +284,15 @@ static void make_bands(int16_t* bands, int start, int stop, int num_bands)
     bands[num_bands-1] = stop - previous;
 }
 
-static int check_n_master(AVCodecContext *avccontext, int n_master, int bs_xover_band)
+static int check_n_master(AVCodecContext *avctx, int n_master, int bs_xover_band)
 {
     // Requirements (14496-3 sp04 p205)
     if (n_master <= 0) {
-        av_log(avccontext, AV_LOG_ERROR, "Invalid n_master: %d\n", n_master);
+        av_log(avctx, AV_LOG_ERROR, "Invalid n_master: %d\n", n_master);
         return -1;
     }
     if (bs_xover_band >= n_master) {
-        av_log(avccontext, AV_LOG_ERROR,
+        av_log(avctx, AV_LOG_ERROR,
                "Invalid bitstream, crossover band index beyond array bounds: %d\n",
                bs_xover_band);
         return -1;
@@ -349,7 +340,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
         sbr_offset_ptr = sbr_offset[5];
         break;
     default:
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Unsupported sample rate for SBR: %d\n", sbr->sample_rate);
         return -1;
     }
@@ -367,7 +358,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
     } else if (spectrum->bs_stop_freq == 15) {
         sbr->k[2] = 3*sbr->k[0];
     } else {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Invalid bs_stop_freq: %d\n", spectrum->bs_stop_freq);
         return -1;
     }
@@ -382,18 +373,17 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
         max_qmf_subbands = 32;
 
     if (sbr->k[2] - sbr->k[0] > max_qmf_subbands) {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Invalid bitstream, too many QMF subbands: %d\n", sbr->k[2] - sbr->k[0]);
         return -1;
     }
 
     if (!spectrum->bs_freq_scale) {
-        unsigned int dk;
-        int k2diff;
+        int dk, k2diff;
 
         dk = spectrum->bs_alter_scale + 1;
         sbr->n_master = ((sbr->k[2] - sbr->k[0] + (dk&2)) >> dk) << 1;
-        if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
+        if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
             return -1;
 
         for (k = 1; k <= sbr->n_master; k++)
@@ -428,7 +418,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
         num_bands_0 = lrintf(half_bands * log2f(sbr->k[1] / (float)sbr->k[0])) * 2;
 
         if (num_bands_0 <= 0) { // Requirements (14496-3 sp04 p205)
-            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
+            av_log(ac->avctx, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
             return -1;
         }
 
@@ -442,7 +432,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
         vk0[0] = sbr->k[0];
         for (k = 1; k <= num_bands_0; k++) {
             if (vk0[k] <= 0) { // Requirements (14496-3 sp04 p205)
-                av_log(ac->avccontext, AV_LOG_ERROR, "Invalid vDk0[%d]: %d\n", k, vk0[k]);
+                av_log(ac->avctx, AV_LOG_ERROR, "Invalid vDk0[%d]: %d\n", k, vk0[k]);
                 return -1;
             }
             vk0[k] += vk0[k-1];
@@ -472,14 +462,14 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
             vk1[0] = sbr->k[1];
             for (k = 1; k <= num_bands_1; k++) {
                 if (vk1[k] <= 0) { // Requirements (14496-3 sp04 p205)
-                    av_log(ac->avccontext, AV_LOG_ERROR, "Invalid vDk1[%d]: %d\n", k, vk1[k]);
+                    av_log(ac->avctx, AV_LOG_ERROR, "Invalid vDk1[%d]: %d\n", k, vk1[k]);
                     return -1;
                 }
                 vk1[k] += vk1[k-1];
             }
 
             sbr->n_master = num_bands_0 + num_bands_1;
-            if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
+            if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
                 return -1;
             memcpy(&sbr->f_master[0],               vk0,
                    (num_bands_0 + 1) * sizeof(sbr->f_master[0]));
@@ -488,7 +478,7 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
 
         } else {
             sbr->n_master = num_bands_0;
-            if (check_n_master(ac->avccontext, sbr->n_master, sbr->spectrum_params.bs_xover_band))
+            if (check_n_master(ac->avctx, sbr->n_master, sbr->spectrum_params.bs_xover_band))
                 return -1;
             memcpy(sbr->f_master, vk0, (num_bands_0 + 1) * sizeof(sbr->f_master[0]));
         }
@@ -524,7 +514,7 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
         // illegal however the Coding Technologies decoder check stream has a final
         // count of 6 patches
         if (sbr->num_patches > 5) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "Too many patches: %d\n", sbr->num_patches);
+            av_log(ac->avctx, AV_LOG_ERROR, "Too many patches: %d\n", sbr->num_patches);
             return -1;
         }
 
@@ -563,12 +553,12 @@ static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
 
     // Requirements (14496-3 sp04 p205)
     if (sbr->kx[1] + sbr->m[1] > 64) {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Stop frequency border too high: %d\n", sbr->kx[1] + sbr->m[1]);
         return -1;
     }
     if (sbr->kx[1] > 32) {
-        av_log(ac->avccontext, AV_LOG_ERROR, "Start frequency border too high: %d\n", sbr->kx[1]);
+        av_log(ac->avctx, AV_LOG_ERROR, "Start frequency border too high: %d\n", sbr->kx[1]);
         return -1;
     }
 
@@ -580,7 +570,7 @@ static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
     sbr->n_q = FFMAX(1, lrintf(sbr->spectrum_params.bs_noise_bands *
                                log2f(sbr->k[2] / (float)sbr->kx[1]))); // 0 <= bs_noise_bands <= 3
     if (sbr->n_q > 5) {
-        av_log(ac->avccontext, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
+        av_log(ac->avctx, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
         return -1;
     }
 
@@ -638,7 +628,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
             ch_data->bs_amp_res = 0;
 
         if (ch_data->bs_num_env > 4) {
-            av_log(ac->avccontext, AV_LOG_ERROR,
+            av_log(ac->avctx, AV_LOG_ERROR,
                    "Invalid bitstream, too many SBR envelopes in FIXFIX type SBR frame: %d\n",
                    ch_data->bs_num_env);
             return -1;
@@ -693,7 +683,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
         ch_data->bs_num_env                 = num_rel_lead + num_rel_trail + 1;
 
         if (ch_data->bs_num_env > 5) {
-            av_log(ac->avccontext, AV_LOG_ERROR,
+            av_log(ac->avctx, AV_LOG_ERROR,
                    "Invalid bitstream, too many SBR envelopes in VARVAR type SBR frame: %d\n",
                    ch_data->bs_num_env);
             return -1;
@@ -714,7 +704,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
     }
 
     if (bs_pointer > ch_data->bs_num_env + 1) {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Invalid bitstream, bs_pointer points to a middle noise border outside the time borders table: %d\n",
                bs_pointer);
         return -1;
@@ -722,7 +712,7 @@ static int read_sbr_grid(AACContext *ac, SpectralBandReplication *sbr,
 
     for (i = 1; i <= ch_data->bs_num_env; i++) {
         if (ch_data->t_env[i-1] > ch_data->t_env[i]) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "Non monotone time borders\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "Non monotone time borders\n");
             return -1;
         }
     }
@@ -903,25 +893,24 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
                                GetBitContext *gb,
                                int bs_extension_id, int *num_bits_left)
 {
-//TODO - implement ps_data for parametric stereo parsing
     switch (bs_extension_id) {
     case EXTENSION_ID_PS:
         if (!ac->m4ac.ps) {
-            av_log(ac->avccontext, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
+            av_log(ac->avctx, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
             *num_bits_left = 0;
         } else {
-#if 0
-            *num_bits_left -= ff_ps_data(gb, ps);
+#if 1
+            *num_bits_left -= ff_ps_read_data(ac->avctx, gb, &sbr->ps, *num_bits_left);
 #else
-            av_log_missing_feature(ac->avccontext, "Parametric Stereo is", 0);
+            av_log_missing_feature(ac->avctx, "Parametric Stereo is", 0);
             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
             *num_bits_left = 0;
 #endif
         }
         break;
     default:
-        av_log_missing_feature(ac->avccontext, "Reserved SBR extensions are", 1);
+        av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
         skip_bits_long(gb, *num_bits_left); // bs_fill_bits
         *num_bits_left = 0;
         break;
@@ -1006,7 +995,7 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
             return get_bits_count(gb) - cnt;
         }
     } else {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
             "Invalid bitstream - cannot apply SBR to element type %d\n", id_aac);
         sbr->start = 0;
         return get_bits_count(gb) - cnt;
@@ -1021,6 +1010,11 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
             num_bits_left -= 2;
             read_sbr_extension(ac, sbr, gb, get_bits(gb, 2), &num_bits_left); // bs_extension_id
         }
+        if (num_bits_left < 0) {
+            av_log(ac->avctx, AV_LOG_ERROR, "SBR Extension over read.\n");
+        }
+        if (num_bits_left > 0)
+            skip_bits(gb, num_bits_left);
     }
 
     return get_bits_count(gb) - cnt;
@@ -1033,7 +1027,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
     if (err >= 0)
         err = sbr_make_f_derived(ac, sbr);
     if (err < 0) {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "SBR reset failed. Switching SBR to pure upsampling mode.\n");
         sbr->start = 0;
     }
@@ -1085,7 +1079,7 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
     bytes_read = ((num_sbr_bits + num_align_bits + 4) >> 3);
 
     if (bytes_read > cnt) {
-        av_log(ac->avccontext, AV_LOG_ERROR,
+        av_log(ac->avctx, AV_LOG_ERROR,
                "Expected to read %d SBR bytes actually read %d.\n", cnt, bytes_read);
     }
     return cnt;
@@ -1139,7 +1133,7 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
  * @param   x       pointer to the beginning of the first sample window
  * @param   W       array of complex-valued samples split into subbands
  */
-static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in, float *x,
+static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in, float *x,
                              float z[320], float W[2][32][32][2],
                              float scale)
 {
@@ -1152,23 +1146,23 @@ static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in
         memcpy(x+288, in, 1024*sizeof(*x));
     for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
                                // are not supported
-        float re, im;
         dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
         for (k = 0; k < 64; k++) {
             float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
-            z[k] = f * analysis_cos_pre[k];
-            z[k+64] = f;
+            z[k] = f;
         }
-        ff_rdft_calc(rdft, z);
-        re = z[0] * 0.5f;
-        im = 0.5f * dsp->scalarproduct_float(z+64, analysis_sin_pre, 64);
-        W[1][i][0][0] = re * analysis_cossin_post[0][0] - im * analysis_cossin_post[0][1];
-        W[1][i][0][1] = re * analysis_cossin_post[0][1] + im * analysis_cossin_post[0][0];
+        //Shuffle to IMDCT
+        z[64] = z[0];
         for (k = 1; k < 32; k++) {
-            re = z[2*k  ] - re;
-            im = z[2*k+1] - im;
-            W[1][i][k][0] = re * analysis_cossin_post[k][0] - im * analysis_cossin_post[k][1];
-            W[1][i][k][1] = re * analysis_cossin_post[k][1] + im * analysis_cossin_post[k][0];
+            z[64+2*k-1] =  z[   k];
+            z[64+2*k  ] = -z[64-k];
+        }
+        z[64+63] = z[32];
+
+        ff_imdct_half(mdct, z, z+64);
+        for (k = 0; k < 32; k++) {
+            W[1][i][k][0] = -z[63-k];
+            W[1][i][k][1] = z[k];
         }
         x += 32;
     }
@@ -1179,7 +1173,7 @@ static void sbr_qmf_analysis(DSPContext *dsp, RDFTContext *rdft, const float *in
  * (14496-3 sp04 p206)
  */
 static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
-                              float *out, float X[2][32][64],
+                              float *out, float X[2][38][64],
                               float mdct_buf[2][64],
                               float *v0, int *v_off, const unsigned int div,
                               float bias, float scale)
@@ -1197,21 +1191,22 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
             *v_off -= 128 >> div;
         }
         v = v0 + *v_off;
-        for (n = 1; n < 64 >> div; n+=2) {
-            X[1][i][n] = -X[1][i][n];
-        }
-        if (div) {
-            memset(X[0][i]+32, 0, 32*sizeof(float));
-            memset(X[1][i]+32, 0, 32*sizeof(float));
-        }
-        ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
-        ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
         if (div) {
             for (n = 0; n < 32; n++) {
-                v[      n] = -mdct_buf[0][63 - 2*n] + mdct_buf[1][2*n    ];
-                v[ 63 - n] =  mdct_buf[0][62 - 2*n] + mdct_buf[1][2*n + 1];
+                X[0][i][   n] = -X[0][i][n];
+                X[0][i][32+n] =  X[1][i][31-n];
+            }
+            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
+            for (n = 0; n < 32; n++) {
+                v[     n] =  mdct_buf[0][63 - 2*n];
+                v[63 - n] = -mdct_buf[0][62 - 2*n];
             }
         } else {
+            for (n = 1; n < 64; n+=2) {
+                X[1][i][n] = -X[1][i][n];
+            }
+            ff_imdct_half(mdct, mdct_buf[0], X[0][i]);
+            ff_imdct_half(mdct, mdct_buf[1], X[1][i]);
             for (n = 0; n < 64; n++) {
                 v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
                 v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
@@ -1380,7 +1375,7 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
             g--;
 
             if (g < 0) {
-                av_log(ac->avccontext, AV_LOG_ERROR,
+                av_log(ac->avctx, AV_LOG_ERROR,
                        "ERROR : no subband found for frequency %d\n", k);
                 return -1;
             }
@@ -1414,7 +1409,7 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
 }
 
 /// Generate the subband filtered lowband
-static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][32][64],
+static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
                      const float X_low[32][40][2], const float Y[2][38][64][2],
                      int ch)
 {
@@ -1436,7 +1431,7 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][32][64],
     }
 
     for (k = 0; k < sbr->kx[1]; k++) {
-        for (i = i_Temp; i < i_f; i++) {
+        for (i = i_Temp; i < 38; i++) {
             X[0][i][k] = X_low[k][i + ENVELOPE_ADJUSTMENT_OFFSET][0];
             X[1][i][k] = X_low[k][i + ENVELOPE_ADJUSTMENT_OFFSET][1];
         }
@@ -1730,7 +1725,7 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
     }
     for (ch = 0; ch < nch; ch++) {
         /* decode channel */
-        sbr_qmf_analysis(&ac->dsp, &sbr->rdft, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
+        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
                          (float*)sbr->qmf_filter_scratch,
                          sbr->data[ch].W, 1/(-1024 * ac->sf_scale));
         sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W);
@@ -1752,6 +1747,16 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
         /* synthesis */
         sbr_x_gen(sbr, sbr->X[ch], sbr->X_low, sbr->data[ch].Y, ch);
     }
+
+    if (ac->m4ac.ps == 1) {
+        if (sbr->ps.start) {
+            ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
+        } else {
+            memcpy(sbr->X[1], sbr->X[0], sizeof(sbr->X[0]));
+        }
+        nch = 2;
+    }
+
     sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
                       sbr->data[0].synthesis_filterbank_samples,
                       &sbr->data[0].synthesis_filterbank_samples_offset,

libavcodec/aactab.c

@@ -29,6 +29,7 @@
 
 #include "libavutil/mem.h"
 #include "aac.h"
+#include "aac_tablegen.h"
 
 #include <stdint.h>
 
@@ -1204,129 +1205,3 @@ const uint8_t ff_tns_max_bands_128[] = {
     9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
 };
 // @}
-
-
-#if CONFIG_HARDCODED_TABLES
-
-/**
- * Table of pow(2, (i - 200)/4.) used for different purposes depending on the
- * range of indices to the table:
- * [ 0, 255] scale factor decoding when using C dsp.float_to_int16
- * [60, 315] scale factor decoding when using SIMD dsp.float_to_int16
- * [45, 300] intensity stereo position decoding mapped in reverse order i.e. 0->300, 1->299, ..., 254->46, 255->45
- */
-const float ff_aac_pow2sf_tab[428] = {
-    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,
-    5.36870912e+08, 6.38450708e+08, 7.59250125e+08, 9.02905651e+08,
-    1.07374182e+09, 1.27690142e+09, 1.51850025e+09, 1.80581130e+09,
-    2.14748365e+09, 2.55380283e+09, 3.03700050e+09, 3.61162260e+09,
-    4.29496730e+09, 5.10760567e+09, 6.07400100e+09, 7.22324521e+09,
-    8.58993459e+09, 1.02152113e+10, 1.21480020e+10, 1.44464904e+10,
-    1.71798692e+10, 2.04304227e+10, 2.42960040e+10, 2.88929808e+10,
-    3.43597384e+10, 4.08608453e+10, 4.85920080e+10, 5.77859616e+10,
-    6.87194767e+10, 8.17216907e+10, 9.71840160e+10, 1.15571923e+11,
-    1.37438953e+11, 1.63443381e+11, 1.94368032e+11, 2.31143847e+11,
-    2.74877907e+11, 3.26886763e+11, 3.88736064e+11, 4.62287693e+11,
-    5.49755814e+11, 6.53773525e+11, 7.77472128e+11, 9.24575386e+11,
-    1.09951163e+12, 1.30754705e+12, 1.55494426e+12, 1.84915077e+12,
-    2.19902326e+12, 2.61509410e+12, 3.10988851e+12, 3.69830155e+12,
-    4.39804651e+12, 5.23018820e+12, 6.21977702e+12, 7.39660309e+12,
-    8.79609302e+12, 1.04603764e+13, 1.24395540e+13, 1.47932062e+13,
-    1.75921860e+13, 2.09207528e+13, 2.48791081e+13, 2.95864124e+13,
-    3.51843721e+13, 4.18415056e+13, 4.97582162e+13, 5.91728247e+13,
-    7.03687442e+13, 8.36830112e+13, 9.95164324e+13, 1.18345649e+14,
-    1.40737488e+14, 1.67366022e+14, 1.99032865e+14, 2.36691299e+14,
-    2.81474977e+14, 3.34732045e+14, 3.98065730e+14, 4.73382598e+14,
-    5.62949953e+14, 6.69464090e+14, 7.96131459e+14, 9.46765196e+14,
-    1.12589991e+15, 1.33892818e+15, 1.59226292e+15, 1.89353039e+15,
-    2.25179981e+15, 2.67785636e+15, 3.18452584e+15, 3.78706078e+15,
-    4.50359963e+15, 5.35571272e+15, 6.36905167e+15, 7.57412156e+15,
-    9.00719925e+15, 1.07114254e+16, 1.27381033e+16, 1.51482431e+16,
-    1.80143985e+16, 2.14228509e+16, 2.54762067e+16, 3.02964863e+16,
-    3.60287970e+16, 4.28457018e+16, 5.09524134e+16, 6.05929725e+16,
-    7.20575940e+16, 8.56914035e+16, 1.01904827e+17, 1.21185945e+17,
-};
-
-#else
-
-float ff_aac_pow2sf_tab[428];
-
-#endif /* CONFIG_HARDCODED_TABLES */

libavcodec/aactab.h

@@ -32,6 +32,7 @@
 
 #include "libavutil/mem.h"
 #include "aac.h"
+#include "aac_tablegen_decl.h"
 
 #include <stdint.h>
 
@@ -73,10 +74,4 @@ extern const uint16_t * const ff_swb_offset_128 [13];
 extern const uint8_t ff_tns_max_bands_1024[13];
 extern const uint8_t ff_tns_max_bands_128 [13];
 
-#if CONFIG_HARDCODED_TABLES
-extern const float ff_aac_pow2sf_tab[428];
-#else
-extern       float ff_aac_pow2sf_tab[428];
-#endif /* CONFIG_HARDCODED_TABLES */
-
 #endif /* AVCODEC_AACTAB_H */

libavcodec/sbr.h

@@ -31,6 +31,7 @@
 
 #include <stdint.h>
 #include "fft.h"
+#include "aacps.h"
 
 /**
  * Spectral Band Replication header - spectrum parameters that invoke a reset if they differ from the previous header.
@@ -133,6 +134,7 @@ typedef struct {
     ///The number of frequency bands in f_master
     unsigned           n_master;
     SBRData            data[2];
+    PSContext          ps;
     ///N_Low and N_High respectively, the number of frequency bands for low and high resolution
     unsigned           n[2];
     ///Number of noise floor bands
@@ -157,7 +159,7 @@ typedef struct {
     ///QMF output of the HF generator
     float              X_high[64][40][2];
     ///QMF values of the reconstructed signal
-    DECLARE_ALIGNED(16, float, X)[2][2][32][64];
+    DECLARE_ALIGNED(16, float, X)[2][2][38][64];
     ///Zeroth coefficient used to filter the subband signals
     float              alpha0[64][2];
     ///First coefficient used to filter the subband signals
@@ -176,7 +178,7 @@ typedef struct {
     float              s_m[7][48];
     float              gain[7][48];
     DECLARE_ALIGNED(16, float, qmf_filter_scratch)[5][64];
-    RDFTContext        rdft;
+    FFTContext         mdct_ana;
     FFTContext         mdct;
 } SpectralBandReplication;