@@ -99,6 +99,40 @@ static VLC vlc_scalefactors;

 static VLC vlc_spectral[11];

+/**

+ * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.

+ *

+ * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.

+ * @param sce_map mono (Single Channel Element) map

+ * @param type speaker type/position for these channels

+ */

+static void decode_channel_map(enum ChannelPosition *cpe_map,

+        enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {

+    while(n--) {

+        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map

+        map[get_bits(gb, 4)] = type;

+    }

+}

+

+/**

+ * Decode program configuration element; reference: table 4.2.

+ *

+ * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.

+ *

+ * @return  Returns error status. 0 - OK, !0 - error

+ */

+static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],

+        GetBitContext * gb) {

+    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;

+

+    skip_bits(gb, 2);  // object_type

+

+    ac->m4ac.sampling_index = get_bits(gb, 4);

+    if(ac->m4ac.sampling_index > 11) {

+        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);

+        return -1;

+    }

+    ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];

     num_front       = get_bits(gb, 4);

     num_side        = get_bits(gb, 4);

     num_back        = get_bits(gb, 4);

@@ -130,6 +164,131 @@ static VLC vlc_spectral[11];

     return 0;

 }

+/**

+ * Set up channel positions based on a default channel configuration

+ * as specified in table 1.17.

+ *

+ * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.

+ *

+ * @return  Returns error status. 0 - OK, !0 - error

+ */

+static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],

+        int channel_config)

+{

+    if(channel_config < 1 || channel_config > 7) {

+        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",

+               channel_config);

+        return -1;

+    }

+

+    /* default channel configurations:

+     *

+     * 1ch : front center (mono)

+     * 2ch : L + R (stereo)

+     * 3ch : front center + L + R

+     * 4ch : front center + L + R + back center

+     * 5ch : front center + L + R + back stereo

+     * 6ch : front center + L + R + back stereo + LFE

+     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE

+     */

+

+    if(channel_config != 2)

+        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)

+    if(channel_config > 1)

+        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)

+    if(channel_config == 4)

+        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center

+    if(channel_config > 4)

+        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]

+                                 = AAC_CHANNEL_BACK;  // back stereo

+    if(channel_config > 5)

+        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE

+    if(channel_config == 7)

+        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right

+

+    return 0;

+}

+

+        return -1;

+    }

+

+    if (get_bits1(gb))       // dependsOnCoreCoder

+        skip_bits(gb, 14);   // coreCoderDelay

+    extension_flag = get_bits1(gb);

+

+    if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||

+       ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)

+        skip_bits(gb, 3);     // layerNr

+

+    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));

+    if (channel_config == 0) {

+        skip_bits(gb, 4);  // element_instance_tag

+        if((ret = decode_pce(ac, new_che_pos, gb)))

+            return ret;

+    } else {

+        if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))

+            return ret;

+    }

+    if((ret = output_configure(ac, ac->che_pos, new_che_pos)))

+        return ret;

+

+    if (extension_flag) {

+        switch (ac->m4ac.object_type) {

+            case AOT_ER_BSAC:

+                skip_bits(gb, 5);    // numOfSubFrame

+                skip_bits(gb, 11);   // layer_length

+                break;

+            case AOT_ER_AAC_LC:

+            case AOT_ER_AAC_LTP:

+            case AOT_ER_AAC_SCALABLE:

+            case AOT_ER_AAC_LD:

+                skip_bits(gb, 3);  /* aacSectionDataResilienceFlag

+                                    * aacScalefactorDataResilienceFlag

+                                    * aacSpectralDataResilienceFlag

+                                    */

+                break;

+        }

+        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)

+    }

+    return 0;

+}

+

+/**

+ * Decode audio specific configuration; reference: table 1.13.

+ *

+ * @param   data        pointer to AVCodecContext extradata

+ * @param   data_size   size of AVCCodecContext extradata

+ *

+ * @return  Returns error status. 0 - OK, !0 - error

+ */

+static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {

+    GetBitContext gb;

+    int i;

+

+    init_get_bits(&gb, data, data_size * 8);

+

+    if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)

+        return -1;

+    if(ac->m4ac.sampling_index > 11) {

+        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);

+        return -1;

+    }

+

+    skip_bits_long(&gb, i);

+

+    switch (ac->m4ac.object_type) {

+    case AOT_AAC_LC:

+        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))

+            return -1;

+        break;

+    default:

+        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",

+               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);

+        return -1;

+    }

+    return 0;

+}

+

 static av_cold int aac_decode_init(AVCodecContext * avccontext) {

     AACContext * ac = avccontext->priv_data;

     int i;

@@ -140,6 +299,7 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {

         decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))

         return -1;

+    avccontext->sample_fmt  = SAMPLE_FMT_S16;

     avccontext->sample_rate = ac->m4ac.sample_rate;

     avccontext->frame_size  = 1024;

@@ -157,6 +317,8 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {

     dsputil_init(&ac->dsp, avccontext);

+    ac->random_state = 0x1f2e3d4c;

+

     // -1024 - Compensate wrong IMDCT method.

     // 32768 - Required to scale values to the correct range for the bias method

     //         for float to int16 conversion.

@@ -188,6 +350,10 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {

     return 0;

 }

+/**

+ * Skip data_stream_element; reference: table 4.10.

+ */

+static void skip_data_stream_element(GetBitContext * gb) {

     int byte_align = get_bits1(gb);

     int count = get_bits(gb, 8);

     if (count == 255)

@@ -198,6 +364,27 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {

 }

 /**

+ * Decode Individual Channel Stream info; reference: table 4.6.

+ *

+ * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.

+ */

+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");

+        memset(ics, 0, sizeof(IndividualChannelStream));

+        return -1;

+    }

+    ics->window_sequence[1] = ics->window_sequence[0];

+    ics->window_sequence[0] = get_bits(gb, 2);

+    ics->use_kb_window[1] = ics->use_kb_window[0];

+    ics->use_kb_window[0] = get_bits1(gb);

+    ics->num_window_groups = 1;

+    ics->group_len[0] = 1;

+

+    return 0;

+}

+

+/**

  * inverse quantization

  *

  * @param   a   quantized value to be dequantized

@@ -210,6 +397,15 @@ static inline float ivquant(int a) {

         return cbrtf(fabsf(a)) * a;

 }

+/**

+ * Decode band types (section_data payload); reference: table 4.46.

+ *

+ * @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

+ *

+ * @return  Returns error status. 0 - OK, !0 - error

+ */

+static int decode_band_types(AACContext * ac, enum BandType band_type[120],

         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;

@@ -232,7 +428,13 @@ static inline float ivquant(int a) {

                     sect_len, ics->max_sfb);

                 return -1;

             }

+        }

+    }

+    return 0;

+}

+/**

+ * Decode scalefactors; reference: table 4.47.

  *

  * @param   mix_gain            channel gain (Not used by AAC bitstream.)

  * @param   global_gain         first scalefactor value as scalefactors are differentially coded

@@ -314,6 +516,16 @@ static void decode_pulses(Pulse * pulse, GetBitContext * gb) {

 }

 /**

+ * Decode Mid/Side data; reference: table 4.54.

+ *

+ * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;

+ *                      [1] mask is decoded from bitstream; [2] mask is all 1s;

+ *                      [3] reserved for scalable AAC

+ */

+static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,

+        int ms_present) {

+

+/**

  * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.

  *

  * @param   pulse   pointer to pulse data struct

@@ -330,10 +542,109 @@ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualCha

 }

 /**

- * Parse Spectral Band Replication extension data; reference: table 4.55.

+ * Decode an individual_channel_stream payload; reference: table 4.44.

+ *

+ * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.

+ * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)

+ *

+ * @return  Returns error status. 0 - OK, !0 - error

+ */

+static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {

+    int icoeffs[1024];

+    Pulse pulse;

+    TemporalNoiseShaping * tns = &sce->tns;

+    IndividualChannelStream * ics = &sce->ics;

+    float * out = sce->coeffs;

+    int global_gain, pulse_present = 0;

+

+    /* These two assignments are to silence some GCC warnings about the

+     * variables being used uninitialised when in fact they always are.

+     */

+    pulse.num_pulse = 0;

+    pulse.start     = 0;

+

+    global_gain = get_bits(gb, 8);

+

+    if (!common_window && !scale_flag) {

+        if (decode_ics_info(ac, ics, gb, 0) < 0)

+            return -1;

+    }

+

+    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)

+        return -1;

+    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)

+        return -1;

+

+    pulse_present = 0;

+    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");

+                return -1;

+            }

+            decode_pulses(&pulse, gb);

+        }

+        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);

+            return -1;

+        }

+    }

+

+    if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)

+        return -1;

+    if (pulse_present)

+        add_pulses(icoeffs, &pulse, ics);

+    dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);

+    return 0;

+}

+

+/**

+ * Decode a channel_pair_element; reference: table 4.4.

+ *

+ * @param   elem_id Identifies the instance of a syntax element.

+ *

+ * @return  Returns error status. 0 - OK, !0 - error

+ */

+static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {

+    int i, ret, common_window, ms_present = 0;

+    ChannelElement * cpe;

+

+    cpe = ac->che[TYPE_CPE][elem_id];

+    common_window = get_bits1(gb);

+    if (common_window) {

+        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))

+            return -1;

+        i = cpe->ch[1].ics.use_kb_window[0];

+        cpe->ch[1].ics = cpe->ch[0].ics;

+        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");

+            return -1;

+        } else if(ms_present)

+            decode_mid_side_stereo(cpe, gb, ms_present);

+    }

+    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))

+        return ret;

+    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))

+        return ret;

+

+    if (common_window && ms_present)

+        apply_mid_side_stereo(cpe);

+

+    if (cpe->ch[1].ics.intensity_present)

+        apply_intensity_stereo(cpe, ms_present);

+    return 0;

+}

+

+/**

+ * Decode 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) {

@@ -343,6 +654,66 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in

     return cnt;

 }

+/**

+ * Decode dynamic range information; reference: table 4.52.

+ *

+ * @param   cnt length of TYPE_FIL syntactic element in bytes

+ *

+ * @return  Returns number of bytes consumed.

+ */

+static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {

+    int n = 1;

+    int drc_num_bands = 1;

+    int i;

+

+    /* pce_tag_present? */

+    if(get_bits1(gb)) {

+        che_drc->pce_instance_tag  = get_bits(gb, 4);

+        skip_bits(gb, 4); // tag_reserved_bits

+        n++;

+    }

+

+    /* excluded_chns_present? */

+    if(get_bits1(gb)) {

+        n += decode_drc_channel_exclusions(che_drc, gb);

+    }

+

+    /* drc_bands_present? */

+    if (get_bits1(gb)) {

+        che_drc->band_incr            = get_bits(gb, 4);

+        che_drc->interpolation_scheme = get_bits(gb, 4);

+        n++;

+        drc_num_bands += che_drc->band_incr;

+        for (i = 0; i < drc_num_bands; i++) {

+            che_drc->band_top[i] = get_bits(gb, 8);

+            n++;

+        }

+    }

+

+    /* prog_ref_level_present? */

+    if (get_bits1(gb)) {

+        che_drc->prog_ref_level = get_bits(gb, 7);

+        skip_bits1(gb); // prog_ref_level_reserved_bits

+        n++;

+    }

+

+    for (i = 0; i < drc_num_bands; i++) {

+        che_drc->dyn_rng_sgn[i] = get_bits1(gb);

+        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);

+        n++;

+    }

+

+    return n;

+}

+

+/**

+ * Decode extension data (incomplete); reference: table 4.51.

+ *

+ * @param   cnt length of TYPE_FIL syntactic element in bytes

+ *

+ * @return Returns number of bytes consumed

+ */

+static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {

     int crc_flag = 0;

     int res = cnt;

     switch (get_bits(gb, 4)) { // extension type

@@ -365,6 +736,21 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in

 }

 /**

+ * Conduct IMDCT and windowing.

+ */

+static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {

+    IndividualChannelStream * ics = &sce->ics;

+    float * in = sce->coeffs;

+    float * out = sce->ret;

+    float * saved = sce->saved;

+    const float * lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;

+    const float * swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;

+    const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;

+    const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;

+    float * buf = ac->buf_mdct;

+    int i;

+

+/**

  * Apply dependent channel coupling (applied before IMDCT).

  *

  * @param   index   index into coupling gain array

@@ -409,6 +795,26 @@ static void apply_independent_coupling(AACContext * ac, SingleChannelElement * s

         sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);

 }

+    if (!ac->is_saved) {

+        ac->is_saved = 1;

+        *data_size = 0;

+        return 0;

+    }

+

+    data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);

+    if(*data_size < data_size_tmp) {

+        av_log(avccontext, 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, 1024, avccontext->channels);

+

+    return buf_size;

+}

+

 static av_cold int aac_decode_close(AVCodecContext * avccontext) {

     AACContext * ac = avccontext->priv_data;

     int i, j;

@@ -43,6 +43,7 @@

         size);

 #define MAX_CHANNELS 64

+#define MAX_ELEM_ID 16

 #define IVQUANT_SIZE 1024

@@ -76,6 +77,17 @@ enum AudioObjectType {

     AOT_SSC,                   ///< N                       SinuSoidal Coding

 };

+enum RawDataBlockType {

+    TYPE_SCE,

+    TYPE_CPE,

+    TYPE_CCE,

+    TYPE_LFE,

+    TYPE_DSE,

+    TYPE_PCE,

+    TYPE_FIL,

+    TYPE_END,

+};

+

 enum ExtensionPayloadID {

     EXT_FILL,

     EXT_FILL_DATA,

@@ -111,6 +123,35 @@ enum ChannelPosition {

     AAC_CHANNEL_CC    = 5,

 };

+/**

+ * The point during decoding at which channel coupling is applied.

+ */

+enum CouplingPoint {

+    BEFORE_TNS,

+    BETWEEN_TNS_AND_IMDCT,

+    AFTER_IMDCT = 3,

+};

+

+/**

+ * Individual Channel Stream

+ */

+

+/**

+ * Dynamic Range Control - decoded from the bitstream but not processed further.

+ */

+typedef struct {

+    int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.

+    int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative

+    int dyn_rng_ctl[17];                            ///< DRC magnitude information

+    int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.

+    int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.

+    int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.

+    int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.

+    int prog_ref_level;                             /**< A reference level for the long-term program audio level for all

+                                                     *   channels combined.

+                                                     */

+} DynamicRangeControl;

+

 typedef struct {

     int num_pulse;

     int start;

@@ -134,9 +175,15 @@ typedef struct {

     int is_saved;                 ///< Set if elements have stored overlap from previous frame.

     DynamicRangeControl che_drc;

+    /**

+     * @defgroup elements

+     * @{

+     */

     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];

+    /** @} */

     /**

      * @defgroup tables   Computed / set up during initialization.

@@ -145,6 +192,7 @@ typedef struct {

     MDCTContext mdct;

     MDCTContext mdct_small;

     DSPContext dsp;

+    int random_state;

     /** @} */

     /**

@@ -32,6 +32,14 @@

 #include <stdint.h>

+const uint8_t ff_aac_num_swb_1024[] = {

+    41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40

+};

+

+const uint8_t ff_aac_num_swb_128[] = {

+    12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15

+};

+

 const uint32_t ff_aac_scalefactor_code[121] = {

     0x3ffe8, 0x3ffe6, 0x3ffe7, 0x3ffe5, 0x7fff5, 0x7fff1, 0x7ffed, 0x7fff6,

     0x7ffee, 0x7ffef, 0x7fff0, 0x7fffc, 0x7fffd, 0x7ffff, 0x7fffe, 0x7fff7,

@@ -796,6 +804,13 @@ const float ff_aac_ivquant_tab[IVQUANT_SIZE] = {

      4064.0312908,  4074.6805676,  4085.3368071,  4096.0000000,

 };

+/**

+ * 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[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,

@@ -35,6 +35,18 @@

 #include <stdint.h>

+/* NOTE:

+ * Tables in this file are used by the AAC decoder and will be used by the AAC

+ * encoder.

+ */

+

+/* @name number of scalefactor window bands for long and short transform windows respectively

+ * @{

+ */

+extern const uint8_t ff_aac_num_swb_1024[];

+extern const uint8_t ff_aac_num_swb_128 [];

+// @}

+

 extern const uint32_t ff_aac_scalefactor_code[121];

 extern const uint8_t  ff_aac_scalefactor_bits[121];