OKed sections of code from the SoC AAC decoder

Originally committed as revision 14626 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 241 insertions, 0 deletions

diff --git a/libavcodec/aac.c b/libavcodec/aac.c
new file mode 100644
index 0000000000..47237ff44e
--- /dev/null
+++ b/libavcodec/aac.c
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+/*
+ * AAC decoder
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file aac.c
+ * AAC decoder
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+/*
+ * supported tools
+ *
+ * Support?             Name
+ * N (code in SoC repo) gain control
+ * Y                    block switching
+ * Y                    window shapes - standard
+ * N                    window shapes - Low Delay
+ * Y                    filterbank - standard
+ * N (code in SoC repo) filterbank - Scalable Sample Rate
+ * Y                    Temporal Noise Shaping
+ * N (code in SoC repo) Long Term Prediction
+ * Y                    intensity stereo
+ * Y                    channel coupling
+ * N                    frequency domain prediction
+ * Y                    Perceptual Noise Substitution
+ * Y                    Mid/Side stereo
+ * N                    Scalable Inverse AAC Quantization
+ * N                    Frequency Selective Switch
+ * N                    upsampling filter
+ * Y                    quantization & coding - AAC
+ * N                    quantization & coding - TwinVQ
+ * N                    quantization & coding - BSAC
+ * N                    AAC Error Resilience tools
+ * N                    Error Resilience payload syntax
+ * N                    Error Protection tool
+ * N                    CELP
+ * N                    Silence Compression
+ * N                    HVXC
+ * N                    HVXC 4kbits/s VR
+ * N                    Structured Audio tools
+ * N                    Structured Audio Sample Bank Format
+ * N                    MIDI
+ * N                    Harmonic and Individual Lines plus Noise
+ * N                    Text-To-Speech Interface
+ * N (in progress)      Spectral Band Replication
+ * Y (not in this code) Layer-1
+ * Y (not in this code) Layer-2
+ * Y (not in this code) Layer-3
+ * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
+ * N (planned)          Parametric Stereo
+ * N                    Direct Stream Transfer
+ *
+ * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
+ *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
+           Parametric Stereo.
+ */
+
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dsputil.h"
+
+#include "aac.h"
+#include "aactab.h"
+#include "mpeg4audio.h"
+
+#include <assert.h>
+#include <errno.h>
+#include <math.h>
+#include <string.h>
+
+#ifndef CONFIG_HARDCODED_TABLES
+    static float ff_aac_ivquant_tab[IVQUANT_SIZE];
+#endif /* CONFIG_HARDCODED_TABLES */
+
+static VLC vlc_scalefactors;
+static VLC vlc_spectral[11];
+
+
+    num_front       = get_bits(gb, 4);
+    num_side        = get_bits(gb, 4);
+    num_back        = get_bits(gb, 4);
+    num_lfe         = get_bits(gb, 2);
+    num_assoc_data  = get_bits(gb, 3);
+    num_cc          = get_bits(gb, 4);
+
+    newpcs->mono_mixdown_tag   = get_bits1(gb) ? get_bits(gb, 4) : -1;
+    newpcs->stereo_mixdown_tag = get_bits1(gb) ? get_bits(gb, 4) : -1;
+
+    if (get_bits1(gb)) {
+        newpcs->mixdown_coeff_index = get_bits(gb, 2);
+        newpcs->pseudo_surround     = get_bits1(gb);
+    }
+
+    program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_FRONT, gb, num_front);
+    program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
+    program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_BACK,  gb, num_back );
+    program_config_element_parse_tags(NULL,                     newpcs->che_type[ID_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
+
+    skip_bits_long(gb, 4 * num_assoc_data);
+
+    program_config_element_parse_tags(newpcs->che_type[ID_CCE], newpcs->che_type[ID_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
+
+    align_get_bits(gb);
+
+    /* comment field, first byte is length */
+    skip_bits_long(gb, 8 * get_bits(gb, 8));
+
+static av_cold int aac_decode_init(AVCodecContext * avccontext) {
+    AACContext * ac = avccontext->priv_data;
+    int i;
+
+    ac->avccontext = avccontext;
+
+    avccontext->sample_rate = ac->m4ac.sample_rate;
+    avccontext->frame_size  = 1024;
+
+    AAC_INIT_VLC_STATIC( 0, 144);
+    AAC_INIT_VLC_STATIC( 1, 114);
+    AAC_INIT_VLC_STATIC( 2, 188);
+    AAC_INIT_VLC_STATIC( 3, 180);
+    AAC_INIT_VLC_STATIC( 4, 172);
+    AAC_INIT_VLC_STATIC( 5, 140);
+    AAC_INIT_VLC_STATIC( 6, 168);
+    AAC_INIT_VLC_STATIC( 7, 114);
+    AAC_INIT_VLC_STATIC( 8, 262);
+    AAC_INIT_VLC_STATIC( 9, 248);
+    AAC_INIT_VLC_STATIC(10, 384);
+
+    dsputil_init(&ac->dsp, avccontext);
+
+    // -1024 - Compensate wrong IMDCT method.
+    // 32768 - Required to scale values to the correct range for the bias method
+    //         for float to int16 conversion.
+
+    if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
+        ac->add_bias = 385.0f;
+        ac->sf_scale = 1. / (-1024. * 32768.);
+        ac->sf_offset = 0;
+    } else {
+        ac->add_bias = 0.0f;
+        ac->sf_scale = 1. / -1024.;
+        ac->sf_offset = 60;
+    }
+
+#ifndef CONFIG_HARDCODED_TABLES
+    for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++)
+        ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] =  cbrt(fabs(i)) * i;
+#endif /* CONFIG_HARDCODED_TABLES */
+
+    INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]),
+        ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
+        ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
+        352);
+
+    ff_mdct_init(&ac->mdct, 11, 1);
+    ff_mdct_init(&ac->mdct_small, 8, 1);
+    return 0;
+}
+
+    int byte_align = get_bits1(gb);
+    int count = get_bits(gb, 8);
+    if (count == 255)
+        count += get_bits(gb, 8);
+    if (byte_align)
+        align_get_bits(gb);
+    skip_bits_long(gb, 8 * count);
+}
+
+/**
+ * inverse quantization
+ *
+ * @param   a   quantized value to be dequantized
+ * @return  Returns dequantized value.
+ */
+static inline float ivquant(int a) {
+    if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1)
+        return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1];
+    else
+        return cbrtf(fabsf(a)) * a;
+}
+
+ * @param   pulse   pointer to pulse data struct
+ * @param   icoef   array of quantized spectral data
+ */
+static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
+    int i, off = ics->swb_offset[pulse->start];
+    for (i = 0; i < pulse->num_pulse; i++) {
+        int ic;
+        off += pulse->offset[i];
+        ic = (icoef[off] - 1)>>31;
+        icoef[off] += (pulse->amp[i]^ic) - ic;
+    }
+}
+
+static av_cold int aac_decode_close(AVCodecContext * avccontext) {
+    AACContext * ac = avccontext->priv_data;
+    int i, j;
+
+    for (i = 0; i < MAX_TAGID; i++) {
+        for(j = 0; j < 4; j++)
+            av_freep(&ac->che[j][i]);
+    }
+
+    ff_mdct_end(&ac->mdct);
+    ff_mdct_end(&ac->mdct_small);
+    av_freep(&ac->interleaved_output);
+    return 0 ;
+}
+
+AVCodec aac_decoder = {
+    "aac",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_AAC,
+    sizeof(AACContext),
+    aac_decode_init,
+    NULL,
+    aac_decode_close,
+    aac_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+};