[FFmpeg-cvslog] r14626 - trunk/libavcodec/aac.c
superdump
subversion
Tue Aug 5 21:32:01 CEST 2008
Author: superdump
Date: Tue Aug 5 21:32:01 2008
New Revision: 14626
Log:
OKed sections of code from the SoC AAC decoder
Added:
trunk/libavcodec/aac.c
Added: trunk/libavcodec/aac.c
==============================================================================
--- (empty file)
+++ trunk/libavcodec/aac.c Tue Aug 5 21:32:01 2008
@@ -0,0 +1,241 @@
+/*
+ * 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"),
+};
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