libfaad2 v2.0rc1 imported

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@10726 b3059339-0415-0410-9bf9-f77b7e298cf2

1 files changed, 357 insertions, 0 deletions

diff --git a/libfaad2/sbr_dec.c b/libfaad2/sbr_dec.c
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+++ b/libfaad2/sbr_dec.c
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+/*
+** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
+** Copyright (C) 2003 M. Bakker, Ahead Software AG, http://www.nero.com
+**  
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+** 
+** This program 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 General Public License for more details.
+** 
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software 
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+**
+** Any non-GPL usage of this software or parts of this software is strictly
+** forbidden.
+**
+** Commercial non-GPL licensing of this software is possible.
+** For more info contact Ahead Software through Mpeg4AAClicense@nero.com.
+**
+** $Id: sbr_dec.c,v 1.5 2003/07/29 08:20:13 menno Exp $
+**/
+
+/*
+   SBR Decoder overview:
+
+   To achieve a synchronized output signal, the following steps have to be
+   acknowledged in the decoder:
+    - The bitstream parser divides the bitstream into two parts; the AAC
+      core coder part and the SBR part.
+    - The SBR bitstream part is fed to the bitstream de-multiplexer followed
+      by de-quantization The raw data is Huffman decoded.
+    - The AAC bitstream part is fed to the AAC core decoder, where the
+      bitstream data of the current frame is decoded, yielding a time domain
+      audio signal block of 1024 samples. The block length could easily be
+      adapted to other sizes e.g. 960.
+    - The core coder audio block is fed to the analysis QMF bank using a
+      delay of 1312 samples.
+    - The analysis QMF bank performs the filtering of the delayed core coder
+      audio signal. The output from the filtering is stored in the matrix
+      Xlow. The output from the analysis QMF bank is delayed tHFGen subband
+      samples, before being fed to the synthesis QMF bank. To achieve
+      synchronization tHFGen = 32, i.e. the value must equal the number of
+      subband samples corresponding to one frame.
+    - The HF generator calculates XHigh given the matrix XLow. The process
+      is guided by the SBR data contained in the current frame.
+    - The envelope adjuster calculates the matrix Y given the matrix XHigh
+      and the SBR envelope data, extracted from the SBR bitstream. To
+      achieve synchronization, tHFAdj has to be set to tHFAdj = 0, i.e. the
+      envelope adjuster operates on data delayed tHFGen subband samples.
+    - The synthesis QMF bank operates on the delayed output from the analysis
+      QMF bank and the output from the envelope adjuster.
+ */
+
+#include "common.h"
+#include "structs.h"
+
+#ifdef SBR_DEC
+
+#include <stdlib.h>
+
+#include "syntax.h"
+#include "bits.h"
+#include "sbr_syntax.h"
+#include "sbr_qmf.h"
+#include "sbr_hfgen.h"
+#include "sbr_hfadj.h"
+
+
+sbr_info *sbrDecodeInit()
+{
+    sbr_info *sbr = malloc(sizeof(sbr_info));
+    memset(sbr, 0, sizeof(sbr_info));
+
+    sbr->bs_freq_scale = 2;
+    sbr->bs_alter_scale = 1;
+    sbr->bs_noise_bands = 2;
+    sbr->bs_limiter_bands = 2;
+    sbr->bs_limiter_gains = 2;
+    sbr->bs_interpol_freq = 1;
+    sbr->bs_smoothing_mode = 1;
+    sbr->bs_start_freq = 5;
+    sbr->bs_amp_res = 1;
+    sbr->bs_samplerate_mode = 1;
+    sbr->prevEnvIsShort[0] = -1;
+    sbr->prevEnvIsShort[1] = -1;
+    sbr->header_count = 0;
+
+    return sbr;
+}
+
+void sbrDecodeEnd(sbr_info *sbr)
+{
+    uint8_t j;
+
+    if (sbr)
+    {
+        qmfa_end(sbr->qmfa[0]);
+        qmfs_end(sbr->qmfs[0]);
+        if (sbr->id_aac == ID_CPE)
+        {
+            qmfa_end(sbr->qmfa[1]);
+            qmfs_end(sbr->qmfs[1]);
+        }
+
+        if (sbr->Xcodec[0]) free(sbr->Xcodec[0]);
+        if (sbr->Xsbr[0]) free(sbr->Xsbr[0]);
+        if (sbr->Xcodec[1]) free(sbr->Xcodec[1]);
+        if (sbr->Xsbr[1]) free(sbr->Xsbr[1]);
+
+        for (j = 0; j < 5; j++)
+        {
+            if (sbr->G_temp_prev[0][j]) free(sbr->G_temp_prev[0][j]);
+            if (sbr->Q_temp_prev[0][j]) free(sbr->Q_temp_prev[0][j]);
+            if (sbr->G_temp_prev[1][j]) free(sbr->G_temp_prev[1][j]);
+            if (sbr->Q_temp_prev[1][j]) free(sbr->Q_temp_prev[1][j]);
+        }
+
+        free(sbr);
+    }
+}
+
+void sbr_save_prev_data(sbr_info *sbr, uint8_t ch)
+{
+    uint8_t i;
+
+    /* save data for next frame */
+    sbr->kx_prev = sbr->kx;
+
+    sbr->L_E_prev[ch] = sbr->L_E[ch];
+
+    sbr->f_prev[ch] = sbr->f[ch][sbr->L_E[ch] - 1];
+    for (i = 0; i < 64; i++)
+    {
+        sbr->E_prev[ch][i] = sbr->E[ch][i][sbr->L_E[ch] - 1];
+        sbr->Q_prev[ch][i] = sbr->Q[ch][i][sbr->L_Q[ch] - 1];
+    }
+
+    for (i = 0; i < 64; i++)
+    {
+        sbr->bs_add_harmonic_prev[ch][i] = sbr->bs_add_harmonic[ch][i];
+    }
+    sbr->bs_add_harmonic_flag_prev[ch] = sbr->bs_add_harmonic_flag[ch];
+
+    if (sbr->l_A[ch] == sbr->L_E[ch])
+        sbr->prevEnvIsShort[ch] = 0;
+    else
+        sbr->prevEnvIsShort[ch] = -1;
+}
+
+
+void sbrDecodeFrame(sbr_info *sbr, real_t *left_channel,
+                    real_t *right_channel, uint8_t id_aac,
+                    uint8_t just_seeked)
+{
+    int16_t i, k, l;
+
+    uint8_t dont_process = 0;
+    uint8_t ch, channels, ret;
+    real_t *ch_buf;
+
+    qmf_t X[32*64];
+#ifdef SBR_LOW_POWER
+    real_t deg[64];
+#endif
+
+    bitfile *ld = (bitfile*)malloc(sizeof(bitfile));
+
+
+    sbr->id_aac = id_aac;
+    channels = (id_aac == ID_SCE) ? 1 : 2;
+
+    /* initialise and read the bitstream */
+    faad_initbits(ld, sbr->data, sbr->data_size);
+
+    ret = sbr_extension_data(ld, sbr, id_aac);
+
+    if (sbr->data) free(sbr->data);
+    sbr->data = NULL;
+
+    ret = ld->error ? ld->error : ret;
+    faad_endbits(ld);
+    if (ld) free(ld);
+    ld = NULL;
+    if (ret || (sbr->header_count == 0))
+    {
+        /* don't process just upsample */
+        dont_process = 1;
+    }
+
+    if (just_seeked)
+        sbr->just_seeked = 1;
+    else
+        sbr->just_seeked = 0;
+
+    for (ch = 0; ch < channels; ch++)
+    {
+        if (sbr->frame == 0)
+        {
+            uint8_t j;
+            sbr->qmfa[ch] = qmfa_init(32);
+            sbr->qmfs[ch] = qmfs_init(64);
+            
+            for (j = 0; j < 5; j++)
+            {
+                sbr->G_temp_prev[ch][j] = malloc(64*sizeof(real_t));
+                sbr->Q_temp_prev[ch][j] = malloc(64*sizeof(real_t));
+            }
+
+            sbr->Xsbr[ch] = malloc((32+tHFGen)*64 * sizeof(qmf_t));
+            sbr->Xcodec[ch] = malloc((32+tHFGen)*32 * sizeof(qmf_t));
+
+            memset(sbr->Xsbr[ch], 0, (32+tHFGen)*64 * sizeof(qmf_t));
+            memset(sbr->Xcodec[ch], 0, (32+tHFGen)*32 * sizeof(qmf_t));
+        }
+
+        if (ch == 0)
+            ch_buf = left_channel;
+        else
+            ch_buf = right_channel;
+
+        for (i = 0; i < tHFAdj; i++)
+        {
+            int8_t j;
+            for (j = sbr->kx_prev; j < sbr->kx; j++)
+            {
+                QMF_RE(sbr->Xcodec[ch][i*32 + j]) = 0;
+#ifndef SBR_LOW_POWER
+                QMF_IM(sbr->Xcodec[ch][i*32 + j]) = 0;
+#endif
+            }
+        }
+
+        /* subband analysis */
+        sbr_qmf_analysis_32(sbr->qmfa[ch], ch_buf, sbr->Xcodec[ch], tHFGen);
+
+        if (!dont_process)
+        {
+            /* insert high frequencies here */
+            /* hf generation using patching */
+            hf_generation(sbr, sbr->Xcodec[ch], sbr->Xsbr[ch]
+#ifdef SBR_LOW_POWER
+                ,deg
+#endif
+                ,ch);
+
+#ifdef SBR_LOW_POWER
+            for (l = sbr->t_E[ch][0]; l < sbr->t_E[ch][sbr->L_E[ch]]; l++)
+            {
+                for (k = 0; k < sbr->kx; k++)
+                {
+                    QMF_RE(sbr->Xsbr[ch][(tHFAdj + l)*64 + k]) = 0;
+                }
+            }
+#endif
+
+            /* hf adjustment */
+            hf_adjustment(sbr, sbr->Xsbr[ch]
+#ifdef SBR_LOW_POWER
+                ,deg
+#endif
+                ,ch);
+        }
+
+        if ((sbr->just_seeked != 0) || dont_process)
+        {
+            for (l = 0; l < 32; l++)
+            {
+                for (k = 0; k < 32; k++)
+                {
+                    QMF_RE(X[l * 64 + k]) = QMF_RE(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
+#ifndef SBR_LOW_POWER
+                    QMF_IM(X[l * 64 + k]) = QMF_IM(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
+#endif
+                }
+                for (k = 32; k < 64; k++)
+                {
+                    QMF_RE(X[l * 64 + k]) = 0;
+#ifndef SBR_LOW_POWER
+                    QMF_IM(X[l * 64 + k]) = 0;
+#endif
+                }
+            }
+        } else {
+            for (l = 0; l < 32; l++)
+            {
+                uint8_t xover_band;
+
+                if (l < sbr->t_E[ch][0])
+                    xover_band = sbr->kx_prev;
+                else
+                    xover_band = sbr->kx;
+
+                for (k = 0; k < xover_band; k++)
+                {
+                    QMF_RE(X[l * 64 + k]) = QMF_RE(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
+#ifndef SBR_LOW_POWER
+                    QMF_IM(X[l * 64 + k]) = QMF_IM(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
+#endif
+                }
+                for (k = xover_band; k < 64; k++)
+                {
+                    QMF_RE(X[l * 64 + k]) = QMF_RE(sbr->Xsbr[ch][(l + tHFAdj)*64 + k]);
+#ifndef SBR_LOW_POWER
+                    QMF_IM(X[l * 64 + k]) = QMF_IM(sbr->Xsbr[ch][(l + tHFAdj)*64 + k]);
+#endif
+                }
+#ifdef SBR_LOW_POWER
+                QMF_RE(X[l * 64 + xover_band - 1]) += QMF_RE(sbr->Xsbr[ch][(l + tHFAdj)*64 + xover_band - 1]);
+#endif
+            }
+        }
+
+        /* subband synthesis */
+        sbr_qmf_synthesis_64(sbr->qmfs[ch], (const complex_t*)X, ch_buf);
+
+        for (i = 0; i < 32; i++)
+        {
+            int8_t j;
+            for (j = 0; j < tHFGen; j++)
+            {
+                QMF_RE(sbr->Xcodec[ch][j*32 + i]) = QMF_RE(sbr->Xcodec[ch][(j+32)*32 + i]);
+#ifndef SBR_LOW_POWER
+                QMF_IM(sbr->Xcodec[ch][j*32 + i]) = QMF_IM(sbr->Xcodec[ch][(j+32)*32 + i]);
+#endif
+            }
+        }
+        for (i = 0; i < 64; i++)
+        {
+            int8_t j;
+            for (j = 0; j < tHFGen; j++)
+            {
+                QMF_RE(sbr->Xsbr[ch][j*64 + i]) = QMF_RE(sbr->Xsbr[ch][(j+32)*64 + i]);
+#ifndef SBR_LOW_POWER
+                QMF_IM(sbr->Xsbr[ch][j*64 + i]) = QMF_IM(sbr->Xsbr[ch][(j+32)*64 + i]);
+#endif
+            }
+        }
+    }
+
+    if (sbr->bs_header_flag)
+        sbr->just_seeked = 0;
+
+    if (sbr->header_count != 0)
+    {
+        for (ch = 0; ch < channels; ch++)
+            sbr_save_prev_data(sbr, ch);
+    }
+
+    sbr->frame++;
+}
+
+#endif