582 lines
17 KiB
C
582 lines
17 KiB
C
/*
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* AAC decoder
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC decoder
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* @author Oded Shimon ( ods15 ods15 dyndns org )
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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*/
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#define FFT_FLOAT 1
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#define FFT_FIXED_32 0
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#define USE_FIXED 0
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "get_bits.h"
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#include "fft.h"
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#include "imdct15.h"
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#include "lpc.h"
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#include "kbdwin.h"
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#include "sinewin.h"
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#include "aac.h"
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#include "aactab.h"
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#include "aacdectab.h"
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#include "cbrt_tablegen.h"
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#include "sbr.h"
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#include "aacsbr.h"
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#include "mpeg4audio.h"
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#include "aacadtsdec.h"
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#include "libavutil/intfloat.h"
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#include <errno.h>
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#include <math.h>
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#include <stdint.h>
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#include <string.h>
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#if ARCH_ARM
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# include "arm/aac.h"
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#elif ARCH_MIPS
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# include "mips/aacdec_mips.h"
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#endif
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static av_always_inline void reset_predict_state(PredictorState *ps)
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{
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ps->r0 = 0.0f;
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ps->r1 = 0.0f;
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ps->cor0 = 0.0f;
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ps->cor1 = 0.0f;
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ps->var0 = 1.0f;
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ps->var1 = 1.0f;
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}
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#ifndef VMUL2
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static inline float *VMUL2(float *dst, const float *v, unsigned idx,
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const float *scale)
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{
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float s = *scale;
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*dst++ = v[idx & 15] * s;
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*dst++ = v[idx>>4 & 15] * s;
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return dst;
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}
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#endif
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#ifndef VMUL4
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static inline float *VMUL4(float *dst, const float *v, unsigned idx,
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const float *scale)
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{
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float s = *scale;
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*dst++ = v[idx & 3] * s;
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*dst++ = v[idx>>2 & 3] * s;
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*dst++ = v[idx>>4 & 3] * s;
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*dst++ = v[idx>>6 & 3] * s;
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return dst;
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}
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#endif
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#ifndef VMUL2S
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static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
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unsigned sign, const float *scale)
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{
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union av_intfloat32 s0, s1;
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s0.f = s1.f = *scale;
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s0.i ^= sign >> 1 << 31;
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s1.i ^= sign << 31;
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*dst++ = v[idx & 15] * s0.f;
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*dst++ = v[idx>>4 & 15] * s1.f;
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return dst;
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}
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#endif
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#ifndef VMUL4S
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static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
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unsigned sign, const float *scale)
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{
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unsigned nz = idx >> 12;
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union av_intfloat32 s = { .f = *scale };
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union av_intfloat32 t;
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t.i = s.i ^ (sign & 1U<<31);
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*dst++ = v[idx & 3] * t.f;
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sign <<= nz & 1; nz >>= 1;
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t.i = s.i ^ (sign & 1U<<31);
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*dst++ = v[idx>>2 & 3] * t.f;
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sign <<= nz & 1; nz >>= 1;
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t.i = s.i ^ (sign & 1U<<31);
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*dst++ = v[idx>>4 & 3] * t.f;
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sign <<= nz & 1;
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t.i = s.i ^ (sign & 1U<<31);
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*dst++ = v[idx>>6 & 3] * t.f;
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return dst;
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}
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#endif
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static av_always_inline float flt16_round(float pf)
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{
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union av_intfloat32 tmp;
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tmp.f = pf;
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tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
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return tmp.f;
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}
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static av_always_inline float flt16_even(float pf)
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{
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union av_intfloat32 tmp;
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tmp.f = pf;
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tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
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return tmp.f;
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}
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static av_always_inline float flt16_trunc(float pf)
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{
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union av_intfloat32 pun;
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pun.f = pf;
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pun.i &= 0xFFFF0000U;
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return pun.f;
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}
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static av_always_inline void predict(PredictorState *ps, float *coef,
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int output_enable)
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{
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const float a = 0.953125; // 61.0 / 64
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const float alpha = 0.90625; // 29.0 / 32
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float e0, e1;
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float pv;
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float k1, k2;
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float r0 = ps->r0, r1 = ps->r1;
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float cor0 = ps->cor0, cor1 = ps->cor1;
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float var0 = ps->var0, var1 = ps->var1;
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k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
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k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
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pv = flt16_round(k1 * r0 + k2 * r1);
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if (output_enable)
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*coef += pv;
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e0 = *coef;
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e1 = e0 - k1 * r0;
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ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
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ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
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ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
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ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
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ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
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ps->r0 = flt16_trunc(a * e0);
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}
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/**
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* Apply dependent channel coupling (applied before IMDCT).
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*
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* @param index index into coupling gain array
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*/
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static void apply_dependent_coupling(AACContext *ac,
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SingleChannelElement *target,
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ChannelElement *cce, int index)
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{
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IndividualChannelStream *ics = &cce->ch[0].ics;
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const uint16_t *offsets = ics->swb_offset;
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float *dest = target->coeffs;
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const float *src = cce->ch[0].coeffs;
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int g, i, group, k, idx = 0;
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if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
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av_log(ac->avctx, AV_LOG_ERROR,
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"Dependent coupling is not supported together with LTP\n");
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return;
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}
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for (g = 0; g < ics->num_window_groups; g++) {
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for (i = 0; i < ics->max_sfb; i++, idx++) {
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if (cce->ch[0].band_type[idx] != ZERO_BT) {
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const float gain = cce->coup.gain[index][idx];
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for (group = 0; group < ics->group_len[g]; group++) {
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for (k = offsets[i]; k < offsets[i + 1]; k++) {
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// FIXME: SIMDify
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dest[group * 128 + k] += gain * src[group * 128 + k];
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}
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}
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}
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}
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dest += ics->group_len[g] * 128;
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src += ics->group_len[g] * 128;
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}
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}
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/**
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* Apply independent channel coupling (applied after IMDCT).
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*
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* @param index index into coupling gain array
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*/
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static void apply_independent_coupling(AACContext *ac,
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SingleChannelElement *target,
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ChannelElement *cce, int index)
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{
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int i;
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const float gain = cce->coup.gain[index][0];
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const float *src = cce->ch[0].ret;
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float *dest = target->ret;
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const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
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for (i = 0; i < len; i++)
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dest[i] += gain * src[i];
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}
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#include "aacdec_template.c"
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#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
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struct LATMContext {
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AACContext aac_ctx; ///< containing AACContext
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int initialized; ///< initialized after a valid extradata was seen
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// parser data
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int audio_mux_version_A; ///< LATM syntax version
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int frame_length_type; ///< 0/1 variable/fixed frame length
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int frame_length; ///< frame length for fixed frame length
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};
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static inline uint32_t latm_get_value(GetBitContext *b)
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{
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int length = get_bits(b, 2);
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return get_bits_long(b, (length+1)*8);
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}
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static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
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GetBitContext *gb, int asclen)
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{
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AACContext *ac = &latmctx->aac_ctx;
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AVCodecContext *avctx = ac->avctx;
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MPEG4AudioConfig m4ac = { 0 };
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int config_start_bit = get_bits_count(gb);
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int sync_extension = 0;
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int bits_consumed, esize;
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if (asclen) {
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sync_extension = 1;
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asclen = FFMIN(asclen, get_bits_left(gb));
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} else
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asclen = get_bits_left(gb);
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if (config_start_bit % 8) {
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avpriv_request_sample(latmctx->aac_ctx.avctx,
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"Non-byte-aligned audio-specific config");
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return AVERROR_PATCHWELCOME;
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}
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if (asclen <= 0)
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return AVERROR_INVALIDDATA;
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bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
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gb->buffer + (config_start_bit / 8),
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asclen, sync_extension);
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if (bits_consumed < 0)
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return AVERROR_INVALIDDATA;
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if (!latmctx->initialized ||
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ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
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ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
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if(latmctx->initialized) {
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av_log(avctx, AV_LOG_INFO, "audio config changed\n");
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} else {
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av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
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}
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latmctx->initialized = 0;
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esize = (bits_consumed+7) / 8;
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if (avctx->extradata_size < esize) {
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av_free(avctx->extradata);
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avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata)
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return AVERROR(ENOMEM);
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}
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avctx->extradata_size = esize;
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memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
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memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
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}
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skip_bits_long(gb, bits_consumed);
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return bits_consumed;
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}
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static int read_stream_mux_config(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int ret, audio_mux_version = get_bits(gb, 1);
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latmctx->audio_mux_version_A = 0;
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if (audio_mux_version)
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latmctx->audio_mux_version_A = get_bits(gb, 1);
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if (!latmctx->audio_mux_version_A) {
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if (audio_mux_version)
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latm_get_value(gb); // taraFullness
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skip_bits(gb, 1); // allStreamSameTimeFraming
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skip_bits(gb, 6); // numSubFrames
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// numPrograms
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if (get_bits(gb, 4)) { // numPrograms
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avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
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return AVERROR_PATCHWELCOME;
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}
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// for each program (which there is only one in DVB)
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// for each layer (which there is only one in DVB)
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if (get_bits(gb, 3)) { // numLayer
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avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
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return AVERROR_PATCHWELCOME;
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}
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// for all but first stream: use_same_config = get_bits(gb, 1);
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if (!audio_mux_version) {
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if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
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return ret;
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} else {
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int ascLen = latm_get_value(gb);
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if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
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return ret;
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ascLen -= ret;
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skip_bits_long(gb, ascLen);
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}
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latmctx->frame_length_type = get_bits(gb, 3);
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switch (latmctx->frame_length_type) {
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case 0:
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skip_bits(gb, 8); // latmBufferFullness
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break;
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case 1:
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latmctx->frame_length = get_bits(gb, 9);
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break;
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case 3:
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case 4:
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case 5:
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skip_bits(gb, 6); // CELP frame length table index
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break;
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case 6:
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case 7:
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skip_bits(gb, 1); // HVXC frame length table index
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break;
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}
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if (get_bits(gb, 1)) { // other data
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if (audio_mux_version) {
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latm_get_value(gb); // other_data_bits
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} else {
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int esc;
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do {
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esc = get_bits(gb, 1);
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skip_bits(gb, 8);
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} while (esc);
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}
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}
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if (get_bits(gb, 1)) // crc present
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skip_bits(gb, 8); // config_crc
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}
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return 0;
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}
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static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
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{
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uint8_t tmp;
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if (ctx->frame_length_type == 0) {
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int mux_slot_length = 0;
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do {
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tmp = get_bits(gb, 8);
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mux_slot_length += tmp;
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} while (tmp == 255);
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return mux_slot_length;
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} else if (ctx->frame_length_type == 1) {
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return ctx->frame_length;
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} else if (ctx->frame_length_type == 3 ||
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ctx->frame_length_type == 5 ||
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ctx->frame_length_type == 7) {
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skip_bits(gb, 2); // mux_slot_length_coded
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}
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return 0;
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}
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static int read_audio_mux_element(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int err;
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uint8_t use_same_mux = get_bits(gb, 1);
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if (!use_same_mux) {
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if ((err = read_stream_mux_config(latmctx, gb)) < 0)
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return err;
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} else if (!latmctx->aac_ctx.avctx->extradata) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
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"no decoder config found\n");
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return AVERROR(EAGAIN);
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}
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if (latmctx->audio_mux_version_A == 0) {
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int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
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if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
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return AVERROR_INVALIDDATA;
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} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
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"frame length mismatch %d << %d\n",
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mux_slot_length_bytes * 8, get_bits_left(gb));
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return AVERROR_INVALIDDATA;
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}
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}
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return 0;
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}
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static int latm_decode_frame(AVCodecContext *avctx, void *out,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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struct LATMContext *latmctx = avctx->priv_data;
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int muxlength, err;
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GetBitContext gb;
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if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
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return err;
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// check for LOAS sync word
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if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
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return AVERROR_INVALIDDATA;
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muxlength = get_bits(&gb, 13) + 3;
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// not enough data, the parser should have sorted this out
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if (muxlength > avpkt->size)
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return AVERROR_INVALIDDATA;
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|
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if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
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return err;
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|
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if (!latmctx->initialized) {
|
|
if (!avctx->extradata) {
|
|
*got_frame_ptr = 0;
|
|
return avpkt->size;
|
|
} else {
|
|
push_output_configuration(&latmctx->aac_ctx);
|
|
if ((err = decode_audio_specific_config(
|
|
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
|
|
avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
|
|
pop_output_configuration(&latmctx->aac_ctx);
|
|
return err;
|
|
}
|
|
latmctx->initialized = 1;
|
|
}
|
|
}
|
|
|
|
if (show_bits(&gb, 12) == 0xfff) {
|
|
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
|
|
"ADTS header detected, probably as result of configuration "
|
|
"misparsing\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
|
|
case AOT_ER_AAC_LC:
|
|
case AOT_ER_AAC_LTP:
|
|
case AOT_ER_AAC_LD:
|
|
case AOT_ER_AAC_ELD:
|
|
err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
|
|
break;
|
|
default:
|
|
err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
|
|
}
|
|
if (err < 0)
|
|
return err;
|
|
|
|
return muxlength;
|
|
}
|
|
|
|
static av_cold int latm_decode_init(AVCodecContext *avctx)
|
|
{
|
|
struct LATMContext *latmctx = avctx->priv_data;
|
|
int ret = aac_decode_init(avctx);
|
|
|
|
if (avctx->extradata_size > 0)
|
|
latmctx->initialized = !ret;
|
|
|
|
return ret;
|
|
}
|
|
|
|
AVCodec ff_aac_decoder = {
|
|
.name = "aac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_AAC,
|
|
.priv_data_size = sizeof(AACContext),
|
|
.init = aac_decode_init,
|
|
.close = aac_decode_close,
|
|
.decode = aac_decode_frame,
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
|
|
},
|
|
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
|
|
.channel_layouts = aac_channel_layout,
|
|
.flush = flush,
|
|
.priv_class = &aac_decoder_class,
|
|
.profiles = profiles,
|
|
};
|
|
|
|
/*
|
|
Note: This decoder filter is intended to decode LATM streams transferred
|
|
in MPEG transport streams which only contain one program.
|
|
To do a more complex LATM demuxing a separate LATM demuxer should be used.
|
|
*/
|
|
AVCodec ff_aac_latm_decoder = {
|
|
.name = "aac_latm",
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_AAC_LATM,
|
|
.priv_data_size = sizeof(struct LATMContext),
|
|
.init = latm_decode_init,
|
|
.close = aac_decode_close,
|
|
.decode = latm_decode_frame,
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
|
|
},
|
|
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
|
|
.channel_layouts = aac_channel_layout,
|
|
.flush = flush,
|
|
.profiles = profiles,
|
|
};
|