3866755d28
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@21304 a1c6a512-1295-4272-9138-f99709370657
1614 lines
48 KiB
C
1614 lines
48 KiB
C
/*
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* WMA compatible decoder
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* Copyright (c) 2002 The FFmpeg Project.
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*
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* This library 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 of the License, or (at your option) any later version.
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*
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* This library 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 this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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/**
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* @file wmadec.c
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* WMA compatible decoder.
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*/
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#include <codecs.h>
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#include <codecs/lib/codeclib.h>
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#include "asf.h"
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#include "wmadec.h"
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#include "wmafixed.h"
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#include "bitstream.h"
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#define VLCBITS 7 /*7 is the lowest without glitching*/
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#define VLCMAX ((22+VLCBITS-1)/VLCBITS)
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#define EXPVLCBITS 7
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#define EXPMAX ((19+EXPVLCBITS-1)/EXPVLCBITS)
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#define HGAINVLCBITS 9
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#define HGAINMAX ((13+HGAINVLCBITS-1)/HGAINVLCBITS)
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typedef struct CoefVLCTable
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{
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int n; /* total number of codes */
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const uint32_t *huffcodes; /* VLC bit values */
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const uint8_t *huffbits; /* VLC bit size */
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const uint16_t *levels; /* table to build run/level tables */
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}
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CoefVLCTable;
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static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len);
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fixed32 coefsarray[MAX_CHANNELS][BLOCK_MAX_SIZE] IBSS_ATTR;
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/*decode and window into IRAM on targets with at least 80KB of codec IRAM*/
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fixed32 frame_out_buf[MAX_CHANNELS][BLOCK_MAX_SIZE * 2] IBSS_ATTR_WMA_LARGE_IRAM;
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//static variables that replace malloced stuff
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fixed32 stat0[2048], stat1[1024], stat2[512], stat3[256], stat4[128]; //these are the MDCT reconstruction windows
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uint16_t *runtabarray[2], *levtabarray[2]; //these are VLC lookup tables
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uint16_t runtab0[1336], runtab1[1336], levtab0[1336], levtab1[1336]; //these could be made smaller since only one can be 1336
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#define VLCBUF1SIZE 4598
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#define VLCBUF2SIZE 3574
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#define VLCBUF3SIZE 360
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#define VLCBUF4SIZE 540
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/*putting these in IRAM actually makes PP slower*/
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VLC_TYPE vlcbuf1[VLCBUF1SIZE][2];
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VLC_TYPE vlcbuf2[VLCBUF2SIZE][2];
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VLC_TYPE vlcbuf3[VLCBUF3SIZE][2];
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VLC_TYPE vlcbuf4[VLCBUF4SIZE][2];
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#include "wmadata.h" // PJJ
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/*
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* Helper functions for wma_window.
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*
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*
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*/
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#ifdef CPU_ARM
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static inline
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void vector_fmul_add_add(fixed32 *dst, const fixed32 *data,
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const fixed32 *window, int n)
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{
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/* Block sizes are always power of two */
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asm volatile (
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"0:"
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"ldmia %[d]!, {r0, r1};"
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"ldmia %[w]!, {r4, r5};"
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/* consume the first data and window value so we can use those
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* registers again */
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"smull r8, r9, r0, r4;"
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"ldmia %[dst], {r0, r4};"
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"add r0, r0, r9, lsl #1;" /* *dst=*dst+(r9<<1)*/
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"smull r8, r9, r1, r5;"
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"add r1, r4, r9, lsl #1;"
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"stmia %[dst]!, {r0, r1};"
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"subs %[n], %[n], #2;"
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"bne 0b;"
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: [d] "+r" (data), [w] "+r" (window), [dst] "+r" (dst), [n] "+r" (n)
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: : "r0", "r1", "r4", "r5", "r8", "r9", "memory", "cc");
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}
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static inline
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void vector_fmul_reverse(fixed32 *dst, const fixed32 *src0, const fixed32 *src1,
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int len)
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{
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/* Block sizes are always power of two */
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asm volatile (
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"add %[s1], %[s1], %[n], lsl #2;"
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"0:"
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"ldmia %[s0]!, {r0, r1};"
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"ldmdb %[s1]!, {r4, r5};"
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"smull r8, r9, r0, r5;"
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"mov r0, r9, lsl #1;"
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"smull r8, r9, r1, r4;"
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"mov r1, r9, lsl #1;"
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"stmia %[dst]!, {r0, r1};"
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"subs %[n], %[n], #2;"
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"bne 0b;"
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: [s0] "+r" (src0), [s1] "+r" (src1), [dst] "+r" (dst), [n] "+r" (len)
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: : "r0", "r1", "r4", "r5", "r8", "r9", "memory", "cc");
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}
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#elif defined(CPU_COLDFIRE)
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static inline
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void vector_fmul_add_add(fixed32 *dst, const fixed32 *data,
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const fixed32 *window, int n)
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{
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/* Block sizes are always power of two. Smallest block is always way bigger
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* than four too.*/
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asm volatile (
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"0:"
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"movem.l (%[d]), %%d0-%%d3;"
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"movem.l (%[w]), %%d4-%%d5/%%a0-%%a1;"
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"mac.l %%d0, %%d4, %%acc0;"
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"mac.l %%d1, %%d5, %%acc1;"
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"mac.l %%d2, %%a0, %%acc2;"
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"mac.l %%d3, %%a1, %%acc3;"
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"lea.l (16, %[d]), %[d];"
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"lea.l (16, %[w]), %[w];"
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"movclr.l %%acc0, %%d0;"
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"movclr.l %%acc1, %%d1;"
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"movclr.l %%acc2, %%d2;"
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"movclr.l %%acc3, %%d3;"
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"movem.l (%[dst]), %%d4-%%d5/%%a0-%%a1;"
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"add.l %%d4, %%d0;"
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"add.l %%d5, %%d1;"
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"add.l %%a0, %%d2;"
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"add.l %%a1, %%d3;"
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"movem.l %%d0-%%d3, (%[dst]);"
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"lea.l (16, %[dst]), %[dst];"
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"subq.l #4, %[n];"
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"jne 0b;"
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: [d] "+a" (data), [w] "+a" (window), [dst] "+a" (dst), [n] "+d" (n)
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: : "d0", "d1", "d2", "d3", "d4", "d5", "a0", "a1", "memory", "cc");
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}
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static inline
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void vector_fmul_reverse(fixed32 *dst, const fixed32 *src0, const fixed32 *src1,
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int len)
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{
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/* Block sizes are always power of two. Smallest block is always way bigger
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* than four too.*/
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asm volatile (
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"lea.l (-16, %[s1], %[n]*4), %[s1];"
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"0:"
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"movem.l (%[s0]), %%d0-%%d3;"
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"movem.l (%[s1]), %%d4-%%d5/%%a0-%%a1;"
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"mac.l %%d0, %%a1, %%acc0;"
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"mac.l %%d1, %%a0, %%acc1;"
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"mac.l %%d2, %%d5, %%acc2;"
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"mac.l %%d3, %%d4, %%acc3;"
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"lea.l (16, %[s0]), %[s0];"
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"lea.l (-16, %[s1]), %[s1];"
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"movclr.l %%acc0, %%d0;"
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"movclr.l %%acc1, %%d1;"
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"movclr.l %%acc2, %%d2;"
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"movclr.l %%acc3, %%d3;"
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"movem.l %%d0-%%d3, (%[dst]);"
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"lea.l (16, %[dst]), %[dst];"
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"subq.l #4, %[n];"
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"jne 0b;"
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: [s0] "+a" (src0), [s1] "+a" (src1), [dst] "+a" (dst), [n] "+d" (len)
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: : "d0", "d1", "d2", "d3", "d4", "d5", "a0", "a1", "memory", "cc");
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}
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#else
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static inline void vector_fmul_add_add(fixed32 *dst, const fixed32 *src0, const fixed32 *src1, int len){
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int i;
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for(i=0; i<len; i++)
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dst[i] = fixmul32b(src0[i], src1[i]) + dst[i];
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}
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static inline void vector_fmul_reverse(fixed32 *dst, const fixed32 *src0, const fixed32 *src1, int len){
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int i;
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src1 += len-1;
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for(i=0; i<len; i++)
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dst[i] = fixmul32b(src0[i], src1[-i]);
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}
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#endif
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/**
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* Apply MDCT window and add into output.
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*
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* We ensure that when the windows overlap their squared sum
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* is always 1 (MDCT reconstruction rule).
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*
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* The Vorbis I spec has a great diagram explaining this process.
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* See section 1.3.2.3 of http://xiph.org/vorbis/doc/Vorbis_I_spec.html
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*/
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static void wma_window(WMADecodeContext *s, fixed32 *in, fixed32 *out)
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{
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//float *in = s->output;
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int block_len, bsize, n;
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/* left part */
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/*previous block was larger, so we'll use the size of the current block to set the window size*/
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if (s->block_len_bits <= s->prev_block_len_bits) {
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block_len = s->block_len;
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bsize = s->frame_len_bits - s->block_len_bits;
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vector_fmul_add_add(out, in, s->windows[bsize], block_len);
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} else {
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/*previous block was smaller or the same size, so use it's size to set the window length*/
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block_len = 1 << s->prev_block_len_bits;
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/*find the middle of the two overlapped blocks, this will be the first overlapped sample*/
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n = (s->block_len - block_len) / 2;
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bsize = s->frame_len_bits - s->prev_block_len_bits;
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vector_fmul_add_add(out+n, in+n, s->windows[bsize], block_len);
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memcpy(out+n+block_len, in+n+block_len, n*sizeof(fixed32));
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}
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/* Advance to the end of the current block and prepare to window it for the next block.
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* Since the window function needs to be reversed, we do it backwards starting with the
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* last sample and moving towards the first
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*/
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out += s->block_len;
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in += s->block_len;
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/* right part */
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if (s->block_len_bits <= s->next_block_len_bits) {
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block_len = s->block_len;
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bsize = s->frame_len_bits - s->block_len_bits;
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vector_fmul_reverse(out, in, s->windows[bsize], block_len);
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} else {
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block_len = 1 << s->next_block_len_bits;
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n = (s->block_len - block_len) / 2;
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bsize = s->frame_len_bits - s->next_block_len_bits;
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memcpy(out, in, n*sizeof(fixed32));
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vector_fmul_reverse(out+n, in+n, s->windows[bsize], block_len);
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memset(out+n+block_len, 0, n*sizeof(fixed32));
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}
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}
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/* XXX: use same run/length optimization as mpeg decoders */
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static void init_coef_vlc(VLC *vlc,
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uint16_t **prun_table, uint16_t **plevel_table,
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const CoefVLCTable *vlc_table, int tab)
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{
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int n = vlc_table->n;
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const uint8_t *table_bits = vlc_table->huffbits;
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const uint32_t *table_codes = vlc_table->huffcodes;
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const uint16_t *levels_table = vlc_table->levels;
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uint16_t *run_table, *level_table;
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const uint16_t *p;
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int i, l, j, level;
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init_vlc(vlc, VLCBITS, n, table_bits, 1, 1, table_codes, 4, 4, 0);
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run_table = runtabarray[tab];
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level_table= levtabarray[tab];
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p = levels_table;
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i = 2;
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level = 1;
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while (i < n)
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{
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l = *p++;
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for(j=0;j<l;++j)
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{
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run_table[i] = j;
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level_table[i] = level;
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++i;
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}
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++level;
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}
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*prun_table = run_table;
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*plevel_table = level_table;
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}
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int wma_decode_init(WMADecodeContext* s, asf_waveformatex_t *wfx)
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{
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//WMADecodeContext *s = avctx->priv_data;
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int i, flags1, flags2;
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fixed32 *window;
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uint8_t *extradata;
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fixed64 bps1;
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fixed32 high_freq;
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fixed64 bps;
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int sample_rate1;
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int coef_vlc_table;
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// int filehandle;
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#ifdef CPU_COLDFIRE
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coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
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#endif
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/*clear stereo setting to avoid glitches when switching stereo->mono*/
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s->channel_coded[0]=0;
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s->channel_coded[1]=0;
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s->ms_stereo=0;
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s->sample_rate = wfx->rate;
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s->nb_channels = wfx->channels;
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s->bit_rate = wfx->bitrate;
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s->block_align = wfx->blockalign;
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s->coefs = &coefsarray;
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s->frame_out = &frame_out_buf;
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if (wfx->codec_id == ASF_CODEC_ID_WMAV1) {
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s->version = 1;
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} else if (wfx->codec_id == ASF_CODEC_ID_WMAV2 ) {
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s->version = 2;
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} else {
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/*one of those other wma flavors that don't have GPLed decoders */
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return -1;
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}
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/* extract flag infos */
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flags1 = 0;
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flags2 = 0;
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extradata = wfx->data;
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if (s->version == 1 && wfx->datalen >= 4) {
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flags1 = extradata[0] | (extradata[1] << 8);
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flags2 = extradata[2] | (extradata[3] << 8);
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}else if (s->version == 2 && wfx->datalen >= 6){
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flags1 = extradata[0] | (extradata[1] << 8) |
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(extradata[2] << 16) | (extradata[3] << 24);
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flags2 = extradata[4] | (extradata[5] << 8);
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}
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s->use_exp_vlc = flags2 & 0x0001;
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s->use_bit_reservoir = flags2 & 0x0002;
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s->use_variable_block_len = flags2 & 0x0004;
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/* compute MDCT block size */
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if (s->sample_rate <= 16000){
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s->frame_len_bits = 9;
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}else if (s->sample_rate <= 22050 ||
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(s->sample_rate <= 32000 && s->version == 1)){
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s->frame_len_bits = 10;
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}else{
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s->frame_len_bits = 11;
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}
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s->frame_len = 1 << s->frame_len_bits;
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if (s-> use_variable_block_len)
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{
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int nb_max, nb;
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nb = ((flags2 >> 3) & 3) + 1;
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if ((s->bit_rate / s->nb_channels) >= 32000)
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{
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nb += 2;
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}
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nb_max = s->frame_len_bits - BLOCK_MIN_BITS; //max is 11-7
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if (nb > nb_max)
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nb = nb_max;
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s->nb_block_sizes = nb + 1;
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}
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else
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{
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s->nb_block_sizes = 1;
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}
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/* init rate dependant parameters */
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s->use_noise_coding = 1;
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high_freq = itofix64(s->sample_rate) >> 1;
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/* if version 2, then the rates are normalized */
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sample_rate1 = s->sample_rate;
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if (s->version == 2)
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{
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if (sample_rate1 >= 44100)
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sample_rate1 = 44100;
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else if (sample_rate1 >= 22050)
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sample_rate1 = 22050;
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else if (sample_rate1 >= 16000)
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sample_rate1 = 16000;
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else if (sample_rate1 >= 11025)
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sample_rate1 = 11025;
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else if (sample_rate1 >= 8000)
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sample_rate1 = 8000;
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}
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fixed64 tmp = itofix64(s->bit_rate);
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fixed64 tmp2 = itofix64(s->nb_channels * s->sample_rate);
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bps = fixdiv64(tmp, tmp2);
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fixed64 tim = bps * s->frame_len;
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fixed64 tmpi = fixdiv64(tim,itofix64(8));
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s->byte_offset_bits = av_log2(fixtoi64(tmpi+0x8000)) + 2;
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/* compute high frequency value and choose if noise coding should
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be activated */
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bps1 = bps;
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if (s->nb_channels == 2)
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bps1 = fixmul32(bps,0x1999a);
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if (sample_rate1 == 44100)
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{
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if (bps1 >= 0x9c29)
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s->use_noise_coding = 0;
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else
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high_freq = fixmul32(high_freq,0x6666);
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}
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else if (sample_rate1 == 22050)
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{
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if (bps1 >= 0x128f6)
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s->use_noise_coding = 0;
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else if (bps1 >= 0xb852)
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high_freq = fixmul32(high_freq,0xb333);
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else
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high_freq = fixmul32(high_freq,0x999a);
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}
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else if (sample_rate1 == 16000)
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{
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if (bps > 0x8000)
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high_freq = fixmul32(high_freq,0x8000);
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else
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high_freq = fixmul32(high_freq,0x4ccd);
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}
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else if (sample_rate1 == 11025)
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{
|
|
high_freq = fixmul32(high_freq,0xb333);
|
|
}
|
|
else if (sample_rate1 == 8000)
|
|
{
|
|
if (bps <= 0xa000)
|
|
{
|
|
high_freq = fixmul32(high_freq,0x8000);
|
|
}
|
|
else if (bps > 0xc000)
|
|
{
|
|
s->use_noise_coding = 0;
|
|
}
|
|
else
|
|
{
|
|
high_freq = fixmul32(high_freq,0xa666);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (bps >= 0xcccd)
|
|
{
|
|
high_freq = fixmul32(high_freq,0xc000);
|
|
}
|
|
else if (bps >= 0x999a)
|
|
{
|
|
high_freq = fixmul32(high_freq,0x999a);
|
|
}
|
|
else
|
|
{
|
|
high_freq = fixmul32(high_freq,0x8000);
|
|
}
|
|
}
|
|
|
|
/* compute the scale factor band sizes for each MDCT block size */
|
|
{
|
|
int a, b, pos, lpos, k, block_len, i, j, n;
|
|
const uint8_t *table;
|
|
|
|
if (s->version == 1)
|
|
{
|
|
s->coefs_start = 3;
|
|
}
|
|
else
|
|
{
|
|
s->coefs_start = 0;
|
|
}
|
|
for(k = 0; k < s->nb_block_sizes; ++k)
|
|
{
|
|
block_len = s->frame_len >> k;
|
|
|
|
if (s->version == 1)
|
|
{
|
|
lpos = 0;
|
|
for(i=0;i<25;++i)
|
|
{
|
|
a = wma_critical_freqs[i];
|
|
b = s->sample_rate;
|
|
pos = ((block_len * 2 * a) + (b >> 1)) / b;
|
|
if (pos > block_len)
|
|
pos = block_len;
|
|
s->exponent_bands[0][i] = pos - lpos;
|
|
if (pos >= block_len)
|
|
{
|
|
++i;
|
|
break;
|
|
}
|
|
lpos = pos;
|
|
}
|
|
s->exponent_sizes[0] = i;
|
|
}
|
|
else
|
|
{
|
|
/* hardcoded tables */
|
|
table = NULL;
|
|
a = s->frame_len_bits - BLOCK_MIN_BITS - k;
|
|
if (a < 3)
|
|
{
|
|
if (s->sample_rate >= 44100)
|
|
table = exponent_band_44100[a];
|
|
else if (s->sample_rate >= 32000)
|
|
table = exponent_band_32000[a];
|
|
else if (s->sample_rate >= 22050)
|
|
table = exponent_band_22050[a];
|
|
}
|
|
if (table)
|
|
{
|
|
n = *table++;
|
|
for(i=0;i<n;++i)
|
|
s->exponent_bands[k][i] = table[i];
|
|
s->exponent_sizes[k] = n;
|
|
}
|
|
else
|
|
{
|
|
j = 0;
|
|
lpos = 0;
|
|
for(i=0;i<25;++i)
|
|
{
|
|
a = wma_critical_freqs[i];
|
|
b = s->sample_rate;
|
|
pos = ((block_len * 2 * a) + (b << 1)) / (4 * b);
|
|
pos <<= 2;
|
|
if (pos > block_len)
|
|
pos = block_len;
|
|
if (pos > lpos)
|
|
s->exponent_bands[k][j++] = pos - lpos;
|
|
if (pos >= block_len)
|
|
break;
|
|
lpos = pos;
|
|
}
|
|
s->exponent_sizes[k] = j;
|
|
}
|
|
}
|
|
|
|
/* max number of coefs */
|
|
s->coefs_end[k] = (s->frame_len - ((s->frame_len * 9) / 100)) >> k;
|
|
/* high freq computation */
|
|
|
|
fixed32 tmp1 = high_freq*2; /* high_freq is a fixed32!*/
|
|
fixed32 tmp2=itofix32(s->sample_rate>>1);
|
|
s->high_band_start[k] = fixtoi32( fixdiv32(tmp1, tmp2) * (block_len>>1) +0x8000);
|
|
|
|
/*
|
|
s->high_band_start[k] = (int)((block_len * 2 * high_freq) /
|
|
s->sample_rate + 0.5);*/
|
|
|
|
n = s->exponent_sizes[k];
|
|
j = 0;
|
|
pos = 0;
|
|
for(i=0;i<n;++i)
|
|
{
|
|
int start, end;
|
|
start = pos;
|
|
pos += s->exponent_bands[k][i];
|
|
end = pos;
|
|
if (start < s->high_band_start[k])
|
|
start = s->high_band_start[k];
|
|
if (end > s->coefs_end[k])
|
|
end = s->coefs_end[k];
|
|
if (end > start)
|
|
s->exponent_high_bands[k][j++] = end - start;
|
|
}
|
|
s->exponent_high_sizes[k] = j;
|
|
}
|
|
}
|
|
|
|
/*Not using the ffmpeg IMDCT anymore*/
|
|
|
|
/* mdct_init_global();
|
|
|
|
for(i = 0; i < s->nb_block_sizes; ++i)
|
|
{
|
|
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1);
|
|
|
|
}
|
|
*/
|
|
|
|
/*ffmpeg uses malloc to only allocate as many window sizes as needed. However, we're really only interested in the worst case memory usage.
|
|
* In the worst case you can have 5 window sizes, 128 doubling up 2048
|
|
* Smaller windows are handled differently.
|
|
* Since we don't have malloc, just statically allocate this
|
|
*/
|
|
fixed32 *temp[5];
|
|
temp[0] = stat0;
|
|
temp[1] = stat1;
|
|
temp[2] = stat2;
|
|
temp[3] = stat3;
|
|
temp[4] = stat4;
|
|
|
|
/* init MDCT windows : simple sinus window */
|
|
for(i = 0; i < s->nb_block_sizes; i++)
|
|
{
|
|
int n, j;
|
|
fixed32 alpha;
|
|
n = 1 << (s->frame_len_bits - i);
|
|
//window = av_malloc(sizeof(fixed32) * n);
|
|
window = temp[i];
|
|
|
|
//fixed32 n2 = itofix32(n<<1); //2x the window length
|
|
//alpha = fixdiv32(M_PI_F, n2); //PI / (2x Window length) == PI<<(s->frame_len_bits - i+1)
|
|
|
|
//alpha = M_PI_F>>(s->frame_len_bits - i+1);
|
|
alpha = (1<<15)>>(s->frame_len_bits - i+1); /* this calculates 0.5/(2*n) */
|
|
for(j=0;j<n;++j)
|
|
{
|
|
fixed32 j2 = itofix32(j) + 0x8000;
|
|
window[j] = fsincos(fixmul32(j2,alpha)<<16, 0); //alpha between 0 and pi/2
|
|
|
|
}
|
|
s->windows[i] = window;
|
|
|
|
}
|
|
|
|
s->reset_block_lengths = 1;
|
|
|
|
if (s->use_noise_coding)
|
|
{
|
|
/* init the noise generator */
|
|
if (s->use_exp_vlc)
|
|
{
|
|
s->noise_mult = 0x51f;
|
|
s->noise_table = noisetable_exp;
|
|
}
|
|
else
|
|
{
|
|
s->noise_mult = 0xa3d;
|
|
/* LSP values are simply 2x the EXP values */
|
|
for (i=0;i<NOISE_TAB_SIZE;++i)
|
|
noisetable_exp[i] = noisetable_exp[i]<< 1;
|
|
s->noise_table = noisetable_exp;
|
|
}
|
|
#if 0
|
|
{
|
|
unsigned int seed;
|
|
fixed32 norm;
|
|
seed = 1;
|
|
norm = 0; // PJJ: near as makes any diff to 0!
|
|
for (i=0;i<NOISE_TAB_SIZE;++i)
|
|
{
|
|
seed = seed * 314159 + 1;
|
|
s->noise_table[i] = itofix32((int)seed) * norm;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
s->hgain_vlc.table = vlcbuf4;
|
|
s->hgain_vlc.table_allocated = VLCBUF4SIZE;
|
|
init_vlc(&s->hgain_vlc, HGAINVLCBITS, sizeof(hgain_huffbits),
|
|
hgain_huffbits, 1, 1,
|
|
hgain_huffcodes, 2, 2, 0);
|
|
}
|
|
|
|
if (s->use_exp_vlc)
|
|
{
|
|
|
|
s->exp_vlc.table = vlcbuf3;
|
|
s->exp_vlc.table_allocated = VLCBUF3SIZE;
|
|
|
|
init_vlc(&s->exp_vlc, EXPVLCBITS, sizeof(scale_huffbits),
|
|
scale_huffbits, 1, 1,
|
|
scale_huffcodes, 4, 4, 0);
|
|
}
|
|
else
|
|
{
|
|
wma_lsp_to_curve_init(s, s->frame_len);
|
|
}
|
|
|
|
/* choose the VLC tables for the coefficients */
|
|
coef_vlc_table = 2;
|
|
if (s->sample_rate >= 32000)
|
|
{
|
|
if (bps1 < 0xb852)
|
|
coef_vlc_table = 0;
|
|
else if (bps1 < 0x128f6)
|
|
coef_vlc_table = 1;
|
|
}
|
|
|
|
runtabarray[0] = runtab0; runtabarray[1] = runtab1;
|
|
levtabarray[0] = levtab0; levtabarray[1] = levtab1;
|
|
|
|
s->coef_vlc[0].table = vlcbuf1;
|
|
s->coef_vlc[0].table_allocated = VLCBUF1SIZE;
|
|
s->coef_vlc[1].table = vlcbuf2;
|
|
s->coef_vlc[1].table_allocated = VLCBUF2SIZE;
|
|
|
|
|
|
init_coef_vlc(&s->coef_vlc[0], &s->run_table[0], &s->level_table[0],
|
|
&coef_vlcs[coef_vlc_table * 2], 0);
|
|
init_coef_vlc(&s->coef_vlc[1], &s->run_table[1], &s->level_table[1],
|
|
&coef_vlcs[coef_vlc_table * 2 + 1], 1);
|
|
|
|
s->last_superframe_len = 0;
|
|
s->last_bitoffset = 0;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/* compute x^-0.25 with an exponent and mantissa table. We use linear
|
|
interpolation to reduce the mantissa table size at a small speed
|
|
expense (linear interpolation approximately doubles the number of
|
|
bits of precision). */
|
|
static inline fixed32 pow_m1_4(WMADecodeContext *s, fixed32 x)
|
|
{
|
|
union {
|
|
float f;
|
|
unsigned int v;
|
|
} u, t;
|
|
unsigned int e, m;
|
|
fixed32 a, b;
|
|
|
|
u.f = fixtof64(x);
|
|
e = u.v >> 23;
|
|
m = (u.v >> (23 - LSP_POW_BITS)) & ((1 << LSP_POW_BITS) - 1);
|
|
/* build interpolation scale: 1 <= t < 2. */
|
|
t.v = ((u.v << LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23);
|
|
a = s->lsp_pow_m_table1[m];
|
|
b = s->lsp_pow_m_table2[m];
|
|
|
|
/* lsp_pow_e_table contains 32.32 format */
|
|
/* TODO: Since we're unlikely have value that cover the whole
|
|
* IEEE754 range, we probably don't need to have all possible exponents */
|
|
|
|
return (lsp_pow_e_table[e] * (a + fixmul32(b, ftofix32(t.f))) >>32);
|
|
}
|
|
|
|
static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len)
|
|
{
|
|
fixed32 wdel, a, b, temp, temp2;
|
|
int i, m;
|
|
|
|
wdel = fixdiv32(M_PI_F, itofix32(frame_len));
|
|
temp = fixdiv32(itofix32(1), itofix32(frame_len));
|
|
for (i=0; i<frame_len; ++i)
|
|
{
|
|
/* TODO: can probably reuse the trig_init values here */
|
|
fsincos((temp*i)<<15, &temp2);
|
|
/* get 3 bits headroom + 1 bit from not doubleing the values */
|
|
s->lsp_cos_table[i] = temp2>>3;
|
|
|
|
}
|
|
/* NOTE: these two tables are needed to avoid two operations in
|
|
pow_m1_4 */
|
|
b = itofix32(1);
|
|
int ix = 0;
|
|
|
|
/*double check this later*/
|
|
for(i=(1 << LSP_POW_BITS) - 1;i>=0;i--)
|
|
{
|
|
m = (1 << LSP_POW_BITS) + i;
|
|
a = pow_a_table[ix++]<<4;
|
|
s->lsp_pow_m_table1[i] = 2 * a - b;
|
|
s->lsp_pow_m_table2[i] = b - a;
|
|
b = a;
|
|
}
|
|
|
|
}
|
|
|
|
/* NOTE: We use the same code as Vorbis here */
|
|
/* XXX: optimize it further with SSE/3Dnow */
|
|
static void wma_lsp_to_curve(WMADecodeContext *s,
|
|
fixed32 *out,
|
|
fixed32 *val_max_ptr,
|
|
int n,
|
|
fixed32 *lsp)
|
|
{
|
|
int i, j;
|
|
fixed32 p, q, w, v, val_max, temp, temp2;
|
|
|
|
val_max = 0;
|
|
for(i=0;i<n;++i)
|
|
{
|
|
/* shift by 2 now to reduce rounding error,
|
|
* we can renormalize right before pow_m1_4
|
|
*/
|
|
|
|
p = 0x8000<<5;
|
|
q = 0x8000<<5;
|
|
w = s->lsp_cos_table[i];
|
|
|
|
for (j=1;j<NB_LSP_COEFS;j+=2)
|
|
{
|
|
/* w is 5.27 format, lsp is in 16.16, temp2 becomes 5.27 format */
|
|
temp2 = ((w - (lsp[j - 1]<<11)));
|
|
temp = q;
|
|
|
|
/* q is 16.16 format, temp2 is 5.27, q becomes 16.16 */
|
|
q = fixmul32b(q, temp2 )<<4;
|
|
p = fixmul32b(p, (w - (lsp[j]<<11)))<<4;
|
|
}
|
|
|
|
/* 2 in 5.27 format is 0x10000000 */
|
|
p = fixmul32(p, fixmul32b(p, (0x10000000 - w)))<<3;
|
|
q = fixmul32(q, fixmul32b(q, (0x10000000 + w)))<<3;
|
|
|
|
v = (p + q) >>9; /* p/q end up as 16.16 */
|
|
v = pow_m1_4(s, v);
|
|
if (v > val_max)
|
|
val_max = v;
|
|
out[i] = v;
|
|
}
|
|
|
|
*val_max_ptr = val_max;
|
|
}
|
|
|
|
/* decode exponents coded with LSP coefficients (same idea as Vorbis) */
|
|
static void decode_exp_lsp(WMADecodeContext *s, int ch)
|
|
{
|
|
fixed32 lsp_coefs[NB_LSP_COEFS];
|
|
int val, i;
|
|
|
|
for (i = 0; i < NB_LSP_COEFS; ++i)
|
|
{
|
|
if (i == 0 || i >= 8)
|
|
val = get_bits(&s->gb, 3);
|
|
else
|
|
val = get_bits(&s->gb, 4);
|
|
lsp_coefs[i] = lsp_codebook[i][val];
|
|
}
|
|
|
|
wma_lsp_to_curve(s,
|
|
s->exponents[ch],
|
|
&s->max_exponent[ch],
|
|
s->block_len,
|
|
lsp_coefs);
|
|
}
|
|
|
|
/* decode exponents coded with VLC codes */
|
|
static int decode_exp_vlc(WMADecodeContext *s, int ch)
|
|
{
|
|
int last_exp, n, code;
|
|
const uint16_t *ptr, *band_ptr;
|
|
fixed32 v, max_scale;
|
|
fixed32 *q,*q_end;
|
|
|
|
/*accommodate the 60 negative indices */
|
|
const fixed32 *pow_10_to_yover16_ptr = &pow_10_to_yover16[61];
|
|
|
|
band_ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
|
|
ptr = band_ptr;
|
|
q = s->exponents[ch];
|
|
q_end = q + s->block_len;
|
|
max_scale = 0;
|
|
|
|
|
|
if (s->version == 1) //wmav1 only
|
|
{
|
|
last_exp = get_bits(&s->gb, 5) + 10;
|
|
/* XXX: use a table */
|
|
v = pow_10_to_yover16_ptr[last_exp];
|
|
max_scale = v;
|
|
n = *ptr++;
|
|
do
|
|
{
|
|
*q++ = v;
|
|
}
|
|
while (--n);
|
|
} else {
|
|
last_exp = 36;
|
|
}
|
|
|
|
while (q < q_end)
|
|
{
|
|
code = get_vlc2(&s->gb, s->exp_vlc.table, EXPVLCBITS, EXPMAX);
|
|
if (code < 0)
|
|
{
|
|
return -1;
|
|
}
|
|
/* NOTE: this offset is the same as MPEG4 AAC ! */
|
|
last_exp += code - 60;
|
|
/* XXX: use a table */
|
|
v = pow_10_to_yover16_ptr[last_exp];
|
|
if (v > max_scale)
|
|
{
|
|
max_scale = v;
|
|
}
|
|
n = *ptr++;
|
|
do
|
|
{
|
|
*q++ = v;
|
|
|
|
}
|
|
while (--n);
|
|
}
|
|
|
|
s->max_exponent[ch] = max_scale;
|
|
return 0;
|
|
}
|
|
|
|
/* return 0 if OK. return 1 if last block of frame. return -1 if
|
|
unrecorrable error. */
|
|
static int wma_decode_block(WMADecodeContext *s, int32_t *scratch_buffer)
|
|
{
|
|
int n, v, a, ch, code, bsize;
|
|
int coef_nb_bits, total_gain;
|
|
int nb_coefs[MAX_CHANNELS];
|
|
fixed32 mdct_norm;
|
|
|
|
/*DEBUGF("***decode_block: %d (%d samples of %d in frame)\n", s->block_num, s->block_len, s->frame_len);*/
|
|
|
|
/* compute current block length */
|
|
if (s->use_variable_block_len)
|
|
{
|
|
n = av_log2(s->nb_block_sizes - 1) + 1;
|
|
|
|
if (s->reset_block_lengths)
|
|
{
|
|
s->reset_block_lengths = 0;
|
|
v = get_bits(&s->gb, n);
|
|
if (v >= s->nb_block_sizes)
|
|
{
|
|
return -2;
|
|
}
|
|
s->prev_block_len_bits = s->frame_len_bits - v;
|
|
v = get_bits(&s->gb, n);
|
|
if (v >= s->nb_block_sizes)
|
|
{
|
|
return -3;
|
|
}
|
|
s->block_len_bits = s->frame_len_bits - v;
|
|
}
|
|
else
|
|
{
|
|
/* update block lengths */
|
|
s->prev_block_len_bits = s->block_len_bits;
|
|
s->block_len_bits = s->next_block_len_bits;
|
|
}
|
|
v = get_bits(&s->gb, n);
|
|
|
|
if (v >= s->nb_block_sizes)
|
|
{
|
|
// rb->splash(HZ*4, "v was %d", v); //5, 7
|
|
return -4; //this is it
|
|
}
|
|
else{
|
|
//rb->splash(HZ, "passed v block (%d)!", v);
|
|
}
|
|
s->next_block_len_bits = s->frame_len_bits - v;
|
|
}
|
|
else
|
|
{
|
|
/* fixed block len */
|
|
s->next_block_len_bits = s->frame_len_bits;
|
|
s->prev_block_len_bits = s->frame_len_bits;
|
|
s->block_len_bits = s->frame_len_bits;
|
|
}
|
|
/* now check if the block length is coherent with the frame length */
|
|
s->block_len = 1 << s->block_len_bits;
|
|
|
|
if ((s->block_pos + s->block_len) > s->frame_len)
|
|
{
|
|
return -5; //oddly 32k sample from tracker fails here
|
|
}
|
|
|
|
if (s->nb_channels == 2)
|
|
{
|
|
s->ms_stereo = get_bits1(&s->gb);
|
|
}
|
|
v = 0;
|
|
for (ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
a = get_bits1(&s->gb);
|
|
s->channel_coded[ch] = a;
|
|
v |= a;
|
|
}
|
|
/* if no channel coded, no need to go further */
|
|
/* XXX: fix potential framing problems */
|
|
if (!v)
|
|
{
|
|
goto next;
|
|
}
|
|
|
|
bsize = s->frame_len_bits - s->block_len_bits;
|
|
|
|
/* read total gain and extract corresponding number of bits for
|
|
coef escape coding */
|
|
total_gain = 1;
|
|
for(;;)
|
|
{
|
|
a = get_bits(&s->gb, 7);
|
|
total_gain += a;
|
|
if (a != 127)
|
|
{
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (total_gain < 15)
|
|
coef_nb_bits = 13;
|
|
else if (total_gain < 32)
|
|
coef_nb_bits = 12;
|
|
else if (total_gain < 40)
|
|
coef_nb_bits = 11;
|
|
else if (total_gain < 45)
|
|
coef_nb_bits = 10;
|
|
else
|
|
coef_nb_bits = 9;
|
|
|
|
/* compute number of coefficients */
|
|
n = s->coefs_end[bsize] - s->coefs_start;
|
|
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
nb_coefs[ch] = n;
|
|
}
|
|
/* complex coding */
|
|
if (s->use_noise_coding)
|
|
{
|
|
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
if (s->channel_coded[ch])
|
|
{
|
|
int i, n, a;
|
|
n = s->exponent_high_sizes[bsize];
|
|
for(i=0;i<n;++i)
|
|
{
|
|
a = get_bits1(&s->gb);
|
|
s->high_band_coded[ch][i] = a;
|
|
/* if noise coding, the coefficients are not transmitted */
|
|
if (a)
|
|
nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
|
|
}
|
|
}
|
|
}
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
if (s->channel_coded[ch])
|
|
{
|
|
int i, n, val, code;
|
|
|
|
n = s->exponent_high_sizes[bsize];
|
|
val = (int)0x80000000;
|
|
for(i=0;i<n;++i)
|
|
{
|
|
if (s->high_band_coded[ch][i])
|
|
{
|
|
if (val == (int)0x80000000)
|
|
{
|
|
val = get_bits(&s->gb, 7) - 19;
|
|
}
|
|
else
|
|
{
|
|
//code = get_vlc(&s->gb, &s->hgain_vlc);
|
|
code = get_vlc2(&s->gb, s->hgain_vlc.table, HGAINVLCBITS, HGAINMAX);
|
|
if (code < 0)
|
|
{
|
|
return -6;
|
|
}
|
|
val += code - 18;
|
|
}
|
|
s->high_band_values[ch][i] = val;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* exponents can be reused in short blocks. */
|
|
if ((s->block_len_bits == s->frame_len_bits) || get_bits1(&s->gb))
|
|
{
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
if (s->channel_coded[ch])
|
|
{
|
|
if (s->use_exp_vlc)
|
|
{
|
|
if (decode_exp_vlc(s, ch) < 0)
|
|
{
|
|
return -7;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
decode_exp_lsp(s, ch);
|
|
}
|
|
s->exponents_bsize[ch] = bsize;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* parse spectral coefficients : just RLE encoding */
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
if (s->channel_coded[ch])
|
|
{
|
|
VLC *coef_vlc;
|
|
int level, run, sign, tindex;
|
|
int16_t *ptr, *eptr;
|
|
const int16_t *level_table, *run_table;
|
|
|
|
/* special VLC tables are used for ms stereo because
|
|
there is potentially less energy there */
|
|
tindex = (ch == 1 && s->ms_stereo);
|
|
coef_vlc = &s->coef_vlc[tindex];
|
|
run_table = s->run_table[tindex];
|
|
level_table = s->level_table[tindex];
|
|
/* XXX: optimize */
|
|
ptr = &s->coefs1[ch][0];
|
|
eptr = ptr + nb_coefs[ch];
|
|
memset(ptr, 0, s->block_len * sizeof(int16_t));
|
|
|
|
for(;;)
|
|
{
|
|
code = get_vlc2(&s->gb, coef_vlc->table, VLCBITS, VLCMAX);
|
|
//code = get_vlc(&s->gb, coef_vlc);
|
|
if (code < 0)
|
|
{
|
|
return -8;
|
|
}
|
|
if (code == 1)
|
|
{
|
|
/* EOB */
|
|
break;
|
|
}
|
|
else if (code == 0)
|
|
{
|
|
/* escape */
|
|
level = get_bits(&s->gb, coef_nb_bits);
|
|
/* NOTE: this is rather suboptimal. reading
|
|
block_len_bits would be better */
|
|
run = get_bits(&s->gb, s->frame_len_bits);
|
|
}
|
|
else
|
|
{
|
|
/* normal code */
|
|
run = run_table[code];
|
|
level = level_table[code];
|
|
}
|
|
sign = get_bits1(&s->gb);
|
|
if (!sign)
|
|
level = -level;
|
|
ptr += run;
|
|
if (ptr >= eptr)
|
|
{
|
|
break;
|
|
}
|
|
*ptr++ = level;
|
|
|
|
|
|
/* NOTE: EOB can be omitted */
|
|
if (ptr >= eptr)
|
|
break;
|
|
}
|
|
}
|
|
if (s->version == 1 && s->nb_channels >= 2)
|
|
{
|
|
align_get_bits(&s->gb);
|
|
}
|
|
}
|
|
|
|
{
|
|
int n4 = s->block_len >> 1;
|
|
|
|
|
|
mdct_norm = 0x10000>>(s->block_len_bits-1);
|
|
|
|
if (s->version == 1)
|
|
{
|
|
mdct_norm *= fixtoi32(fixsqrt32(itofix32(n4)));
|
|
}
|
|
}
|
|
|
|
|
|
/* finally compute the MDCT coefficients */
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
if (s->channel_coded[ch])
|
|
{
|
|
int16_t *coefs1;
|
|
fixed32 *exponents;
|
|
fixed32 *coefs, atemp;
|
|
fixed64 mult;
|
|
fixed64 mult1;
|
|
fixed32 noise, temp1, temp2, mult2;
|
|
int i, j, n, n1, last_high_band, esize;
|
|
fixed32 exp_power[HIGH_BAND_MAX_SIZE];
|
|
|
|
//total_gain, coefs1, mdctnorm are lossless
|
|
|
|
coefs1 = s->coefs1[ch];
|
|
exponents = s->exponents[ch];
|
|
esize = s->exponents_bsize[ch];
|
|
coefs = (*(s->coefs))[ch];
|
|
n=0;
|
|
|
|
/*
|
|
* The calculation of coefs has a shift right by 2 built in. This
|
|
* prepares samples for the Tremor IMDCT which uses a slightly
|
|
* different fixed format then the ffmpeg one. If the old ffmpeg
|
|
* imdct is used, each shift storing into coefs should be reduced
|
|
* by 1.
|
|
* See SVN logs for details.
|
|
*/
|
|
|
|
|
|
if (s->use_noise_coding)
|
|
{
|
|
/*TODO: mult should be converted to 32 bit to speed up noise coding*/
|
|
|
|
mult = fixdiv64(pow_table[total_gain+20],Fixed32To64(s->max_exponent[ch]));
|
|
mult = mult* mdct_norm;
|
|
mult1 = mult;
|
|
|
|
/* very low freqs : noise */
|
|
for(i = 0;i < s->coefs_start; ++i)
|
|
{
|
|
*coefs++ = fixmul32( (fixmul32(s->noise_table[s->noise_index],
|
|
exponents[i<<bsize>>esize])>>4),Fixed32From64(mult1)) >>2;
|
|
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
|
|
}
|
|
|
|
n1 = s->exponent_high_sizes[bsize];
|
|
|
|
/* compute power of high bands */
|
|
exponents = s->exponents[ch] +(s->high_band_start[bsize]<<bsize);
|
|
last_high_band = 0; /* avoid warning */
|
|
for (j=0;j<n1;++j)
|
|
{
|
|
n = s->exponent_high_bands[s->frame_len_bits -
|
|
s->block_len_bits][j];
|
|
if (s->high_band_coded[ch][j])
|
|
{
|
|
fixed32 e2, v;
|
|
e2 = 0;
|
|
for(i = 0;i < n; ++i)
|
|
{
|
|
/*v is normalized later on so its fixed format is irrelevant*/
|
|
v = exponents[i<<bsize>>esize]>>4;
|
|
e2 += fixmul32(v, v)>>3;
|
|
}
|
|
exp_power[j] = e2/n; /*n is an int...*/
|
|
last_high_band = j;
|
|
}
|
|
exponents += n<<bsize;
|
|
}
|
|
|
|
/* main freqs and high freqs */
|
|
exponents = s->exponents[ch] + (s->coefs_start<<bsize);
|
|
for(j=-1;j<n1;++j)
|
|
{
|
|
if (j < 0)
|
|
{
|
|
n = s->high_band_start[bsize] -
|
|
s->coefs_start;
|
|
}
|
|
else
|
|
{
|
|
n = s->exponent_high_bands[s->frame_len_bits -
|
|
s->block_len_bits][j];
|
|
}
|
|
if (j >= 0 && s->high_band_coded[ch][j])
|
|
{
|
|
/* use noise with specified power */
|
|
fixed32 tmp = fixdiv32(exp_power[j],exp_power[last_high_band]);
|
|
|
|
/*mult1 is 48.16, pow_table is 48.16*/
|
|
mult1 = fixmul32(fixsqrt32(tmp),
|
|
pow_table[s->high_band_values[ch][j]+20]) >> 16;
|
|
|
|
/*this step has a fairly high degree of error for some reason*/
|
|
mult1 = fixdiv64(mult1,fixmul32(s->max_exponent[ch],s->noise_mult));
|
|
mult1 = mult1*mdct_norm>>PRECISION;
|
|
for(i = 0;i < n; ++i)
|
|
{
|
|
noise = s->noise_table[s->noise_index];
|
|
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
|
|
*coefs++ = fixmul32((fixmul32(exponents[i<<bsize>>esize],noise)>>4),
|
|
Fixed32From64(mult1)) >>2;
|
|
|
|
}
|
|
exponents += n<<bsize;
|
|
}
|
|
else
|
|
{
|
|
/* coded values + small noise */
|
|
for(i = 0;i < n; ++i)
|
|
{
|
|
noise = s->noise_table[s->noise_index];
|
|
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
|
|
|
|
/*don't forget to renormalize the noise*/
|
|
temp1 = (((int32_t)*coefs1++)<<16) + (noise>>4);
|
|
temp2 = fixmul32(exponents[i<<bsize>>esize], mult>>18);
|
|
*coefs++ = fixmul32(temp1, temp2);
|
|
}
|
|
exponents += n<<bsize;
|
|
}
|
|
}
|
|
|
|
/* very high freqs : noise */
|
|
n = s->block_len - s->coefs_end[bsize];
|
|
mult2 = fixmul32(mult>>16,exponents[((-1<<bsize))>>esize]) ;
|
|
for (i = 0; i < n; ++i)
|
|
{
|
|
/*renormalize the noise product and then reduce to 14.18 precison*/
|
|
*coefs++ = fixmul32(s->noise_table[s->noise_index],mult2) >>6;
|
|
|
|
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/*Noise coding not used, simply convert from exp to fixed representation*/
|
|
|
|
fixed32 mult3 = (fixed32)(fixdiv64(pow_table[total_gain+20],
|
|
Fixed32To64(s->max_exponent[ch])));
|
|
mult3 = fixmul32(mult3, mdct_norm);
|
|
|
|
/*zero the first 3 coefficients for WMA V1, does nothing otherwise*/
|
|
for(i=0; i<s->coefs_start; i++)
|
|
*coefs++=0;
|
|
|
|
n = nb_coefs[ch];
|
|
|
|
/* XXX: optimize more, unrolling this loop in asm
|
|
might be a good idea */
|
|
|
|
for(i = 0;i < n; ++i)
|
|
{
|
|
/*ffmpeg imdct needs 15.17, while tremor 14.18*/
|
|
atemp = (coefs1[i] * mult3)>>2;
|
|
*coefs++=fixmul32(atemp,exponents[i<<bsize>>esize]);
|
|
}
|
|
n = s->block_len - s->coefs_end[bsize];
|
|
memset(coefs, 0, n*sizeof(fixed32));
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
|
|
if (s->ms_stereo && s->channel_coded[1])
|
|
{
|
|
fixed32 a, b;
|
|
int i;
|
|
fixed32 (*coefs)[MAX_CHANNELS][BLOCK_MAX_SIZE] = (s->coefs);
|
|
|
|
/* nominal case for ms stereo: we do it before mdct */
|
|
/* no need to optimize this case because it should almost
|
|
never happen */
|
|
if (!s->channel_coded[0])
|
|
{
|
|
memset((*(s->coefs))[0], 0, sizeof(fixed32) * s->block_len);
|
|
s->channel_coded[0] = 1;
|
|
}
|
|
|
|
for(i = 0; i < s->block_len; ++i)
|
|
{
|
|
a = (*coefs)[0][i];
|
|
b = (*coefs)[1][i];
|
|
(*coefs)[0][i] = a + b;
|
|
(*coefs)[1][i] = a - b;
|
|
}
|
|
}
|
|
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
if (s->channel_coded[ch])
|
|
{
|
|
int n4, index;
|
|
|
|
n4 = s->block_len >>1;
|
|
|
|
/*faster IMDCT from Vorbis*/
|
|
mdct_backward( (1 << (s->block_len_bits+1)), (int32_t*)(*(s->coefs))[ch], (int32_t*)scratch_buffer);
|
|
|
|
/*slower but more easily understood IMDCT from FFMPEG*/
|
|
//ff_imdct_calc(&s->mdct_ctx[bsize],
|
|
// output,
|
|
// (*(s->coefs))[ch]);
|
|
|
|
|
|
/* add in the frame */
|
|
index = (s->frame_len / 2) + s->block_pos - n4;
|
|
wma_window(s, scratch_buffer, &((*s->frame_out)[ch][index]));
|
|
|
|
|
|
|
|
/* specific fast case for ms-stereo : add to second
|
|
channel if it is not coded */
|
|
if (s->ms_stereo && !s->channel_coded[1])
|
|
{
|
|
wma_window(s, scratch_buffer, &((*s->frame_out)[1][index]));
|
|
}
|
|
}
|
|
}
|
|
next:
|
|
/* update block number */
|
|
++s->block_num;
|
|
s->block_pos += s->block_len;
|
|
if (s->block_pos >= s->frame_len)
|
|
{
|
|
return 1;
|
|
}
|
|
else
|
|
{
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
/* decode a frame of frame_len samples */
|
|
static int wma_decode_frame(WMADecodeContext *s, int32_t *samples)
|
|
{
|
|
int ret, i, n, ch, incr;
|
|
int32_t *ptr;
|
|
fixed32 *iptr;
|
|
|
|
/* read each block */
|
|
s->block_num = 0;
|
|
s->block_pos = 0;
|
|
|
|
|
|
for(;;)
|
|
{
|
|
ret = wma_decode_block(s, samples);
|
|
if (ret < 0)
|
|
{
|
|
|
|
DEBUGF("wma_decode_block failed with code %d\n", ret);
|
|
return -1;
|
|
}
|
|
if (ret)
|
|
{
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* return frame with full 30-bit precision */
|
|
n = s->frame_len;
|
|
incr = s->nb_channels;
|
|
for(ch = 0; ch < s->nb_channels; ++ch)
|
|
{
|
|
ptr = samples + ch;
|
|
iptr = &((*s->frame_out)[ch][0]);
|
|
|
|
for (i=0;i<n;++i)
|
|
{
|
|
*ptr = (*iptr++);
|
|
ptr += incr;
|
|
}
|
|
|
|
memmove(&((*s->frame_out)[ch][0]), &((*s->frame_out)[ch][s->frame_len]),
|
|
s->frame_len * sizeof(fixed32));
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Initialise the superframe decoding */
|
|
|
|
int wma_decode_superframe_init(WMADecodeContext* s,
|
|
const uint8_t *buf, /*input*/
|
|
int buf_size)
|
|
{
|
|
if (buf_size==0)
|
|
{
|
|
s->last_superframe_len = 0;
|
|
return 0;
|
|
}
|
|
|
|
s->current_frame = 0;
|
|
|
|
init_get_bits(&s->gb, buf, buf_size*8);
|
|
|
|
if (s->use_bit_reservoir)
|
|
{
|
|
/* read super frame header */
|
|
skip_bits(&s->gb, 4); /* super frame index */
|
|
s->nb_frames = get_bits(&s->gb, 4);
|
|
|
|
if (s->last_superframe_len == 0)
|
|
s->nb_frames --;
|
|
else if (s->nb_frames == 0)
|
|
s->nb_frames++;
|
|
|
|
s->bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3);
|
|
} else {
|
|
s->nb_frames = 1;
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
|
|
/* Decode a single frame in the current superframe - return -1 if
|
|
there was a decoding error, or the number of samples decoded.
|
|
*/
|
|
|
|
int wma_decode_superframe_frame(WMADecodeContext* s,
|
|
int32_t* samples, /*output*/
|
|
const uint8_t *buf, /*input*/
|
|
int buf_size)
|
|
{
|
|
int pos, len;
|
|
uint8_t *q;
|
|
int done = 0;
|
|
if ((s->use_bit_reservoir) && (s->current_frame == 0))
|
|
{
|
|
if (s->last_superframe_len > 0)
|
|
{
|
|
/* add s->bit_offset bits to last frame */
|
|
if ((s->last_superframe_len + ((s->bit_offset + 7) >> 3)) >
|
|
MAX_CODED_SUPERFRAME_SIZE)
|
|
{
|
|
DEBUGF("superframe size too large error\n");
|
|
goto fail;
|
|
}
|
|
q = s->last_superframe + s->last_superframe_len;
|
|
len = s->bit_offset;
|
|
while (len > 7)
|
|
{
|
|
*q++ = (get_bits)(&s->gb, 8);
|
|
len -= 8;
|
|
}
|
|
if (len > 0)
|
|
{
|
|
*q++ = (get_bits)(&s->gb, len) << (8 - len);
|
|
}
|
|
|
|
/* XXX: s->bit_offset bits into last frame */
|
|
init_get_bits(&s->gb, s->last_superframe, MAX_CODED_SUPERFRAME_SIZE*8);
|
|
/* skip unused bits */
|
|
if (s->last_bitoffset > 0)
|
|
skip_bits(&s->gb, s->last_bitoffset);
|
|
|
|
/* this frame is stored in the last superframe and in the
|
|
current one */
|
|
if (wma_decode_frame(s, samples) < 0)
|
|
{
|
|
goto fail;
|
|
}
|
|
done = 1;
|
|
}
|
|
|
|
/* read each frame starting from s->bit_offset */
|
|
pos = s->bit_offset + 4 + 4 + s->byte_offset_bits + 3;
|
|
init_get_bits(&s->gb, buf + (pos >> 3), (MAX_CODED_SUPERFRAME_SIZE - (pos >> 3))*8);
|
|
len = pos & 7;
|
|
if (len > 0)
|
|
skip_bits(&s->gb, len);
|
|
|
|
s->reset_block_lengths = 1;
|
|
}
|
|
|
|
/* If we haven't decoded a frame yet, do it now */
|
|
if (!done)
|
|
{
|
|
if (wma_decode_frame(s, samples) < 0)
|
|
{
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
s->current_frame++;
|
|
|
|
if ((s->use_bit_reservoir) && (s->current_frame == s->nb_frames))
|
|
{
|
|
/* we copy the end of the frame in the last frame buffer */
|
|
pos = get_bits_count(&s->gb) + ((s->bit_offset + 4 + 4 + s->byte_offset_bits + 3) & ~7);
|
|
s->last_bitoffset = pos & 7;
|
|
pos >>= 3;
|
|
len = buf_size - pos;
|
|
if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0)
|
|
{
|
|
DEBUGF("superframe size too large error after decoding\n");
|
|
goto fail;
|
|
}
|
|
s->last_superframe_len = len;
|
|
memcpy(s->last_superframe, buf + pos, len);
|
|
}
|
|
|
|
return s->frame_len;
|
|
|
|
fail:
|
|
/* when error, we reset the bit reservoir */
|
|
|
|
s->last_superframe_len = 0;
|
|
return -1;
|
|
}
|
|
|