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Patch by Mohamed Tarek from FS#10182: 1) Move the main() test program from cook.c to a new main.c; 2) Move some common definitions from cook.c to cook.h. No functional changes.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@20898 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Dave Chapman 2009-05-09 23:24:02 +00:00
parent 868652abaa
commit 7ba6ef42de
4 changed files with 241 additions and 182 deletions
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2
apps/codecs/libcook/Makefile.test
View file

@ -1,5 +1,5 @@
CFLAGS = -Wall -O3 CFLAGS = -Wall -O3
OBJS = bitstream.o cook.o fft.o libavutil/log.o mdct.o libavutil/mem.o libavutil/lfg.o libavutil/md5.o rm2wav.o OBJS = main.o bitstream.o cook.o fft.o libavutil/log.o mdct.o libavutil/mem.o libavutil/lfg.o libavutil/md5.o rm2wav.o
cooktest: $(OBJS) cooktest: $(OBJS)
gcc -o cooktest $(OBJS) -lm gcc -o cooktest $(OBJS) -lm

185
apps/codecs/libcook/cook.c
View file

@ -45,18 +45,9 @@
#include <math.h> #include <math.h>
#include <stddef.h> #include <stddef.h>
#include <stdio.h> #include <stdio.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <unistd.h>
#include "libavutil/lfg.h"
#include "bitstream.h"
#include "dsputil.h"
#include "bytestream.h"
#include "cook.h"
#include "cookdata.h" #include "cookdata.h"
#include "rm2wav.h"
/* The following table is taken from libavutil/mathematics.c */ /* The following table is taken from libavutil/mathematics.c */
const uint8_t ff_log2_tab[256]={ const uint8_t ff_log2_tab[256]={
@ -82,90 +73,6 @@ const uint8_t ff_log2_tab[256]={
//#define DUMP_RAW_FRAMES //#define DUMP_RAW_FRAMES
#define DEBUGF(message,args ...) av_log(NULL,AV_LOG_ERROR,message,## args) #define DEBUGF(message,args ...) av_log(NULL,AV_LOG_ERROR,message,## args)
typedef struct {
int *now;
int *previous;
} cook_gains;
typedef struct cook {
/*
* The following 5 functions provide the lowlevel arithmetic on
* the internal audio buffers.
*/
void (* scalar_dequant)(struct cook *q, int index, int quant_index,
int* subband_coef_index, int* subband_coef_sign,
float* mlt_p);
void (* decouple) (struct cook *q,
int subband,
float f1, float f2,
float *decode_buffer,
float *mlt_buffer1, float *mlt_buffer2);
void (* imlt_window) (struct cook *q, float *buffer1,
cook_gains *gains_ptr, float *previous_buffer);
void (* interpolate) (struct cook *q, float* buffer,
int gain_index, int gain_index_next);
void (* saturate_output) (struct cook *q, int chan, int16_t *out);
GetBitContext gb;
int frame_number;
int block_align;
int extradata_size;
/* stream data */
int nb_channels;
int joint_stereo;
int bit_rate;
int sample_rate;
int samples_per_channel;
int samples_per_frame;
int subbands;
int log2_numvector_size;
int numvector_size; //1 << log2_numvector_size;
int js_subband_start;
int total_subbands;
int num_vectors;
int bits_per_subpacket;
int cookversion;
/* states */
AVLFG random_state;
/* transform data */
MDCTContext mdct_ctx;
float* mlt_window;
/* gain buffers */
cook_gains gains1;
cook_gains gains2;
int gain_1[9];
int gain_2[9];
int gain_3[9];
int gain_4[9];
/* VLC data */
int js_vlc_bits;
VLC envelope_quant_index[13];
VLC sqvh[7]; //scalar quantization
VLC ccpl; //channel coupling
/* generatable tables and related variables */
int gain_size_factor;
float gain_table[23];
/* data buffers */
uint8_t* decoded_bytes_buffer;
float mono_mdct_output[2048] __attribute__ ((aligned(16))); //DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
float decode_buffer_1[1024];
float decode_buffer_2[1024];
float decode_buffer_0[1060]; /* static allocation for joint decode */
const float *cplscales[5];
} COOKContext;
static float pow2tab[127]; static float pow2tab[127];
static float rootpow2tab[127]; static float rootpow2tab[127];
@ -335,7 +242,7 @@ static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes)
* Cook uninit * Cook uninit
*/ */
static av_cold int cook_decode_close(COOKContext *q) av_cold int cook_decode_close(COOKContext *q)
{ {
int i; int i;
//COOKContext *q = avctx->priv_data; //COOKContext *q = avctx->priv_data;
@ -997,7 +904,7 @@ static int decode_subpacket(COOKContext *q, const uint8_t *inbuffer,
* @param rmctx pointer to the RMContext * @param rmctx pointer to the RMContext
*/ */
static int cook_decode_frame(RMContext *rmctx,COOKContext *q, int cook_decode_frame(RMContext *rmctx,COOKContext *q,
int16_t *outbuffer, int *data_size, int16_t *outbuffer, int *data_size,
const uint8_t *inbuffer, int buf_size) { const uint8_t *inbuffer, int buf_size) {
//COOKContext *q = avctx->priv_data; //COOKContext *q = avctx->priv_data;
@ -1053,7 +960,7 @@ static av_cold int cook_count_channels(unsigned int mask){
* Cook initialization * Cook initialization
*/ */
static av_cold int cook_decode_init(RMContext *rmctx, COOKContext *q) av_cold int cook_decode_init(RMContext *rmctx, COOKContext *q)
{ {
/* cook extradata */ /* cook extradata */
q->cookversion = rmctx->cook_version; q->cookversion = rmctx->cook_version;
@ -1203,87 +1110,3 @@ static av_cold int cook_decode_init(RMContext *rmctx, COOKContext *q)
return 0; return 0;
} }
int main(int argc, char *argv[])
{
int fd, fd_dec;
int res, datasize,x,i;
int nb_frames = 0;
#ifdef DUMP_RAW_FRAMES
char filename[15];
int fd_out;
#endif
int16_t outbuf[2048];
uint8_t inbuf[1024];
uint16_t fs,sps,h;
uint32_t packet_count;
COOKContext q;
RMContext rmctx;
RMPacket pkt;
memset(&q,0,sizeof(COOKContext));
memset(&rmctx,0,sizeof(RMContext));
memset(&pkt,0,sizeof(RMPacket));
if (argc != 2) {
av_log(NULL,AV_LOG_ERROR,"Incorrect number of arguments\n");
return -1;
}
fd = open(argv[1],O_RDONLY);
if (fd < 0) {
av_log(NULL,AV_LOG_ERROR,"Error opening file %s\n", argv[1]);
return -1;
}
fd_dec = open_wav("output.wav");
if (fd_dec < 0) {
av_log(NULL,AV_LOG_ERROR,"Error creating output file\n");
return -1;
}
res = real_parse_header(fd, &rmctx);
packet_count = rmctx.nb_packets;
rmctx.audio_framesize = rmctx.block_align;
rmctx.block_align = rmctx.sub_packet_size;
fs = rmctx.audio_framesize;
sps= rmctx.block_align;
h = rmctx.sub_packet_h;
cook_decode_init(&rmctx,&q);
av_log(NULL,AV_LOG_ERROR,"nb_frames = %d\n",nb_frames);
x = 0;
if(packet_count % h)
{
packet_count += h - (packet_count % h);
rmctx.nb_packets = packet_count;
}
while(packet_count)
{
memset(pkt.data,0,sizeof(pkt.data));
rm_get_packet(fd, &rmctx, &pkt);
DEBUGF("total frames = %d packet count = %d output counter = %d \n",rmctx.audio_pkt_cnt*(fs/sps), packet_count,rmctx.audio_pkt_cnt);
for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
{
/* output raw audio frames that are sent to the decoder into separate files */
#ifdef DUMP_RAW_FRAMES
snprintf(filename,sizeof(filename),"dump%d.raw",++x);
fd_out = open(filename,O_WRONLY|O_CREAT|O_APPEND);
write(fd_out,pkt.data+i*sps,sps);
close(fd_out);
#endif
memcpy(inbuf,pkt.data+i*sps,sps);
nb_frames = cook_decode_frame(&rmctx,&q, outbuf, &datasize, inbuf , rmctx.block_align);
rmctx.frame_number++;
write(fd_dec,outbuf,datasize);
}
packet_count -= rmctx.audio_pkt_cnt;
rmctx.audio_pkt_cnt = 0;
}
cook_decode_close(&q);
close_wav(fd_dec,&rmctx);
close(fd);
return 0;
}

122
apps/codecs/libcook/cook.h Normal file
View file

@ -0,0 +1,122 @@
/*
* COOK compatible decoder
* Copyright (c) 2003 Sascha Sommer
* Copyright (c) 2005 Benjamin Larsson
*
* This file is taken from FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef _COOK_H
#define _COOK_H
#include <stdint.h>
#include "libavutil/lfg.h"
#include "bitstream.h"
#include "dsputil.h"
#include "bytestream.h"
#include "rm2wav.h"
typedef struct {
int *now;
int *previous;
} cook_gains;
typedef struct cook {
/*
* The following 5 functions provide the lowlevel arithmetic on
* the internal audio buffers.
*/
void (* scalar_dequant)(struct cook *q, int index, int quant_index,
int* subband_coef_index, int* subband_coef_sign,
float* mlt_p);
void (* decouple) (struct cook *q,
int subband,
float f1, float f2,
float *decode_buffer,
float *mlt_buffer1, float *mlt_buffer2);
void (* imlt_window) (struct cook *q, float *buffer1,
cook_gains *gains_ptr, float *previous_buffer);
void (* interpolate) (struct cook *q, float* buffer,
int gain_index, int gain_index_next);
void (* saturate_output) (struct cook *q, int chan, int16_t *out);
GetBitContext gb;
int frame_number;
int block_align;
int extradata_size;
/* stream data */
int nb_channels;
int joint_stereo;
int bit_rate;
int sample_rate;
int samples_per_channel;
int samples_per_frame;
int subbands;
int log2_numvector_size;
int numvector_size; //1 << log2_numvector_size;
int js_subband_start;
int total_subbands;
int num_vectors;
int bits_per_subpacket;
int cookversion;
/* states */
AVLFG random_state;
/* transform data */
MDCTContext mdct_ctx;
float* mlt_window;
/* gain buffers */
cook_gains gains1;
cook_gains gains2;
int gain_1[9];
int gain_2[9];
int gain_3[9];
int gain_4[9];
/* VLC data */
int js_vlc_bits;
VLC envelope_quant_index[13];
VLC sqvh[7]; //scalar quantization
VLC ccpl; //channel coupling
/* generatable tables and related variables */
int gain_size_factor;
float gain_table[23];
/* data buffers */
uint8_t* decoded_bytes_buffer;
float mono_mdct_output[2048] __attribute__ ((aligned(16))); //DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
float decode_buffer_1[1024];
float decode_buffer_2[1024];
float decode_buffer_0[1060]; /* static allocation for joint decode */
const float *cplscales[5];
} COOKContext;
av_cold int cook_decode_init(RMContext *rmctx, COOKContext *q);
int cook_decode_frame(RMContext *rmctx,COOKContext *q,
int16_t *outbuffer, int *data_size,
const uint8_t *inbuffer, int buf_size);
av_cold int cook_decode_close(COOKContext *q);
#endif

114
apps/codecs/libcook/main.c Normal file
View file

@ -0,0 +1,114 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2009 Mohamed Tarek
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <stdint.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <unistd.h>
#include "rm2wav.h"
#include "cook.h"
//#define DUMP_RAW_FRAMES
#define DEBUGF(message,args ...) av_log(NULL,AV_LOG_ERROR,message,## args)
int main(int argc, char *argv[])
{
int fd, fd_dec;
int res, datasize,x,i;
int nb_frames = 0;
#ifdef DUMP_RAW_FRAMES
char filename[15];
int fd_out;
#endif
int16_t outbuf[2048];
uint8_t inbuf[1024];
uint16_t fs,sps,h;
uint32_t packet_count;
COOKContext q;
RMContext rmctx;
RMPacket pkt;
memset(&q,0,sizeof(COOKContext));
memset(&rmctx,0,sizeof(RMContext));
memset(&pkt,0,sizeof(RMPacket));
if (argc != 2) {
DEBUGF("Incorrect number of arguments\n");
return -1;
}
fd = open(argv[1],O_RDONLY);
if (fd < 0) {
DEBUGF("Error opening file %s\n", argv[1]);
return -1;
}
fd_dec = open_wav("output.wav");
if (fd_dec < 0) {
DEBUGF("Error creating output file\n");
return -1;
}
res = real_parse_header(fd, &rmctx);
packet_count = rmctx.nb_packets;
rmctx.audio_framesize = rmctx.block_align;
rmctx.block_align = rmctx.sub_packet_size;
fs = rmctx.audio_framesize;
sps= rmctx.block_align;
h = rmctx.sub_packet_h;
cook_decode_init(&rmctx,&q);
av_log(NULL,AV_LOG_ERROR,"nb_frames = %d\n",nb_frames);
x = 0;
if(packet_count % h)
{
packet_count += h - (packet_count % h);
rmctx.nb_packets = packet_count;
}
while(packet_count)
{
memset(pkt.data,0,sizeof(pkt.data));
rm_get_packet(fd, &rmctx, &pkt);
DEBUGF("total frames = %d packet count = %d output counter = %d \n",rmctx.audio_pkt_cnt*(fs/sps), packet_count,rmctx.audio_pkt_cnt);
for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
{
/* output raw audio frames that are sent to the decoder into separate files */
#ifdef DUMP_RAW_FRAMES
snprintf(filename,sizeof(filename),"dump%d.raw",++x);
fd_out = open(filename,O_WRONLY|O_CREAT|O_APPEND);
write(fd_out,pkt.data+i*sps,sps);
close(fd_out);
#endif
memcpy(inbuf,pkt.data+i*sps,sps);
nb_frames = cook_decode_frame(&rmctx,&q, outbuf, &datasize, inbuf , rmctx.block_align);
rmctx.frame_number++;
write(fd_dec,outbuf,datasize);
}
packet_count -= rmctx.audio_pkt_cnt;
rmctx.audio_pkt_cnt = 0;
}
cook_decode_close(&q);
close_wav(fd_dec,&rmctx);
close(fd);
return 0;
}