apps/codecs/a52_rm.c

   1 /***************************************************************************
   2  *             __________               __   ___.
   3  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
   4  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
   5  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
   6  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
   7  *                     \/            \/     \/    \/            \/
   8  * $Id$
   9  *
  10  * Copyright (C) 2009 Mohamed Tarek
  11  *
  12  * This program is free software; you can redistribute it and/or
  13  * modify it under the terms of the GNU General Public License
  14  * as published by the Free Software Foundation; either version 2
  15  * of the License, or (at your option) any later version.
  16  *
  17  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  18  * KIND, either express or implied.
  19  *
  20  ****************************************************************************/
  21
  22 #include "codeclib.h"
  23 #include <codecs/librm/rm.h>
  24 #include <inttypes.h>  /* Needed by a52.h */
  25 #include <codecs/liba52/config-a52.h>
  26 #include <codecs/liba52/a52.h>
  27
  28 CODEC_HEADER
  29
  30 #define BUFFER_SIZE 4096
  31
  32 #define A52_SAMPLESPERFRAME (6*256)
  33
  34 static a52_state_t *state;
  35 static unsigned long samplesdone;
  36 static unsigned long frequency;
  37 static RMContext rmctx;
  38 static RMPacket pkt;
  39
  40 static void init_rm(RMContext *rmctx)
  41 {
  42     memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
  43 }
  44
  45 /* used outside liba52 */
  46 static uint8_t buf[3840] IBSS_ATTR;
  47
  48 /* The following two functions, a52_decode_data and output_audio are taken from apps/codecs/a52.c */
  49 static inline void output_audio(sample_t *samples)
  50 {
  51     ci->yield();
  52     ci->pcmbuf_insert(&samples[0], &samples[256], 256);
  53 }
  54
  55 static void a52_decode_data(uint8_t *start, uint8_t *end)
  56 {
  57     static uint8_t *bufptr = buf;
  58     static uint8_t *bufpos = buf + 7;
  59     /*
  60      * sample_rate and flags are static because this routine could
  61      * exit between the a52_syncinfo() and the ao_setup(), and we want
  62      * to have the same values when we get back !
  63      */
  64     static int sample_rate;
  65     static int flags;
  66     int bit_rate;
  67     int len;
  68
  69     while (1) {
  70         len = end - start;
  71         if (!len)
  72             break;
  73         if (len > bufpos - bufptr)
  74             len = bufpos - bufptr;
  75         memcpy(bufptr, start, len);
  76         bufptr += len;
  77         start += len;
  78         if (bufptr == bufpos) {
  79             if (bufpos == buf + 7) {
  80                 int length;
  81
  82                 length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate);
  83                 if (!length) {
  84                     //DEBUGF("skip\n");
  85                     for (bufptr = buf; bufptr < buf + 6; bufptr++)
  86                         bufptr[0] = bufptr[1];
  87                     continue;
  88                 }
  89                 bufpos = buf + length;
  90             } else {
  91                 /* Unity gain is 1 << 26, and we want to end up on 28 bits
  92                    of precision instead of the default 30.
  93                  */
  94                 level_t level = 1 << 24;
  95                 sample_t bias = 0;
  96                 int i;
  97
  98                 /* This is the configuration for the downmixing: */
  99                 flags = A52_STEREO | A52_ADJUST_LEVEL;
 100
 101                 if (a52_frame(state, buf, &flags, &level, bias))
 102                     goto error;
 103                 a52_dynrng(state, NULL, NULL);
 104                 frequency = sample_rate;
 105
 106                 /* An A52 frame consists of 6 blocks of 256 samples
 107                    So we decode and output them one block at a time */
 108                 for (i = 0; i < 6; i++) {
 109                     if (a52_block(state))
 110                         goto error;
 111                     output_audio(a52_samples(state));
 112                     samplesdone += 256;
 113                 }
 114                 ci->set_elapsed(samplesdone/(frequency/1000));
 115                 bufptr = buf;
 116                 bufpos = buf + 7;
 117                 continue;
 118             error:
 119                 //logf("Error decoding A52 stream\n");
 120                 bufptr = buf;
 121                 bufpos = buf + 7;
 122             }
 123         }
 124     }
 125 }
 126
 127
 128 /* this is the codec entry point */
 129 enum codec_status codec_main(void)
 130 {
 131     size_t n;
 132     uint8_t *filebuf;
 133     int retval, consumed, packet_offset;
 134     int playback_on = -1;
 135     size_t resume_offset = ci->id3->offset;
 136
 137     /* Generic codec initialisation */
 138     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
 139     ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
 140
 141 next_track:
 142     if (codec_init()) {
 143         retval = CODEC_ERROR;
 144         goto exit;
 145     }
 146
 147     while (!ci->taginfo_ready)
 148         ci->yield();
 149
 150     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
 151     codec_set_replaygain(ci->id3);
 152
 153     /* Intializations */
 154     state = a52_init(0);
 155     ci->memset(&rmctx,0,sizeof(RMContext));
 156     ci->memset(&pkt,0,sizeof(RMPacket));
 157     init_rm(&rmctx);
 158
 159     /* check for a mid-track resume and force a seek time accordingly */
 160     if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) {
 161         resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
 162         /* put number of subpackets to skip in resume_offset */
 163         resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE);
 164         ci->seek_time =  (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);
 165     }
 166
 167     /* Seek to the first packet */
 168     ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE );
 169
 170     /* The main decoding loop */
 171     while((unsigned)rmctx.audio_pkt_cnt < rmctx.nb_packets) {
 172         ci->yield();
 173         if (ci->stop_codec || ci->new_track)
 174             break;
 175
 176         if (ci->seek_time) {
 177             packet_offset = ci->seek_time / ((rmctx.block_align*8*1000)/rmctx.bit_rate);
 178             ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE));
 179             rmctx.audio_pkt_cnt = packet_offset;
 180             samplesdone = (rmctx.sample_rate/1000 * ci->seek_time);
 181             ci->seek_complete();
 182         }
 183
 184         filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE);
 185         consumed = rm_get_packet(&filebuf, &rmctx, &pkt);
 186
 187         if(consumed < 0 && playback_on != 0) {
 188             if(playback_on == -1) {
 189             /* Error only if packet-parsing failed and playback hadn't started */
 190                 DEBUGF("rm_get_packet failed\n");
 191                 return CODEC_ERROR;
 192             }
 193             else {
 194                 retval = CODEC_OK;
 195                 goto exit;
 196             }
 197         }
 198
 199         playback_on = 1;
 200         a52_decode_data(filebuf, filebuf + rmctx.block_align);
 201         ci->advance_buffer(pkt.length);
 202     }
 203
 204     retval = CODEC_OK;
 205
 206     if (ci->request_next_track())
 207         goto next_track;
 208
 209 exit:
 210     a52_free(state);
 211     return retval;
 212 }