/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2008 Dominik Wenger * * 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 "codeclib.h" #include "libasap/asap.h" CODEC_HEADER #define CHUNK_SIZE (1024*2) static byte samples[CHUNK_SIZE] IBSS_ATTR; /* The sample buffer */ static ASAP_State asap IBSS_ATTR; /* asap codec state */ /* this is the codec entry point */ enum codec_status codec_main(enum codec_entry_call_reason reason) { /* Nothing to do */ return CODEC_OK; (void)reason; } /* this is called for each file to process */ enum codec_status codec_run(void) { int n_bytes; int song; int duration; char* module; int bytesPerSample =2; intptr_t param; if (codec_init()) { DEBUGF("codec init failed\n"); return CODEC_ERROR; } codec_set_replaygain(ci->id3); int bytes_done =0; size_t filesize; ci->seek_buffer(0); module = ci->request_buffer(&filesize, ci->filesize); if (!module || (size_t)filesize < (size_t)ci->filesize) { DEBUGF("loading error\n"); return CODEC_ERROR; } /*Init ASAP */ if (!ASAP_Load(&asap, ci->id3->path, module, filesize)) { DEBUGF("%s: format not supported",ci->id3->path); return CODEC_ERROR; } /* Make use of 44.1khz */ ci->configure(DSP_SET_FREQUENCY, 44100); /* Sample depth is 16 bit little endian */ ci->configure(DSP_SET_SAMPLE_DEPTH, 16); /* Stereo or Mono output ? */ if(asap.module_info->channels ==1) { ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO); bytesPerSample = 2; } else { ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED); bytesPerSample = 4; } /* reset eleapsed */ ci->set_elapsed(0); song = asap.module_info->default_song; duration = asap.module_info->durations[song]; if (duration < 0) duration = 180 * 1000; /* set id3 length, because metadata parse might not have done it */ ci->id3->length = duration; ASAP_PlaySong(&asap, song, duration); ASAP_MutePokeyChannels(&asap, 0); /* The main decoder loop */ while (1) { enum codec_command_action action = ci->get_command(¶m); if (action == CODEC_ACTION_HALT) break; if (action == CODEC_ACTION_SEEK_TIME) { /* New time is ready in param */ /* seek to pos */ ASAP_Seek(&asap,param); /* update bytes_done */ bytes_done = param*44.1*2; /* update elapsed */ ci->set_elapsed((bytes_done / 2) / 44.1); /* seek ready */ ci->seek_complete(); } /* Generate a buffer full of Audio */ #ifdef ROCKBOX_LITTLE_ENDIAN n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_LE); #else n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_BE); #endif ci->pcmbuf_insert(samples, NULL, n_bytes /bytesPerSample); bytes_done += n_bytes; ci->set_elapsed((bytes_done / 2) / 44.1); if(n_bytes != sizeof(samples)) break; } return CODEC_OK; }