6427d127aa
Use a smaller PCM buffer on targets with 2MB or less ram. (FS#9703) git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19743 a1c6a512-1295-4272-9138-f99709370657
134 lines
3.9 KiB
C
134 lines
3.9 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2008 Dominik Wenger
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "codeclib.h"
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#include "libasap/asap.h"
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CODEC_HEADER
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#define CHUNK_SIZE (1024*2)
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static byte samples[CHUNK_SIZE] IBSS_ATTR; /* The sample buffer */
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static ASAP_State asap; /* asap codec state */
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/* this is the codec entry point */
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enum codec_status codec_main(void)
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{
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int n_bytes;
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int song;
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int duration;
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char* module;
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int bytesPerSample =2;
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next_track:
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if (codec_init()) {
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DEBUGF("codec init failed\n");
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return CODEC_ERROR;
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}
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while (!*ci->taginfo_ready && !ci->stop_codec)
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ci->sleep(1);
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codec_set_replaygain(ci->id3);
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int bytes_done =0;
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size_t filesize;
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ci->seek_buffer(0);
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module = ci->request_buffer(&filesize, ci->filesize);
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if (!module || (size_t)filesize < (size_t)ci->filesize)
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{
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DEBUGF("loading error\n");
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return CODEC_ERROR;
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}
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/*Init ASAP */
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if (!ASAP_Load(&asap, ci->id3->path, module, filesize))
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{
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DEBUGF("%s: format not supported",ci->id3->path);
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return CODEC_ERROR;
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}
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/* Make use of 44.1khz */
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ci->configure(DSP_SET_FREQUENCY, 44100);
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/* Sample depth is 16 bit little endian */
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ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
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/* Stereo or Mono output ? */
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if(asap.module_info.channels ==1)
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{
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ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
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bytesPerSample = 2;
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}
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else
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{
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ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
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bytesPerSample = 4;
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}
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/* reset eleapsed */
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ci->set_elapsed(0);
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song = asap.module_info.default_song;
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duration = asap.module_info.durations[song];
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if (duration < 0)
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duration = 180 * 1000;
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ASAP_PlaySong(&asap, song, duration);
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ASAP_MutePokeyChannels(&asap, 0);
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/* The main decoder loop */
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while (1) {
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ci->yield();
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if (ci->stop_codec || ci->new_track)
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break;
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if (ci->seek_time) {
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/* New time is ready in ci->seek_time */
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/* seek to pos */
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ASAP_Seek(&asap,ci->seek_time);
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/* update elapsed */
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ci->set_elapsed(ci->seek_time);
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/* update bytes_done */
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bytes_done = ci->seek_time*44.1*2;
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/* seek ready */
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ci->seek_complete();
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}
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/* Generate a buffer full of Audio */
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#ifdef ROCKBOX_LITTLE_ENDIAN
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n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_LE);
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#else
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n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_BE);
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#endif
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ci->pcmbuf_insert(samples, NULL, n_bytes /bytesPerSample);
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bytes_done += n_bytes;
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ci->set_elapsed((bytes_done / 2) / 44.1);
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if(n_bytes != sizeof(samples))
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break;
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}
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if (ci->request_next_track())
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goto next_track;
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return CODEC_OK;
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}
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