/*************************************************************************** * __________ __ ___. * 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 "codeclib.h" #include #include /* Needed by a52.h */ #include #include CODEC_HEADER #define BUFFER_SIZE 4096 #define A52_SAMPLESPERFRAME (6*256) static a52_state_t *state; static unsigned long samplesdone; static unsigned long frequency; static RMContext rmctx; static RMPacket pkt; static void init_rm(RMContext *rmctx) { memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext)); } /* used outside liba52 */ static uint8_t buf[3840] IBSS_ATTR; /* The following two functions, a52_decode_data and output_audio are taken from apps/codecs/a52.c */ static inline void output_audio(sample_t *samples) { ci->yield(); ci->pcmbuf_insert(&samples[0], &samples[256], 256); } static void a52_decode_data(uint8_t *start, uint8_t *end) { static uint8_t *bufptr = buf; static uint8_t *bufpos = buf + 7; /* * sample_rate and flags are static because this routine could * exit between the a52_syncinfo() and the ao_setup(), and we want * to have the same values when we get back ! */ static int sample_rate; static int flags; int bit_rate; int len; while (1) { len = end - start; if (!len) break; if (len > bufpos - bufptr) len = bufpos - bufptr; memcpy(bufptr, start, len); bufptr += len; start += len; if (bufptr == bufpos) { if (bufpos == buf + 7) { int length; length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate); if (!length) { //DEBUGF("skip\n"); for (bufptr = buf; bufptr < buf + 6; bufptr++) bufptr[0] = bufptr[1]; continue; } bufpos = buf + length; } else { /* Unity gain is 1 << 26, and we want to end up on 28 bits of precision instead of the default 30. */ level_t level = 1 << 24; sample_t bias = 0; int i; /* This is the configuration for the downmixing: */ flags = A52_STEREO | A52_ADJUST_LEVEL; if (a52_frame(state, buf, &flags, &level, bias)) goto error; a52_dynrng(state, NULL, NULL); frequency = sample_rate; /* An A52 frame consists of 6 blocks of 256 samples So we decode and output them one block at a time */ for (i = 0; i < 6; i++) { if (a52_block(state)) goto error; output_audio(a52_samples(state)); samplesdone += 256; } ci->set_elapsed(samplesdone/(frequency/1000)); bufptr = buf; bufpos = buf + 7; continue; error: //logf("Error decoding A52 stream\n"); bufptr = buf; bufpos = buf + 7; } } } } /* this is the codec entry point */ enum codec_status codec_main(enum codec_entry_call_reason reason) { if (reason == CODEC_LOAD) { /* Generic codec initialisation */ ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED); ci->configure(DSP_SET_SAMPLE_DEPTH, 28); } else if (reason == CODEC_UNLOAD) { if (state) a52_free(state); } return CODEC_OK; } /* this is called for each file to process */ enum codec_status codec_run(void) { size_t n; uint8_t *filebuf; int consumed, packet_offset; int playback_on = -1; size_t resume_offset; intptr_t param; enum codec_command_action action = CODEC_ACTION_NULL; if (codec_init()) { return CODEC_ERROR; } resume_offset = ci->id3->offset; ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); codec_set_replaygain(ci->id3); ci->seek_buffer(ci->id3->first_frame_offset); /* Intializations */ state = a52_init(0); ci->memset(&rmctx,0,sizeof(RMContext)); ci->memset(&pkt,0,sizeof(RMPacket)); init_rm(&rmctx); samplesdone = 0; /* check for a mid-track resume and force a seek time accordingly */ if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) { resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE; /* put number of subpackets to skip in resume_offset */ resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE); param = (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate); action = CODEC_ACTION_SEEK_TIME; } else { /* Seek to the first packet */ ci->set_elapsed(0); ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE ); } /* The main decoding loop */ while((unsigned)rmctx.audio_pkt_cnt < rmctx.nb_packets) { if (action == CODEC_ACTION_NULL) action = ci->get_command(¶m); if (action == CODEC_ACTION_HALT) break; if (action == CODEC_ACTION_SEEK_TIME) { packet_offset = param / ((rmctx.block_align*8*1000)/rmctx.bit_rate); ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE)); rmctx.audio_pkt_cnt = packet_offset; samplesdone = (rmctx.sample_rate/1000 * param); ci->set_elapsed(samplesdone/(frequency/1000)); ci->seek_complete(); } action = CODEC_ACTION_NULL; filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE); consumed = rm_get_packet(&filebuf, &rmctx, &pkt); if(consumed < 0 && playback_on != 0) { if(playback_on == -1) { /* Error only if packet-parsing failed and playback hadn't started */ DEBUGF("rm_get_packet failed\n"); return CODEC_ERROR; } else { break; } } playback_on = 1; a52_decode_data(filebuf, filebuf + rmctx.block_align); ci->advance_buffer(pkt.length); } return CODEC_OK; }