From 26cee86a0ca354ac15d46fb92db0cc9a776dd4b2 Mon Sep 17 00:00:00 2001 From: Mohamed Tarek Date: Tue, 4 Aug 2009 13:54:06 +0000 Subject: Add support for AC3 audio in RM container. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22155 a1c6a512-1295-4272-9138-f99709370657 --- apps/codecs/dnet.c | 190 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 190 insertions(+) create mode 100644 apps/codecs/dnet.c (limited to 'apps/codecs/dnet.c') diff --git a/apps/codecs/dnet.c b/apps/codecs/dnet.c new file mode 100644 index 0000000000..12352ed903 --- /dev/null +++ b/apps/codecs/dnet.c @@ -0,0 +1,190 @@ +/*************************************************************************** + * __________ __ ___. + * 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; +unsigned long samplesdone; +unsigned long frequency; +RMContext rmctx; +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; + +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(void) +{ + size_t n; + uint8_t *filebuf; + int retval, consumed, packet_offset; + + /* Generic codec initialisation */ + ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED); + ci->configure(DSP_SET_SAMPLE_DEPTH, 28); + +next_track: + if (codec_init()) { + retval = CODEC_ERROR; + goto exit; + } + + while (!ci->taginfo_ready) + ci->yield(); + + ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); + codec_set_replaygain(ci->id3); + + /* Intializations */ + state = a52_init(0); + ci->memset(&rmctx,0,sizeof(RMContext)); + ci->memset(&pkt,0,sizeof(RMPacket)); + init_rm(&rmctx); + + /* Seek to the first packet */ + ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE ); + + /* The main decoding loop */ + while(pkt.timestamp < rmctx.duration) { + ci->yield(); + if (ci->stop_codec || ci->new_track) + break; + + if (ci->seek_time) { + packet_offset = ci->seek_time / (((rmctx.block_align + PACKET_HEADER_SIZE)*8*1000)/rmctx.bit_rate); + ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE)); + samplesdone = A52_SAMPLESPERFRAME * packet_offset; + ci->seek_complete(); + } + + filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE); + consumed = rm_get_packet(&filebuf, &rmctx, &pkt); + if(consumed < 0) { + DEBUGF("rm_get_packet failed\n"); + return CODEC_ERROR; + } + a52_decode_data(filebuf, filebuf + rmctx.block_align); + ci->advance_buffer(pkt.length); + } + + retval = CODEC_OK; + + if (ci->request_next_track()) + goto next_track; + +exit: + a52_free(state); + return retval; +} -- cgit v1.2.3