From 010bb8e6bad91c2c03a37f2a5f51820910500bdc Mon Sep 17 00:00:00 2001 From: Mohamed Tarek Date: Tue, 4 Aug 2009 15:05:56 +0000 Subject: Rename codecs/dnet.c to codecs/a52_rm.c git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22158 a1c6a512-1295-4272-9138-f99709370657 --- apps/codecs/SOURCES | 2 +- apps/codecs/a52_rm.c | 190 ++++++++++++++++++++++++++++++++++++++++++++++++ apps/codecs/codecs.make | 2 +- apps/codecs/dnet.c | 190 ------------------------------------------------ apps/metadata.c | 2 +- 5 files changed, 193 insertions(+), 193 deletions(-) create mode 100644 apps/codecs/a52_rm.c delete mode 100644 apps/codecs/dnet.c diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES index 92e4d2d..4e6c891 100644 --- a/apps/codecs/SOURCES +++ b/apps/codecs/SOURCES @@ -11,7 +11,7 @@ alac.c #endif cook.c raac.c -dnet.c +a52_rm.c mpc.c wma.c sid.c diff --git a/apps/codecs/a52_rm.c b/apps/codecs/a52_rm.c new file mode 100644 index 0000000..12352ed --- /dev/null +++ b/apps/codecs/a52_rm.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; +} diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make index 41f5bdc..ba5f0ed 100644 --- a/apps/codecs/codecs.make +++ b/apps/codecs/codecs.make @@ -77,7 +77,7 @@ $(CODECDIR)/wavpack_enc.codec: $(CODECDIR)/libwavpack.a $(CODECDIR)/asap.codec : $(CODECDIR)/libasap.a $(CODECDIR)/cook.codec : $(CODECDIR)/libcook.a $(CODECDIR)/librm.a $(CODECDIR)/raac.codec : $(CODECDIR)/libfaad.a $(CODECDIR)/librm.a -$(CODECDIR)/dnet.codec : $(CODECDIR)/liba52.a $(CODECDIR)/librm.a +$(CODECDIR)/a52_rm.codec : $(CODECDIR)/liba52.a $(CODECDIR)/librm.a $(CODECS): $(CODECLIB) # this must be last in codec dependency list diff --git a/apps/codecs/dnet.c b/apps/codecs/dnet.c deleted file mode 100644 index 12352ed..0000000 --- a/apps/codecs/dnet.c +++ /dev/null @@ -1,190 +0,0 @@ -/*************************************************************************** - * __________ __ ___. - * 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; -} diff --git a/apps/metadata.c b/apps/metadata.c index 6354764..8414bf0 100644 --- a/apps/metadata.c +++ b/apps/metadata.c @@ -123,7 +123,7 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] = AFMT_ENTRY("RAAC", "raac", NULL, "rm\0ra\0rmvb\0" ), /* AC3 in RM/RA */ [AFMT_RM_AC3] = - AFMT_ENTRY("AC3", "dnet", NULL, "rm\0ra\0rmvb\0" ), + AFMT_ENTRY("AC3", "a52_rm", NULL, "rm\0ra\0rmvb\0" ), #endif }; -- cgit v1.1