f40bfc9267
Change-Id: Id7f4717d51ed02d67cb9f9cb3c0ada4a81843f97 Reviewed-on: http://gerrit.rockbox.org/137 Reviewed-by: Nils Wallménius <nils@rockbox.org> Tested-by: Nils Wallménius <nils@rockbox.org>
250 lines
7.1 KiB
C
250 lines
7.1 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) 2010 Yoshihisa Uchida
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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 "adpcm_seek.h"
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#include "support_formats.h"
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/*
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* YAMAHA ADPCM
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*
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* References
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* [1] YAMAHA, YAMAHA ADPCM ACM Driver Version 1.0.0.0, 2005
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* [2] BlendWorks, YM2608 ADPCM,
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* http://web.archive.org/web/20050208190547/www.memb.jp/~dearna/ma/ym2608/adpcm.html
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* [3] Naoyuki Sawa, ADPCM no shikumi #1,
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* http://www.piece-me.org/piece-lab/adpcm/adpcm1.html
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* [4] ffmpeg source code, libavcodec/adpcm.c
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*/
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/* ADPCM data block layout
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*
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* when the block header exists. (for example, encoding by YAMAHA ADPCM ACM Driver)
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* blockAlign = (frequency / 60 + 4) * channels.
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*
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* block
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* <Mono> (channels = 1)
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* int16_t first value (Little endian)
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* uint16_t first predictor (Little endian)
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* uint8_t ADPCM data (1st data: 0-3 bit, 2nd data: 4-7 bit)
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* ....
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*
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* <Stereo> (channels = 2)
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* int16_t Left channel first value (Little endian)
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* uint16_t Left channel first predictor (Little endian)
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* int16_t Right channel first value (Little endian)
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* uint16_t Right channel first predictor (Little endian)
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* uint8_t ADPCM data (Left channel: 0-3 bit, Right channel: 4-7 bit)
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* ....
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*
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* when the block header does not exist. (for example, encoding by ffmpeg)
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* blockAlign = 8000
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*
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* block
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* <Mono> (channels = 1)
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* uint8_t ADPCM data (1st data: 0-3 bit, 2nd data: 4-7 bit)
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* ....
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*
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* <Stereo> (channels = 2)
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* uint8_t ADPCM data (Left channel: 0-3 bit, Right channel: 4-7 bit)
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* ....
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*/
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static const int32_t amplification_table[] ICONST_ATTR = {
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230, 230, 230, 230, 307, 409, 512, 614, 230, 230, 230, 230, 307, 409, 512, 614
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};
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static bool has_block_header = false;
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static struct adpcm_data cur_data;
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static int blocksperchunk;
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static struct pcm_format *fmt;
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static bool set_format(struct pcm_format *format)
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{
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fmt = format;
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if (fmt->channels == 0)
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{
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DEBUGF("CODEC_ERROR: channels is 0\n");
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return false;
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}
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if (fmt->bitspersample != 4)
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{
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DEBUGF("CODEC_ERROR: yamaha adpcm must be 4 bitspersample: %d\n",
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fmt->bitspersample);
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return false;
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}
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/* check exists block header */
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if (fmt->blockalign == ((ci->id3->frequency / 60) + 4) * fmt->channels)
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{
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has_block_header = true;
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/* chunksize = about 1/30 [sec] data */
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fmt->chunksize = fmt->blockalign;
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blocksperchunk = 1;
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}
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else
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{
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uint32_t max_chunk_count;
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has_block_header = false;
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/* blockalign = 2 * channels samples */
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fmt->blockalign = fmt->channels;
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fmt->samplesperblock = 2;
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/* chunksize = about 1/32[sec] data */
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blocksperchunk = ci->id3->frequency >> 6;
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fmt->chunksize = blocksperchunk * fmt->blockalign;
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max_chunk_count = (uint64_t)ci->id3->length * ci->id3->frequency
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/ (2000LL * fmt->chunksize / fmt->channels);
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/* initialize seek table */
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init_seek_table(max_chunk_count);
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/* add first data */
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add_adpcm_data(&cur_data);
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}
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return true;
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}
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static int16_t create_pcmdata(int ch, uint8_t nibble)
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{
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int32_t tmp_pcmdata = cur_data.pcmdata[ch];
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int32_t step = cur_data.step[ch];
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int32_t delta = step >> 3;
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if (nibble & 4) delta += step;
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if (nibble & 2) delta += (step >> 1);
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if (nibble & 1) delta += (step >> 2);
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if (nibble & 0x08)
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tmp_pcmdata -= delta;
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else
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tmp_pcmdata += delta;
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CLIP(tmp_pcmdata, -32768, 32767);
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cur_data.pcmdata[ch] = tmp_pcmdata;
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step = (step * amplification_table[nibble & 0x07]) >> 8;
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CLIP(step, 127, 24576);
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cur_data.step[ch] = step;
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return cur_data.pcmdata[ch];
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}
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static int decode(const uint8_t *inbuf, size_t inbufsize,
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int32_t *outbuf, int *outbufcount)
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{
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int ch;
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size_t nsamples = 0;
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/* read block header */
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if (has_block_header)
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{
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for (ch = 0; ch < fmt->channels; ch++)
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{
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cur_data.pcmdata[ch] = inbuf[0] | (SE(inbuf[1]) << 8);
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cur_data.step[ch] = inbuf[2] | (inbuf[3] << 8);
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inbuf += 4;
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inbufsize -= 4;
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}
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}
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/* read block data */
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ch = fmt->channels - 1;
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while (inbufsize)
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{
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*outbuf++ = create_pcmdata(0, *inbuf ) << (PCM_OUTPUT_DEPTH - 16);
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*outbuf++ = create_pcmdata(ch, *inbuf >> 4) << (PCM_OUTPUT_DEPTH - 16);
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nsamples += 2;
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inbuf++;
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inbufsize--;
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}
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if (fmt->channels == 2)
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nsamples >>= 1;
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*outbufcount = nsamples;
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if (!has_block_header)
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add_adpcm_data(&cur_data);
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return CODEC_OK;
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}
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static int decode_for_seek(const uint8_t *inbuf, size_t inbufsize)
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{
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int ch = fmt->channels - 1;
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while (inbufsize)
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{
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create_pcmdata(0, *inbuf );
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create_pcmdata(ch, *inbuf >> 4);
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inbuf++;
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inbufsize--;
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}
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add_adpcm_data(&cur_data);
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return CODEC_OK;
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}
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static struct pcm_pos *get_seek_pos(uint32_t seek_val, int seek_mode,
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uint8_t *(*read_buffer)(size_t *realsize))
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{
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static struct pcm_pos newpos;
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uint32_t new_count = (seek_mode == PCM_SEEK_TIME)?
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((uint64_t)seek_val * ci->id3->frequency / 1000LL)
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/ (blocksperchunk * fmt->samplesperblock) :
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seek_val / (unsigned long)fmt->chunksize;
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if (!has_block_header)
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{
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new_count = seek(new_count, &cur_data, read_buffer, &decode_for_seek);
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}
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newpos.pos = new_count * fmt->chunksize;
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newpos.samples = new_count * blocksperchunk * fmt->samplesperblock;
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return &newpos;
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}
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static struct pcm_codec codec = {
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set_format,
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get_seek_pos,
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decode,
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};
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const struct pcm_codec *get_yamaha_adpcm_codec(void)
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{
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/* initialize first step, pcm data */
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cur_data.pcmdata[0] = 0;
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cur_data.pcmdata[1] = 0;
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cur_data.step[0] = 127;
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cur_data.step[1] = 127;
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return &codec;
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}
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