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/*
* RoQ audio encoder
*
* Copyright (c) 2005 Eric Lasota
* Based on RoQ specs (c)2001 Tim Ferguson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "bytestream.h"
#define ROQ_FIRST_FRAME_SIZE (735*8)
#define ROQ_FRAME_SIZE 735
#define MAX_DPCM (127*127)
static unsigned char dpcmValues[MAX_DPCM];
typedef struct
{
short lastSample[2];
} ROQDPCMContext;
static av_cold void roq_dpcm_table_init(void)
{
int i;
/* Create a table of quick DPCM values */
for (i=0; i<MAX_DPCM; i++) {
int s= ff_sqrt(i);
int mid= s*s + s;
dpcmValues[i]= s + (i>mid);
}
}
static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
return -1;
}
if (avctx->sample_rate != 22050) {
av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
return -1;
}
if (avctx->sample_fmt != SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
return -1;
}
roq_dpcm_table_init();
avctx->frame_size = ROQ_FIRST_FRAME_SIZE;
context->lastSample[0] = context->lastSample[1] = 0;
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
}
static unsigned char dpcm_predict(short *previous, short current)
{
int diff;
int negative;
int result;
int predicted;
diff = current - *previous;
negative = diff<0;
diff = FFABS(diff);
if (diff >= MAX_DPCM)
result = 127;
else
result = dpcmValues[diff];
/* See if this overflows */
retry:
diff = result*result;
if (negative)
diff = -diff;
predicted = *previous + diff;
/* If it overflows, back off a step */
if (predicted > 32767 || predicted < -32768) {
result--;
goto retry;
}
/* Add the sign bit */
result |= negative << 7; //if (negative) result |= 128;
*previous = predicted;
return result;
}
static int roq_dpcm_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
int i, samples, stereo, ch;
short *in;
unsigned char *out;
ROQDPCMContext *context = avctx->priv_data;
stereo = (avctx->channels == 2);
if (stereo) {
context->lastSample[0] &= 0xFF00;
context->lastSample[1] &= 0xFF00;
}
out = frame;
in = data;
bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
bytestream_put_byte(&out, 0x10);
bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
if (stereo) {
bytestream_put_byte(&out, (context->lastSample[1])>>8);
bytestream_put_byte(&out, (context->lastSample[0])>>8);
} else
bytestream_put_le16(&out, context->lastSample[0]);
/* Write the actual samples */
samples = avctx->frame_size;
for (i=0; i<samples; i++)
for (ch=0; ch<avctx->channels; ch++)
*out++ = dpcm_predict(&context->lastSample[ch], *in++);
/* Use smaller frames from now on */
avctx->frame_size = ROQ_FRAME_SIZE;
/* Return the result size */
return out - frame;
}
static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
AVCodec roq_dpcm_encoder = {
"roq_dpcm",
CODEC_TYPE_AUDIO,
CODEC_ID_ROQ_DPCM,
sizeof(ROQDPCMContext),
roq_dpcm_encode_init,
roq_dpcm_encode_frame,
roq_dpcm_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
};
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