mirror of
https://github.com/xenia-project/FFmpeg.git
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2912e87a6c
Signed-off-by: Mans Rullgard <mans@mansr.com>
168 lines
4.2 KiB
C
168 lines
4.2 KiB
C
/*
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* RoQ audio encoder
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*
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* Copyright (c) 2005 Eric Lasota
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* Based on RoQ specs (c)2001 Tim Ferguson
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intmath.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#define ROQ_FIRST_FRAME_SIZE (735*8)
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#define ROQ_FRAME_SIZE 735
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#define MAX_DPCM (127*127)
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typedef struct
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{
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short lastSample[2];
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} ROQDPCMContext;
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static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
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{
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ROQDPCMContext *context = avctx->priv_data;
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if (avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
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return -1;
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}
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if (avctx->sample_rate != 22050) {
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av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
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return -1;
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}
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if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
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av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
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return -1;
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}
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avctx->frame_size = ROQ_FIRST_FRAME_SIZE;
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context->lastSample[0] = context->lastSample[1] = 0;
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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return 0;
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}
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static unsigned char dpcm_predict(short *previous, short current)
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{
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int diff;
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int negative;
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int result;
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int predicted;
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diff = current - *previous;
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negative = diff<0;
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diff = FFABS(diff);
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if (diff >= MAX_DPCM)
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result = 127;
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else {
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result = ff_sqrt(diff);
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result += diff > result*result+result;
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}
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/* See if this overflows */
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retry:
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diff = result*result;
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if (negative)
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diff = -diff;
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predicted = *previous + diff;
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/* If it overflows, back off a step */
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if (predicted > 32767 || predicted < -32768) {
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result--;
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goto retry;
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}
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/* Add the sign bit */
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result |= negative << 7; //if (negative) result |= 128;
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*previous = predicted;
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return result;
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}
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static int roq_dpcm_encode_frame(AVCodecContext *avctx,
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unsigned char *frame, int buf_size, void *data)
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{
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int i, samples, stereo, ch;
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const short *in;
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unsigned char *out;
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ROQDPCMContext *context = avctx->priv_data;
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stereo = (avctx->channels == 2);
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if (stereo) {
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context->lastSample[0] &= 0xFF00;
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context->lastSample[1] &= 0xFF00;
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}
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out = frame;
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in = data;
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bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
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bytestream_put_byte(&out, 0x10);
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bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
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if (stereo) {
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bytestream_put_byte(&out, (context->lastSample[1])>>8);
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bytestream_put_byte(&out, (context->lastSample[0])>>8);
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} else
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bytestream_put_le16(&out, context->lastSample[0]);
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/* Write the actual samples */
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samples = avctx->frame_size;
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for (i=0; i<samples; i++)
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for (ch=0; ch<avctx->channels; ch++)
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*out++ = dpcm_predict(&context->lastSample[ch], *in++);
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/* Use smaller frames from now on */
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avctx->frame_size = ROQ_FRAME_SIZE;
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/* Return the result size */
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return out - frame;
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}
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static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
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{
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av_freep(&avctx->coded_frame);
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return 0;
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}
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AVCodec ff_roq_dpcm_encoder = {
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"roq_dpcm",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_ROQ_DPCM,
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sizeof(ROQDPCMContext),
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roq_dpcm_encode_init,
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roq_dpcm_encode_frame,
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roq_dpcm_encode_close,
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NULL,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
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};
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