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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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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"
#include "internal.h"
#include "mathops.h"
#define ROQ_FRAME_SIZE 735
#define ROQ_HEADER_SIZE 8
#define MAX_DPCM (127*127)
typedef struct
{
short lastSample[2];
int input_frames;
int buffered_samples;
int16_t *frame_buffer;
int64_t first_pts;
} ROQDPCMContext;
static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&context->frame_buffer);
return 0;
}
static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
int ret;
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
return AVERROR(EINVAL);
}
if (avctx->sample_rate != 22050) {
av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
return AVERROR(EINVAL);
}
avctx->frame_size = ROQ_FRAME_SIZE;
avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
(22050 / ROQ_FRAME_SIZE) * 8;
context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
sizeof(*context->frame_buffer));
if (!context->frame_buffer) {
ret = AVERROR(ENOMEM);
goto error;
}
context->lastSample[0] = context->lastSample[1] = 0;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
roq_dpcm_encode_close(avctx);
return ret;
}
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 = ff_sqrt(diff);
result += diff > result*result+result;
}
/* 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, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int i, stereo, data_size, ret;
const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
uint8_t *out;
ROQDPCMContext *context = avctx->priv_data;
stereo = (avctx->channels == 2);
if (!in && context->input_frames >= 8)
return 0;
if (in && context->input_frames < 8) {
memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
in, avctx->frame_size * avctx->channels * sizeof(*in));
context->buffered_samples += avctx->frame_size;
if (context->input_frames == 0)
context->first_pts = frame->pts;
if (context->input_frames < 7) {
context->input_frames++;
return 0;
}
in = context->frame_buffer;
}
if (stereo) {
context->lastSample[0] &= 0xFF00;
context->lastSample[1] &= 0xFF00;
}
if (context->input_frames == 7 || !in)
data_size = avctx->channels * context->buffered_samples;
else
data_size = avctx->channels * avctx->frame_size;
if ((ret = ff_alloc_packet(avpkt, ROQ_HEADER_SIZE + data_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
out = avpkt->data;
bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
bytestream_put_byte(&out, 0x10);
bytestream_put_le32(&out, data_size);
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 */
for (i = 0; i < data_size; i++)
*out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
avpkt->duration = data_size / avctx->channels;
context->input_frames++;
if (!in)
context->input_frames = FFMAX(context->input_frames, 8);
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_roq_dpcm_encoder = {
.name = "roq_dpcm",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ROQ_DPCM,
.priv_data_size = sizeof(ROQDPCMContext),
.init = roq_dpcm_encode_init,
.encode2 = roq_dpcm_encode_frame,
.close = roq_dpcm_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
};
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