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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" |
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#include "internal.h" |
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#include "mathops.h" |
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#define ROQ_FRAME_SIZE 735 |
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#define ROQ_HEADER_SIZE 8 |
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#define MAX_DPCM (127*127)
|
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typedef struct ROQDPCMContext { |
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short lastSample[2]; |
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int input_frames;
int buffered_samples;
int16_t *frame_buffer; |
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int64_t first_pts; |
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} ROQDPCMContext; |
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|
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static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
av_freep(&context->frame_buffer);
return 0;
}
|
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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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int ret; |
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if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n"); |
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return AVERROR(EINVAL); |
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}
if (avctx->sample_rate != 22050) {
av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n"); |
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return AVERROR(EINVAL); |
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}
|
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avctx->frame_size = ROQ_FRAME_SIZE; |
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avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
(22050 / ROQ_FRAME_SIZE) * 8; |
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context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
sizeof(*context->frame_buffer));
if (!context->frame_buffer) {
ret = AVERROR(ENOMEM);
goto error;
} |
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context->lastSample[0] = context->lastSample[1] = 0;
return 0; |
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error:
roq_dpcm_encode_close(avctx);
return ret; |
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}
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; |
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else {
result = ff_sqrt(diff);
result += diff > result*result+result;
} |
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/* 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;
}
|
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static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr) |
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{ |
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int i, stereo, data_size, ret;
const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
uint8_t *out; |
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ROQDPCMContext *context = avctx->priv_data; |
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stereo = (avctx->channels == 2);
|
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if (!in && context->input_frames >= 8) |
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return 0;
|
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if (in && context->input_frames < 8) { |
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memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
in, avctx->frame_size * avctx->channels * sizeof(*in));
context->buffered_samples += avctx->frame_size; |
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if (context->input_frames == 0)
context->first_pts = frame->pts; |
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if (context->input_frames < 7) {
context->input_frames++;
return 0;
} |
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} |
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if (context->input_frames < 8) |
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in = context->frame_buffer;
|
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if (stereo) {
context->lastSample[0] &= 0xFF00;
context->lastSample[1] &= 0xFF00;
}
|
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if (context->input_frames == 7) |
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data_size = avctx->channels * context->buffered_samples;
else
data_size = avctx->channels * avctx->frame_size;
|
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if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size, 0)) < 0) |
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return ret;
out = avpkt->data; |
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bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
bytestream_put_byte(&out, 0x10); |
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bytestream_put_le32(&out, data_size); |
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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 */ |
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for (i = 0; i < data_size; i++)
*out++ = dpcm_predict(&context->lastSample[i & 1], *in++); |
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|
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avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
avpkt->duration = data_size / avctx->channels;
|
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context->input_frames++; |
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if (!in) |
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context->input_frames = FFMAX(context->input_frames, 8); |
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|
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*got_packet_ptr = 1;
return 0; |
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}
|
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AVCodec ff_roq_dpcm_encoder = { |
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.name = "roq_dpcm", |
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.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_ROQ_DPCM, |
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.priv_data_size = sizeof(ROQDPCMContext),
.init = roq_dpcm_encode_init, |
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.encode2 = roq_dpcm_encode_frame, |
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.close = roq_dpcm_encode_close, |
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.capabilities = AV_CODEC_CAP_DELAY, |
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE }, |
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}; |