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167 lines
4.1 KiB
167 lines
4.1 KiB
/* |
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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 != 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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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 roq_dpcm_encoder = { |
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"roq_dpcm", |
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CODEC_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 SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
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.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"), |
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};
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