Mercurial > libavcodec.hg
view roqaudioenc.c @ 11709:6f9b4c452130 libavcodec
Optimize decoding high freqs.
this is 10-20cpu cycles faster on duron (whole is about 50-60 cpu cylses)
I wonder why gcc isnt doing this on its own ...
author | michael |
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date | Tue, 11 May 2010 21:10:55 +0000 |
parents | 8a4984c5cacc |
children | dde20597f15e |
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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 "libavutil/intmath.h" #include "avcodec.h" #include "bytestream.h" #define ROQ_FIRST_FRAME_SIZE (735*8) #define ROQ_FRAME_SIZE 735 #define MAX_DPCM (127*127) typedef struct { short lastSample[2]; } ROQDPCMContext; 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; } 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 = 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, 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", AVMEDIA_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"), };