mirror of https://github.com/FFmpeg/FFmpeg.git
* qatar/master: mss3: use standard zigzag table mss3: split DSP functions that are used in MTS2(MSS4) into separate file motion-test: do not use getopt() tcp: add initial timeout limit for incoming connections configure: Change the rdtsc check to a linker check avconv: propagate fatal errors from lavfi. lavfi: add error handling to filter_samples(). fate-run: make avconv() properly deal with multiple inputs. asplit: don't leak the input buffer. af_resample: fix request_frame() behavior. af_asyncts: fix request_frame() behavior. libx264: support aspect ratio switching matroskadec: honor error_recognition when encountering unknown elements. lavr: resampling: add support for s32p, fltp, and dblp internal sample formats lavr: resampling: add filter type and Kaiser window beta to AVOptions lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format lavr: mix: validate internal sample format in ff_audio_mix_init() Conflicts: ffmpeg.c ffplay.c libavcodec/libx264.c libavfilter/audio.c libavfilter/split.c libavformat/tcp.c tests/fate-run.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>pull/59/head
Michael Niedermayer
13 years ago
commit
f8911b987d
45 changed files with 648 additions and 339 deletions
@ -0,0 +1,114 @@ |
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/*
|
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* Common stuff for some Microsoft Screen codecs |
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* Copyright (C) 2012 Konstantin Shishkov |
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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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|
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#include <stdint.h> |
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#include "libavutil/common.h" |
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#include "mss34dsp.h" |
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static const uint8_t luma_quant[64] = { |
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16, 11, 10, 16, 24, 40, 51, 61, |
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12, 12, 14, 19, 26, 58, 60, 55, |
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14, 13, 16, 24, 40, 57, 69, 56, |
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14, 17, 22, 29, 51, 87, 80, 62, |
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18, 22, 37, 56, 68, 109, 103, 77, |
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24, 35, 55, 64, 81, 104, 113, 92, |
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49, 64, 78, 87, 103, 121, 120, 101, |
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72, 92, 95, 98, 112, 100, 103, 99 |
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}; |
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static const uint8_t chroma_quant[64] = { |
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17, 18, 24, 47, 99, 99, 99, 99, |
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18, 21, 26, 66, 99, 99, 99, 99, |
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24, 26, 56, 99, 99, 99, 99, 99, |
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47, 66, 99, 99, 99, 99, 99, 99, |
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99, 99, 99, 99, 99, 99, 99, 99, |
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99, 99, 99, 99, 99, 99, 99, 99, |
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99, 99, 99, 99, 99, 99, 99, 99, |
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99, 99, 99, 99, 99, 99, 99, 99 |
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}; |
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void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma) |
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{ |
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int i; |
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const uint8_t *qsrc = luma ? luma_quant : chroma_quant; |
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if (quality >= 50) { |
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int scale = 200 - 2 * quality; |
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for (i = 0; i < 64; i++) |
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qmat[i] = (qsrc[i] * scale + 50) / 100; |
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} else { |
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for (i = 0; i < 64; i++) |
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qmat[i] = (5000 * qsrc[i] / quality + 50) / 100; |
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} |
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} |
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#define DCT_TEMPLATE(blk, step, SOP, shift) \ |
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const int t0 = -39409 * blk[7 * step] - 58980 * blk[1 * step]; \
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const int t1 = 39410 * blk[1 * step] - 58980 * blk[7 * step]; \
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const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
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const int t3 = 33410 * blk[3 * step] - 167963 * blk[5 * step]; \
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const int t4 = blk[3 * step] + blk[7 * step]; \
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const int t5 = blk[1 * step] + blk[5 * step]; \
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const int t6 = 77062 * t4 + 51491 * t5; \
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const int t7 = 77062 * t5 - 51491 * t4; \
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const int t8 = 35470 * blk[2 * step] - 85623 * blk[6 * step]; \
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const int t9 = 35470 * blk[6 * step] + 85623 * blk[2 * step]; \
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const int tA = SOP(blk[0 * step] - blk[4 * step]); \
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const int tB = SOP(blk[0 * step] + blk[4 * step]); \
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\
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blk[0 * step] = ( t1 + t6 + t9 + tB) >> shift; \
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blk[1 * step] = ( t3 + t7 + t8 + tA) >> shift; \
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blk[2 * step] = ( t2 + t6 - t8 + tA) >> shift; \
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blk[3 * step] = ( t0 + t7 - t9 + tB) >> shift; \
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blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift; \
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blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift; \
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blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift; \
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blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift; \
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#define SOP_ROW(a) ((a) << 16) + 0x2000 |
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#define SOP_COL(a) ((a + 32) << 16) |
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void ff_mss34_dct_put(uint8_t *dst, int stride, int *block) |
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{ |
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int i, j; |
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int *ptr; |
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ptr = block; |
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for (i = 0; i < 8; i++) { |
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DCT_TEMPLATE(ptr, 1, SOP_ROW, 13); |
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ptr += 8; |
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} |
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ptr = block; |
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for (i = 0; i < 8; i++) { |
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DCT_TEMPLATE(ptr, 8, SOP_COL, 22); |
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ptr++; |
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} |
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ptr = block; |
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for (j = 0; j < 8; j++) { |
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for (i = 0; i < 8; i++) |
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dst[i] = av_clip_uint8(ptr[i] + 128); |
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dst += stride; |
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ptr += 8; |
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} |
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} |
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@ -0,0 +1,45 @@ |
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/*
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* Common stuff for some Microsoft Screen codecs |
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* Copyright (C) 2012 Konstantin Shishkov |
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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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#ifndef AVCODEC_MSS34DSP_H |
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#define AVCODEC_MSS34DSP_H |
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#include <stdint.h> |
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/**
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* Generate quantisation matrix for given quality. |
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* |
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* @param qmat destination matrix |
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* @param quality quality setting (1-100) |
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* @param luma generate quantisation matrix for luma or chroma |
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*/ |
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void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma); |
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/**
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* Transform and output DCT block. |
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* |
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* @param dst output plane |
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* @param stride output plane stride |
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* @param block block to transform and output |
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*/ |
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void ff_mss34_dct_put(uint8_t *dst, int stride, int *block); |
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#endif /* AVCODEC_MSS34DSP_H */ |
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@ -0,0 +1,102 @@ |
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/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> |
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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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#if defined(CONFIG_RESAMPLE_DBL) |
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#define SET_TYPE(func) func ## _dbl |
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#define FELEM double |
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#define FELEM2 double |
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#define FELEML double |
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#define OUT(d, v) d = v |
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#define DBL_TO_FELEM(d, v) d = v |
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#elif defined(CONFIG_RESAMPLE_FLT) |
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#define SET_TYPE(func) func ## _flt |
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#define FELEM float |
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#define FELEM2 float |
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#define FELEML float |
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#define OUT(d, v) d = v |
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#define DBL_TO_FELEM(d, v) d = v |
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#elif defined(CONFIG_RESAMPLE_S32) |
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#define SET_TYPE(func) func ## _s32 |
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#define FELEM int32_t |
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#define FELEM2 int64_t |
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#define FELEML int64_t |
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#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30) |
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#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30))); |
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#else |
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#define SET_TYPE(func) func ## _s16 |
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#define FELEM int16_t |
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#define FELEM2 int32_t |
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#define FELEML int64_t |
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#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15) |
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#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15))) |
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#endif |
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static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter, |
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void *dst0, int dst_index, const void *src0, |
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int src_size, int index, int frac) |
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{ |
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FELEM *dst = dst0; |
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const FELEM *src = src0; |
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if (no_filter) { |
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dst[dst_index] = src[index]; |
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} else { |
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int i; |
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int sample_index = index >> c->phase_shift; |
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FELEM2 val = 0; |
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FELEM *filter = ((FELEM *)c->filter_bank) + |
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c->filter_length * (index & c->phase_mask); |
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if (sample_index < 0) { |
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for (i = 0; i < c->filter_length; i++) |
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val += src[FFABS(sample_index + i) % src_size] * |
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(FELEM2)filter[i]; |
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} else if (c->linear) { |
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FELEM2 v2 = 0; |
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for (i = 0; i < c->filter_length; i++) { |
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val += src[abs(sample_index + i)] * (FELEM2)filter[i]; |
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v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length]; |
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} |
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val += (v2 - val) * (FELEML)frac / c->src_incr; |
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} else { |
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for (i = 0; i < c->filter_length; i++) |
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val += src[sample_index + i] * (FELEM2)filter[i]; |
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} |
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OUT(dst[dst_index], val); |
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} |
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} |
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static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase, |
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int tap_count) |
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{ |
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int i; |
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FELEM *filter = ((FELEM *)filter0) + phase * tap_count; |
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for (i = 0; i < tap_count; i++) { |
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DBL_TO_FELEM(filter[i], tab[i]); |
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} |
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} |
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#undef SET_TYPE |
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#undef FELEM |
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#undef FELEM2 |
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#undef FELEML |
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#undef OUT |
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#undef DBL_TO_FELEM |
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