diff --git a/ffserver.c b/ffserver.c index 46f89fb743..000350cbf3 100644 --- a/ffserver.c +++ b/ffserver.c @@ -522,6 +522,7 @@ static int socket_open_listen(struct sockaddr_in *my_addr) tmp = 1; setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp)); + my_addr->sin_family = AF_INET; if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) { char bindmsg[32]; snprintf(bindmsg, sizeof(bindmsg), "bind(port %d)", ntohs(my_addr->sin_port)); diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c index cf609e74f1..a0cd5cc77e 100644 --- a/libavcodec/adpcm.c +++ b/libavcodec/adpcm.c @@ -42,31 +42,35 @@ * Features and limitations: * * Reference documents: - * http://www.pcisys.net/~melanson/codecs/simpleaudio.html - * http://www.geocities.com/SiliconValley/8682/aud3.txt - * http://openquicktime.sourceforge.net/plugins.htm - * XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html - * http://www.cs.ucla.edu/~leec/mediabench/applications.html - * SoX source code http://home.sprynet.com/~cbagwell/sox.html + * http://wiki.multimedia.cx/index.php?title=Category:ADPCM_Audio_Codecs + * http://www.pcisys.net/~melanson/codecs/simpleaudio.html [dead] + * http://www.geocities.com/SiliconValley/8682/aud3.txt [dead] + * http://openquicktime.sourceforge.net/ + * XAnim sources (xa_codec.c) http://xanim.polter.net/ + * http://www.cs.ucla.edu/~leec/mediabench/applications.html [dead] + * SoX source code http://sox.sourceforge.net/ * * CD-ROM XA: - * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html - * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html + * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html [dead] + * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html [dead] * readstr http://www.geocities.co.jp/Playtown/2004/ */ /* These are for CD-ROM XA ADPCM */ static const int xa_adpcm_table[5][2] = { - { 0, 0 }, - { 60, 0 }, - { 115, -52 }, - { 98, -55 }, - { 122, -60 } + { 0, 0 }, + { 60, 0 }, + { 115, -52 }, + { 98, -55 }, + { 122, -60 } }; static const int ea_adpcm_table[] = { - 0, 240, 460, 392, 0, 0, -208, -220, 0, 1, - 3, 4, 7, 8, 10, 11, 0, -1, -3, -4 + 0, 240, 460, 392, + 0, 0, -208, -220, + 0, 1, 3, 4, + 7, 8, 10, 11, + 0, -1, -3, -4 }; // padded to zero where table size is less then 16 @@ -336,27 +340,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, ADPCMDecodeContext *c = avctx->priv_data; ADPCMChannelStatus *cs; int n, m, channel, i; - int block_predictor[2]; short *samples; short *samples_end; const uint8_t *src; int st; /* stereo */ - - /* DK3 ADPCM accounting variables */ - unsigned char last_byte = 0; - unsigned char nibble; - int decode_top_nibble_next = 0; - int diff_channel; - - /* EA ADPCM state variables */ uint32_t samples_in_chunk; - int32_t previous_left_sample, previous_right_sample; - int32_t current_left_sample, current_right_sample; - int32_t next_left_sample, next_right_sample; - int32_t coeff1l, coeff2l, coeff1r, coeff2r; - uint8_t shift_left, shift_right; int count1, count2; - int coeff[2][2], shift[2];//used in EA MAXIS ADPCM if (!buf_size) return 0; @@ -376,7 +365,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, switch(avctx->codec->id) { case CODEC_ID_ADPCM_IMA_QT: - n = buf_size - 2*avctx->channels; + /* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples). + Channel data is interleaved per-chunk. */ + if (buf_size / 34 < avctx->channels) { + av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); + return AVERROR(EINVAL); + } for (channel = 0; channel < avctx->channels; channel++) { int16_t predictor; int step_index; @@ -409,7 +403,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, samples = (short*)data + channel; - for(m=32; n>0 && m>0; n--, m--) { /* in QuickTime, IMA is encoded by chuncks of 34 bytes (=64 samples) */ + for (m = 0; m < 32; m++) { *samples = adpcm_ima_qt_expand_nibble(cs, src[0] & 0x0F, 3); samples += avctx->channels; *samples = adpcm_ima_qt_expand_nibble(cs, src[0] >> 4 , 3); @@ -439,60 +433,66 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } while(src < buf + buf_size){ - for(m=0; m<4; m++){ - for(i=0; i<=st; i++) - *samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] & 0x0F, 3); - for(i=0; i<=st; i++) - *samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] >> 4 , 3); - src++; + for (i = 0; i < avctx->channels; i++) { + cs = &c->status[i]; + for (m = 0; m < 4; m++) { + uint8_t v = *src++; + *samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 3); + samples += avctx->channels; + *samples = adpcm_ima_expand_nibble(cs, v >> 4 , 3); + samples += avctx->channels; + } + samples -= 8 * avctx->channels - 1; } - src += 4*st; + samples += 7 * avctx->channels; } break; case CODEC_ID_ADPCM_4XM: - cs = &(c->status[0]); - c->status[0].predictor= (int16_t)bytestream_get_le16(&src); - if(st){ - c->status[1].predictor= (int16_t)bytestream_get_le16(&src); + for (i = 0; i < avctx->channels; i++) + c->status[i].predictor= (int16_t)bytestream_get_le16(&src); + + for (i = 0; i < avctx->channels; i++) { + c->status[i].step_index= (int16_t)bytestream_get_le16(&src); + c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88); } - c->status[0].step_index= (int16_t)bytestream_get_le16(&src); - if(st){ - c->status[1].step_index= (int16_t)bytestream_get_le16(&src); - } - if (cs->step_index < 0) cs->step_index = 0; - if (cs->step_index > 88) cs->step_index = 88; m= (buf_size - (src - buf))>>st; - for(i=0; istatus[0], src[i] & 0x0F, 4); - if (st) - *samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] & 0x0F, 4); - *samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] >> 4, 4); - if (st) - *samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] >> 4, 4); + + for (i = 0; i < avctx->channels; i++) { + samples = (short*)data + i; + cs = &c->status[i]; + for (n = 0; n < m; n++) { + uint8_t v = *src++; + *samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 4); + samples += avctx->channels; + *samples = adpcm_ima_expand_nibble(cs, v >> 4 , 4); + samples += avctx->channels; + } } - - src += m<channels - 1); break; case CODEC_ID_ADPCM_MS: + { + int block_predictor; + if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; n = buf_size - 7 * avctx->channels; if (n < 0) return -1; - block_predictor[0] = av_clip(*src++, 0, 6); - block_predictor[1] = 0; - if (st) - block_predictor[1] = av_clip(*src++, 0, 6); + + block_predictor = av_clip(*src++, 0, 6); + c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor]; + c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor]; + if (st) { + block_predictor = av_clip(*src++, 0, 6); + c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor]; + c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor]; + } c->status[0].idelta = (int16_t)bytestream_get_le16(&src); if (st){ c->status[1].idelta = (int16_t)bytestream_get_le16(&src); } - c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[0]]; - c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[0]]; - c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[1]]; - c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[1]]; c->status[0].sample1 = bytestream_get_le16(&src); if (st) c->status[1].sample1 = bytestream_get_le16(&src); @@ -509,39 +509,37 @@ static int adpcm_decode_frame(AVCodecContext *avctx, src ++; } break; + } case CODEC_ID_ADPCM_IMA_DK4: if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; - c->status[0].predictor = (int16_t)bytestream_get_le16(&src); - c->status[0].step_index = *src++; - src++; - *samples++ = c->status[0].predictor; - if (st) { - c->status[1].predictor = (int16_t)bytestream_get_le16(&src); - c->status[1].step_index = *src++; - src++; - *samples++ = c->status[1].predictor; + n = buf_size - 4 * avctx->channels; + if (n < 0) { + av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); + return AVERROR(EINVAL); } - while (src < buf + buf_size) { - - /* take care of the top nibble (always left or mono channel) */ - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4, 3); - - /* take care of the bottom nibble, which is right sample for - * stereo, or another mono sample */ - if (st) - *samples++ = adpcm_ima_expand_nibble(&c->status[1], - src[0] & 0x0F, 3); - else - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] & 0x0F, 3); + for (channel = 0; channel < avctx->channels; channel++) { + cs = &c->status[channel]; + cs->predictor = (int16_t)bytestream_get_le16(&src); + cs->step_index = *src++; src++; + *samples++ = cs->predictor; + } + while (n-- > 0) { + uint8_t v = *src++; + *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3); + *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3); } break; case CODEC_ID_ADPCM_IMA_DK3: + { + unsigned char last_byte = 0; + unsigned char nibble; + int decode_top_nibble_next = 0; + int diff_channel; + if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; @@ -586,50 +584,41 @@ static int adpcm_decode_frame(AVCodecContext *avctx, *samples++ = c->status[0].predictor - c->status[1].predictor; } break; + } case CODEC_ID_ADPCM_IMA_ISS: - c->status[0].predictor = (int16_t)AV_RL16(src + 0); - c->status[0].step_index = src[2]; - src += 4; - if(st) { - c->status[1].predictor = (int16_t)AV_RL16(src + 0); - c->status[1].step_index = src[2]; - src += 4; + n = buf_size - 4 * avctx->channels; + if (n < 0) { + av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); + return AVERROR(EINVAL); } - while (src < buf + buf_size) { - - if (st) { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[1], - src[0] & 0x0F, 3); - } else { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] & 0x0F, 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - } - + for (channel = 0; channel < avctx->channels; channel++) { + cs = &c->status[channel]; + cs->predictor = (int16_t)bytestream_get_le16(&src); + cs->step_index = *src++; src++; } + + while (n-- > 0) { + uint8_t v1, v2; + uint8_t v = *src++; + /* nibbles are swapped for mono */ + if (st) { + v1 = v >> 4; + v2 = v & 0x0F; + } else { + v2 = v >> 4; + v1 = v & 0x0F; + } + *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v1, 3); + *samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3); + } break; case CODEC_ID_ADPCM_IMA_WS: - /* no per-block initialization; just start decoding the data */ while (src < buf + buf_size) { - - if (st) { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[1], - src[0] & 0x0F, 3); - } else { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] & 0x0F, 3); - } - - src++; + uint8_t v = *src++; + *samples++ = adpcm_ima_expand_nibble(&c->status[0], v >> 4 , 3); + *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3); } break; case CODEC_ID_ADPCM_XA: @@ -668,6 +657,13 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } break; case CODEC_ID_ADPCM_EA: + { + int32_t previous_left_sample, previous_right_sample; + int32_t current_left_sample, current_right_sample; + int32_t next_left_sample, next_right_sample; + int32_t coeff1l, coeff2l, coeff1r, coeff2r; + uint8_t shift_left, shift_right; + /* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces, each coding 28 stereo samples. */ if (buf_size < 12) { @@ -721,7 +717,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx, src += 2; // Skip terminating 0x0000 break; + } case CODEC_ID_ADPCM_EA_MAXIS_XA: + { + int coeff[2][2], shift[2]; + for(channel = 0; channel < avctx->channels; channel++) { for (i=0; i<2; i++) coeff[channel][i] = ea_adpcm_table[(*src >> 4) + 4*i]; @@ -743,6 +743,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, src+=avctx->channels; } break; + } case CODEC_ID_ADPCM_EA_R1: case CODEC_ID_ADPCM_EA_R2: case CODEC_ID_ADPCM_EA_R3: { @@ -885,18 +886,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx, break; case CODEC_ID_ADPCM_CT: while (src < buf + buf_size) { - if (st) { - *samples++ = adpcm_ct_expand_nibble(&c->status[0], - src[0] >> 4); - *samples++ = adpcm_ct_expand_nibble(&c->status[1], - src[0] & 0x0F); - } else { - *samples++ = adpcm_ct_expand_nibble(&c->status[0], - src[0] >> 4); - *samples++ = adpcm_ct_expand_nibble(&c->status[0], - src[0] & 0x0F); - } - src++; + uint8_t v = *src++; + *samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4 ); + *samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F); } break; case CODEC_ID_ADPCM_SBPRO_4: @@ -1004,18 +996,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } case CODEC_ID_ADPCM_YAMAHA: while (src < buf + buf_size) { - if (st) { - *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], - src[0] & 0x0F); - *samples++ = adpcm_yamaha_expand_nibble(&c->status[1], - src[0] >> 4 ); - } else { - *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], - src[0] & 0x0F); - *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], - src[0] >> 4 ); - } - src++; + uint8_t v = *src++; + *samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F); + *samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4 ); } break; case CODEC_ID_ADPCM_THP: diff --git a/libavcodec/adpcm_data.c b/libavcodec/adpcm_data.c index 3654a8d67d..f19d622d3b 100644 --- a/libavcodec/adpcm_data.c +++ b/libavcodec/adpcm_data.c @@ -38,14 +38,14 @@ const int8_t ff_adpcm_index_table[16] = { * this table, but such deviations are negligible: */ const int16_t ff_adpcm_step_table[89] = { - 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, - 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, - 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, - 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, - 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, - 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, - 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, - 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 }; @@ -53,18 +53,18 @@ const int16_t ff_adpcm_step_table[89] = { /* ff_adpcm_AdaptationTable[], ff_adpcm_AdaptCoeff1[], and ff_adpcm_AdaptCoeff2[] are from libsndfile */ const int16_t ff_adpcm_AdaptationTable[] = { - 230, 230, 230, 230, 307, 409, 512, 614, - 768, 614, 512, 409, 307, 230, 230, 230 + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 }; /** Divided by 4 to fit in 8-bit integers */ const uint8_t ff_adpcm_AdaptCoeff1[] = { - 64, 128, 0, 48, 60, 115, 98 + 64, 128, 0, 48, 60, 115, 98 }; /** Divided by 4 to fit in 8-bit integers */ const int8_t ff_adpcm_AdaptCoeff2[] = { - 0, -64, 0, 16, 0, -52, -58 + 0, -64, 0, 16, 0, -52, -58 }; const int16_t ff_adpcm_yamaha_indexscale[] = { @@ -73,6 +73,6 @@ const int16_t ff_adpcm_yamaha_indexscale[] = { }; const int8_t ff_adpcm_yamaha_difflookup[] = { - 1, 3, 5, 7, 9, 11, 13, 15, + 1, 3, 5, 7, 9, 11, 13, 15, -1, -3, -5, -7, -9, -11, -13, -15 }; diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c index 14978f0218..c193f5c7ef 100644 --- a/libavcodec/adpcmenc.c +++ b/libavcodec/adpcmenc.c @@ -32,13 +32,7 @@ * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood) * by Mike Melanson (melanson@pcisys.net) * - * Reference documents: - * http://www.pcisys.net/~melanson/codecs/simpleaudio.html - * http://www.geocities.com/SiliconValley/8682/aud3.txt - * http://openquicktime.sourceforge.net/plugins.htm - * XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html - * http://www.cs.ucla.edu/~leec/mediabench/applications.html - * SoX source code http://home.sprynet.com/~cbagwell/sox.html + * See ADPCM decoder reference documents for codec information. */ typedef struct TrellisPath { diff --git a/libavcodec/arm/dca.h b/libavcodec/arm/dca.h new file mode 100644 index 0000000000..c4c024a36a --- /dev/null +++ b/libavcodec/arm/dca.h @@ -0,0 +1,49 @@ +/* + * Copyright (c) 2011 Mans Rullgard + * + * 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 + */ + +#ifndef AVCODEC_ARM_DCA_H +#define AVCODEC_ARM_DCA_H + +#include +#include "config.h" + +#if HAVE_NEON && HAVE_INLINE_ASM + +#define int8x8_fmul_int32 int8x8_fmul_int32 +static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale) +{ + __asm__ ("vcvt.f32.s32 %2, %2, #4 \n" + "vld1.8 {d0}, [%1,:64] \n" + "vmovl.s8 q0, d0 \n" + "vmovl.s16 q1, d1 \n" + "vmovl.s16 q0, d0 \n" + "vcvt.f32.s32 q0, q0 \n" + "vcvt.f32.s32 q1, q1 \n" + "vmul.f32 q0, q0, %y2 \n" + "vmul.f32 q1, q1, %y2 \n" + "vst1.32 {q0-q1}, [%m0,:128] \n" + : "=Um"(*(float (*)[8])dst) + : "r"(src), "x"(scale) + : "d0", "d1", "d2", "d3"); +} + +#endif + +#endif /* AVCODEC_ARM_DCA_H */ diff --git a/libavcodec/dca.c b/libavcodec/dca.c index ace89d436f..8c3cc4b720 100644 --- a/libavcodec/dca.c +++ b/libavcodec/dca.c @@ -42,6 +42,10 @@ #include "dcadsp.h" #include "fmtconvert.h" +#if ARCH_ARM +# include "arm/dca.h" +#endif + //#define TRACE #define DCA_PRIM_CHANNELS_MAX (7) @@ -320,7 +324,7 @@ typedef struct { int lfe_scale_factor; /* Subband samples history (for ADPCM) */ - float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; + DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512]; DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32]; int hist_index[DCA_PRIM_CHANNELS_MAX]; @@ -1057,6 +1061,16 @@ static int decode_blockcode(int code, int levels, int *values) static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; +#ifndef int8x8_fmul_int32 +static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale) +{ + float fscale = scale / 16.0; + int i; + for (i = 0; i < 8; i++) + dst[i] = src[i] * fscale; +} +#endif + static int dca_subsubframe(DCAContext * s, int base_channel, int block_index) { int k, l; @@ -1161,19 +1175,16 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index) for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { /* 1 vector -> 32 samples but we only need the 8 samples * for this subsubframe. */ - int m; + int hfvq = s->high_freq_vq[k][l]; if (!s->debug_flag & 0x01) { av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); s->debug_flag |= 0x01; } - for (m = 0; m < 8; m++) { - subband_samples[k][l][m] = - high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 + - m] - * (float) s->scale_factor[k][l][0] / 16.0; - } + int8x8_fmul_int32(subband_samples[k][l], + &high_freq_vq[hfvq][subsubframe * 8], + s->scale_factor[k][l][0]); } } diff --git a/libavcodec/dcadata.h b/libavcodec/dcadata.h index e8a31fd0a1..02dbb0fe54 100644 --- a/libavcodec/dcadata.h +++ b/libavcodec/dcadata.h @@ -4224,7 +4224,7 @@ static const float lossless_quant_d[32] = { /* Vector quantization tables */ -static const int8_t high_freq_vq[1024][32] = +DECLARE_ALIGNED(8, static const int8_t, high_freq_vq)[1024][32] = { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, diff --git a/libavcodec/dpcm.c b/libavcodec/dpcm.c index d9c15246e9..8f6cd8e115 100644 --- a/libavcodec/dpcm.c +++ b/libavcodec/dpcm.c @@ -39,17 +39,16 @@ #include "libavutil/intreadwrite.h" #include "avcodec.h" +#include "bytestream.h" typedef struct DPCMContext { int channels; - short roq_square_array[256]; - long sample[2];//for SOL_DPCM - const int *sol_table;//for SOL_DPCM + int16_t roq_square_array[256]; + int sample[2]; ///< previous sample (for SOL_DPCM) + const int8_t *sol_table; ///< delta table for SOL_DPCM } DPCMContext; -#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; - -static const int interplay_delta_table[] = { +static const int16_t interplay_delta_table[] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, @@ -85,15 +84,17 @@ static const int interplay_delta_table[] = { }; -static const int sol_table_old[16] = - { 0x0, 0x1, 0x2 , 0x3, 0x6, 0xA, 0xF, 0x15, - -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0}; +static const int8_t sol_table_old[16] = { + 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, + -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0 +}; -static const int sol_table_new[16] = - { 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, - 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15}; +static const int8_t sol_table_new[16] = { + 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, + 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15 +}; -static const int sol_table_16[128] = { +static const int16_t sol_table_16[128] = { 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080, 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120, 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0, @@ -110,12 +111,15 @@ static const int sol_table_16[128] = { }; - static av_cold int dpcm_decode_init(AVCodecContext *avctx) { DPCMContext *s = avctx->priv_data; int i; - short square; + + if (avctx->channels < 1 || avctx->channels > 2) { + av_log(avctx, AV_LOG_INFO, "invalid number of channels\n"); + return AVERROR(EINVAL); + } s->channels = avctx->channels; s->sample[0] = s->sample[1] = 0; @@ -125,25 +129,23 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx) case CODEC_ID_ROQ_DPCM: /* initialize square table */ for (i = 0; i < 128; i++) { - square = i * i; - s->roq_square_array[i] = square; + int16_t square = i * i; + s->roq_square_array[i ] = square; s->roq_square_array[i + 128] = -square; } break; - case CODEC_ID_SOL_DPCM: switch(avctx->codec_tag){ case 1: - s->sol_table=sol_table_old; + s->sol_table = sol_table_old; s->sample[0] = s->sample[1] = 0x80; break; case 2: - s->sol_table=sol_table_new; + s->sol_table = sol_table_new; s->sample[0] = s->sample[1] = 0x80; break; case 3: - s->sol_table=sol_table_16; break; default: av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n"); @@ -155,146 +157,160 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx) break; } - avctx->sample_fmt = AV_SAMPLE_FMT_S16; + if (avctx->codec->id == CODEC_ID_SOL_DPCM && avctx->codec_tag != 3) + avctx->sample_fmt = AV_SAMPLE_FMT_U8; + else + avctx->sample_fmt = AV_SAMPLE_FMT_S16; + return 0; } -static int dpcm_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, + +static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; + const uint8_t *buf_end = buf + buf_size; DPCMContext *s = avctx->priv_data; - int in, out = 0; + int out = 0; int predictor[2]; - int channel_number = 0; - short *output_samples = data; - int shift[2]; - unsigned char byte; - short diff; + int ch = 0; + int stereo = s->channels - 1; + int16_t *output_samples = data; if (!buf_size) return 0; - // almost every DPCM variant expands one byte of data into two - if(*data_size/2 < buf_size) - return -1; + /* calculate output size */ + switch(avctx->codec->id) { + case CODEC_ID_ROQ_DPCM: + out = buf_size - 8; + break; + case CODEC_ID_INTERPLAY_DPCM: + out = buf_size - 6 - s->channels; + break; + case CODEC_ID_XAN_DPCM: + out = buf_size - 2 * s->channels; + break; + case CODEC_ID_SOL_DPCM: + if (avctx->codec_tag != 3) + out = buf_size * 2; + else + out = buf_size; + break; + } + out *= av_get_bytes_per_sample(avctx->sample_fmt); + if (out < 0) { + av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); + return AVERROR(EINVAL); + } + if (*data_size < out) { + av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n"); + return AVERROR(EINVAL); + } switch(avctx->codec->id) { case CODEC_ID_ROQ_DPCM: - if (s->channels == 1) - predictor[0] = AV_RL16(&buf[6]); - else { - predictor[0] = buf[7] << 8; - predictor[1] = buf[6] << 8; + buf += 6; + + if (stereo) { + predictor[1] = (int16_t)(bytestream_get_byte(&buf) << 8); + predictor[0] = (int16_t)(bytestream_get_byte(&buf) << 8); + } else { + predictor[0] = (int16_t)bytestream_get_le16(&buf); } - SE_16BIT(predictor[0]); - SE_16BIT(predictor[1]); /* decode the samples */ - for (in = 8, out = 0; in < buf_size; in++, out++) { - predictor[channel_number] += s->roq_square_array[buf[in]]; - predictor[channel_number] = av_clip_int16(predictor[channel_number]); - output_samples[out] = predictor[channel_number]; + while (buf < buf_end) { + predictor[ch] += s->roq_square_array[*buf++]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } break; case CODEC_ID_INTERPLAY_DPCM: - in = 6; /* skip over the stream mask and stream length */ - predictor[0] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[0]) - output_samples[out++] = predictor[0]; - if (s->channels == 2) { - predictor[1] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[1]) - output_samples[out++] = predictor[1]; + buf += 6; /* skip over the stream mask and stream length */ + + for (ch = 0; ch < s->channels; ch++) { + predictor[ch] = (int16_t)bytestream_get_le16(&buf); + *output_samples++ = predictor[ch]; } - while (in < buf_size) { - predictor[channel_number] += interplay_delta_table[buf[in++]]; - predictor[channel_number] = av_clip_int16(predictor[channel_number]); - output_samples[out++] = predictor[channel_number]; + ch = 0; + while (buf < buf_end) { + predictor[ch] += interplay_delta_table[*buf++]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } - break; case CODEC_ID_XAN_DPCM: - in = 0; - shift[0] = shift[1] = 4; - predictor[0] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[0]); - if (s->channels == 2) { - predictor[1] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[1]); - } + { + int shift[2] = { 4, 4 }; - while (in < buf_size) { - byte = buf[in++]; - diff = (byte & 0xFC) << 8; - if ((byte & 0x03) == 3) - shift[channel_number]++; + for (ch = 0; ch < s->channels; ch++) + predictor[ch] = (int16_t)bytestream_get_le16(&buf); + + ch = 0; + while (buf < buf_end) { + uint8_t n = *buf++; + int16_t diff = (n & 0xFC) << 8; + if ((n & 0x03) == 3) + shift[ch]++; else - shift[channel_number] -= (2 * (byte & 3)); + shift[ch] -= (2 * (n & 3)); /* saturate the shifter to a lower limit of 0 */ - if (shift[channel_number] < 0) - shift[channel_number] = 0; + if (shift[ch] < 0) + shift[ch] = 0; - diff >>= shift[channel_number]; - predictor[channel_number] += diff; + diff >>= shift[ch]; + predictor[ch] += diff; - predictor[channel_number] = av_clip_int16(predictor[channel_number]); - output_samples[out++] = predictor[channel_number]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } break; + } case CODEC_ID_SOL_DPCM: - in = 0; if (avctx->codec_tag != 3) { - if(*data_size/4 < buf_size) - return -1; - while (in < buf_size) { - int n1, n2; - n1 = (buf[in] >> 4) & 0xF; - n2 = buf[in++] & 0xF; - s->sample[0] += s->sol_table[n1]; - if (s->sample[0] < 0) s->sample[0] = 0; - if (s->sample[0] > 255) s->sample[0] = 255; - output_samples[out++] = (s->sample[0] - 128) << 8; - s->sample[s->channels - 1] += s->sol_table[n2]; - if (s->sample[s->channels - 1] < 0) s->sample[s->channels - 1] = 0; - if (s->sample[s->channels - 1] > 255) s->sample[s->channels - 1] = 255; - output_samples[out++] = (s->sample[s->channels - 1] - 128) << 8; + uint8_t *output_samples_u8 = data; + while (buf < buf_end) { + uint8_t n = *buf++; + + s->sample[0] += s->sol_table[n >> 4]; + s->sample[0] = av_clip_uint8(s->sample[0]); + *output_samples_u8++ = s->sample[0]; + + s->sample[stereo] += s->sol_table[n & 0x0F]; + s->sample[stereo] = av_clip_uint8(s->sample[stereo]); + *output_samples_u8++ = s->sample[stereo]; } } else { - while (in < buf_size) { - int n; - n = buf[in++]; - if (n & 0x80) s->sample[channel_number] -= s->sol_table[n & 0x7F]; - else s->sample[channel_number] += s->sol_table[n & 0x7F]; - s->sample[channel_number] = av_clip_int16(s->sample[channel_number]); - output_samples[out++] = s->sample[channel_number]; + while (buf < buf_end) { + uint8_t n = *buf++; + if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F]; + else s->sample[ch] += sol_table_16[n & 0x7F]; + s->sample[ch] = av_clip_int16(s->sample[ch]); + *output_samples++ = s->sample[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } } break; } - *data_size = out * sizeof(short); + *data_size = out; return buf_size; } @@ -310,6 +326,6 @@ AVCodec ff_ ## name_ ## _decoder = { \ } DPCM_DECODER(CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay"); -DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ"); -DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol"); -DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan"); +DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ"); +DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol"); +DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan"); diff --git a/libavcodec/dxva2_h264.c b/libavcodec/dxva2_h264.c index 3d5af31757..a707e63a54 100644 --- a/libavcodec/dxva2_h264.c +++ b/libavcodec/dxva2_h264.c @@ -162,18 +162,18 @@ static void fill_scaling_lists(struct dxva_context *ctx, const H264Context *h, D for (j = 0; j < 16; j++) qm->bScalingLists4x4[i][j] = h->pps.scaling_matrix4[i][j]; - for (j = 0; j < 64; j++) { - qm->bScalingLists8x8[0][j] = h->pps.scaling_matrix8[0][j]; - qm->bScalingLists8x8[1][j] = h->pps.scaling_matrix8[3][j]; + for (i = 0; i < 64; i++) { + qm->bScalingLists8x8[0][i] = h->pps.scaling_matrix8[0][i]; + qm->bScalingLists8x8[1][i] = h->pps.scaling_matrix8[3][i]; } } else { for (i = 0; i < 6; i++) for (j = 0; j < 16; j++) qm->bScalingLists4x4[i][j] = h->pps.scaling_matrix4[i][zigzag_scan[j]]; - for (j = 0; j < 64; j++) { - qm->bScalingLists8x8[0][j] = h->pps.scaling_matrix8[0][ff_zigzag_direct[j]]; - qm->bScalingLists8x8[1][j] = h->pps.scaling_matrix8[3][ff_zigzag_direct[j]]; + for (i = 0; i < 64; i++) { + qm->bScalingLists8x8[0][i] = h->pps.scaling_matrix8[0][ff_zigzag_direct[i]]; + qm->bScalingLists8x8[1][i] = h->pps.scaling_matrix8[3][ff_zigzag_direct[i]]; } } } diff --git a/libavcodec/proresdec_lgpl.c b/libavcodec/proresdec_lgpl.c index a3d762bf08..89e5582a37 100644 --- a/libavcodec/proresdec_lgpl.c +++ b/libavcodec/proresdec_lgpl.c @@ -427,13 +427,13 @@ static inline void decode_ac_coeffs(GetBitContext *gb, DCTELEM *out, lev_cb_index = lev_to_cb_index[FFMIN(level, 9)]; bits_left = get_bits_left(gb); - if (bits_left <= 8 && !show_bits(gb, bits_left)) + if (bits_left <= 0 || (bits_left <= 8 && !show_bits(gb, bits_left))) return; run = decode_vlc_codeword(gb, ac_codebook[run_cb_index]); bits_left = get_bits_left(gb); - if (bits_left <= 8 && !show_bits(gb, bits_left)) + if (bits_left <= 0 || (bits_left <= 8 && !show_bits(gb, bits_left))) return; level = decode_vlc_codeword(gb, ac_codebook[lev_cb_index]) + 1; diff --git a/libavcodec/utils.c b/libavcodec/utils.c index 0ac81ba333..8af4c338fd 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -823,6 +823,11 @@ int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *sa avctx->pkt = avpkt; + if (!avpkt->data && avpkt->size) { + av_log(avctx, AV_LOG_ERROR, "invalid packet: NULL data, size != 0\n"); + return AVERROR(EINVAL); + } + if((avctx->codec->capabilities & CODEC_CAP_DELAY) || avpkt->size){ //FIXME remove the check below _after_ ensuring that all audio check that the available space is enough if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){ diff --git a/libavformat/latmenc.c b/libavformat/latmenc.c index 9299cc3a7b..c461ac39dc 100644 --- a/libavformat/latmenc.c +++ b/libavformat/latmenc.c @@ -120,7 +120,7 @@ static int latm_write_frame_header(AVFormatContext *s, PutBitContext *bs) } put_bits(bs, 3, 0); /* frameLengthType */ - put_bits(bs, 8, 0); /* latmBufferFullness */ + put_bits(bs, 8, 0xff); /* latmBufferFullness */ put_bits(bs, 1, 0); /* otherDataPresent */ put_bits(bs, 1, 0); /* crcCheckPresent */ diff --git a/libavformat/mpeg.c b/libavformat/mpeg.c index ad2358124e..6fb1c5c31b 100644 --- a/libavformat/mpeg.c +++ b/libavformat/mpeg.c @@ -49,6 +49,10 @@ static int check_pes(uint8_t *p, uint8_t *end){ return pes1||pes2; } +static int check_pack_header(const uint8_t *buf) { + return (buf[1] & 0xC0) == 0x40 || (buf[1] & 0xF0) == 0x20; +} + static int mpegps_probe(AVProbeData *p) { uint32_t code= -1; @@ -61,9 +65,10 @@ static int mpegps_probe(AVProbeData *p) if ((code & 0xffffff00) == 0x100) { int len= p->buf[i+1] << 8 | p->buf[i+2]; int pes= check_pes(p->buf+i, p->buf+p->buf_size); + int pack = check_pack_header(p->buf+i); if(code == SYSTEM_HEADER_START_CODE) sys++; - else if(code == PACK_START_CODE) pspack++; + else if(code == PACK_START_CODE && pack) pspack++; else if((code & 0xf0) == VIDEO_ID && pes) vid++; // skip pes payload to avoid start code emulation for private // and audio streams diff --git a/libavformat/utils.c b/libavformat/utils.c index 93a3f69503..65c244a48f 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -3535,7 +3535,7 @@ void av_dump_format(AVFormatContext *ic, int is_output) { int i; - uint8_t *printed = av_mallocz(ic->nb_streams); + uint8_t *printed = ic->nb_streams ? av_mallocz(ic->nb_streams) : NULL; if (ic->nb_streams && !printed) return;