March 2009 - FFmpeg-soc - lists.ffmpeg.org

/soc/libavfilter is not compiling
by Juergen_Meissner＠spiegel.de 27 Mar '09

27 Mar '09

/soc/libavfilter based on /ffmpeg/trunk r18133 is not compiling. r18133 has a moving target in it: external swscale is constantly changing. I propose to base /soc/libavfilter on /ffmpeg/bases/0.5 instead of r18133. Should I supply a patch? See: $ ~/ffmpeg/r18133_soc_libavfilter/ffmpeg $ svn update -r 18133 Fetching external item into 'libswscale' External at revision 29067. At revision 18133. $ ~/ffmpeg/r18133_soc_libavfilter/ffmpeg $ make /home/meissner/ffmpeg/r18133_soc_libavfilter/ffmpeg/version.sh "/home/meissner/f fmpeg/r18133_soc_libavfilter/ffmpeg" version.h make: *** No rule to make target `libswscale/x86/cavsdsp_mmx.o', needed by `libs wscale/libswscale.a'. Stop. mfG J.Meissner

2 1

[soc]: r4187 - libavfilter/vf_setpts.c
by victor 26 Mar '09

26 Mar '09

Author: victor Date: Thu Mar 26 22:54:53 2009 New Revision: 4187 Log: The const_names array should be private to this filter. Modified: libavfilter/vf_setpts.c Modified: libavfilter/vf_setpts.c ============================================================================== --- libavfilter/vf_setpts.c Tue Mar 24 18:31:21 2009 (r4186) +++ libavfilter/vf_setpts.c Thu Mar 26 22:54:53 2009 (r4187) @@ -40,7 +40,7 @@ #include "avfilter.h" -const char *const_names[]={ +static const char *const_names[]={ "PI", "E", "AVTB", ///< AV_TIME_BASE

1 0

[soc]: r4184 - dcaenc/dcaenc.c
by bwolowiec 24 Mar '09

24 Mar '09

Author: bwolowiec Date: Tue Mar 24 18:23:40 2009 New Revision: 4184 Log: Add DCA_SUBBANDS constant Modified: dcaenc/dcaenc.c Modified: dcaenc/dcaenc.c ============================================================================== --- dcaenc/dcaenc.c Tue Mar 24 18:20:31 2009 (r4183) +++ dcaenc/dcaenc.c Tue Mar 24 18:23:40 2009 (r4184) @@ -30,6 +30,7 @@ #define DCA_MAX_FRAME_SIZE (16383) #define DCA_HEADER_SIZE 13 +#define DCA_SUBBANDS 32 ///< Subband activity count #define SUBFRAMES 2 #define SUBSUBFRAMES 2 #define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8) @@ -223,14 +224,14 @@ static void put_primary_audio_header(DCA /* Number of primary audio channels */ put_bits(&c->pb, 3, c->prim_channels-1); - /* Subband activity count: 27 + 27 */ + /* Subband activity count */ for(ch=0; ch<c->prim_channels; ch++){ - put_bits(&c->pb, 5, 25); + put_bits(&c->pb, 5, DCA_SUBBANDS-2); } - /* High frequency VQ start subband: 27, 27 */ + /* High frequency VQ start subband */ for(ch=0; ch<c->prim_channels; ch++){ - put_bits(&c->pb, 5, 26); + put_bits(&c->pb, 5, DCA_SUBBANDS-1); } /* Joint intensity coding index: 0, 0 */ @@ -284,26 +285,26 @@ static void put_subframe(DCAContext *c, /* Prediction mode: no ADPCM, in each channel and subband */ for (ch = 0; ch < c->prim_channels; ch++) - for (sub = 0; sub < 27; sub++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) put_bits(&c->pb, 1, 0); /* Prediction VQ addres: not transmitted */ /* Bit allocation index: 19 = "16 bits", for each channel and subband */ for (ch = 0; ch < c->prim_channels; ch++) - for (sub = 0; sub < 27; sub++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) put_bits(&c->pb, 5, 19); if(SUBSUBFRAMES>1){ /* Transition mode: none for each channel and subband */ for (ch = 0; ch < c->prim_channels; ch++) - for (sub = 0; sub < 27; sub++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */ } /* Scale factors: the same for each channel and subband, encoded according to Table D.1.2 */ for (ch = 0; ch < c->prim_channels; ch++) - for (sub = 0; sub < 27; sub++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) put_bits(&c->pb, 7, 110); /* Joint subband scale factor codebook select: not transmitted */ @@ -317,7 +318,7 @@ static void put_subframe(DCAContext *c, for (ss = 0; ss < SUBSUBFRAMES ; ss++) for (ch = 0; ch < c->prim_channels; ch++) - for (sub = 0; sub < 27; sub++) + for (sub = 0; sub < DCA_SUBBANDS; sub++) for (i = 0; i < 8; i++) put_bits(&c->pb, 16, quantize(subband_data[ss * 8 + i][ch][sub])); /* DSYNC */

3 2

[soc]: r4186 - dcaenc/dcaenc.c
by bwolowiec 24 Mar '09

24 Mar '09

Author: bwolowiec Date: Tue Mar 24 18:31:21 2009 New Revision: 4186 Log: cosmetics Modified: dcaenc/dcaenc.c Modified: dcaenc/dcaenc.c ============================================================================== --- dcaenc/dcaenc.c Tue Mar 24 18:29:49 2009 (r4185) +++ dcaenc/dcaenc.c Tue Mar 24 18:31:21 2009 (r4186) @@ -151,7 +151,7 @@ static void put_frame_header(DCAContext /* Number of PCM sample blocks */ put_bits(&c->pb, 7, PCM_SAMPLES-1); - /* Primary frame byte size: 7168 */ + /* Primary frame byte size */ put_bits(&c->pb, 14, c->frame_size-1); /* Audio channel arrangement: L + R (stereo) */ @@ -285,7 +285,6 @@ static inline uint32_t quantize(int32_t return sample; } - static void put_subframe(DCAContext *c, int32_t subband_data[8*SUBSUBFRAMES][MAX_CHANNELS][32]) { int i, j, sub, ss, ch; @@ -345,7 +344,7 @@ static void put_subframe(DCAContext *c, /* Stde information CRC check word: not transmitted */ /* VQ encoded high frequency subbands: not transmitted */ /* LFE data: none */ - /* Audio data: 4 subsubframes */ + /* Audio data (subsubframes) */ for (ss = 0; ss < SUBSUBFRAMES ; ss++) for (ch = 0; ch < c->prim_channels; ch++)

1 0

[soc]: r4185 - dcaenc/dcaenc.c
by bwolowiec 24 Mar '09

24 Mar '09

Author: bwolowiec Date: Tue Mar 24 18:29:49 2009 New Revision: 4185 Log: Add QUANTIZER_BITS and use of factor_scale Modified: dcaenc/dcaenc.c Modified: dcaenc/dcaenc.c ============================================================================== --- dcaenc/dcaenc.c Tue Mar 24 18:23:40 2009 (r4184) +++ dcaenc/dcaenc.c Tue Mar 24 18:29:49 2009 (r4185) @@ -25,12 +25,16 @@ #include "dcaenc.h" #include "dcadata.h" +#undef NDEBUG +#include <assert.h> + #define MAX_CHANNELS (6) #define DCA_SUBBANDS_32 (32) #define DCA_MAX_FRAME_SIZE (16383) #define DCA_HEADER_SIZE 13 #define DCA_SUBBANDS 32 ///< Subband activity count +#define QUANTIZER_BITS 16 #define SUBFRAMES 2 #define SUBSUBFRAMES 2 #define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8) @@ -42,6 +46,7 @@ typedef struct { int frame_size; int prim_channels; int sample_rate_code; + int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32]; int32_t pcm[DCA_SUBBANDS_32]; int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */ @@ -266,17 +271,26 @@ static void put_primary_audio_header(DCA /* Scale factor adjustment index: not transmitted */ } -/* TODO: don't hardcode 16-bit quantization */ -static uint32_t quantize(int32_t d) +/** + * 8-23 bits quantization + * @param sample + * @param bits + */ +static inline uint32_t quantize(int32_t sample, int bits) { - d = d >> 16; - return d & 0xfffe; //XXX: this is done to avoid false syncwords + assert(sample < 1<<(bits-1)); + assert(sample >= -(1<<(bits-1))); + sample &= sample & ((1<<bits)-1); + //sample &= sample & ((1<<bits)-2); //XXX: this is done to avoid false syncwords + return sample; } static void put_subframe(DCAContext *c, int32_t subband_data[8*SUBSUBFRAMES][MAX_CHANNELS][32]) { - int i, sub, ss, ch; + int i, j, sub, ss, ch; + const uint32_t *scale_table = scale_factor_quant7; + /* Subsubframes count */ put_bits(&c->pb, 2, SUBSUBFRAMES -1); @@ -289,10 +303,10 @@ static void put_subframe(DCAContext *c, put_bits(&c->pb, 1, 0); /* Prediction VQ addres: not transmitted */ - /* Bit allocation index: 19 = "16 bits", for each channel and subband */ + /* Bit allocation index */ for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < DCA_SUBBANDS; sub++) - put_bits(&c->pb, 5, 19); + put_bits(&c->pb, 5, QUANTIZER_BITS+3); if(SUBSUBFRAMES>1){ /* Transition mode: none for each channel and subband */ @@ -301,11 +315,28 @@ static void put_subframe(DCAContext *c, put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */ } + /* Determine scale_factor */ + for(ch=0; ch<c->prim_channels; ch++) + for(sub=0; sub<DCA_SUBBANDS; sub++){ + int m=0; + for(i=0; i<8*SUBSUBFRAMES; i++) + m = FFMAX(m, FFABS(subband_data[i][ch][sub])); + m /= (lossy_quant[QUANTIZER_BITS]<<(QUANTIZER_BITS-1)) >> 16; //XXX + i=0; j=128; + while(i<j){ + int q=(i+j)>>1; + if(m < scale_table[q]) j=q; + else i=q+1; + } + assert(i<128); + c->scale_factor[ch][sub] = i; + } + /* Scale factors: the same for each channel and subband, encoded according to Table D.1.2 */ for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < DCA_SUBBANDS; sub++) - put_bits(&c->pb, 7, 110); + put_bits(&c->pb, 7, c->scale_factor[ch][sub]); /* Joint subband scale factor codebook select: not transmitted */ /* Scale factors for joint subband coding: not transmitted */ @@ -319,8 +350,14 @@ static void put_subframe(DCAContext *c, for (ss = 0; ss < SUBSUBFRAMES ; ss++) for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < DCA_SUBBANDS; sub++) - for (i = 0; i < 8; i++) - put_bits(&c->pb, 16, quantize(subband_data[ss * 8 + i][ch][sub])); + for (i = 0; i < 8; i++){ + ///XXX float is unnecessary + float sample = subband_data[ss * 8 + i][ch][sub]; + sample /= scale_table[c->scale_factor[ch][sub]]; + sample *= 1<<16; + sample /= lossy_quant[QUANTIZER_BITS]; + put_bits(&c->pb, QUANTIZER_BITS, quantize(sample, QUANTIZER_BITS)); + } /* DSYNC */ put_bits(&c->pb, 16, 0xffff); }

1 0

[soc]: r4183 - dcaenc/dcaenc.c
by bwolowiec 24 Mar '09

24 Mar '09

Author: bwolowiec Date: Tue Mar 24 18:20:31 2009 New Revision: 4183 Log: make qmf_init static Modified: dcaenc/dcaenc.c Modified: dcaenc/dcaenc.c ============================================================================== --- dcaenc/dcaenc.c Sun Mar 22 18:13:06 2009 (r4182) +++ dcaenc/dcaenc.c Tue Mar 24 18:20:31 2009 (r4183) @@ -59,7 +59,7 @@ static inline int32_t mul32(int32_t a, i /* Integer version of the cosine modulated Pseudo QMF */ -void qmf_init(void) +static void qmf_init(void) { int i; int32_t c[17], s[17];

1 0

DCA encoder (part 2)
by Bartlomiej Wolowiec 24 Mar '09

24 Mar '09

Hi, here is the next part of improvement of DCA encoder. Main changes are: use of scale factors and add constant QUANTIZER_BITS, which determines how many bits are used for mantissa. Now it has also a good quality with 8 bits/sample. -- Bartlomiej Wolowiec

2 1

DCA encoder
by Bartlomiej Wolowiec 24 Mar '09

24 Mar '09

Hi, I had some free time recently, so I had written some code for FFmpeg :) I improved significantly DCA encoder. It works for any SUB(SUB)FRAMES and any sample rate. Also files are played correctly with libdts. I will try to make LFE and some extensions works soon. -- Bartlomiej Wolowiec

3 5

[soc]: r4179 - in libavfilter: checkout.sh diffs/01_ffplay_filters.diff diffs/02_ffmpeg_filters.diff
by stefano 22 Mar '09

22 Mar '09

Author: stefano Date: Sun Mar 22 01:05:42 2009 New Revision: 4179 Log: Update to FFmpeg r18133. Modified: libavfilter/checkout.sh libavfilter/diffs/01_ffplay_filters.diff libavfilter/diffs/02_ffmpeg_filters.diff Modified: libavfilter/checkout.sh ============================================================================== --- libavfilter/checkout.sh Thu Mar 19 07:49:16 2009 (r4178) +++ libavfilter/checkout.sh Sun Mar 22 01:05:42 2009 (r4179) @@ -1,7 +1,7 @@ #! /bin/sh echo "checking out pristine ffmpeg" -svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk/ ffmpeg -r17792 +svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk/ ffmpeg -r18133 echo "patching ffmpeg" for diff in $(ls $(pwd)/diffs/*.diff); do patch -d ffmpeg -p0 -i $diff; done Modified: libavfilter/diffs/01_ffplay_filters.diff ============================================================================== --- libavfilter/diffs/01_ffplay_filters.diff Thu Mar 19 07:49:16 2009 (r4178) +++ libavfilter/diffs/01_ffplay_filters.diff Sun Mar 22 01:05:42 2009 (r4179) @@ -1,6 +1,6 @@ Index: ffplay.c =================================================================== ---- ffplay.c (revision 17789) +--- ffplay.c (revision 18133) +++ ffplay.c (working copy) @@ -29,6 +29,12 @@ #include "libavcodec/audioconvert.h" @@ -38,7 +38,7 @@ Index: ffplay.c } VideoPicture; typedef struct SubPicture { -@@ -172,6 +185,10 @@ +@@ -173,6 +186,10 @@ // QETimer *video_timer; char filename[1024]; int width, height, xleft, ytop; @@ -49,7 +49,7 @@ Index: ffplay.c } VideoState; static void show_help(void); -@@ -212,6 +229,9 @@ +@@ -213,6 +230,9 @@ static int error_recognition = FF_ER_CAREFUL; static int error_concealment = 3; static int decoder_reorder_pts= 0; @@ -59,7 +59,7 @@ Index: ffplay.c /* current context */ static int is_full_screen; -@@ -661,6 +681,13 @@ +@@ -662,6 +682,13 @@ vp = &is->pictq[is->pictq_rindex]; if (vp->bmp) { @@ -73,7 +73,7 @@ Index: ffplay.c /* XXX: use variable in the frame */ if (is->video_st->sample_aspect_ratio.num) aspect_ratio = av_q2d(is->video_st->sample_aspect_ratio); -@@ -668,9 +695,10 @@ +@@ -669,9 +696,10 @@ aspect_ratio = av_q2d(is->video_st->codec->sample_aspect_ratio); else aspect_ratio = 0; @@ -85,7 +85,7 @@ Index: ffplay.c /* if an active format is indicated, then it overrides the mpeg format */ #if 0 -@@ -873,9 +901,15 @@ +@@ -874,9 +902,15 @@ } else if(!is_full_screen && screen_width){ w = screen_width; h = screen_height; @@ -101,7 +101,7 @@ Index: ffplay.c } else { w = 640; h = 480; -@@ -1188,6 +1222,20 @@ +@@ -1189,6 +1223,20 @@ if (vp->bmp) SDL_FreeYUVOverlay(vp->bmp); @@ -122,7 +122,7 @@ Index: ffplay.c #if 0 /* XXX: use generic function */ /* XXX: disable overlay if no hardware acceleration or if RGB format */ -@@ -1205,12 +1253,9 @@ +@@ -1206,12 +1254,9 @@ break; } #endif @@ -136,7 +136,7 @@ Index: ffplay.c SDL_LockMutex(is->pictq_mutex); vp->allocated = 1; -@@ -1227,8 +1272,11 @@ +@@ -1228,8 +1273,11 @@ VideoPicture *vp; int dst_pix_fmt; AVPicture pict; @@ -149,7 +149,7 @@ Index: ffplay.c /* wait until we have space to put a new picture */ SDL_LockMutex(is->pictq_mutex); while (is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE && -@@ -1244,8 +1292,13 @@ +@@ -1245,8 +1293,13 @@ /* alloc or resize hardware picture buffer */ if (!vp->bmp || @@ -163,7 +163,7 @@ Index: ffplay.c SDL_Event event; vp->allocated = 0; -@@ -1269,6 +1322,12 @@ +@@ -1270,6 +1323,12 @@ /* if the frame is not skipped, then display it */ if (vp->bmp) { @@ -176,7 +176,7 @@ Index: ffplay.c /* get a pointer on the bitmap */ SDL_LockYUVOverlay (vp->bmp); -@@ -1280,18 +1339,30 @@ +@@ -1281,18 +1340,30 @@ pict.linesize[0] = vp->bmp->pitches[0]; pict.linesize[1] = vp->bmp->pitches[2]; pict.linesize[2] = vp->bmp->pitches[1]; @@ -211,7 +211,7 @@ Index: ffplay.c /* update the bitmap content */ SDL_UnlockYUVOverlay(vp->bmp); -@@ -1346,54 +1417,267 @@ +@@ -1347,54 +1418,267 @@ return queue_picture(is, src_frame, pts); } @@ -510,7 +510,7 @@ Index: ffplay.c av_free(frame); return 0; } -@@ -2175,6 +2459,12 @@ +@@ -2171,6 +2455,12 @@ /* free all pictures */ for(i=0;i<VIDEO_PICTURE_QUEUE_SIZE; i++) { vp = &is->pictq[i]; @@ -523,7 +523,7 @@ Index: ffplay.c if (vp->bmp) { SDL_FreeYUVOverlay(vp->bmp); vp->bmp = NULL; -@@ -2513,6 +2803,9 @@ +@@ -2509,6 +2799,9 @@ { "ec", OPT_INT | HAS_ARG | OPT_EXPERT, {(void*)&error_concealment}, "set error concealment options", "bit_mask" }, { "sync", HAS_ARG | OPT_FUNC2 | OPT_EXPERT, {(void*)opt_sync}, "set audio-video sync. type (type=audio/video/ext)", "type" }, { "threads", HAS_ARG | OPT_FUNC2 | OPT_EXPERT, {(void*)opt_thread_count}, "thread count", "count" }, @@ -533,7 +533,7 @@ Index: ffplay.c { "default", OPT_FUNC2 | HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {(void*)opt_default}, "generic catch all option", "" }, { NULL, }, }; -@@ -2561,7 +2854,9 @@ +@@ -2557,7 +2850,9 @@ avcodec_opts[i]= avcodec_alloc_context2(i); } avformat_opts = avformat_alloc_context(); Modified: libavfilter/diffs/02_ffmpeg_filters.diff ============================================================================== --- libavfilter/diffs/02_ffmpeg_filters.diff Thu Mar 19 07:49:16 2009 (r4178) +++ libavfilter/diffs/02_ffmpeg_filters.diff Sun Mar 22 01:05:42 2009 (r4179) @@ -1,6 +1,6 @@ Index: ffmpeg.c =================================================================== ---- ffmpeg.c (revision 17789) +--- ffmpeg.c (revision 18133) +++ ffmpeg.c (working copy) @@ -40,6 +40,13 @@ #include "libavutil/avstring.h" @@ -26,7 +26,7 @@ Index: ffmpeg.c static int intra_only = 0; static int audio_sample_rate = 44100; -@@ -280,6 +290,13 @@ +@@ -281,6 +291,13 @@ is not defined */ int64_t pts; /* current pts */ int is_start; /* is 1 at the start and after a discontinuity */ @@ -40,7 +40,7 @@ Index: ffmpeg.c } AVInputStream; typedef struct AVInputFile { -@@ -295,6 +312,183 @@ +@@ -296,6 +313,183 @@ static struct termios oldtty; #endif @@ -224,7 +224,7 @@ Index: ffmpeg.c static void term_exit(void) { #if HAVE_TERMIOS_H -@@ -879,6 +1073,9 @@ +@@ -880,6 +1074,9 @@ if (nb_frames <= 0) return; @@ -234,7 +234,7 @@ Index: ffmpeg.c if (ost->video_crop) { if (av_picture_crop((AVPicture *)&picture_crop_temp, (AVPicture *)in_picture, dec->pix_fmt, ost->topBand, ost->leftBand) < 0) { fprintf(stderr, "error cropping picture\n"); -@@ -890,6 +1087,7 @@ +@@ -891,6 +1088,7 @@ } else { formatted_picture = in_picture; } @@ -242,7 +242,7 @@ Index: ffmpeg.c final_picture = formatted_picture; padding_src = formatted_picture; -@@ -907,12 +1105,14 @@ +@@ -908,12 +1106,14 @@ } } @@ -257,7 +257,7 @@ Index: ffmpeg.c if (ost->video_pad) { av_picture_pad((AVPicture*)final_picture, (AVPicture *)padding_src, -@@ -1183,6 +1383,9 @@ +@@ -1184,6 +1384,9 @@ static short *samples= NULL; AVSubtitle subtitle, *subtitle_to_free; int got_subtitle; @@ -267,7 +267,7 @@ Index: ffmpeg.c if(ist->next_pts == AV_NOPTS_VALUE) ist->next_pts= ist->pts; -@@ -1302,6 +1505,15 @@ +@@ -1303,6 +1506,15 @@ &buffer_to_free); } @@ -283,7 +283,7 @@ Index: ffmpeg.c // preprocess audio (volume) if (ist->st->codec->codec_type == CODEC_TYPE_AUDIO) { if (audio_volume != 256) { -@@ -1323,10 +1535,18 @@ +@@ -1324,10 +1536,18 @@ if (pts > now) usleep(pts - now); } @@ -303,7 +303,7 @@ Index: ffmpeg.c for(i=0;i<nb_ostreams;i++) { int frame_size; -@@ -1349,6 +1569,9 @@ +@@ -1350,6 +1570,9 @@ do_audio_out(os, ost, ist, data_buf, data_size); break; case CODEC_TYPE_VIDEO: @@ -313,7 +313,7 @@ Index: ffmpeg.c do_video_out(os, ost, ist, &picture, &frame_size); if (vstats_filename && frame_size) do_video_stats(os, ost, frame_size); -@@ -1406,7 +1629,17 @@ +@@ -1407,7 +1630,17 @@ av_free_packet(&opkt); } } @@ -331,7 +331,7 @@ Index: ffmpeg.c av_free(buffer_to_free); /* XXX: allocate the subtitles in the codec ? */ if (subtitle_to_free) { -@@ -1837,10 +2070,21 @@ +@@ -1840,10 +2073,21 @@ fprintf(stderr, "Cannot get resampling context\n"); av_exit(1); } @@ -353,7 +353,7 @@ Index: ffmpeg.c break; case CODEC_TYPE_SUBTITLE: ost->encoding_needed = 1; -@@ -3825,6 +4069,9 @@ +@@ -3860,6 +4104,9 @@ { "psnr", OPT_BOOL | OPT_EXPERT | OPT_VIDEO, {(void*)&do_psnr}, "calculate PSNR of compressed frames" }, { "vstats", OPT_EXPERT | OPT_VIDEO, {(void*)&opt_vstats}, "dump video coding statistics to file" }, { "vstats_file", HAS_ARG | OPT_EXPERT | OPT_VIDEO, {(void*)opt_vstats_file}, "dump video coding statistics to file", "file" },

5 7

[soc]: r4182 - dcaenc/dcaenc.c
by bwolowiec 22 Mar '09

22 Mar '09

Author: bwolowiec Date: Sun Mar 22 18:13:06 2009 New Revision: 4182 Log: Improve DCA encoder. Modified: dcaenc/dcaenc.c Modified: dcaenc/dcaenc.c ============================================================================== --- dcaenc/dcaenc.c Sun Mar 22 17:24:02 2009 (r4181) +++ dcaenc/dcaenc.c Sun Mar 22 18:13:06 2009 (r4182) @@ -23,19 +23,27 @@ #include "avcodec.h" #include "bitstream.h" #include "dcaenc.h" +#include "dcadata.h" -#define MAX_CHANNELS (2) +#define MAX_CHANNELS (6) #define DCA_SUBBANDS_32 (32) #define DCA_MAX_FRAME_SIZE (16383) -#define FIXED_FRAME_SIZE (7167) +#define DCA_HEADER_SIZE 13 + +#define SUBFRAMES 2 +#define SUBSUBFRAMES 2 +#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8) typedef struct { PutBitContext pb; int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */ int start[MAX_CHANNELS]; + int frame_size; + int prim_channels; + int sample_rate_code; int32_t pcm[DCA_SUBBANDS_32]; - int32_t subband[64][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */ + int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */ } DCAContext; static int32_t cos_table[128]; @@ -134,21 +142,19 @@ static void put_frame_header(DCAContext /* CRC is not present */ put_bits(&c->pb, 1, 0); - /* Number of PCM sample blocks: 64 - (larger values are unusable with 1:1 compression due to high bitrate - and frame size limitation) */ - put_bits(&c->pb, 7, 63); + /* Number of PCM sample blocks */ + put_bits(&c->pb, 7, PCM_SAMPLES-1); /* Primary frame byte size: 7168 */ - put_bits(&c->pb, 14, FIXED_FRAME_SIZE); + put_bits(&c->pb, 14, c->frame_size-1); /* Audio channel arrangement: L + R (stereo) */ - put_bits(&c->pb, 6, 2); + put_bits(&c->pb, 6, c->prim_channels==2?2:9); //FIXME - /* Core audio sampling frequency: 44100 Hz */ - put_bits(&c->pb, 4, 8); + /* Core audio sampling frequency */ + put_bits(&c->pb, 4, c->sample_rate_code); - /* Transmission bit rate: 1411.2 kbps */ + /* Transmission bit rate: 1411.2 kbps */ //FIXME put_bits(&c->pb, 5, 0x16); /* Embedded down mix: disabled */ @@ -206,48 +212,55 @@ static void put_frame_header(DCAContext static void put_primary_audio_header(DCAContext *c) { - /* Number of subframes: 2 */ - put_bits(&c->pb, 4, 1); + /* From dca.c */ + static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; + static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; - /* Number of primary audio channels: 2 */ - put_bits(&c->pb, 3, 1); + int ch, i; + /* Number of subframes */ + put_bits(&c->pb, 4, SUBFRAMES-1); + + /* Number of primary audio channels */ + put_bits(&c->pb, 3, c->prim_channels-1); /* Subband activity count: 27 + 27 */ - put_bits(&c->pb, 5, 25); - put_bits(&c->pb, 5, 25); + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, 5, 25); + } /* High frequency VQ start subband: 27, 27 */ - put_bits(&c->pb, 5, 26); - put_bits(&c->pb, 5, 26); + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, 5, 26); + } /* Joint intensity coding index: 0, 0 */ - put_bits(&c->pb, 3, 0); - put_bits(&c->pb, 3, 0); + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, 3, 0); + } /* Transient mode codebook: A4, A4 (arbitrary) */ - put_bits(&c->pb, 2, 0); - put_bits(&c->pb, 2, 0); + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, 2, 0); + } /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */ - put_bits(&c->pb, 3, 6); - put_bits(&c->pb, 3, 6); + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, 3, 6); + } /* Bit allocation quantizer select: linear 5-bit */ - put_bits(&c->pb, 3, 6); - put_bits(&c->pb, 3, 6); + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, 3, 6); + } /* Quantization index codebook select: dummy data to avoid transmission of scale factor adjustment */ - put_bits(&c->pb, 1, 1); put_bits(&c->pb, 1, 1); - put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3); - put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3); - put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3); - put_bits(&c->pb, 2, 3); put_bits(&c->pb, 2, 3); - put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7); - put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7); - put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7); - put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7); - put_bits(&c->pb, 3, 7); put_bits(&c->pb, 3, 7); + + for(i=1; i<11; i++){ + for(ch=0; ch<c->prim_channels; ch++){ + put_bits(&c->pb, bitlen[i], thr[i]); + } + } /* Scale factor adjustment index: not transmitted */ } @@ -260,35 +273,36 @@ static uint32_t quantize(int32_t d) } -static void put_subframe(DCAContext *c, int32_t subband_data[32][2][32]) +static void put_subframe(DCAContext *c, int32_t subband_data[8*SUBSUBFRAMES][MAX_CHANNELS][32]) { int i, sub, ss, ch; - - /* Subsubframes count: 4 */ - put_bits(&c->pb, 2, 3); + /* Subsubframes count */ + put_bits(&c->pb, 2, SUBSUBFRAMES -1); /* Partial subsubframe sample count: dummy */ put_bits(&c->pb, 3, 0); /* Prediction mode: no ADPCM, in each channel and subband */ - for (ch = 0; ch < 2; ch++) + for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < 27; sub++) put_bits(&c->pb, 1, 0); /* Prediction VQ addres: not transmitted */ /* Bit allocation index: 19 = "16 bits", for each channel and subband */ - for (ch = 0; ch < 2; ch++) + for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < 27; sub++) put_bits(&c->pb, 5, 19); - /* Transition mode: none for each channel and subband */ - for (ch = 0; ch < 2; ch++) - for (sub = 0; sub < 27; sub++) - put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */ + if(SUBSUBFRAMES>1){ + /* Transition mode: none for each channel and subband */ + for (ch = 0; ch < c->prim_channels; ch++) + for (sub = 0; sub < 27; sub++) + put_bits(&c->pb, 1, 0); /* according to Huffman codebook A4 */ + } /* Scale factors: the same for each channel and subband, encoded according to Table D.1.2 */ - for (ch = 0; ch < 2; ch++) + for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < 27; sub++) put_bits(&c->pb, 7, 110); @@ -301,8 +315,8 @@ static void put_subframe(DCAContext *c, /* LFE data: none */ /* Audio data: 4 subsubframes */ - for (ss = 0; ss < 4 ; ss++) - for (ch = 0; ch < 2; ch++) + for (ss = 0; ss < SUBSUBFRAMES ; ss++) + for (ch = 0; ch < c->prim_channels; ch++) for (sub = 0; sub < 27; sub++) for (i = 0; i < 8; i++) put_bits(&c->pb, 16, quantize(subband_data[ss * 8 + i][ch][sub])); @@ -310,17 +324,21 @@ static void put_subframe(DCAContext *c, put_bits(&c->pb, 16, 0xffff); } -void put_frame(DCAContext *c, int32_t subband_data[64][2][32], uint8_t *frame) +void put_frame(DCAContext *c, int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32], uint8_t *frame) { - int channel; - init_put_bits(&c->pb, frame, DCA_MAX_FRAME_SIZE); + int i; + init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE); - put_frame_header(c); put_primary_audio_header(c); - for (channel=0 ; channel<2; channel++) - put_subframe(c, &subband_data[32 * channel]); + for(i=0; i<SUBFRAMES; i++) + put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i]); flush_put_bits(&c->pb); + c->frame_size = (put_bits_count(&c->pb)>>3) + DCA_HEADER_SIZE; + + init_put_bits(&c->pb, frame, DCA_HEADER_SIZE); + put_frame_header(c); + flush_put_bits(&c->pb); } static int DCA_encode_frame(AVCodecContext *avctx, @@ -333,11 +351,11 @@ static int DCA_encode_frame(AVCodecConte // if (buf_size < MAX_CHANNELS*2048*sizeof(int16_t)) // return -1; - for (i = 0; i < 64; i ++) /* i is the decimated sample number */ - for (channel=0; channel<2 ; channel++) { + for (i = 0; i < PCM_SAMPLES; i ++) /* i is the decimated sample number */ + for (channel=0; channel<c->prim_channels; channel++) { /* Get 32 PCM samples */ for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */ - c->pcm[k] = samples[2 * (32*i+k) + channel] << 16; + c->pcm[k] = samples[avctx->channels * (32*i+k) + channel] << 16; } /* Put subband samples into the proper place */ qmf_decompose(c, c->pcm, &c->subband[i][channel][0], channel); @@ -345,18 +363,31 @@ static int DCA_encode_frame(AVCodecConte put_frame(c, c->subband, frame); - return put_bits_count(&c->pb)>>3; + return c->frame_size; } static int DCA_encode_init(AVCodecContext *avctx) { - //DCAContext *c = avctx->priv_data; + DCAContext *c = avctx->priv_data; + int i; - if(avctx->channels != 2 || avctx->sample_rate != 44100) { - av_log(avctx, AV_LOG_ERROR, "Only 44.1 kHz stereo is supported at the moment!\n"); + c->prim_channels = FFMIN(avctx->channels, 5); //XXX only 5 channels + + for(i=0; i<16; i++){ + if(dca_sample_rates[i] == avctx->sample_rate) + break; + } + if(i==16){ + av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported\n", avctx->sample_rate); return -1; } + c->sample_rate_code = i; - avctx->frame_size = 2048; + if(avctx->channels != 2 && avctx->channels != 6) { + av_log(avctx, AV_LOG_ERROR, "Only stereo and 5.1 supported at the moment!\n"); + return -1; + } + + avctx->frame_size = 32 * PCM_SAMPLES; qmf_init(); return 0;

1 0

[soc]: r4181 - libavfilter/doc/vfilters.texi
by victor 22 Mar '09

22 Mar '09

Author: victor Date: Sun Mar 22 17:24:02 2009 New Revision: 4181 Log: Forgot to escape the equal sign. Added a warning as penitence. Modified: libavfilter/doc/vfilters.texi Modified: libavfilter/doc/vfilters.texi ============================================================================== --- libavfilter/doc/vfilters.texi Sun Mar 22 17:15:05 2009 (r4180) +++ libavfilter/doc/vfilters.texi Sun Mar 22 17:24:02 2009 (r4181) @@ -156,12 +156,13 @@ Rotate video by a chosen amount in degre @section scale @example -./ffmpeg -i in.avi -vfilters "scale=200:100:sws_flags=bicubic" out.avi +./ffmpeg -i in.avi -vfilters "scale=200:100:sws_flags\=bicubic" out.avi @end example Scale to new width/height and/or convert pixel format. The two first parameter are the desired width and height. If the third parameter @var{sws_flags} is absent, then @var{bilinear} is assumed. +Be aware that the equal sign after @var{sws_flags} must be escaped, as in example. @section setpts

1 0

[soc]: r4180 - in libavfilter: README checkout.sh doc doc/vfilters.texi
by victor 22 Mar '09

22 Mar '09

Author: victor Date: Sun Mar 22 17:15:05 2009 New Revision: 4180 Log: Initial documentation of video filters. Added: libavfilter/doc/ libavfilter/doc/vfilters.texi Modified: libavfilter/README libavfilter/checkout.sh Modified: libavfilter/README ============================================================================== --- libavfilter/README Sun Mar 22 01:05:42 2009 (r4179) +++ libavfilter/README Sun Mar 22 17:15:05 2009 (r4180) @@ -68,22 +68,7 @@ of the filters. One of the interesting features of libavfilter is the ability for filters to have multiple inputs and multiple outputs. Unfortunately, ffplay cannot feed multiple video streams to the filter system, and cannot handle multiple output -streams. To illustrate the sorts of things that are possible, however, we can -use a complex filter graph. For example, the following graph: - -input --> split ---> fifo ----------------------> overlay --> output - | ^ - |------> fifo --> crop --> vflip --------| - -splits the stream into two streams, sends one stream through the crop filter -and the vflip filter before merging it back with the other stream by overlaying -it on top. You can use the following command to do this filtering - -./ffmpeg -i input_video.avi -s 240x320 -vfilters "[in]split [T1], fifo, [T2]overlay='0:240'[out]; [T1]fifo, crop=0:0:-1:240, vflip[T2]" out.avi - -where input_video.avi has a vertical resolution of 480 pixels, the result -should be that in output the top half of the video is mirroed onto the bottom -half. +streams. KNOWN ISSUES: - if for some reason the filter chain cannot be setup (due to an unknown Modified: libavfilter/checkout.sh ============================================================================== --- libavfilter/checkout.sh Sun Mar 22 01:05:42 2009 (r4179) +++ libavfilter/checkout.sh Sun Mar 22 17:15:05 2009 (r4180) @@ -8,3 +8,6 @@ for diff in $(ls $(pwd)/diffs/*.diff); d echo "copying files to libavfilter" find $(pwd) -maxdepth 1 -type f -not -name $(basename $0) -exec cp {} ffmpeg/libavfilter \; + +echo "copying libavfilter doc files" +cp -p $(pwd)/doc/* ffmpeg/doc Added: libavfilter/doc/vfilters.texi ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ libavfilter/doc/vfilters.texi Sun Mar 22 17:15:05 2009 (r4180) @@ -0,0 +1,242 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Video Filters Documentation +@titlepage +@sp 7 +@center @titlefont{Video Filters Documentation} +@sp 3 +@end titlepage + + +@chapter Introduction + +Libavfilter is the filtering API of FFmpeg. It is the substitute of the +now deprecated 'vhooks' and started as a Google Summer of Code project. +To get libavfilter you must download it into some directory of your choice by: +@example + svn checkout svn://svn.mplayerhq.hu/soc/libavfilter +@end example + +And then read @url{http://svn.mplayerhq.hu/soc/libavfilter/README?view=co} +to learn how to integrate it into ffmpeg and ffplay. + +The former video hook modules allowed a list of filters, that would run in the +order that they were specified on the ffmpeg command line. +Each filter had one input and one output. + +In libavfilter, it is possible for filters to have multiple inputs and +multiple outputs. +To illustrate the sorts of things that are possible, we can +use a complex filter graph. For example, the following one: + +@example +input --> split ---> fifo ----------------------> overlay --> output + | ^ + |------> fifo --> crop --> vflip --------| +@end example + +splits the stream into two streams, sends one stream through the crop filter +and the vflip filter before merging it back with the other stream by +overlaying it on top. You can use the following command to do this filtering + +@example +./ffmpeg -i in.avi -s 240x320 -vfilters "[in]split [T1], fifo, [T2]overlay='0:240'[out]; [T1]fifo, crop=0:0:-1:240, vflip[T2]" out.avi +@end example + +where input_video.avi has a vertical resolution of 480 pixels, the result +should be that in output the top half of the video is mirrored onto the bottom +half. + +Video filters are loaded using the @var{-vfilters} option to ffmpeg. The +filters in the same linear chain are separated by commas. In our example, +@var{split, fifo, overlay} are in one linear chain, and @var{fifo, crop, vflip} +are in another. +The points where the linear chains join are labeled by names enclosed in +square brackets. In our example, that is @var{[T1]} and @var{[T2]}. +The magic labels @var{[in]} and @var{[out]} are respectively the points +where video is input and output. +After the filter name, you may find the equal sign and a semicolon separated +parameter list. + +There are also the so called @var{source filters} that do not have a video input, +and we expect in future some @var{sink filters} that will not have video output. + +@chapter Available video filters + +When you configure your ffmpeg build, you can disable any of the existing video +filters. +The configure output will show the video filters included in your build. + +These are the currently available video filters: + +@section crop + +@example +./ffmpeg -i in.avi -vfilters "crop=0:0:-1:240" out.avi +@end example + +Crop the input video to x:y:width:height. +The -1 in width or height means that dimension in the input video. + +@section drawbox + +@example +./ffmpeg -i in.avi -vfilters "drawbox=10:20:200:60:red" out.avi +@end example + +Draw a box with x:y:width:height dimensions in a chosen color. + +@section fifo + +@example +./ffmpeg -i in.avi -vfilters "fifo" out.avi +@end example + +@section format + +@example +./ffmpeg -i in.avi -vfilters "format=yuv420p" out.avi +@end example + +Convert video to one of the specified filters formats (libavfilter will +try to pick one that is supported as an input to the next filter). + +@section fps + +@example +./ffmpeg -i in.avi -vfilters "fps=30000:1001" out.avi +@end example + +Modifies the framerate of the input video to a chosen one. + +@section hflip + +@example +./ffmpeg -i in.avi -vfilters "hflip" out.avi +@end example + +Flip the video horizontally. + +@section negate + +@example +./ffmpeg -i in.avi -vfilters "negate" out.avi +@end example + +Convert a video to its negative. + +@section noformat + +@example +./ffmpeg -i in.avi -vfilters "noformat=yuv420p" out.avi +@end example + +Force libavfilter not to send any of the specified pixel formats to the +next filter. + +@section overlay + +Overlay one video on top of another (see the example in the introduction +section). +Be aware that frames are taken +from each input video in timestamp order, hence, if their initial +timestamps differ, it is a a good idea to pass the two +inputs through a @var{setpts=PTS-STARTPTS} filter to have them +begin in the same zero timestamp, as it does the example for +the @var{movie} filter. + +@section rotate + +@example +./ffmpeg -i in.avi -vfilters "rotate=30" out.avi +@end example + +Rotate video by a chosen amount in degrees. By default, 45 degrees. + +@section scale + +@example +./ffmpeg -i in.avi -vfilters "scale=200:100:sws_flags=bicubic" out.avi +@end example + +Scale to new width/height and/or convert pixel format. +The two first parameter are the desired width and height. +If the third parameter @var{sws_flags} is absent, then @var{bilinear} is assumed. + +@section setpts + +@example +# Start counting PTS from zero +ffmpeg -i input.avi -vfilters setpts=PTS-STARTPTS output.avi + +# Fast motion +ffmpeg -i input.avi -vfilters setpts=0.5*PTS output.avi + +# Fixed rate 25 fps +ffmpeg -i input.avi -vfilters setpts=N*AVTB/25 output.avi + +# Fixed rate 25 fps with some jitter +ffmpeg -i input.avi -vfilters 'setpts=AVTB/25*(N+0.05*sin(N*2*PI/25))' output.avi +@end example + +Modifies the presentation timestamp (PTS) of the input video. + +@section slicify + +@example +./ffmpeg -i in.avi -vfilters "slicify=32" out.avi +@end example + +Pass on input video to next video filter as multiple slices. +The parameter is the slice height (16 if not specified). +Adding this in the beginning of filter chains should make filtering +faster due to the better use of the memory cache. + +@section split + +See the example in the introduction section. +The input video is passed on to two outputs. + +@section transpose + +@example +./ffmpeg -i in.avi -vfilters "transpose" out.avi +@end example + +Transpose (line => column) input video to next video filter. + +@section vflip + +@example +./ffmpeg -i in.avi -vfilters "vflip" out.avi +@end example + +Flip the video vertically. + +@section buffer + +This input filter is used by the client application to feed pictures to +the filter chain. See ffmpeg.c for an usage example. + +@section movie + +The parameters of the movie filter are +@example + seekpoint in microseconds : string format : string filename +@end example + +We can overlay a second movie on top of a main one as in this graph: + +@example + input -----------> deltapts0 --> overlay --> output + ^ + movie --> scale--> deltapts1 ------| +@end example + +To do that + +@example +ffmpeg -i in.avi -s 240x320 -vfilters "[in]setpts=PTS-STARTPTS, [T1]overlay=16:16[out]; movie=3200000:avi:in.avi, scale=180:144, setpts=PTS-STARTPTS[T1]" -y out.avi +@end example + +@bye

1 0

Interested in NEW Seeking API of GSoC09
by zhentan feng 20 Mar '09

20 Mar '09

hi, My name is zhentan feng, and I am a student from china. I have finished last year's soc project MXF muxer mentored by Baptiste. This summer, I plan to apply a soc project of ffmpeg again, and i am interested in the NEW Seeking API project. should I start from qualification task again or beginning to collect information for the new project? sincerely yours zhentan feng -- Best wishes~

1 0

[soc]: r4178 - aacenc/psymodel.c
by kostya 19 Mar '09

19 Mar '09

Author: kostya Date: Thu Mar 19 07:49:16 2009 New Revision: 4178 Log: Make frequency cutoff coefficient dependent on encoding parameters Modified: aacenc/psymodel.c Modified: aacenc/psymodel.c ============================================================================== --- aacenc/psymodel.c Wed Mar 18 22:19:58 2009 (r4177) +++ aacenc/psymodel.c Thu Mar 19 07:49:16 2009 (r4178) @@ -80,10 +80,17 @@ av_cold struct FFPsyPreprocessContext* f { FFPsyPreprocessContext *ctx; int i; + float cutoff_coeff; ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); ctx->avctx = avctx; + + if(avctx->flags & CODEC_FLAG_QSCALE) + cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8); + else + cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels); + ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS, - FILT_ORDER, 0.25, 0.0, 0.0); + FILT_ORDER, cutoff_coeff, 0.0, 0.0); if(ctx->fcoeffs){ ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels); for(i = 0; i < avctx->channels; i++)

1 0

[soc]: r4177 - wmapro/wma3dec.c
by faust3 18 Mar '09

18 Mar '09

Author: faust3 Date: Wed Mar 18 22:19:58 2009 New Revision: 4177 Log: do not recalculate the minimum subframe length in wma_decode_tilehdr Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Wed Mar 18 22:05:49 2009 (r4176) +++ wmapro/wma3dec.c Wed Mar 18 22:19:58 2009 (r4177) @@ -453,7 +453,7 @@ static int wma_decode_tilehdr(WMA3Decode int min_channel_len = s->samples_per_frame; int read_channel_mask = 1; int channels_for_cur_subframe = 0; - int subframe_len = s->samples_per_frame / s->max_num_subframes; + int subframe_len; if(fixed_channel_layout){ read_channel_mask = 0; @@ -472,7 +472,7 @@ static int wma_decode_tilehdr(WMA3Decode } if(channels_for_cur_subframe == 1 || - subframe_len * channels_for_cur_subframe == missing_samples) + s->min_samples_per_subframe * channels_for_cur_subframe == missing_samples) read_channel_mask = 0; } @@ -488,9 +488,8 @@ static int wma_decode_tilehdr(WMA3Decode } } - /** if the frames are not evenly split, get the next subframe - length from the bitstream */ - if(subframe_len * channels_for_cur_subframe != missing_samples){ + /** if we have the choice get next subframe length from the bitstream */ + if(s->min_samples_per_subframe * channels_for_cur_subframe != missing_samples){ int log2_subframe_len = 0; /* 1 bit indicates if the subframe length is zero */ if(subframe_len_zero_bit){ @@ -517,7 +516,8 @@ static int wma_decode_tilehdr(WMA3Decode "broken frame: subframe_len %i\n", subframe_len); return -1; } - } + }else + subframe_len = s->min_samples_per_subframe; for(c=0; c<s->num_channels;c++){ WMA3ChannelCtx* chan = &s->channel[c];

1 0

[soc]: r4176 - in wmapro: wma3.h wma3dec.c
by faust3 18 Mar '09

18 Mar '09

Author: faust3 Date: Wed Mar 18 22:05:49 2009 New Revision: 4176 Log: simplified wma_decode_tilehdr a bit Modified: wmapro/wma3.h wmapro/wma3dec.c Modified: wmapro/wma3.h ============================================================================== --- wmapro/wma3.h Tue Mar 17 18:35:02 2009 (r4175) +++ wmapro/wma3.h Wed Mar 18 22:05:49 2009 (r4176) @@ -137,7 +137,6 @@ typedef struct WMA3DecodeContext { int16_t* samples; //< current samplebuffer pointer int16_t* samples_end; //< maximum samplebuffer pointer uint8_t drc_gain; //< gain for the DRC tool - uint8_t no_tiling; //< frames contain subframes int skip_frame; //< skip output step int parsed_all_subframes; //< all subframes decoded? Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Tue Mar 17 18:35:02 2009 (r4175) +++ wmapro/wma3dec.c Wed Mar 18 22:05:49 2009 (r4176) @@ -433,8 +433,10 @@ static int wma_decode_tilehdr(WMA3Decode } }else{ /** subframe length and number of subframes is not constant */ int subframe_len_bits = 0; /** bits needed for the subframe length */ - int subframe_len_zero_bit = 0; /** how many of the length bits indicate - if the subframe is zero */ + int subframe_len_zero_bit = 0; /** first bit indicates if length is zero */ + int fixed_channel_layout; /** all channels have the same subframe layout */ + + fixed_channel_layout = get_bits1(&s->getbit); /** calculate subframe len bits */ if(s->lossless) @@ -447,63 +449,48 @@ static int wma_decode_tilehdr(WMA3Decode /** loop until the frame data is split between the subframes */ while(missing_samples > 0){ - int64_t tileinfo = -1; + unsigned int channel_mask = 0; int min_channel_len = s->samples_per_frame; - int num_subframes_per_channel = 0; - int num_channels = 0; + int read_channel_mask = 1; + int channels_for_cur_subframe = 0; int subframe_len = s->samples_per_frame / s->max_num_subframes; - /** find channel with the smallest overall length */ - for(c=0;c<s->num_channels;c++){ - if(min_channel_len > s->channel[c].channel_len) - min_channel_len = s->channel[c].channel_len; - } - - /** check if this is the start of a new frame */ - if(missing_samples == s->num_channels * s->samples_per_frame){ - s->no_tiling = get_bits1(&s->getbit); - } - - if(s->no_tiling){ - num_subframes_per_channel = 1; - num_channels = s->num_channels; + if(fixed_channel_layout){ + read_channel_mask = 0; + channels_for_cur_subframe = s->num_channels; + min_channel_len = s->channel[0].channel_len; }else{ - /** count how many channels have the minimum length */ + /** find channels with the smallest overall length */ for(c=0;c<s->num_channels;c++){ - if(min_channel_len == s->channel[c].channel_len){ - ++num_channels; + if(s->channel[c].channel_len <= min_channel_len){ + if(s->channel[c].channel_len < min_channel_len){ + channels_for_cur_subframe = 0; + min_channel_len = s->channel[c].channel_len; + } + ++channels_for_cur_subframe; } } - if(num_channels <= 1) - num_subframes_per_channel = 1; - } - /** maximum number of subframes, evenly split */ - if(subframe_len == missing_samples / num_channels){ - num_subframes_per_channel = 1; + if(channels_for_cur_subframe == 1 || + subframe_len * channels_for_cur_subframe == missing_samples) + read_channel_mask = 0; } - /** if there might be multiple subframes per channel */ - if(num_subframes_per_channel != 1){ - int total_num_bits = num_channels; - tileinfo = 0; - /** For every channel with the minimum length, 1 bit - is transmitted that informs us if the channel - contains a subframe with the next subframe_len. */ - while(total_num_bits){ - int num_bits = total_num_bits; - if(num_bits > 32) - num_bits = 32; - tileinfo |= get_bits_long(&s->getbit,num_bits); - total_num_bits -= num_bits; - num_bits = total_num_bits; - tileinfo <<= (num_bits > 32)? 32 : num_bits; + /** For every channel with the minimum length, 1 bit + might be transmitted that informs us if the channel + contains a subframe with the next subframe_len. */ + if(read_channel_mask){ + channel_mask = get_bits(&s->getbit,channels_for_cur_subframe); + if(!channel_mask){ + av_log(s->avctx, AV_LOG_ERROR, + "broken frame: zero frames for subframe_len\n"); + return -1; } } /** if the frames are not evenly split, get the next subframe length from the bitstream */ - if(subframe_len != missing_samples / num_channels){ + if(subframe_len * channels_for_cur_subframe != missing_samples){ int log2_subframe_len = 0; /* 1 bit indicates if the subframe length is zero */ if(subframe_len_zero_bit){ @@ -522,28 +509,23 @@ static int wma_decode_tilehdr(WMA3Decode }else subframe_len = s->samples_per_frame / (1 << log2_subframe_len); - } - /** sanity check the length */ - if(subframe_len < s->min_samples_per_subframe - || subframe_len > s->samples_per_frame){ - av_log(s->avctx, AV_LOG_ERROR, + /** sanity check the length */ + if(subframe_len < s->min_samples_per_subframe + || subframe_len > s->samples_per_frame){ + av_log(s->avctx, AV_LOG_ERROR, "broken frame: subframe_len %i\n", subframe_len); - return -1; + return -1; + } } + for(c=0; c<s->num_channels;c++){ WMA3ChannelCtx* chan = &s->channel[c]; - if(chan->num_subframes > 32){ - av_log(s->avctx, AV_LOG_ERROR, - "broken frame: num subframes %i\n", - chan->num_subframes); - return -1; - } /** add subframes to the individual channels */ if(min_channel_len == chan->channel_len){ - --num_channels; - if(tileinfo & (1<<num_channels)){ + --channels_for_cur_subframe; + if(!read_channel_mask || channel_mask & (1<<channels_for_cur_subframe)){ if(chan->num_subframes > 31){ av_log(s->avctx, AV_LOG_ERROR, "broken frame: num subframes > 31\n");

1 0

[soc]: r4172 - aacenc/aacenc.c
by kostya 17 Mar '09

17 Mar '09

Author: kostya Date: Tue Mar 17 18:24:05 2009 New Revision: 4172 Log: Fill all elements in ff_aac_pow2sf_tab[]; its size has been increased Modified: aacenc/aacenc.c Modified: aacenc/aacenc.c ============================================================================== --- aacenc/aacenc.c Tue Mar 17 18:19:21 2009 (r4171) +++ aacenc/aacenc.c Tue Mar 17 18:24:05 2009 (r4172) @@ -197,7 +197,7 @@ static av_cold int aac_encode_init(AVCod s->coder = &ff_aac_coders[0]; #if !CONFIG_HARDCODED_TABLES - for (i = 0; i < 316; i++) + for (i = 0; i < 428; i++) ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); #endif /* CONFIG_HARDCODED_TABLES */

3 2

[soc]: r4175 - in aacenc: aaccoder.c aacenc.c
by kostya 17 Mar '09

17 Mar '09

Author: kostya Date: Tue Mar 17 18:35:02 2009 New Revision: 4175 Log: Spread TODOs between corresponding files Modified: aacenc/aaccoder.c aacenc/aacenc.c Modified: aacenc/aaccoder.c ============================================================================== --- aacenc/aaccoder.c Tue Mar 17 18:32:36 2009 (r4174) +++ aacenc/aaccoder.c Tue Mar 17 18:35:02 2009 (r4175) @@ -24,6 +24,12 @@ * AAC coefficients encoder */ +/*********************************** + * TODOs: + * speedup quantizer selection + * add sane pulse detection + ***********************************/ + #include "avcodec.h" #include "bitstream.h" #include "aac.h" Modified: aacenc/aacenc.c ============================================================================== --- aacenc/aacenc.c Tue Mar 17 18:32:36 2009 (r4174) +++ aacenc/aacenc.c Tue Mar 17 18:35:02 2009 (r4175) @@ -26,8 +26,6 @@ /*********************************** * TODOs: - * speedup quantizer selection - * add sane pulse detection * add temporal noise shaping ***********************************/

1 0

[soc]: r4174 - aacenc/aaccoder.c
by kostya 17 Mar '09

17 Mar '09

Author: kostya Date: Tue Mar 17 18:32:36 2009 New Revision: 4174 Log: Introduce a new function for choosing codebooks with emphasis on minimizing distortion with intended use in non-optimal but fast coding. Modified: aacenc/aaccoder.c Modified: aacenc/aaccoder.c ============================================================================== --- aacenc/aaccoder.c Tue Mar 17 18:26:55 2009 (r4173) +++ aacenc/aaccoder.c Tue Mar 17 18:32:36 2009 (r4174) @@ -562,6 +562,13 @@ static void encode_window_bands_info(AAC } } +static void encode_window_bands_info_fixed(AACEncContext *s, SingleChannelElement *sce, + int win, int group_len, const float lambda) +{ + encode_window_bands_info(s, sce, win, group_len, 1.0f); +} + + typedef struct TrellisPath { float cost; int prev;

1 0

[soc]: r4173 - in aacenc: aaccoder.c aacenc.c aacenc.h
by kostya 17 Mar '09

17 Mar '09

Author: kostya Date: Tue Mar 17 18:26:55 2009 New Revision: 4173 Log: Pass lambda as parameter to encode_window_bands_info() as well Modified: aacenc/aaccoder.c aacenc/aacenc.c aacenc/aacenc.h Modified: aacenc/aaccoder.c ============================================================================== --- aacenc/aaccoder.c Tue Mar 17 18:24:05 2009 (r4172) +++ aacenc/aaccoder.c Tue Mar 17 18:26:55 2009 (r4173) @@ -467,7 +467,7 @@ typedef struct BandCodingPath { * Encode band info for single window group bands. */ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce, - int win, int group_len) + int win, int group_len, const float lambda) { BandCodingPath path[120][12]; int w, swb, cb, start, start2, size; @@ -502,7 +502,7 @@ static void encode_window_bands_info(AAC FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(win+w)*16+swb]; rd += quantize_band_cost(sce->coeffs + start + w*128, size, sce->sf_idx[(win+w)*16+swb], cb, - s->lambda / band->threshold, INFINITY, NULL); + lambda / band->threshold, INFINITY, NULL); } if( run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run] != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1]) Modified: aacenc/aacenc.c ============================================================================== --- aacenc/aacenc.c Tue Mar 17 18:24:05 2009 (r4172) +++ aacenc/aacenc.c Tue Mar 17 18:26:55 2009 (r4173) @@ -362,7 +362,7 @@ static void encode_band_info(AACEncConte int w; for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){ - s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w]); + s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda); } } Modified: aacenc/aacenc.h ============================================================================== --- aacenc/aacenc.h Tue Mar 17 18:24:05 2009 (r4172) +++ aacenc/aacenc.h Tue Mar 17 18:26:55 2009 (r4173) @@ -36,7 +36,7 @@ typedef struct AACCoefficientsEncoder{ void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s, SingleChannelElement *sce, const float lambda); void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce, - int win, int group_len); + int win, int group_len, const float lambda); void (*quantize_and_encode_band)(PutBitContext *pb, const float *in, int size, int scale_idx, int cb, const float lambda); void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);

1 0

[soc]: r4171 - aacenc/aaccoder.c
by kostya 17 Mar '09

17 Mar '09

Author: kostya Date: Tue Mar 17 18:19:21 2009 New Revision: 4171 Log: Use lambda value passed as argument instead of one taken from the context Modified: aacenc/aaccoder.c Modified: aacenc/aaccoder.c ============================================================================== --- aacenc/aaccoder.c Thu Mar 5 17:12:01 2009 (r4170) +++ aacenc/aaccoder.c Tue Mar 17 18:19:21 2009 (r4171) @@ -632,7 +632,7 @@ static void search_for_quantizers_anmr(A int cb; for(cb = 0; cb <= ESC_BT; cb++){ dists[cb] += quantize_band_cost(coefs + w2*128, sce->ics.swb_sizes[g], - q, cb, s->lambda / band->threshold, INFINITY, NULL); + q, cb, lambda / band->threshold, INFINITY, NULL); } } dist = dists[0];

1 0

FFmpeg Release 0.5 is announced and already in ffmpeg/branches/0.5. hurry up for checking in soc/libavfilter!
by Jürgen Meissner 11 Mar '09

11 Mar '09

Currently there's not too much in ffmpeg/libavfilter, soc/livabfilter/checkout.sh does include much usefull stuff. I appreciated very much, if this could be in the FFmpeg 0.5 Release. mfG J.Meissner

3 2

[soc]: r4170 - libavfilter/Makefile
by victor 06 Mar '09

06 Mar '09

Author: victor Date: Thu Mar 5 17:12:01 2009 New Revision: 4170 Log: CONFIG_SWSCALE disappeared in FFmpeg r17786. Modified: libavfilter/Makefile Modified: libavfilter/Makefile ============================================================================== --- libavfilter/Makefile Thu Mar 5 00:00:57 2009 (r4169) +++ libavfilter/Makefile Thu Mar 5 17:12:01 2009 (r4170) @@ -2,7 +2,7 @@ include $(SUBDIR)../config.mak NAME = avfilter FFLIBS = avcodec avutil -FFLIBS-$(CONFIG_SWSCALE) += swscale +FFLIBS-$(CONFIG_SCALE_FILTER) += swscale FFLIBS-$(CONFIG_AVFILTER_LAVF) += avformat OBJS = allfilters.o \

5 5

[soc]: r4169 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Thu Mar 5 00:00:57 2009 New Revision: 4169 Log: Fix broken compilation. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 23:59:24 2009 (r4168) +++ amr/amrnbfloatdec.c Thu Mar 5 00:00:57 2009 (r4169) @@ -694,7 +694,7 @@ static float fixed_gain_prediction(float * @return fixed gain smoothed */ -static float fixed_gain_smooth(AMRContext *p , const float *lsp, const float *lsp_avg +static float fixed_gain_smooth(AMRContext *p , const float *lsp, const float *lsp_avg, const enum Mode mode) { float diff = 0.0; int i;

1 0

[soc]: r4168 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 23:59:24 2009 New Revision: 4168 Log: The hangover period (diff > 0.65) must be updated for every mode, not just the ones for which the fixed gain is smoothed. See 3GPP TS 26.090 eq 71. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 22:44:08 2009 (r4167) +++ amr/amrnbfloatdec.c Wed Mar 4 23:59:24 2009 (r4168) @@ -694,7 +694,8 @@ static float fixed_gain_prediction(float * @return fixed gain smoothed */ -static float fixed_gain_smooth(AMRContext *p , const float *lsp, const float *lsp_avg) { +static float fixed_gain_smooth(AMRContext *p , const float *lsp, const float *lsp_avg + const enum Mode mode) { float diff = 0.0; int i; @@ -704,7 +705,8 @@ static float fixed_gain_smooth(AMRContex } // if diff has been >0.65 for 10 frames (40 subframes) no smoothing is applied - if((p->diff_count = diff > 0.65 ? p->diff_count+1 : 0) < 40) { + if((p->diff_count = diff > 0.65 ? p->diff_count+1 : 0) < 40 && + (mode < MODE_74 || mode == MODE_102)) { // calculate the fixed gain smoothing factor (k_m) const float smoothing_factor = FFMIN(1.0, FFMAX(0.0, 4.0*diff - 1.6)); // calculate the mean fixed gain for the current subframe @@ -971,8 +973,7 @@ static int amrnb_decode_frame(AVCodecCon } // smooth fixed gain - if(p->cur_frame_mode < MODE_74 || p->cur_frame_mode == MODE_102) - p->fixed_gain[4] = fixed_gain_smooth(p, p->lsp[subframe], p->lsp_avg); + p->fixed_gain[4] = fixed_gain_smooth(p, p->lsp[subframe], p->lsp_avg, p->cur_frame_mode); apply_ir_filter(p, p->fixed_vector);

1 0

[soc] [propchange]: r4161 - svn:log
by diego 04 Mar '09

04 Mar '09

Author: diego Revision: 4161 Property Name: svn:log Action: modified Property diff: --- old property value +++ new property value @@ -1 +1 @@ -Use memmove instead of assignements. +Use memmove instead of assignments.

2 1

[soc]: r4161 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 20:52:49 2009 New Revision: 4161 Log: Use memmove instead of assignements. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 20:02:04 2009 (r4160) +++ amr/amrnbfloatdec.c Wed Mar 4 20:52:49 2009 (r4161) @@ -719,9 +719,7 @@ static void decode_gains(AMRContext *p, * fixed_gain_prediction(p->fixed_vector, p->prediction_error, mode); // update quantified prediction error energy history - p->prediction_error[0] = p->prediction_error[1]; - p->prediction_error[1] = p->prediction_error[2]; - p->prediction_error[2] = p->prediction_error[3]; + memmove(&p->prediction_error[0], &p->prediction_error[1], 3 * sizeof(p->prediction_error[0])); p->prediction_error[3] = 20.0*log10f(fixed_gain_factor); }

2 1

[soc] [propchange]: r4160 - svn:log
by diego 04 Mar '09

04 Mar '09

Author: diego Revision: 4160 Property Name: svn:log Action: modified Property diff: --- old property value +++ new property value @@ -1 +1 @@ -Merge 2 assignements. +Merge 2 assignments.

1 0

[soc]: r4167 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 22:44:08 2009 New Revision: 4167 Log: Replace division by multiplication in fixed_gain_smooth. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 22:42:26 2009 (r4166) +++ amr/amrnbfloatdec.c Wed Mar 4 22:44:08 2009 (r4167) @@ -706,9 +706,9 @@ static float fixed_gain_smooth(AMRContex // if diff has been >0.65 for 10 frames (40 subframes) no smoothing is applied if((p->diff_count = diff > 0.65 ? p->diff_count+1 : 0) < 40) { // calculate the fixed gain smoothing factor (k_m) - const float smoothing_factor = FFMIN(0.25, FFMAX(0.0, diff - 0.4))/0.25; + const float smoothing_factor = FFMIN(1.0, FFMAX(0.0, 4.0*diff - 1.6)); // calculate the mean fixed gain for the current subframe - const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + p->fixed_gain[2] + p->fixed_gain[3] + p->fixed_gain[4])/5.0; + const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + p->fixed_gain[2] + p->fixed_gain[3] + p->fixed_gain[4])*0.2; // calculate the smoothed fixed gain p->fixed_gain[4] = smoothing_factor*p->fixed_gain[4] + (1.0 - smoothing_factor)*fixed_gain_mean; }

1 0

[soc]: r4166 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 22:42:26 2009 New Revision: 4166 Log: cosmetics: use ? : instead of if else for more compact code. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 22:38:51 2009 (r4165) +++ amr/amrnbfloatdec.c Wed Mar 4 22:42:26 2009 (r4166) @@ -704,13 +704,7 @@ static float fixed_gain_smooth(AMRContex } // if diff has been >0.65 for 10 frames (40 subframes) no smoothing is applied - if(diff > 0.65) { - p->diff_count++; - }else { - p->diff_count = 0; - } - - if(p->diff_count < 40) { + if((p->diff_count = diff > 0.65 ? p->diff_count+1 : 0) < 40) { // calculate the fixed gain smoothing factor (k_m) const float smoothing_factor = FFMIN(0.25, FFMAX(0.0, diff - 0.4))/0.25; // calculate the mean fixed gain for the current subframe

1 0

[soc]: r4165 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 22:38:51 2009 New Revision: 4165 Log: Move variable declaration and add const. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 22:35:32 2009 (r4164) +++ amr/amrnbfloatdec.c Wed Mar 4 22:38:51 2009 (r4165) @@ -696,7 +696,6 @@ static float fixed_gain_prediction(float static float fixed_gain_smooth(AMRContext *p , const float *lsp, const float *lsp_avg) { float diff = 0.0; - float smoothing_factor = 0.0; int i; for(i=0; i<LP_FILTER_ORDER; i++) { @@ -712,11 +711,10 @@ static float fixed_gain_smooth(AMRContex } if(p->diff_count < 40) { - float fixed_gain_mean; // calculate the fixed gain smoothing factor (k_m) - smoothing_factor = FFMIN(0.25, FFMAX(0.0, diff - 0.4))/0.25; + const float smoothing_factor = FFMIN(0.25, FFMAX(0.0, diff - 0.4))/0.25; // calculate the mean fixed gain for the current subframe - fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + p->fixed_gain[2] + p->fixed_gain[3] + p->fixed_gain[4])/5.0; + const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + p->fixed_gain[2] + p->fixed_gain[3] + p->fixed_gain[4])/5.0; // calculate the smoothed fixed gain p->fixed_gain[4] = smoothing_factor*p->fixed_gain[4] + (1.0 - smoothing_factor)*fixed_gain_mean; }

1 0

[soc]: r4164 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 22:35:32 2009 New Revision: 4164 Log: Move smoothing of fixed gain into its own function fixed_gain_smooth. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 21:25:29 2009 (r4163) +++ amr/amrnbfloatdec.c Wed Mar 4 22:35:32 2009 (r4164) @@ -685,6 +685,44 @@ static float fixed_gain_prediction(float } /** + * fixed gain smoothing + * + * @param p the context + * @param lsp lsp coefficients for the current subframe + * @param lsp_avg averaged lsp coefficients + * + * @return fixed gain smoothed + */ + +static float fixed_gain_smooth(AMRContext *p , const float *lsp, const float *lsp_avg) { + float diff = 0.0; + float smoothing_factor = 0.0; + int i; + + for(i=0; i<LP_FILTER_ORDER; i++) { + // calculate diff + diff += fabs(lsp_avg[i]-lsp[i])/lsp_avg[i]; + } + + // if diff has been >0.65 for 10 frames (40 subframes) no smoothing is applied + if(diff > 0.65) { + p->diff_count++; + }else { + p->diff_count = 0; + } + + if(p->diff_count < 40) { + float fixed_gain_mean; + // calculate the fixed gain smoothing factor (k_m) + smoothing_factor = FFMIN(0.25, FFMAX(0.0, diff - 0.4))/0.25; + // calculate the mean fixed gain for the current subframe + fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + p->fixed_gain[2] + p->fixed_gain[3] + p->fixed_gain[4])/5.0; + // calculate the smoothed fixed gain + p->fixed_gain[4] = smoothing_factor*p->fixed_gain[4] + (1.0 - smoothing_factor)*fixed_gain_mean; + } + return p->fixed_gain[4]; +} +/** * Decode fixed and pitch gains. * * @param p the context @@ -941,32 +979,8 @@ static int amrnb_decode_frame(AVCodecCon } // smooth fixed gain - if(p->cur_frame_mode < MODE_74 || p->cur_frame_mode == MODE_102) { - float diff = 0.0; - float smoothing_factor = 0.0; - - for(i=0; i<LP_FILTER_ORDER; i++) { - // calculate diff - diff += fabs(p->lsp_avg[i]-p->lsp[subframe][i])/p->lsp_avg[i]; - } - - // if diff has been >0.65 for 10 frames (40 subframes) no smoothing is applied - if(diff > 0.65) { - p->diff_count++; - }else { - p->diff_count = 0; - } - - if(p->diff_count < 40) { - float fixed_gain_mean; - // calculate the fixed gain smoothing factor (k_m) - smoothing_factor = FFMIN(0.25, FFMAX(0.0, diff - 0.4))/0.25; - // calculate the mean fixed gain for the current subframe - fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + p->fixed_gain[2] + p->fixed_gain[3] + p->fixed_gain[4])/5.0; - // calculate the smoothed fixed gain - p->fixed_gain[4] = smoothing_factor*p->fixed_gain[4] + (1.0 - smoothing_factor)*fixed_gain_mean; - } - } + if(p->cur_frame_mode < MODE_74 || p->cur_frame_mode == MODE_102) + p->fixed_gain[4] = fixed_gain_smooth(p, p->lsp[subframe], p->lsp_avg); apply_ir_filter(p, p->fixed_vector);

1 0

[soc]: r4163 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 21:25:29 2009 New Revision: 4163 Log: Simplify further gain decoding. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 21:13:10 2009 (r4162) +++ amr/amrnbfloatdec.c Wed Mar 4 21:25:29 2009 (r4163) @@ -701,17 +701,14 @@ static void decode_gains(AMRContext *p, if(mode == MODE_122 || mode == MODE_795) { p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain]; fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]; - }else if(mode >= MODE_67) { - p->pitch_gain[4] = gains_high[amr_subframe->p_gain][0]; - fixed_gain_factor = gains_high[amr_subframe->p_gain][1]; - }else if(mode >= MODE_515) { - p->pitch_gain[4] = gains_low[amr_subframe->p_gain][0]; - fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; }else { - // gain index is only coded in subframes 0,2 - const int index = (p->frame.subframe[subframe&2].p_gain << 1) + (subframe&1); - p->pitch_gain[4] = gains_MODE_475[index][0]; - fixed_gain_factor = gains_MODE_475[index][1]; + // gain index is only coded in subframes 0,2 for MODE_475 + const float *gains = mode >= MODE_67 ? gains_high[amr_subframe->p_gain] : + mode >= MODE_515 ? gains_low [amr_subframe->p_gain] : + gains_MODE_475[(p->frame.subframe[subframe&2].p_gain << 1) + (subframe&1)]; + + p->pitch_gain[4] = gains[0]; + fixed_gain_factor = gains[1]; } // ^g_c = g_c' * ^gamma_gc

1 0

[soc]: r4162 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 21:13:10 2009 New Revision: 4162 Log: Cosmetics: replace unmappable char γ by gamma. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 20:52:49 2009 (r4161) +++ amr/amrnbfloatdec.c Wed Mar 4 21:13:10 2009 (r4162) @@ -66,7 +66,7 @@ typedef struct AMRContext { float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic code book (fixed) vector - float prediction_error[4]; ///< quantified prediction errors {20log10(^γ_gc)} for previous four subframes + float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes @@ -695,7 +695,7 @@ static float fixed_gain_prediction(float static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, const enum Mode mode, const int subframe) { - float fixed_gain_factor; // fixed gain correction factor {^γ_gc} for the current frame + float fixed_gain_factor; // fixed gain correction factor {^gamma_gc} for the current frame // decode pitch gain and fixed gain correction factor if(mode == MODE_122 || mode == MODE_795) {

1 0

[soc]: r4160 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 20:02:04 2009 New Revision: 4160 Log: Merge 2 assignements. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 20:00:17 2009 (r4159) +++ amr/amrnbfloatdec.c Wed Mar 4 20:02:04 2009 (r4160) @@ -1,4 +1,4 @@ -/* +/* * AMR narrowband decoder (floating point) * Copyright (c) 2006-2007 Robert Swain * @@ -714,11 +714,9 @@ static void decode_gains(AMRContext *p, fixed_gain_factor = gains_MODE_475[index][1]; } - // calculate the predicted fixed gain g_c' - p->fixed_gain[4] = fixed_gain_prediction(p->fixed_vector, p->prediction_error, mode); - // ^g_c = g_c' * ^gamma_gc - p->fixed_gain[4] *= fixed_gain_factor; + p->fixed_gain[4] = fixed_gain_factor + * fixed_gain_prediction(p->fixed_vector, p->prediction_error, mode); // update quantified prediction error energy history p->prediction_error[0] = p->prediction_error[1];

1 0

[soc]: r4159 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 20:00:17 2009 New Revision: 4159 Log: Remove fixed_gain_factor from the AMRContext since it can be a local variable. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 19:53:10 2009 (r4158) +++ amr/amrnbfloatdec.c Wed Mar 4 20:00:17 2009 (r4159) @@ -68,7 +68,6 @@ typedef struct AMRContext { float prediction_error[4]; ///< quantified prediction errors {20log10(^γ_gc)} for previous four subframes float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes - float fixed_gain_factor; ///< fixed gain correction factor {^γ_gc} for the current frame float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes float beta; ///< beta = pitch_gain, bounded by [0.0,1.0] for 12.2 kbps or [0.0,0.8] for other modes @@ -696,34 +695,36 @@ static float fixed_gain_prediction(float static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, const enum Mode mode, const int subframe) { + float fixed_gain_factor; // fixed gain correction factor {^γ_gc} for the current frame + // decode pitch gain and fixed gain correction factor if(mode == MODE_122 || mode == MODE_795) { p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain]; - p->fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]; + fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]; }else if(mode >= MODE_67) { p->pitch_gain[4] = gains_high[amr_subframe->p_gain][0]; - p->fixed_gain_factor = gains_high[amr_subframe->p_gain][1]; + fixed_gain_factor = gains_high[amr_subframe->p_gain][1]; }else if(mode >= MODE_515) { p->pitch_gain[4] = gains_low[amr_subframe->p_gain][0]; - p->fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; + fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; }else { // gain index is only coded in subframes 0,2 const int index = (p->frame.subframe[subframe&2].p_gain << 1) + (subframe&1); p->pitch_gain[4] = gains_MODE_475[index][0]; - p->fixed_gain_factor = gains_MODE_475[index][1]; + fixed_gain_factor = gains_MODE_475[index][1]; } // calculate the predicted fixed gain g_c' p->fixed_gain[4] = fixed_gain_prediction(p->fixed_vector, p->prediction_error, mode); // ^g_c = g_c' * ^gamma_gc - p->fixed_gain[4] *= p->fixed_gain_factor; + p->fixed_gain[4] *= fixed_gain_factor; // update quantified prediction error energy history p->prediction_error[0] = p->prediction_error[1]; p->prediction_error[1] = p->prediction_error[2]; p->prediction_error[2] = p->prediction_error[3]; - p->prediction_error[3] = 20.0*log10f(p->fixed_gain_factor); + p->prediction_error[3] = 20.0*log10f(fixed_gain_factor); } /// @}

1 0

[soc]: r4158 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 19:53:10 2009 New Revision: 4158 Log: Remove ugly static variable gains_index_MODE_475 in decode_gains. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 19:44:41 2009 (r4157) +++ amr/amrnbfloatdec.c Wed Mar 4 19:53:10 2009 (r4158) @@ -696,8 +696,6 @@ static float fixed_gain_prediction(float static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, const enum Mode mode, const int subframe) { - static int gains_index_MODE_475; - // decode pitch gain and fixed gain correction factor if(mode == MODE_122 || mode == MODE_795) { p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain]; @@ -710,9 +708,9 @@ static void decode_gains(AMRContext *p, p->fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; }else { // gain index is only coded in subframes 0,2 - gains_index_MODE_475 = subframe&1 ? gains_index_MODE_475+1 : amr_subframe->p_gain<<1; - p->pitch_gain[4] = gains_MODE_475[gains_index_MODE_475][0]; - p->fixed_gain_factor = gains_MODE_475[gains_index_MODE_475][1]; + const int index = (p->frame.subframe[subframe&2].p_gain << 1) + (subframe&1); + p->pitch_gain[4] = gains_MODE_475[index][0]; + p->fixed_gain_factor = gains_MODE_475[index][1]; } // calculate the predicted fixed gain g_c'

1 0

[soc]: r4157 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 19:44:41 2009 New Revision: 4157 Log: Simplify decoding gains for MODE_475. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 19:34:41 2009 (r4156) +++ amr/amrnbfloatdec.c Wed Mar 4 19:44:41 2009 (r4157) @@ -1,4 +1,4 @@ -/* +/* * AMR narrowband decoder (floating point) * Copyright (c) 2006-2007 Robert Swain * @@ -710,11 +710,9 @@ static void decode_gains(AMRContext *p, p->fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; }else { // gain index is only coded in subframes 0,2 - if(!(subframe&1)) { - gains_index_MODE_475 = amr_subframe->p_gain<<1; - } - p->pitch_gain[4] = gains_MODE_475[gains_index_MODE_475 + (subframe&1)][0]; - p->fixed_gain_factor = gains_MODE_475[gains_index_MODE_475 + (subframe&1)][1]; + gains_index_MODE_475 = subframe&1 ? gains_index_MODE_475+1 : amr_subframe->p_gain<<1; + p->pitch_gain[4] = gains_MODE_475[gains_index_MODE_475][0]; + p->fixed_gain_factor = gains_MODE_475[gains_index_MODE_475][1]; } // calculate the predicted fixed gain g_c'

1 0

[soc]: r4156 - amr/amrnbfloatdec.c
by kmalaussene 04 Mar '09

04 Mar '09

Author: kmalaussene Date: Wed Mar 4 19:34:41 2009 New Revision: 4156 Log: Factor out gains decoding into decode_gains function, to make the main loop more readable and allow further simplifications. Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Wed Mar 4 01:06:52 2009 (r4155) +++ amr/amrnbfloatdec.c Wed Mar 4 19:34:41 2009 (r4156) @@ -685,6 +685,51 @@ static float fixed_gain_prediction(float return powf(10.0, 0.05*(energy_pred + energy_mean[mode] - energy_fixed_mean)); } +/** + * Decode fixed and pitch gains. + * + * @param p the context + * @param amr_subframe unpacked amr subframe + * @param mode mode of the current frame + * @param subframe current subframe number + */ + +static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, const enum Mode mode, const int subframe) +{ + static int gains_index_MODE_475; + + // decode pitch gain and fixed gain correction factor + if(mode == MODE_122 || mode == MODE_795) { + p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain]; + p->fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]; + }else if(mode >= MODE_67) { + p->pitch_gain[4] = gains_high[amr_subframe->p_gain][0]; + p->fixed_gain_factor = gains_high[amr_subframe->p_gain][1]; + }else if(mode >= MODE_515) { + p->pitch_gain[4] = gains_low[amr_subframe->p_gain][0]; + p->fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; + }else { + // gain index is only coded in subframes 0,2 + if(!(subframe&1)) { + gains_index_MODE_475 = amr_subframe->p_gain<<1; + } + p->pitch_gain[4] = gains_MODE_475[gains_index_MODE_475 + (subframe&1)][0]; + p->fixed_gain_factor = gains_MODE_475[gains_index_MODE_475 + (subframe&1)][1]; + } + + // calculate the predicted fixed gain g_c' + p->fixed_gain[4] = fixed_gain_prediction(p->fixed_vector, p->prediction_error, mode); + + // ^g_c = g_c' * ^gamma_gc + p->fixed_gain[4] *= p->fixed_gain_factor; + + // update quantified prediction error energy history + p->prediction_error[0] = p->prediction_error[1]; + p->prediction_error[1] = p->prediction_error[2]; + p->prediction_error[2] = p->prediction_error[3]; + p->prediction_error[3] = 20.0*log10f(p->fixed_gain_factor); +} + /// @} @@ -835,12 +880,6 @@ static void update_state(AMRContext *p) memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE], (PITCH_LAG_MAX + LP_FILTER_ORDER + 1)*sizeof(float)); - // update quantified prediction error energy history - p->prediction_error[0] = p->prediction_error[1]; - p->prediction_error[1] = p->prediction_error[2]; - p->prediction_error[2] = p->prediction_error[3]; - p->prediction_error[3] = 20.0*log10f(p->fixed_gain_factor); - // update gain history memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4*sizeof(float)); memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4*sizeof(float)); @@ -861,7 +900,6 @@ static int amrnb_decode_frame(AVCodecCon AMRContext *p = avctx->priv_data; // pointer to private data float *buf_out = data; // pointer to the output data buffer int i, subframe; // counters - int gains_index_MODE_475 = 0; // MODE_475 gains index coded every other subframe enum Mode speech_mode = MODE_475; // ??? // decode the bitstream to AMR parameters @@ -899,34 +937,7 @@ static int amrnb_decode_frame(AVCodecCon decode_fixed_vector(p->fixed_vector, amr_subframe->pulses, p->cur_frame_mode, subframe); -/*** gain decoding ***/ - - // calculate the predicted fixed gain g_c' - p->fixed_gain[4] = fixed_gain_prediction(p->fixed_vector, p->prediction_error, p->cur_frame_mode); - - // decode pitch gain and fixed gain correction factor - if(p->cur_frame_mode == MODE_122 || p->cur_frame_mode == MODE_795) { - p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain]; - p->fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]; - }else if(p->cur_frame_mode >= MODE_67) { - p->pitch_gain[4] = gains_high[amr_subframe->p_gain][0]; - p->fixed_gain_factor = gains_high[amr_subframe->p_gain][1]; - }else if(p->cur_frame_mode >= MODE_515) { - p->pitch_gain[4] = gains_low[amr_subframe->p_gain][0]; - p->fixed_gain_factor = gains_low[amr_subframe->p_gain][1]; - }else { - // gain index is only coded in subframes 0,2 - if(!(subframe&1)) { - gains_index_MODE_475 = amr_subframe->p_gain<<1; - } - p->pitch_gain[4] = gains_MODE_475[gains_index_MODE_475 + (subframe&1)][0]; - p->fixed_gain_factor = gains_MODE_475[gains_index_MODE_475 + (subframe&1)][1]; - } - - // ^g_c = g_c' * ^gamma_gc - p->fixed_gain[4] *= p->fixed_gain_factor; - -/*** end of gain decoding ***/ + decode_gains(p, amr_subframe, p->cur_frame_mode, subframe); /*** pre-processing ***/

1 0

[soc]: r4155 - libavfilter/vf_scale.c
by stefano 04 Mar '09

04 Mar '09

Author: stefano Date: Wed Mar 4 01:06:52 2009 New Revision: 4155 Log: Use av_gcd() rather than av_reduce() when computing outpic->pixel_aspect, simplify. Modified: libavfilter/vf_scale.c Modified: libavfilter/vf_scale.c ============================================================================== --- libavfilter/vf_scale.c Wed Mar 4 01:03:07 2009 (r4154) +++ libavfilter/vf_scale.c Wed Mar 4 01:06:52 2009 (r4155) @@ -128,15 +128,17 @@ static void start_frame(AVFilterLink *li { ScaleContext *scale = link->dst->priv; AVFilterLink *out = link->dst->outputs[0]; + int64_t gcd; out->outpic = avfilter_get_video_buffer(out, AV_PERM_WRITE); out->outpic->pts = picref->pts; out->outpic->pixel_aspect.num = picref->pixel_aspect.num * out->h * link->w; out->outpic->pixel_aspect.den = picref->pixel_aspect.den * out->w * link->h; - av_reduce(&out->outpic->pixel_aspect.num, &out->outpic->pixel_aspect.den, - out->outpic->pixel_aspect.num, out->outpic->pixel_aspect.den, - FFMAX(out->outpic->pixel_aspect.num, out->outpic->pixel_aspect.den)); + + gcd = av_gcd(out->outpic->pixel_aspect.num, out->outpic->pixel_aspect.den); + out->outpic->pixel_aspect.num /= gcd; + out->outpic->pixel_aspect.den /= gcd; avfilter_start_frame(out, avfilter_ref_pic(out->outpic, ~0));

1 0

[soc]: r4154 - libavfilter/vf_scale.c
by stefano 04 Mar '09

04 Mar '09

Author: stefano Date: Wed Mar 4 01:03:07 2009 New Revision: 4154 Log: Use sscanf() for parsing the sws_opts string, simplify. Modified: libavfilter/vf_scale.c Modified: libavfilter/vf_scale.c ============================================================================== --- libavfilter/vf_scale.c Wed Mar 4 00:58:05 2009 (r4153) +++ libavfilter/vf_scale.c Wed Mar 4 01:03:07 2009 (r4154) @@ -57,12 +57,7 @@ static av_cold int init(AVFilterContext if ((p = strstr(sws_opts, "sws_flags="))) { char sws_flags[256]; - int i = 0; - p = strchr(sws_opts, '='); - p++; - while (*p && *p != ':' && i < sizeof(sws_flags) - 1) - sws_flags[i++] = *p++; - sws_flags[i] = 0; + sscanf(p, "sws_flags=%255[^:]", sws_flags); if (av_set_string3(scale->sws, "sws_flags", sws_flags, 1, NULL) < 0) { sws_freeContext(scale->sws);

1 0

[soc]: r4153 - in libavfilter: checkout.sh diffs/01_ffplay_filters.diff diffs/02_ffmpeg_filters.diff
by stefano 03 Mar '09

03 Mar '09

Author: stefano Date: Wed Mar 4 00:58:05 2009 New Revision: 4153 Log: Update to FFmpeg r17792. Modified: libavfilter/checkout.sh libavfilter/diffs/01_ffplay_filters.diff libavfilter/diffs/02_ffmpeg_filters.diff Modified: libavfilter/checkout.sh ============================================================================== --- libavfilter/checkout.sh Tue Mar 3 21:57:40 2009 (r4152) +++ libavfilter/checkout.sh Wed Mar 4 00:58:05 2009 (r4153) @@ -1,7 +1,7 @@ #! /bin/sh echo "checking out pristine ffmpeg" -svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk/ ffmpeg -r17547 +svn checkout svn://svn.ffmpeg.org/ffmpeg/trunk/ ffmpeg -r17792 echo "patching ffmpeg" for diff in $(ls $(pwd)/diffs/*.diff); do patch -d ffmpeg -p0 -i $diff; done Modified: libavfilter/diffs/01_ffplay_filters.diff ============================================================================== --- libavfilter/diffs/01_ffplay_filters.diff Tue Mar 3 21:57:40 2009 (r4152) +++ libavfilter/diffs/01_ffplay_filters.diff Wed Mar 4 00:58:05 2009 (r4153) @@ -1,6 +1,6 @@ Index: ffplay.c =================================================================== ---- ffplay.c (revision 17547) +--- ffplay.c (revision 17789) +++ ffplay.c (working copy) @@ -29,6 +29,12 @@ #include "libavcodec/audioconvert.h" @@ -429,7 +429,7 @@ Index: ffplay.c + inputs->pad_idx = 0; + inputs->next = NULL; + -+ if (avfilter_parse_graph(graph, vfilters, inputs, outputs, NULL) < 0) ++ if (avfilter_graph_parse(graph, vfilters, inputs, outputs, NULL) < 0) + goto the_end; + } else { + if(avfilter_link(filt_src, 0, filt_out, 0) < 0) goto the_end; @@ -505,12 +505,12 @@ Index: ffplay.c } the_end: + #if CONFIG_AVFILTER -+ avfilter_destroy_graph(graph); ++ avfilter_graph_destroy(graph); + #endif av_free(frame); return 0; } -@@ -2189,6 +2473,12 @@ +@@ -2175,6 +2459,12 @@ /* free all pictures */ for(i=0;i<VIDEO_PICTURE_QUEUE_SIZE; i++) { vp = &is->pictq[i]; @@ -523,7 +523,7 @@ Index: ffplay.c if (vp->bmp) { SDL_FreeYUVOverlay(vp->bmp); vp->bmp = NULL; -@@ -2527,6 +2817,9 @@ +@@ -2513,6 +2803,9 @@ { "ec", OPT_INT | HAS_ARG | OPT_EXPERT, {(void*)&error_concealment}, "set error concealment options", "bit_mask" }, { "sync", HAS_ARG | OPT_FUNC2 | OPT_EXPERT, {(void*)opt_sync}, "set audio-video sync. type (type=audio/video/ext)", "type" }, { "threads", HAS_ARG | OPT_FUNC2 | OPT_EXPERT, {(void*)opt_thread_count}, "thread count", "count" }, @@ -533,8 +533,8 @@ Index: ffplay.c { "default", OPT_FUNC2 | HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {(void*)opt_default}, "generic catch all option", "" }, { NULL, }, }; -@@ -2575,7 +2868,9 @@ - avctx_opts[i]= avcodec_alloc_context2(i); +@@ -2561,7 +2854,9 @@ + avcodec_opts[i]= avcodec_alloc_context2(i); } avformat_opts = avformat_alloc_context(); +#if !CONFIG_AVFILTER Modified: libavfilter/diffs/02_ffmpeg_filters.diff ============================================================================== --- libavfilter/diffs/02_ffmpeg_filters.diff Tue Mar 3 21:57:40 2009 (r4152) +++ libavfilter/diffs/02_ffmpeg_filters.diff Wed Mar 4 00:58:05 2009 (r4153) @@ -1,8 +1,8 @@ Index: ffmpeg.c =================================================================== ---- ffmpeg.c (revision 17547) +--- ffmpeg.c (revision 17789) +++ ffmpeg.c (working copy) -@@ -41,6 +41,13 @@ +@@ -40,6 +40,13 @@ #include "libavutil/avstring.h" #include "libavformat/os_support.h" @@ -16,7 +16,7 @@ Index: ffmpeg.c #if HAVE_SYS_RESOURCE_H #include <sys/types.h> #include <sys/resource.h> -@@ -148,6 +155,9 @@ +@@ -147,6 +154,9 @@ static int loop_input = 0; static int loop_output = AVFMT_NOOUTPUTLOOP; static int qp_hist = 0; @@ -26,7 +26,7 @@ Index: ffmpeg.c static int intra_only = 0; static int audio_sample_rate = 44100; -@@ -286,6 +296,13 @@ +@@ -280,6 +290,13 @@ is not defined */ int64_t pts; /* current pts */ int is_start; /* is 1 at the start and after a discontinuity */ @@ -40,7 +40,7 @@ Index: ffmpeg.c } AVInputStream; typedef struct AVInputFile { -@@ -301,6 +318,183 @@ +@@ -295,6 +312,183 @@ static struct termios oldtty; #endif @@ -193,7 +193,7 @@ Index: ffmpeg.c + inputs->pad_idx = 0; + inputs->next = NULL; + -+ if (avfilter_parse_graph(filt_graph_all, vfilters, inputs, outputs, NULL) < 0) ++ if (avfilter_graph_parse(filt_graph_all, vfilters, inputs, outputs, NULL) < 0) + return -1; + } else { + if(avfilter_link(curr_filter, 0, ist->out_video_filter, 0) < 0) @@ -224,7 +224,7 @@ Index: ffmpeg.c static void term_exit(void) { #if HAVE_TERMIOS_H -@@ -886,6 +1080,9 @@ +@@ -879,6 +1073,9 @@ if (nb_frames <= 0) return; @@ -234,7 +234,7 @@ Index: ffmpeg.c if (ost->video_crop) { if (av_picture_crop((AVPicture *)&picture_crop_temp, (AVPicture *)in_picture, dec->pix_fmt, ost->topBand, ost->leftBand) < 0) { fprintf(stderr, "error cropping picture\n"); -@@ -897,6 +1094,7 @@ +@@ -890,6 +1087,7 @@ } else { formatted_picture = in_picture; } @@ -242,7 +242,7 @@ Index: ffmpeg.c final_picture = formatted_picture; padding_src = formatted_picture; -@@ -914,12 +1112,14 @@ +@@ -907,12 +1105,14 @@ } } @@ -257,7 +257,7 @@ Index: ffmpeg.c if (ost->video_pad) { av_picture_pad((AVPicture*)final_picture, (AVPicture *)padding_src, -@@ -1190,6 +1390,9 @@ +@@ -1183,6 +1383,9 @@ static short *samples= NULL; AVSubtitle subtitle, *subtitle_to_free; int got_subtitle; @@ -267,7 +267,7 @@ Index: ffmpeg.c if(ist->next_pts == AV_NOPTS_VALUE) ist->next_pts= ist->pts; -@@ -1307,6 +1510,15 @@ +@@ -1302,6 +1505,15 @@ &buffer_to_free); } @@ -283,7 +283,7 @@ Index: ffmpeg.c // preprocess audio (volume) if (ist->st->codec->codec_type == CODEC_TYPE_AUDIO) { if (audio_volume != 256) { -@@ -1328,10 +1540,18 @@ +@@ -1323,10 +1535,18 @@ if (pts > now) usleep(pts - now); } @@ -303,7 +303,7 @@ Index: ffmpeg.c for(i=0;i<nb_ostreams;i++) { int frame_size; -@@ -1354,6 +1574,9 @@ +@@ -1349,6 +1569,9 @@ do_audio_out(os, ost, ist, data_buf, data_size); break; case CODEC_TYPE_VIDEO: @@ -313,7 +313,7 @@ Index: ffmpeg.c do_video_out(os, ost, ist, &picture, &frame_size); if (vstats_filename && frame_size) do_video_stats(os, ost, frame_size); -@@ -1411,7 +1634,17 @@ +@@ -1406,7 +1629,17 @@ av_free_packet(&opkt); } } @@ -331,7 +331,7 @@ Index: ffmpeg.c av_free(buffer_to_free); /* XXX: allocate the subtitles in the codec ? */ if (subtitle_to_free) { -@@ -1842,10 +2075,21 @@ +@@ -1837,10 +2070,21 @@ fprintf(stderr, "Cannot get resampling context\n"); av_exit(1); } @@ -353,10 +353,10 @@ Index: ffmpeg.c break; case CODEC_TYPE_SUBTITLE: ost->encoding_needed = 1; -@@ -3835,6 +4079,9 @@ - #if CONFIG_VHOOK - { "vhook", HAS_ARG | OPT_EXPERT | OPT_VIDEO, {(void*)add_frame_hooker}, "insert video processing module", "module" }, - #endif +@@ -3825,6 +4069,9 @@ + { "psnr", OPT_BOOL | OPT_EXPERT | OPT_VIDEO, {(void*)&do_psnr}, "calculate PSNR of compressed frames" }, + { "vstats", OPT_EXPERT | OPT_VIDEO, {(void*)&opt_vstats}, "dump video coding statistics to file" }, + { "vstats_file", HAS_ARG | OPT_EXPERT | OPT_VIDEO, {(void*)opt_vstats_file}, "dump video coding statistics to file", "file" }, +#if CONFIG_AVFILTER + { "vfilters", OPT_STRING | HAS_ARG, {(void*)&vfilters}, "video filters", "filter list" }, +#endif

1 0

[soc]: r4152 - in wmapro: wma3.h wma3dec.c
by faust3 03 Mar '09

03 Mar '09

Author: faust3 Date: Tue Mar 3 21:57:40 2009 New Revision: 4152 Log: moved bitstream reader context to the decoder context and its init to wma_decode_frame Modified: wmapro/wma3.h wmapro/wma3dec.c Modified: wmapro/wma3.h ============================================================================== --- wmapro/wma3.h Tue Mar 3 21:25:50 2009 (r4151) +++ wmapro/wma3.h Tue Mar 3 21:57:40 2009 (r4152) @@ -132,7 +132,7 @@ typedef struct WMA3DecodeContext { /** frame decode state */ unsigned int frame_num; //< current frame number - GetBitContext* getbit; //< bitstream reader context + GetBitContext getbit; //< bitstream reader context int buf_bit_size; //< buffer size in bits int16_t* samples; //< current samplebuffer pointer int16_t* samples_end; //< maximum samplebuffer pointer Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Tue Mar 3 21:25:50 2009 (r4151) +++ wmapro/wma3dec.c Tue Mar 3 21:57:40 2009 (r4152) @@ -461,7 +461,7 @@ static int wma_decode_tilehdr(WMA3Decode /** check if this is the start of a new frame */ if(missing_samples == s->num_channels * s->samples_per_frame){ - s->no_tiling = get_bits1(s->getbit); + s->no_tiling = get_bits1(&s->getbit); } if(s->no_tiling){ @@ -494,7 +494,7 @@ static int wma_decode_tilehdr(WMA3Decode int num_bits = total_num_bits; if(num_bits > 32) num_bits = 32; - tileinfo |= get_bits_long(s->getbit,num_bits); + tileinfo |= get_bits_long(&s->getbit,num_bits); total_num_bits -= num_bits; num_bits = total_num_bits; tileinfo <<= (num_bits > 32)? 32 : num_bits; @@ -507,13 +507,13 @@ static int wma_decode_tilehdr(WMA3Decode int log2_subframe_len = 0; /* 1 bit indicates if the subframe length is zero */ if(subframe_len_zero_bit){ - if(get_bits1(s->getbit)){ + if(get_bits1(&s->getbit)){ log2_subframe_len = - get_bits(s->getbit,subframe_len_bits-1); + get_bits(&s->getbit,subframe_len_bits-1); ++log2_subframe_len; } }else - log2_subframe_len = get_bits(s->getbit,subframe_len_bits); + log2_subframe_len = get_bits(&s->getbit,subframe_len_bits); if(s->lossless){ subframe_len = @@ -600,7 +600,7 @@ static int wma_decode_channel_transform( }else{ int remaining_channels = s->channels_for_cur_subframe; - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ av_log(s->avctx,AV_LOG_ERROR,"unsupported channel transform bit\n"); return 0; } @@ -618,7 +618,7 @@ static int wma_decode_channel_transform( for(i=0;i<s->channels_for_cur_subframe;i++){ int channel_idx = s->channel_indexes_for_cur_subframe[i]; if(!s->channel[channel_idx].grouped - && get_bits(s->getbit,1)){ + && get_bits(&s->getbit,1)){ ++chgroup->num_channels; s->channel[channel_idx].grouped = 1; chgroup->use_channel[channel_idx] = 1; @@ -643,8 +643,8 @@ static int wma_decode_channel_transform( chgroup->decorrelation_matrix[0] = 1.0; }else if(chgroup->num_channels == 2){ - if(get_bits(s->getbit,1)){ - if(!get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ + if(!get_bits(&s->getbit,1)){ chgroup->no_rotation = 1; chgroup->transform = 2; chgroup->decorrelation_matrix[0] = 1.0; @@ -661,8 +661,8 @@ static int wma_decode_channel_transform( chgroup->decorrelation_matrix[3] = 0.70703125; } }else{ - if(get_bits(s->getbit,1)){ - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ + if(get_bits(&s->getbit,1)){ chgroup->no_rotation = 0; chgroup->transform = 0; }else{ @@ -692,10 +692,10 @@ static int wma_decode_channel_transform( int n_offset = chgroup->num_channels * (chgroup->num_channels - 1) / 2; int i; for(i=0;i<n_offset;i++){ - chgroup->rotation_offset[i] = get_bits(s->getbit,6); + chgroup->rotation_offset[i] = get_bits(&s->getbit,6); } for(i=0;i<chgroup->num_channels;i++) - chgroup->positive[i] = get_bits(s->getbit,1); + chgroup->positive[i] = get_bits(&s->getbit,1); } /* decode transform on / off */ @@ -705,11 +705,11 @@ static int wma_decode_channel_transform( for(i=0;i<s->num_bands;i++) chgroup->transform_band[i] = 1; }else{ - if(get_bits(s->getbit,1) == 0){ + if(get_bits(&s->getbit,1) == 0){ int i; // transform works on individual scale factor bands for(i=0;i< s->num_bands;i++){ - chgroup->transform_band[i] = get_bits(s->getbit,1); + chgroup->transform_band[i] = get_bits(&s->getbit,1); } }else{ int i; @@ -727,16 +727,16 @@ static int wma_decode_channel_transform( static unsigned int wma_get_large_val(WMA3DecodeContext* s) { int n_bits = 8; - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ n_bits += 8; - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ n_bits += 8; - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ n_bits += 7; } } } - return get_bits_long(s->getbit,n_bits); + return get_bits_long(&s->getbit,n_bits); } static inline void wma_get_vec4(WMA3DecodeContext *s,int* vals,int* masks) @@ -744,18 +744,18 @@ static inline void wma_get_vec4(WMA3Deco unsigned int idx; int i = 0; // read 4 values - idx = get_vlc2(s->getbit, s->vec4_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC4_MAXBITS+VLCBITS-1)/VLCBITS)); + idx = get_vlc2(&s->getbit, s->vec4_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC4_MAXBITS+VLCBITS-1)/VLCBITS)); if ( idx == FF_WMA3_HUFF_VEC4_SIZE - 1 ) { while(i < 4){ - idx = get_vlc2(s->getbit, s->vec2_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC2_MAXBITS+VLCBITS-1)/VLCBITS)); + idx = get_vlc2(&s->getbit, s->vec2_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC2_MAXBITS+VLCBITS-1)/VLCBITS)); if ( idx == FF_WMA3_HUFF_VEC2_SIZE - 1 ){ - vals[i] = get_vlc2(s->getbit, s->vec1_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)); + vals[i] = get_vlc2(&s->getbit, s->vec1_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)); if(vals[i] == FF_WMA3_HUFF_VEC1_SIZE - 1) vals[i] += wma_get_large_val(s); - vals[i+1] = get_vlc2(s->getbit, s->vec1_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)); + vals[i+1] = get_vlc2(&s->getbit, s->vec1_vlc.table, VLCBITS, ((FF_WMA3_HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)); if(vals[i+1] == FF_WMA3_HUFF_VEC1_SIZE - 1) vals[i+1] += wma_get_large_val(s); }else{ @@ -774,13 +774,13 @@ static inline void wma_get_vec4(WMA3Deco } if(vals[0]) - masks[0] = get_bits(s->getbit,1); + masks[0] = get_bits(&s->getbit,1); if(vals[1]) - masks[1] = get_bits(s->getbit,1); + masks[1] = get_bits(&s->getbit,1); if(vals[2]) - masks[2] = get_bits(s->getbit,1); + masks[2] = get_bits(&s->getbit,1); if(vals[3]) - masks[3] = get_bits(s->getbit,1); + masks[3] = get_bits(&s->getbit,1); } static int decode_coeffs(WMA3DecodeContext *s, int c) @@ -797,7 +797,7 @@ static int decode_coeffs(WMA3DecodeConte av_log(s->avctx,AV_LOG_DEBUG,"decode coefficients for channel %i\n",c); - vlctable = get_bits(s->getbit, 1); + vlctable = get_bits(&s->getbit, 1); vlc = &s->coef_vlc[vlctable]; vlcmax = s->coef_max[vlctable]; @@ -814,28 +814,28 @@ static int decode_coeffs(WMA3DecodeConte unsigned int idx; int mask; int val; - idx = get_vlc2(s->getbit, vlc->table, VLCBITS, vlcmax); + idx = get_vlc2(&s->getbit, vlc->table, VLCBITS, vlcmax); if( idx == 1) return 0; else if( !idx ){ val = wma_get_large_val(s); /** escape decode */ - if(get_bits(s->getbit,1)){ - if(get_bits(s->getbit,1)){ - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ + if(get_bits(&s->getbit,1)){ + if(get_bits(&s->getbit,1)){ av_log(s->avctx,AV_LOG_ERROR,"broken escape sequence\n"); return 0; }else - cur_coeff += get_bits(s->getbit,s->esc_len) + 4; + cur_coeff += get_bits(&s->getbit,s->esc_len) + 4; }else - cur_coeff += get_bits(s->getbit,2) + 1; + cur_coeff += get_bits(&s->getbit,2) + 1; } }else{ cur_coeff += run[idx]; val = level[idx]; } - mask = get_bits(s->getbit,1) - 1; + mask = get_bits(&s->getbit,1) - 1; ci->coeffs[cur_coeff] = (val^mask) - mask; ++cur_coeff; }else{ @@ -892,7 +892,7 @@ static int wma_decode_scale_factors(WMA3 } if(s->channel[c].cur_subframe > 0){ - s->channel[c].transmit_sf = get_bits(s->getbit,1); + s->channel[c].transmit_sf = get_bits(&s->getbit,1); }else s->channel[c].transmit_sf = 1; @@ -901,9 +901,9 @@ static int wma_decode_scale_factors(WMA3 if(!s->channel[c].reuse_sf){ int i; - s->channel[c].scale_factor_step = get_bits(s->getbit,2) + 1; + s->channel[c].scale_factor_step = get_bits(&s->getbit,2) + 1; for(i=0;i<s->num_bands;i++){ - int val = get_vlc2(s->getbit, s->sf_vlc.table, SCALEVLCBITS, ((FF_WMA3_HUFF_SCALE_MAXBITS+SCALEVLCBITS-1)/SCALEVLCBITS)); // DPCM-coded + int val = get_vlc2(&s->getbit, s->sf_vlc.table, SCALEVLCBITS, ((FF_WMA3_HUFF_SCALE_MAXBITS+SCALEVLCBITS-1)/SCALEVLCBITS)); // DPCM-coded if(!i) s->channel[c].scale_factors[i] = 45 / s->channel[c].scale_factor_step + val - 60; else @@ -920,10 +920,10 @@ static int wma_decode_scale_factors(WMA3 short level_mask; short val; - idx = get_vlc2(s->getbit, s->sf_rl_vlc.table, VLCBITS, ((FF_WMA3_HUFF_SCALE_RL_MAXBITS+VLCBITS-1)/VLCBITS)); + idx = get_vlc2(&s->getbit, s->sf_rl_vlc.table, VLCBITS, ((FF_WMA3_HUFF_SCALE_RL_MAXBITS+VLCBITS-1)/VLCBITS)); if( !idx ){ - uint32_t mask = get_bits(s->getbit,14); + uint32_t mask = get_bits(&s->getbit,14); level_mask = mask >> 6; val = (mask & 1) - 1; skip = (0x3f & mask)>>1; @@ -932,7 +932,7 @@ static int wma_decode_scale_factors(WMA3 }else{ skip = ff_wma3_scale_rl_run[idx]; level_mask = ff_wma3_scale_rl_level[idx]; - val = get_bits(s->getbit,1)-1; + val = get_bits(&s->getbit,1)-1; } i += skip; @@ -1104,8 +1104,7 @@ static int wma_decode_subframe(WMA3Decod int total_samples = s->samples_per_frame * s->num_channels; int transmit_coeffs = 0; - bitstreamcounter = get_bits_count(s->getbit); - + bitstreamcounter = get_bits_count(&s->getbit); /** reset channel context and find the next block offset and size == the next block of the channel with the smallest number of decoded samples @@ -1177,20 +1176,20 @@ static int wma_decode_subframe(WMA3Decod s->esc_len = av_log2(s->subframe_len - 1) + 1; /** skip extended header if any */ - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ int num_fill_bits; - if(!(num_fill_bits = get_bits(s->getbit,2))){ - num_fill_bits = get_bits(s->getbit,4); - num_fill_bits = get_bits(s->getbit,num_fill_bits) + 1; + if(!(num_fill_bits = get_bits(&s->getbit,2))){ + num_fill_bits = get_bits(&s->getbit,4); + num_fill_bits = get_bits(&s->getbit,num_fill_bits) + 1; } if(num_fill_bits >= 0){ - skip_bits(s->getbit,num_fill_bits); + skip_bits(&s->getbit,num_fill_bits); } } /** no idea for what the following bit is used */ - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ av_log(s->avctx,AV_LOG_ERROR,"reserved bit set\n"); return 0; } @@ -1202,7 +1201,7 @@ static int wma_decode_subframe(WMA3Decod for(i=0;i<s->channels_for_cur_subframe;i++){ int c = s->channel_indexes_for_cur_subframe[i]; - if((s->channel[c].transmit_coefs = get_bits(s->getbit,1))) + if((s->channel[c].transmit_coefs = get_bits(&s->getbit,1))) transmit_coeffs = 1; } @@ -1212,12 +1211,12 @@ static int wma_decode_subframe(WMA3Decod int quant; int sign = 1; int large_quant = 0; - if((get_bits(s->getbit,1))){ /** FIXME: might influence how often getvec4 may be called */ + if((get_bits(&s->getbit,1))){ /** FIXME: might influence how often getvec4 may be called */ av_log(s->avctx,AV_LOG_ERROR,"unsupported quant step coding\n"); return 0; } /** decode quantization step */ - quant = get_bits(s->getbit,6); + quant = get_bits(&s->getbit,6); if(quant & 0x20){ quant |= 0xFFFFFFC0u; sign = -1; @@ -1226,7 +1225,7 @@ static int wma_decode_subframe(WMA3Decod if(quant <= -32 || quant > 30) large_quant = 1; while(large_quant){ - quant = get_bits(s->getbit,5); + quant = get_bits(&s->getbit,5); if(quant != 31){ s->quant_step += quant * sign; break; @@ -1244,13 +1243,13 @@ static int wma_decode_subframe(WMA3Decod if(s->channels_for_cur_subframe == 1) s->channel[s->channel_indexes_for_cur_subframe[0]].quant_step_modifier = 0; else{ - int modifier_len = get_bits(s->getbit,3); + int modifier_len = get_bits(&s->getbit,3); for(i=0;i<s->channels_for_cur_subframe;i++){ int c = s->channel_indexes_for_cur_subframe[i]; s->channel[c].quant_step_modifier = 0; - if(get_bits(s->getbit,1)){ + if(get_bits(&s->getbit,1)){ if(modifier_len) - s->channel[c].quant_step_modifier = get_bits(s->getbit,modifier_len) + 1; + s->channel[c].quant_step_modifier = get_bits(&s->getbit,modifier_len) + 1; else s->channel[c].quant_step_modifier = 1; }else @@ -1264,7 +1263,7 @@ static int wma_decode_subframe(WMA3Decod return 0; } - av_log(s->avctx,AV_LOG_DEBUG,"BITSTREAM: subframe header length was %i\n",get_bits_count(s->getbit) - bitstreamcounter); + av_log(s->avctx,AV_LOG_DEBUG,"BITSTREAM: subframe header length was %i\n",get_bits_count(&s->getbit) - bitstreamcounter); /** parse coefficients */ for(i=0;i<s->channels_for_cur_subframe;i++){ @@ -1273,7 +1272,7 @@ static int wma_decode_subframe(WMA3Decod decode_coeffs(s,c); } - av_log(s->avctx,AV_LOG_DEBUG,"BITSTREAM: subframe length was %i\n",get_bits_count(s->getbit) - bitstreamcounter); + av_log(s->avctx,AV_LOG_DEBUG,"BITSTREAM: subframe length was %i\n",get_bits_count(&s->getbit) - bitstreamcounter); if(transmit_coeffs){ wma_inverse_channel_transform(s); @@ -1340,14 +1339,14 @@ static int wma_decode_subframe(WMA3Decod *@return 0 if the trailer bit indicates that this is the last frame, * 1 if there are more frames */ -static int wma_decode_frame(WMA3DecodeContext *s,GetBitContext* gb) +static int wma_decode_frame(WMA3DecodeContext *s) { - unsigned int gb_start_count = get_bits_count(gb); + GetBitContext* gb = &s->getbit; int more_frames = 0; int len = 0; int i; - s->getbit = gb; + init_get_bits(&s->getbit, s->frame_data,s->prev_packet_bit_size); /** check for potential output buffer overflow */ if(s->samples + s->num_channels * s->samples_per_frame > s->samples_end){ @@ -1455,12 +1454,12 @@ static int wma_decode_frame(WMA3DecodeCo // FIXME: remove - av_log(s->avctx,AV_LOG_DEBUG,"frame[%i] skipping %i bits\n",s->frame_num,len - (get_bits_count(gb) - gb_start_count) - 1); - if(len - (get_bits_count(gb) - gb_start_count) - 1 != 1) + av_log(s->avctx,AV_LOG_DEBUG,"frame[%i] skipping %i bits\n",s->frame_num,len - get_bits_count(gb) - 1); + if(len != get_bits_count(gb) + 2) assert(0); /** skip the rest of the frame data */ - skip_bits_long(gb,len - (get_bits_count(gb) - gb_start_count) - 1); + skip_bits_long(gb,len - get_bits_count(gb) - 1); /** decode trailer bit */ more_frames = get_bits1(gb); @@ -1580,12 +1579,8 @@ static int wma3_decode_packet(AVCodecCon s->prev_packet_bit_size); /** decode the cross packet frame if it is valid */ - if(!s->packet_loss){ - GetBitContext gb_prev; - init_get_bits(&gb_prev, s->frame_data, - s->prev_packet_bit_size); - wma_decode_frame(s,&gb_prev); - } + if(!s->packet_loss) + wma_decode_frame(s); }else if(s->prev_packet_bit_size){ av_log(avctx, AV_LOG_DEBUG, "ignoring %x previously saved bits\n", s->prev_packet_bit_size); @@ -1600,14 +1595,11 @@ static int wma3_decode_packet(AVCodecCon /** there is enough data for a full frame */ if(remaining_bits(s,&gb) >= frame_size){ - GetBitContext gb_prev; s->prev_packet_bit_size = 0; save_bits(s,&gb,frame_size); - init_get_bits(&gb_prev, s->frame_data, - s->prev_packet_bit_size); /** decode the frame */ - more_frames = wma_decode_frame(s,&gb_prev); + more_frames = wma_decode_frame(s); if(!more_frames){ av_log(avctx, AV_LOG_DEBUG, "no more frames\n");

1 0

[soc]: r4151 - wmapro/wma3dec.c
by faust3 03 Mar '09

03 Mar '09

Author: faust3 Date: Tue Mar 3 21:25:50 2009 New Revision: 4151 Log: always copy the input buffer to a temporary buffer before starting the decoding Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Tue Mar 3 20:59:51 2009 (r4150) +++ wmapro/wma3dec.c Tue Mar 3 21:25:50 2009 (r4151) @@ -1600,9 +1600,14 @@ static int wma3_decode_packet(AVCodecCon /** there is enough data for a full frame */ if(remaining_bits(s,&gb) >= frame_size){ + GetBitContext gb_prev; + s->prev_packet_bit_size = 0; + save_bits(s,&gb,frame_size); + init_get_bits(&gb_prev, s->frame_data, + s->prev_packet_bit_size); /** decode the frame */ - more_frames = wma_decode_frame(s,&gb); + more_frames = wma_decode_frame(s,&gb_prev); if(!more_frames){ av_log(avctx, AV_LOG_DEBUG, "no more frames\n"); @@ -1612,6 +1617,7 @@ static int wma3_decode_packet(AVCodecCon } if(!s->packet_loss){ + s->prev_packet_bit_size = 0; /** save the rest of the data so that it can be decoded with the next packet */ save_bits(s,&gb,remaining_bits(s,&gb));

1 0

[soc]: r4150 - in wmapro: wma3.h wma3dec.c
by faust3 03 Mar '09

03 Mar '09

Author: faust3 Date: Tue Mar 3 20:59:51 2009 New Revision: 4150 Log: use a fixed size buffer for cross packet frames Modified: wmapro/wma3.h wmapro/wma3dec.c Modified: wmapro/wma3.h ============================================================================== --- wmapro/wma3.h Mon Mar 2 23:19:00 2009 (r4149) +++ wmapro/wma3.h Tue Mar 3 20:59:51 2009 (r4150) @@ -35,6 +35,7 @@ #define MAX_CHANNELS 8 //< max number of handled channels #define MAX_SUBFRAMES 32 //< max number of subframes per channel #define MAX_BANDS 29 //< max number of scale factor bands +#define MAX_FRAMESIZE 16384 //< maximum compressed frame size /* size of block defines taken from wma.h */ #define BLOCK_MIN_BITS 7 //< log2 of min block size @@ -88,6 +89,8 @@ typedef struct WMA3DecodeContext { /** generic decoder variables */ AVCodecContext* avctx; //< codec context for av_log DSPContext dsp; //< accelerated dsp functions + uint8_t frame_data[MAX_FRAMESIZE + + FF_INPUT_BUFFER_PADDING_SIZE];//< compressed frame data MDCTContext mdct_ctx[BLOCK_NB_SIZES]; //< MDCT context per block size DECLARE_ALIGNED_16(float, tmp[BLOCK_MAX_SIZE]); //< imdct output buffer float* windows[BLOCK_NB_SIZES]; //< window per block size @@ -122,7 +125,6 @@ typedef struct WMA3DecodeContext { /** packet decode state */ uint8_t packet_sequence_number; //< current packet number int prev_packet_bit_size; //< saved number of bits - uint8_t* prev_packet_data; //< prev frame data uint8_t bit5; //< padding bit? (CBR files) uint8_t bit6; //< unknown uint8_t packet_loss; //< set in case of bitstream error Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Mon Mar 2 23:19:00 2009 (r4149) +++ wmapro/wma3dec.c Tue Mar 3 20:59:51 2009 (r4150) @@ -105,7 +105,6 @@ static av_cold int wma3_decode_end(AVCod WMA3DecodeContext *s = avctx->priv_data; int i; - av_free(s->prev_packet_data); av_free(s->num_sfb); av_free(s->sfb_offsets); av_free(s->subwoofer_cutoffs); @@ -1494,32 +1493,31 @@ static void save_bits(WMA3DecodeContext int pos = (s->prev_packet_bit_size - bit_offset) / 8; s->prev_packet_bit_size += len; - if(len <= 0) + if(len <= 0 || buflen > MAX_FRAMESIZE){ + av_log(s->avctx, AV_LOG_ERROR, "input buffer to small\n"); + s->packet_loss = 1; return; - - /** increase length if needed */ - s->prev_packet_data = av_realloc(s->prev_packet_data,buflen + - FF_INPUT_BUFFER_PADDING_SIZE); + } /** byte align prev_frame buffer */ if(bit_offset){ int missing = 8 - bit_offset; if(len < missing) missing = len; - s->prev_packet_data[pos++] |= + s->frame_data[pos++] |= get_bits(gb, missing) << (8 - bit_offset - missing); len -= missing; } /** copy full bytes */ while(len > 7){ - s->prev_packet_data[pos++] = get_bits(gb,8); + s->frame_data[pos++] = get_bits(gb,8); len -= 8; } /** copy remaining bits */ if(len > 0) - s->prev_packet_data[pos++] = get_bits(gb,len) << (8 - len); + s->frame_data[pos++] = get_bits(gb,len) << (8 - len); } /** @@ -1584,7 +1582,7 @@ static int wma3_decode_packet(AVCodecCon /** decode the cross packet frame if it is valid */ if(!s->packet_loss){ GetBitContext gb_prev; - init_get_bits(&gb_prev, s->prev_packet_data, + init_get_bits(&gb_prev, s->frame_data, s->prev_packet_bit_size); wma_decode_frame(s,&gb_prev); }

1 0

libavfilter - filter for .ass subtitle rendering using libass
by Alexey Lebedeff 02 Mar '09

02 Mar '09

I've made a simple filter which renders subtitles using libass, and now using ffmpeg with it to create .mp4 files for streaming onto PS3. Before that, I made half a dozen of unsuccessful attempts using mplayer/mencoder. So for me it is the only working method for making .ass hardsubs on linux server. However, there is some license problems: libass is under GPL, not LGPL as the rest of libavfilter/ffmpeg. If there is any interest in it, I'll do some cleanup to code, properly integrate it into build process, and share the results.

7 14

[soc]: r4149 - libavfilter/vf_drawbox.c
by stefano 02 Mar '09

02 Mar '09

Author: stefano Date: Mon Mar 2 23:19:00 2009 New Revision: 4149 Log: Apply whitespace cosmetics. Modified: libavfilter/vf_drawbox.c Modified: libavfilter/vf_drawbox.c ============================================================================== --- libavfilter/vf_drawbox.c Mon Mar 2 21:07:15 2009 (r4148) +++ libavfilter/vf_drawbox.c Mon Mar 2 23:19:00 2009 (r4149) @@ -50,9 +50,9 @@ static box_color colors[] = typedef struct { - int x,y,w,h; + int x, y, w, h; box_color color; - int vsub,hsub; //< chroma subsampling + int vsub, hsub; //< chroma subsampling } BoxContext; static void box_strlwr(char *str) @@ -126,7 +126,7 @@ static void start_frame(AVFilterLink *li static void draw_box(AVFilterPicRef *pic, BoxContext* context, box_color color) { - int x,y; + int x, y; int channel; unsigned char *row[4]; int xb = context->x; @@ -152,7 +152,7 @@ static void draw_box(AVFilterPicRef *pic static void end_frame(AVFilterLink *link) { BoxContext *context = link->dst->priv; - AVFilterLink* output = link->dst->outputs[0]; + AVFilterLink *output = link->dst->outputs[0]; AVFilterPicRef *pic = link->cur_pic; draw_box(pic,context,context->color);

1 0

[soc]: r4148 - wmapro/wma3dec.c
by faust3 02 Mar '09

02 Mar '09

Author: faust3 Date: Mon Mar 2 21:07:15 2009 New Revision: 4148 Log: always decode block_align bytes Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Mon Mar 2 20:53:23 2009 (r4147) +++ wmapro/wma3dec.c Mon Mar 2 21:07:15 2009 (r4148) @@ -1552,6 +1552,8 @@ static int wma3_decode_packet(AVCodecCon if(buf_size < avctx->block_align) return 0; + buf_size = avctx->block_align; + /** parse packet header */ init_get_bits(&gb, buf, s->buf_bit_size); packet_sequence_number = get_bits(&gb, 4);

1 0

[soc]: r4147 - wmapro/wma3dec.c
by faust3 02 Mar '09

02 Mar '09

Author: faust3 Date: Mon Mar 2 20:53:23 2009 New Revision: 4147 Log: be less verbose about the no more frames condition Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Mon Mar 2 20:44:12 2009 (r4146) +++ wmapro/wma3dec.c Mon Mar 2 20:53:23 2009 (r4147) @@ -1587,7 +1587,7 @@ static int wma3_decode_packet(AVCodecCon wma_decode_frame(s,&gb_prev); } }else if(s->prev_packet_bit_size){ - av_log(avctx, AV_LOG_ERROR, "ignoring %x previously saved bits\n", + av_log(avctx, AV_LOG_DEBUG, "ignoring %x previously saved bits\n", s->prev_packet_bit_size); } @@ -1605,7 +1605,7 @@ static int wma3_decode_packet(AVCodecCon more_frames = wma_decode_frame(s,&gb); if(!more_frames){ - av_log(avctx, AV_LOG_ERROR, "no more frames\n"); + av_log(avctx, AV_LOG_DEBUG, "no more frames\n"); } }else more_frames = 0;

1 0

[soc]: r4146 - wmapro/wma3dec.c
by faust3 02 Mar '09

02 Mar '09

Author: faust3 Date: Mon Mar 2 20:44:12 2009 New Revision: 4146 Log: len_prefix should be boolean Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Mon Mar 2 20:38:19 2009 (r4145) +++ wmapro/wma3dec.c Mon Mar 2 20:44:12 2009 (r4146) @@ -172,7 +172,7 @@ static av_cold int wma3_decode_init(AVCo /** frame info */ s->skip_frame = 1; /** skip first frame */ s->packet_loss = 1; - s->len_prefix = s->decode_flags & 0x40; + s->len_prefix = (s->decode_flags & 0x40) >> 6; if(!s->len_prefix){ av_log(avctx, AV_LOG_ERROR, "file has no length prefix, please report\n");

1 0

[soc]: r4145 - wmapro/wma3dec.c
by faust3 02 Mar '09

02 Mar '09

Author: faust3 Date: Mon Mar 2 20:38:19 2009 New Revision: 4145 Log: implemented flush Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Mon Mar 2 19:47:55 2009 (r4144) +++ wmapro/wma3dec.c Mon Mar 2 20:38:19 2009 (r4145) @@ -171,6 +171,7 @@ static av_cold int wma3_decode_init(AVCo /** frame info */ s->skip_frame = 1; /** skip first frame */ + s->packet_loss = 1; s->len_prefix = s->decode_flags & 0x40; if(!s->len_prefix){ @@ -1563,10 +1564,11 @@ static int wma3_decode_packet(AVCodecCon num_bits_prev_frame); /** check for packet loss */ - if (((s->packet_sequence_number + 1)&0xF) != packet_sequence_number) { + if (!s->packet_loss && + ((s->packet_sequence_number + 1)&0xF) != packet_sequence_number) { s->packet_loss = 1; - av_log(avctx, AV_LOG_ERROR, "!!Packet loss detected! seq %x vs %x\n", - s->packet_sequence_number,avctx->frame_number&0xF); + av_log(avctx, AV_LOG_ERROR, "Packet loss detected! seq %x vs %x\n", + s->packet_sequence_number,packet_sequence_number); } s->packet_sequence_number = packet_sequence_number; @@ -1613,9 +1615,6 @@ static int wma3_decode_packet(AVCodecCon /** save the rest of the data so that it can be decoded with the next packet */ save_bits(s,&gb,remaining_bits(s,&gb)); - }else{ - s->prev_packet_bit_size = 0; - s->packet_loss = 0; } *data_size = (int8_t *)s->samples - (int8_t *)data; @@ -1623,6 +1622,16 @@ static int wma3_decode_packet(AVCodecCon return avctx->block_align; } +static void wma3_flush(AVCodecContext *avctx) +{ + WMA3DecodeContext *s = avctx->priv_data; + int i; + for(i=0;i<s->num_channels;i++) + memset(s->channel[i].out, 0, sizeof(s->channel[0].out)); + s->packet_loss = 1; +} + + /** *@brief WMA9 decoder */ @@ -1636,5 +1645,6 @@ AVCodec wmapro_decoder = NULL, wma3_decode_end, wma3_decode_packet, + .flush= wma3_flush, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"), };

1 0

[soc]: r4144 - wmapro/wma3dec.c
by faust3 02 Mar '09

02 Mar '09

Author: faust3 Date: Mon Mar 2 19:47:55 2009 New Revision: 4144 Log: fixed seeking Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Mon Mar 2 19:18:20 2009 (r4143) +++ wmapro/wma3dec.c Mon Mar 2 19:47:55 2009 (r4144) @@ -1538,6 +1538,7 @@ static int wma3_decode_packet(AVCodecCon WMA3DecodeContext *s = avctx->priv_data; int more_frames=1; int num_bits_prev_frame; + int packet_sequence_number; s->samples = data; s->samples_end = (int16_t*)((int8_t*)data + *data_size); @@ -1552,7 +1553,7 @@ static int wma3_decode_packet(AVCodecCon /** parse packet header */ init_get_bits(&gb, buf, s->buf_bit_size); - s->packet_sequence_number = get_bits(&gb, 4); + packet_sequence_number = get_bits(&gb, 4); s->bit5 = get_bits1(&gb); s->bit6 = get_bits1(&gb); @@ -1562,11 +1563,12 @@ static int wma3_decode_packet(AVCodecCon num_bits_prev_frame); /** check for packet loss */ - if (s->packet_sequence_number != (avctx->frame_number&0xF)) { + if (((s->packet_sequence_number + 1)&0xF) != packet_sequence_number) { s->packet_loss = 1; av_log(avctx, AV_LOG_ERROR, "!!Packet loss detected! seq %x vs %x\n", s->packet_sequence_number,avctx->frame_number&0xF); } + s->packet_sequence_number = packet_sequence_number; if (num_bits_prev_frame > 0) { /** append the previous frame data to the remaining data from the @@ -1607,12 +1609,14 @@ static int wma3_decode_packet(AVCodecCon more_frames = 0; } - if(s->packet_loss == 1 && !s->negative_quantstep) - assert(0); - - /** save the rest of the data so that it can be decoded - with the next packet */ - save_bits(s,&gb,remaining_bits(s,&gb)); + if(!s->packet_loss){ + /** save the rest of the data so that it can be decoded + with the next packet */ + save_bits(s,&gb,remaining_bits(s,&gb)); + }else{ + s->prev_packet_bit_size = 0; + s->packet_loss = 0; + } *data_size = (int8_t *)s->samples - (int8_t *)data;

1 0

[soc]: r4143 - wmapro/wma3dec.c
by faust3 02 Mar '09

02 Mar '09

Author: faust3 Date: Mon Mar 2 19:18:20 2009 New Revision: 4143 Log: leave main decoding loop if packet loss was detected Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Sun Mar 1 21:57:34 2009 (r4142) +++ wmapro/wma3dec.c Mon Mar 2 19:18:20 2009 (r4143) @@ -1591,7 +1591,7 @@ static int wma3_decode_packet(AVCodecCon s->prev_packet_bit_size = 0; s->packet_loss = 0; /** decode the rest of the packet */ - while(more_frames && remaining_bits(s,&gb) > s->log2_frame_size){ + while(!s->packet_loss && more_frames && remaining_bits(s,&gb) > s->log2_frame_size){ int frame_size = show_bits(&gb, s->log2_frame_size); /** there is enough data for a full frame */

1 0

[soc]: r4142 - wmapro/wma3dec.c
by faust3 01 Mar '09

01 Mar '09

Author: faust3 Date: Sun Mar 1 21:57:34 2009 New Revision: 4142 Log: 10l: fixed decoding on x86_64 Modified: wmapro/wma3dec.c Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Sun Mar 1 14:48:44 2009 (r4141) +++ wmapro/wma3dec.c Sun Mar 1 21:57:34 2009 (r4142) @@ -336,7 +336,7 @@ static av_cold int wma3_decode_init(AVCo } /** set up decorrelation matrixes */ - s->def_decorrelation_mat = av_mallocz(sizeof(int) * (s->num_channels + 1)); + s->def_decorrelation_mat = av_mallocz(sizeof(float*) * (s->num_channels + 1)); if(!s->def_decorrelation_mat){ av_log(avctx, AV_LOG_ERROR, "failed to allocate decorrelation matrix\n"); wma3_decode_end(avctx); @@ -346,7 +346,7 @@ static av_cold int wma3_decode_init(AVCo s->def_decorrelation_mat[0] = 0; for(i=1;i<=s->num_channels;i++){ const float* tab = ff_wma3_default_decorrelation_matrices; - s->def_decorrelation_mat[i] = av_mallocz(sizeof(float) * i); + s->def_decorrelation_mat[i] = av_mallocz(sizeof(float*) * i); if(!s->def_decorrelation_mat[i]){ av_log(avctx, AV_LOG_ERROR, "failed to set up decorrelation matrix\n"); wma3_decode_end(avctx);

1 0

[soc]: r4140 - wmapro/mplayer.patch
by faust3 01 Mar '09

01 Mar '09

Author: faust3 Date: Sun Mar 1 14:33:49 2009 New Revision: 4140 Log: added codecs.conf patch for MPlayer Added: wmapro/mplayer.patch Added: wmapro/mplayer.patch ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ wmapro/mplayer.patch Sun Mar 1 14:33:49 2009 (r4140) @@ -0,0 +1,19 @@ +Index: etc/codecs.conf +=================================================================== +--- etc/codecs.conf (revision 28740) ++++ etc/codecs.conf (working copy) +@@ -3595,6 +3595,14 @@ + driver ffmpeg + dll "wmav2" + ++audiocodec ffwmapro ++ info "WMA Pro audio (FFmpeg)" ++ status untested ++ format 0x162 ++ fourcc "WMA3" ++ driver ffmpeg ++ dll "wmapro" ++ + audiocodec ffmac3 + info "Macintosh Audio Compression and Expansion 3:1" + status untested

3 2

[soc]: r4141 - wmapro/mplayer.patch
by faust3 01 Mar '09

01 Mar '09

Author: faust3 Date: Sun Mar 1 14:48:44 2009 New Revision: 4141 Log: changed codecs.conf info as suggested by Diego Modified: wmapro/mplayer.patch Modified: wmapro/mplayer.patch ============================================================================== --- wmapro/mplayer.patch Sun Mar 1 14:33:49 2009 (r4140) +++ wmapro/mplayer.patch Sun Mar 1 14:48:44 2009 (r4141) @@ -7,7 +7,7 @@ Index: etc/codecs.conf dll "wmav2" +audiocodec ffwmapro -+ info "WMA Pro audio (FFmpeg)" ++ info "FFmpeg WMA Pro audio" + status untested + format 0x162 + fourcc "WMA3"

1 0