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  • Flash : Adobe publie à son tour une lettre ouverte et lance une campagne de publicité pour répondre

    Mise à jour du 14/05/10 Flash : Adobe publie à son tour une lettre ouverte Et lance une campagne de publicité pour répondre à Apple La mode est aux lettres ouvertes. Après celle de Steve Jobs, qui accusait Flash d'être une technologie fermée et qui justifiait son choix de l'exclure de l'iPad, c'est au tour d'Adobe d'utiliser cette forme d'expression. Que nous dit cette lettre ? Objectivement, rien de bien nouveau. « Nos pensées sur les marchés ouverts » commence par quelques généralités (le web c'est l'innovation, etc.), pour finalement en venir à la position d'Adobe que la société défend depuis qu...

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  • Windows 8 sera adapté aux puces ARM, Microsoft fait la première démonstration officielle de l'OS au CES de Las Vegas

    Windows 8 sera adapté aux puces ARM Microsoft fait la première démonstration officielle de l'OS au CES de Las Vegas Mise à jour du 06/01/2011 par Idelways Après plus d'un an de spéculations et de fuites involontaires (ou pas), Steve Ballmer vient de faire la première démonstration du successeur de Windows 7, et a annoncé que l'OS supportera en natif les puces ARM. On sait donc désormais que Windows 8 (bien que Ballmer ait fait très attention d'éviter de lui donner le moindre nom) fonctionnera sur une architecture System-on-a-chip (SoC), fournie dans un premier temps par NVIDIA et Texas Instruments. Le PDG ...

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  • Windows Phone 8 : les terminaux se vendent quatre fois plus que l'an dernier, selon le PDG de Microsoft

    Windows Phone 8 : les terminaux se vendent quatre fois plus que l'an dernier selon le PDG de Microsoft Les systèmes d'exploitation de Microsoft ont connu un bon, voire même très bon démarrage. Après Windows 8 qui a fait état de 40 millions de licences vendues en seulement un mois, c'est au tour de Windows Phone 8 de livrer ses chiffres. Steve Ballmer, PDG de Microsoft, a déclaré lors d'une assemblée générale des actionnaires que Windows Phone se vendait quatre fois plus qu'à la même période de l'année dernière. Le PDG de Microsoft se félicite du bon accueil des terminaux Windows Phone 8X de HTC et L...

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  • Les tablettes vont-elles réussir leur percée en entreprise ? 38 % des professionnels prêts à remplacer leur notebook par un de ces appareils

    Les tablettes vont-elles réussir leur percée en entreprise ? 38% des professionnels serait prêt à remplacer leur notebook par l'un de ces appareils C'est un scénario que plusieurs experts envisagent depuis des mois, et c'est aussi une possibilité que Steve Ballmer réfute : les tablettes pourraient remplacer les ordinateurs et les netbooks. A moins d'être sourd et aveugle, tout le monde a constaté le succès phénoménal qu'ont rencontré ces appareils. La société de conseil ChangeWave s'est penchée sur la question, mais depuis le point de vue des professionnels. Elle a ainsi interrogé 1641 employés à propos de leur intérêt pour les tablettes, dans le cadre de leurs fonctions. Et les résultats sont clairs :...

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  • Map a GoogleDocs Spreadsheet to a SQL table

    - by justSteve
    I'm looking for a generalized plan of attack at this problem: Upon saving a spreadsheet in my Google Docs account a SQL 08 table is updated. Or perhaps via spreadsheet's new 'Form' input method. But the overarching goal is letting google docs function as a front-end editor to sql. thx --steve...

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  • ffmpeg - How to determine if -movflags faststart is enabled? PHP

    - by IIIOXIII
    While I am able to encode an mp4 file which I can plan on my local windows machine, I am having trouble encoding files to mp4 which are readable when streaming by safari, etc. After a bit of reading, I believe my issue is that I must move the metadata from the end of the file to the beginning in order for the converted mp4 files to be streamable. To that end, I am trying to find out if the build of ffmpeg that I am currently using is able to use the -movflags faststart option through php - as my current outputted mp4 files are not working when streamed online. This is the way I am now echoing the -help, -formats, -codecs, but I am not seeing anything about -movflags faststart in any of the lists: exec($ffmpegPath." -help", $codecArr); for($ii=0;$ii<count($codecArr);$ii++){ echo $codecArr[$ii].'</br>'; } Is there a similar method of determining if -movflags fastart is available to my ffmpeg build? Any other way? Should it be listed with any of the previously suggested commands? -help/-formats? Can someone that knows it is enabled in their version of ffmpeg check to see if it is listed under -help or -formats, etc.? TIA. EDIT: COMPLETE CONSOLE OUTPUT FOR BOTH THE CONVERSION COMMAND AND -MOVFLAGS COMMAND BELOW: COMMAND: ffmpeg_new -i C:\vidtests\Wildlife.wmv -s 640x480 C:\vidtests\Wildlife.mp4 OUTPUT: ffmpeg version N-54207-ge59fb3f Copyright (c) 2000-2013 the FFmpeg developers built on Jun 25 2013 21:55:00 with gcc 4.7.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo- amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs -- enable-libxvid --enable-zlib libavutil 52. 37.101 / 52. 37.101 libavcodec 55. 17.100 / 55. 17.100 libavformat 55. 10.100 / 55. 10.100 libavdevice 55. 2.100 / 55. 2.100 libavfilter 3. 77.101 / 3. 77.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [asf @ 00000000002ed760] Stream #0: not enough frames to estimate rate; consider increasing probesize Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, asf, from 'C:\vidtests\Wildlife.wmv' : Metadata: SfOriginalFPS : 299700 WMFSDKVersion : 11.0.6001.7000 WMFSDKNeeded : 0.0.0.0000 comment : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball title : Wildlife in HD copyright : -¬ 2008 Microsoft Corporation IsVBR : 0 DeviceConformanceTemplate: AP@L3 Duration: 00:00:30.09, start: 0.000000, bitrate: 6977 kb/s Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, stereo, fltp , 192 kb/s Stream #0:1(eng): Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1280x7 20, 5942 kb/s, 29.97 tbr, 1k tbn, 1k tbc [libx264 @ 00000000002e6980] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64 [libx264 @ 00000000002e6980] profile High, level 3.0 [libx264 @ 00000000002e6980] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC cod ec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 r ef=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed _ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pski p=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 deci mate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_ adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=2 5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.6 0 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'C:\vidtests\Wildlife.mp4': Metadata: SfOriginalFPS : 299700 WMFSDKVersion : 11.0.6001.7000 WMFSDKNeeded : 0.0.0.0000 comment : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball title : Wildlife in HD copyright : -¬ 2008 Microsoft Corporation IsVBR : 0 DeviceConformanceTemplate: AP@L3 encoder : Lavf55.10.100 Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 6 40x480, q=-1--1, 30k tbn, 29.97 tbc Stream #0:1(eng): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (vc1 -> libx264) Stream #0:0 -> #0:1 (wmav2 -> libvo_aacenc) Press [q] to stop, [?] for help frame= 53 fps= 49 q=29.0 size= 0kB time=00:00:00.13 bitrate= 2.9kbits/ frame= 63 fps= 40 q=29.0 size= 0kB time=00:00:00.46 bitrate= 0.8kbits/ frame= 74 fps= 35 q=29.0 size= 0kB time=00:00:00.83 bitrate= 0.5kbits/ frame= 85 fps= 32 q=29.0 size= 0kB time=00:00:01.20 bitrate= 0.3kbits/ frame= 95 fps= 30 q=29.0 size= 0kB time=00:00:01.53 bitrate= 0.3kbits/ frame= 107 fps= 28 q=29.0 size= 0kB time=00:00:01.93 bitrate= 0.2kbits/ Queue input is backward in time [mp4 @ 00000000003ef800] Non-monotonous DTS in output stream 0:1; previous: 7616 , current: 7063; changing to 7617. This may result in incorrect timestamps in th e output file. frame= 118 fps= 28 q=29.0 size= 113kB time=00:00:02.30 bitrate= 402.6kbits/ frame= 129 fps= 26 q=29.0 size= 219kB time=00:00:02.66 bitrate= 670.7kbits/ frame= 141 fps= 26 q=29.0 size= 264kB time=00:00:03.06 bitrate= 704.2kbits/ frame= 152 fps= 25 q=29.0 size= 328kB time=00:00:03.43 bitrate= 782.2kbits/ frame= 163 fps= 25 q=29.0 size= 431kB time=00:00:03.80 bitrate= 928.1kbits/ frame= 174 fps= 24 q=29.0 size= 568kB time=00:00:04.17 bitrate=1116.3kbits/ frame= 190 fps= 25 q=29.0 size= 781kB time=00:00:04.70 bitrate=1359.9kbits/ frame= 204 fps= 25 q=29.0 size= 1006kB time=00:00:05.17 bitrate=1593.1kbits/ frame= 218 fps= 25 q=29.0 size= 1058kB time=00:00:05.63 bitrate=1536.8kbits/ frame= 229 fps= 25 q=29.0 size= 1093kB time=00:00:06.00 bitrate=1490.9kbits/ frame= 239 fps= 24 q=29.0 size= 1118kB time=00:00:06.33 bitrate=1444.4kbits/ frame= 251 fps= 24 q=29.0 size= 1150kB time=00:00:06.74 bitrate=1397.9kbits/ frame= 265 fps= 24 q=29.0 size= 1234kB time=00:00:07.20 bitrate=1402.3kbits/ frame= 278 fps= 25 q=29.0 size= 1332kB time=00:00:07.64 bitrate=1428.3kbits/ frame= 294 fps= 25 q=29.0 size= 1403kB time=00:00:08.17 bitrate=1405.7kbits/ frame= 308 fps= 25 q=29.0 size= 1547kB time=00:00:08.64 bitrate=1466.4kbits/ frame= 323 fps= 25 q=29.0 size= 1595kB time=00:00:09.14 bitrate=1429.5kbits/ frame= 337 fps= 25 q=29.0 size= 1702kB time=00:00:09.60 bitrate=1450.7kbits/ frame= 351 fps= 25 q=29.0 size= 1755kB time=00:00:10.07 bitrate=1427.1kbits/ frame= 365 fps= 25 q=29.0 size= 1820kB time=00:00:10.54 bitrate=1414.1kbits/ frame= 381 fps= 25 q=29.0 size= 1852kB time=00:00:11.07 bitrate=1369.6kbits/ frame= 396 fps= 26 q=29.0 size= 1893kB time=00:00:11.57 bitrate=1339.5kbits/ frame= 409 fps= 26 q=29.0 size= 1923kB time=00:00:12.01 bitrate=1311.8kbits/ frame= 421 fps= 25 q=29.0 size= 1967kB time=00:00:12.41 bitrate=1298.3kbits/ frame= 434 fps= 25 q=29.0 size= 1998kB time=00:00:12.84 bitrate=1274.0kbits/ frame= 445 fps= 25 q=29.0 size= 2018kB time=00:00:13.21 bitrate=1251.3kbits/ frame= 458 fps= 25 q=29.0 size= 2048kB time=00:00:13.64 bitrate=1229.5kbits/ frame= 471 fps= 25 q=29.0 size= 2067kB time=00:00:14.08 bitrate=1202.3kbits/ frame= 484 fps= 25 q=29.0 size= 2189kB time=00:00:14.51 bitrate=1235.5kbits/ frame= 497 fps= 25 q=29.0 size= 2260kB time=00:00:14.94 bitrate=1238.3kbits/ frame= 509 fps= 25 q=29.0 size= 2311kB time=00:00:15.34 bitrate=1233.3kbits/ frame= 523 fps= 25 q=29.0 size= 2429kB time=00:00:15.81 bitrate=1258.1kbits/ frame= 535 fps= 25 q=29.0 size= 2541kB time=00:00:16.21 bitrate=1283.5kbits/ frame= 548 fps= 25 q=29.0 size= 2718kB time=00:00:16.64 bitrate=1337.5kbits/ frame= 560 fps= 25 q=29.0 size= 2845kB time=00:00:17.05 bitrate=1367.1kbits/ frame= 571 fps= 25 q=29.0 size= 2965kB time=00:00:17.41 bitrate=1394.6kbits/ frame= 580 fps= 25 q=29.0 size= 3025kB time=00:00:17.71 bitrate=1398.7kbits/ frame= 588 fps= 25 q=29.0 size= 3098kB time=00:00:17.98 bitrate=1411.1kbits/ frame= 597 fps= 25 q=29.0 size= 3183kB time=00:00:18.28 bitrate=1426.1kbits/ frame= 606 fps= 24 q=29.0 size= 3279kB time=00:00:18.58 bitrate=1445.2kbits/ frame= 616 fps= 24 q=29.0 size= 3441kB time=00:00:18.91 bitrate=1489.9kbits/ frame= 626 fps= 24 q=29.0 size= 3650kB time=00:00:19.25 bitrate=1553.0kbits/ frame= 638 fps= 24 q=29.0 size= 3826kB time=00:00:19.65 bitrate=1594.7kbits/ frame= 649 fps= 24 q=29.0 size= 3950kB time=00:00:20.02 bitrate=1616.3kbits/ frame= 660 fps= 24 q=29.0 size= 4067kB time=00:00:20.38 bitrate=1634.1kbits/ frame= 669 fps= 24 q=29.0 size= 4121kB time=00:00:20.68 bitrate=1631.8kbits/ frame= 682 fps= 24 q=29.0 size= 4274kB time=00:00:21.12 bitrate=1657.9kbits/ frame= 696 fps= 24 q=29.0 size= 4446kB time=00:00:21.58 bitrate=1687.1kbits/ frame= 709 fps= 24 q=29.0 size= 4590kB time=00:00:22.02 bitrate=1707.3kbits/ frame= 719 fps= 24 q=29.0 size= 4772kB time=00:00:22.35 bitrate=1748.5kbits/ frame= 732 fps= 24 q=29.0 size= 4852kB time=00:00:22.78 bitrate=1744.3kbits/ frame= 744 fps= 24 q=29.0 size= 4973kB time=00:00:23.18 bitrate=1756.9kbits/ frame= 756 fps= 24 q=29.0 size= 5099kB time=00:00:23.59 bitrate=1770.8kbits/ frame= 768 fps= 24 q=29.0 size= 5149kB time=00:00:23.99 bitrate=1758.4kbits/ frame= 780 fps= 24 q=29.0 size= 5227kB time=00:00:24.39 bitrate=1755.7kbits/ frame= 797 fps= 24 q=29.0 size= 5377kB time=00:00:24.95 bitrate=1765.0kbits/ frame= 813 fps= 24 q=29.0 size= 5507kB time=00:00:25.49 bitrate=1769.5kbits/ frame= 828 fps= 24 q=29.0 size= 5634kB time=00:00:25.99 bitrate=1775.5kbits/ frame= 843 fps= 24 q=29.0 size= 5701kB time=00:00:26.49 bitrate=1762.9kbits/ frame= 859 fps= 24 q=29.0 size= 5830kB time=00:00:27.02 bitrate=1767.0kbits/ frame= 872 fps= 24 q=29.0 size= 5926kB time=00:00:27.46 bitrate=1767.7kbits/ frame= 888 fps= 24 q=29.0 size= 6014kB time=00:00:27.99 bitrate=1759.7kbits/ frame= 900 fps= 24 q=29.0 size= 6332kB time=00:00:28.39 bitrate=1826.9kbits/ frame= 901 fps= 24 q=-1.0 Lsize= 6717kB time=00:00:30.10 bitrate=1828.0kbits /s video:6211kB audio:472kB subtitle:0 global headers:0kB muxing overhead 0.513217% [libx264 @ 00000000002e6980] frame I:8 Avg QP:21.77 size: 39744 [libx264 @ 00000000002e6980] frame P:433 Avg QP:25.69 size: 11490 [libx264 @ 00000000002e6980] frame B:460 Avg QP:29.25 size: 2319 [libx264 @ 00000000002e6980] consecutive B-frames: 5.4% 78.6% 2.7% 13.3% [libx264 @ 00000000002e6980] mb I I16..4: 21.8% 48.8% 29.5% [libx264 @ 00000000002e6980] mb P I16..4: 0.7% 4.0% 1.3% P16..4: 37.1% 22.2 % 15.5% 0.0% 0.0% skip:19.2% [libx264 @ 00000000002e6980] mb B I16..4: 0.1% 0.5% 0.2% B16..8: 43.5% 7.0 % 2.1% direct: 2.2% skip:44.5% L0:36.4% L1:52.7% BI:10.9% [libx264 @ 00000000002e6980] 8x8 transform intra:62.8% inter:56.2% [libx264 @ 00000000002e6980] coded y,uvDC,uvAC intra: 74.2% 78.8% 44.0% inter: 2 3.6% 14.5% 1.0% [libx264 @ 00000000002e6980] i16 v,h,dc,p: 48% 24% 9% 20% [libx264 @ 00000000002e6980] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 16% 17% 15% 7% 8% 11% 8% 10% 8% [libx264 @ 00000000002e6980] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 17% 15% 7% 10% 11% 8% 7% 7% [libx264 @ 00000000002e6980] i8c dc,h,v,p: 53% 21% 18% 7% [libx264 @ 00000000002e6980] Weighted P-Frames: Y:0.7% UV:0.0% [libx264 @ 00000000002e6980] ref P L0: 62.4% 19.0% 12.0% 6.6% 0.0% [libx264 @ 00000000002e6980] ref B L0: 90.5% 8.9% 0.7% [libx264 @ 00000000002e6980] ref B L1: 97.9% 2.1% [libx264 @ 00000000002e6980] kb/s:1692.37 AND THE –MOVFLAGS COMMAND: C:\XSITE\SITE>ffmpeg_new -i C:\vidtests\Wildlife.mp4 -movflags faststart C:\vidtests\Wildlife_fs.mp4 AND THE –MOVFLAGS OUTPUT ffmpeg version N-54207-ge59fb3f Copyright (c) 2000-2013 the FFmpeg developers built on Jun 25 2013 21:55:00 with gcc 4.7.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo- amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs -- enable-libxvid --enable-zlib libavutil 52. 37.101 / 52. 37.101 libavcodec 55. 17.100 / 55. 17.100 libavformat 55. 10.100 / 55. 10.100 libavdevice 55. 2.100 / 55. 2.100 libavfilter 3. 77.101 / 3. 77.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\vidtests\Wildlife.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 title : Wildlife in HD encoder : Lavf55.10.100 comment : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball copyright : -¬ 2008 Microsoft Corporation Duration: 00:00:30.13, start: 0.036281, bitrate: 1826 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480, 1692 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 12 8 kb/s Metadata: handler_name : SoundHandler [libx264 @ 0000000004360620] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64 [libx264 @ 0000000004360620] profile High, level 3.0 [libx264 @ 0000000004360620] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC cod ec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 r ef=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed _ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pski p=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 deci mate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_ adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=2 5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.6 0 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'C:\vidtests\Wildlife_fs.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 title : Wildlife in HD copyright : -¬ 2008 Microsoft Corporation comment : Footage: Small World Productions, Inc; Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball encoder : Lavf55.10.100 Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 6 40x480, q=-1--1, 30k tbn, 29.97 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 128 kb/s Metadata: handler_name : SoundHandler Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (aac -> libvo_aacenc) Press [q] to stop, [?] for help frame= 52 fps=0.0 q=29.0 size= 29kB time=00:00:01.76 bitrate= 133.9kbits/ frame= 63 fps= 60 q=29.0 size= 104kB time=00:00:02.14 bitrate= 397.2kbits/ frame= 74 fps= 47 q=29.0 size= 176kB time=00:00:02.51 bitrate= 573.2kbits/ frame= 87 fps= 41 q=29.0 size= 265kB time=00:00:02.93 bitrate= 741.2kbits/ frame= 101 fps= 37 q=29.0 size= 358kB time=00:00:03.39 bitrate= 862.8kbits/ frame= 113 fps= 34 q=29.0 size= 437kB time=00:00:03.79 bitrate= 943.7kbits/ frame= 125 fps= 33 q=29.0 size= 520kB time=00:00:04.20 bitrate=1012.2kbits/ frame= 138 fps= 32 q=29.0 size= 606kB time=00:00:04.64 bitrate=1069.8kbits/ frame= 151 fps= 31 q=29.0 size= 696kB time=00:00:05.06 bitrate=1124.3kbits/ frame= 163 fps= 30 q=29.0 size= 780kB time=00:00:05.47 bitrate=1166.4kbits/ frame= 176 fps= 30 q=29.0 size= 919kB time=00:00:05.90 bitrate=1273.9kbits/ frame= 196 fps= 31 q=29.0 size= 994kB time=00:00:06.57 bitrate=1237.4kbits/ frame= 213 fps= 31 q=29.0 size= 1097kB time=00:00:07.13 bitrate=1258.8kbits/ frame= 225 fps= 30 q=29.0 size= 1204kB time=00:00:07.53 bitrate=1309.8kbits/ frame= 236 fps= 30 q=29.0 size= 1323kB time=00:00:07.91 bitrate=1369.4kbits/ frame= 249 fps= 29 q=29.0 size= 1451kB time=00:00:08.34 bitrate=1424.6kbits/ frame= 263 fps= 29 q=29.0 size= 1574kB time=00:00:08.82 bitrate=1461.3kbits/ frame= 278 fps= 29 q=29.0 size= 1610kB time=00:00:09.30 bitrate=1416.9kbits/ frame= 296 fps= 30 q=29.0 size= 1655kB time=00:00:09.91 bitrate=1368.0kbits/ frame= 313 fps= 30 q=29.0 size= 1697kB time=00:00:10.48 bitrate=1326.4kbits/ 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frame= 531 fps= 30 q=29.0 size= 2681kB time=00:00:17.73 bitrate=1238.2kbits/ frame= 546 fps= 30 q=29.0 size= 2758kB time=00:00:18.24 bitrate=1238.2kbits/ frame= 561 fps= 30 q=29.0 size= 2824kB time=00:00:18.75 bitrate=1233.4kbits/ frame= 576 fps= 30 q=29.0 size= 2955kB time=00:00:19.25 bitrate=1256.8kbits/ frame= 586 fps= 29 q=29.0 size= 3061kB time=00:00:19.59 bitrate=1279.6kbits/ frame= 598 fps= 29 q=29.0 size= 3217kB time=00:00:19.99 bitrate=1318.4kbits/ frame= 610 fps= 29 q=29.0 size= 3354kB time=00:00:20.39 bitrate=1347.2kbits/ frame= 622 fps= 29 q=29.0 size= 3483kB time=00:00:20.78 bitrate=1372.6kbits/ frame= 634 fps= 29 q=29.0 size= 3593kB time=00:00:21.19 bitrate=1388.6kbits/ frame= 648 fps= 29 q=29.0 size= 3708kB time=00:00:21.66 bitrate=1402.3kbits/ frame= 661 fps= 29 q=29.0 size= 3811kB time=00:00:22.08 bitrate=1413.5kbits/ frame= 674 fps= 29 q=29.0 size= 3978kB time=00:00:22.53 bitrate=1446.3kbits/ frame= 690 fps= 29 q=29.0 size= 4133kB time=00:00:23.05 bitrate=1468.4kbits/ frame= 706 fps= 29 q=29.0 size= 4263kB time=00:00:23.58 bitrate=1480.4kbits/ frame= 721 fps= 29 q=29.0 size= 4391kB time=00:00:24.08 bitrate=1493.8kbits/ frame= 735 fps= 29 q=29.0 size= 4524kB time=00:00:24.55 bitrate=1509.4kbits/ frame= 748 fps= 29 q=29.0 size= 4661kB time=00:00:24.98 bitrate=1528.2kbits/ frame= 763 fps= 29 q=29.0 size= 4835kB time=00:00:25.50 bitrate=1553.1kbits/ frame= 778 fps= 29 q=29.0 size= 4993kB time=00:00:25.99 bitrate=1573.6kbits/ frame= 795 fps= 29 q=29.0 size= 5149kB time=00:00:26.56 bitrate=1588.1kbits/ frame= 814 fps= 29 q=29.0 size= 5258kB time=00:00:27.18 bitrate=1584.4kbits/ frame= 833 fps= 29 q=29.0 size= 5368kB time=00:00:27.82 bitrate=1580.2kbits/ frame= 851 fps= 29 q=29.0 size= 5469kB time=00:00:28.43 bitrate=1575.9kbits/ frame= 870 fps= 29 q=29.0 size= 5567kB time=00:00:29.05 bitrate=1569.5kbits/ frame= 889 fps= 29 q=29.0 size= 5688kB time=00:00:29.70 bitrate=1568.4kbits/ Starting second pass: moving header on top of the file frame= 902 fps= 28 q=-1.0 Lsize= 6109kB time=00:00:30.14 bitrate=1659.8kbits /s dup=1 drop=0 video:5602kB audio:472kB subtitle:0 global headers:0kB muxing overhead 0.566600% [libx264 @ 0000000004360620] frame I:8 Avg QP:20.52 size: 39667 [libx264 @ 0000000004360620] frame P:419 Avg QP:25.06 size: 10524 [libx264 @ 0000000004360620] frame B:475 Avg QP:29.03 size: 2123 [libx264 @ 0000000004360620] consecutive B-frames: 3.2% 79.6% 0.3% 16.9% [libx264 @ 0000000004360620] mb I I16..4: 20.7% 52.3% 26.9% [libx264 @ 0000000004360620] mb P I16..4: 0.7% 4.2% 1.1% P16..4: 39.4% 21.4 % 13.8% 0.0% 0.0% skip:19.3% [libx264 @ 0000000004360620] mb B I16..4: 0.1% 0.9% 0.3% B16..8: 41.8% 6.4 % 1.7% direct: 1.7% skip:47.1% L0:36.4% L1:53.3% BI:10.3% [libx264 @ 0000000004360620] 8x8 transform intra:65.7% inter:58.8% [libx264 @ 0000000004360620] coded y,uvDC,uvAC intra: 71.2% 76.6% 35.7% inter: 2 0.7% 13.0% 0.5% [libx264 @ 0000000004360620] i16 v,h,dc,p: 48% 24% 8% 20% [libx264 @ 0000000004360620] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 18% 15% 6% 8% 11% 8% 10% 8% [libx264 @ 0000000004360620] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 16% 15% 7% 10% 11% 8% 8% 7% [libx264 @ 0000000004360620] i8c dc,h,v,p: 51% 22% 19% 9% [libx264 @ 0000000004360620] Weighted P-Frames: Y:0.7% UV:0.0% [libx264 @ 0000000004360620] ref P L0: 63.4% 19.7% 11.0% 5.9% 0.0% [libx264 @ 0000000004360620] ref B L0: 90.7% 8.7% 0.7% [libx264 @ 0000000004360620] ref B L1: 98.4% 1.6% [libx264 @ 0000000004360620] kb/s:1524.54

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  • Guaranteed semaphore order?

    - by Steve_
    The documentation for the .NET Semaphore class states that: There is no guaranteed order, such as FIFO or LIFO, in which blocked threads enter the semaphore. In this case, if I want a guaranteed order (either FIFO or LIFO), what are my options? Is this something that just isn't easily possible? Would I have to write my own Semaphore? I assume that would be pretty advanced? Thanks, Steve

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  • PHP Equivalent of Java Type-Casting Solution

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    Since PHP has no custom-class type-casting, how would I go about doing the PHP equivalent of this Java code: CustomBaseObject cusBaseObject = cusBaseObjectDao.readCustomBaseObjectById(id); ((CustomChildObject) cusBaseObject).setChildAttribute1(value1); ((CustomChildObject) cusBaseObject).setChildAttribute2(value2); In my case, it would very nice if I could do this. However, trying this without type-casting support, it gives me an error that the methods do not exist for the object. Thanks, Steve

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  • XVal validation framework

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    Hi, I started using this validation framework: http://blog.stevensanderson.com/2009/01/10/xval-a-validation-framework-for-aspnet-mvc/ Steve mentions that the DataAnnotationsValidationRunner implementation is slightly naive. Why is this and are there better implementations? I am also looking for a good example that uses this framework to validate database dependent business rules. Any pointers would be very much appreciated. Thanks. Christian

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  • Conceal packet loss in PCM stream

    - by ZeroDefect
    I am looking to use 'Packet Loss Concealment' to conceal lost PCM frames in an audio stream. Unfortunately, I cannot find a library that is accessible without all the licensing restrictions and code bloat (...up for some suggestions though). I have located some GPL code written by Steve Underwood for the Asterisk project which implements PLC. There are several limitations; although, as Steve suggests in his code, his algorithm can be applied to different streams with a bit of work. Currently, the code works with 8kHz 16-bit signed mono streams. Variations of the code can be found through a simple search of Google Code Search. My hope is that I can adapt the code to work with other streams. Initially, the goal is to adjust the algorithm for 8+ kHz, 16-bit signed, multichannel audio (all in a C++ environment). Eventually, I'm looking to make the code available under the GPL license in hopes that it could be of benefit to others... Attached is the code below with my efforts. The code includes a main function that will "drop" a number of frames with a given probability. Unfortunately, the code does not quite work as expected. I'm receiving EXC_BAD_ACCESS when running in gdb, but I don't get a trace from gdb when using 'bt' command. Clearly, I'm trampimg on memory some where but not sure exactly where. When I comment out the *amdf_pitch* function, the code runs without crashing... int main (int argc, char *argv[]) { std::ifstream fin("C:\\cc32kHz.pcm"); if(!fin.is_open()) { std::cout << "Failed to open input file" << std::endl; return 1; } std::ofstream fout_repaired("C:\\cc32kHz_repaired.pcm"); if(!fout_repaired.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } std::ofstream fout_lossy("C:\\cc32kHz_lossy.pcm"); if(!fout_lossy.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } audio::PcmConcealer Concealer; Concealer.Init(1, 16, 32000); //Generate random numbers; srand( time(NULL) ); int value = 0; int probability = 5; while(!fin.eof()) { char arr[2]; fin.read(arr, 2); //Generate's random number; value = rand() % 100 + 1; if(value <= probability) { char blank[2] = {0x00, 0x00}; fout_lossy.write(blank, 2); //Fill in data; Concealer.Fill((int16_t *)blank, 1); fout_repaired.write(blank, 2); } else { //Write data to file; fout_repaired.write(arr, 2); fout_lossy.write(arr, 2); Concealer.Receive((int16_t *)arr, 1); } } fin.close(); fout_repaired.close(); fout_lossy.close(); return 0; } PcmConcealer.hpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #ifndef __PCMCONCEALER_HPP__ #define __PCMCONCEALER_HPP__ /** 1. What does it do? The packet loss concealment module provides a suitable synthetic fill-in signal, to minimise the audible effect of lost packets in VoIP applications. It is not tied to any particular codec, and could be used with almost any codec which does not specify its own procedure for packet loss concealment. Where a codec specific concealment procedure exists, the algorithm is usually built around knowledge of the characteristics of the particular codec. It will, therefore, generally give better results for that particular codec than this generic concealer will. 2. How does it work? While good packets are being received, the plc_rx() routine keeps a record of the trailing section of the known speech signal. If a packet is missed, plc_fillin() is called to produce a synthetic replacement for the real speech signal. The average mean difference function (AMDF) is applied to the last known good signal, to determine its effective pitch. Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech will be repeated over and over until the real speech resumes. However, several refinements are needed to obtain smooth pleasant sounding results. - The two ends of the stored cycle of speech will not always fit together smoothly. This can cause roughness, or even clicks, at the joins between cycles. To soften this, the 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e. in total, the last 5/4 pitch periods of real speech are used). - The start of the synthetic speech will not always fit together smoothly with the tail of real speech passed on before the erasure was identified. Ideally, we would like to modify the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However, it is too late for that. We could have delayed the real speech a little, but that would require more buffer manipulation, and hurt the efficiency of the no-lost-packets case (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed, and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result seems quite acceptable. - As we progress into the erasure, the chances of the synthetic signal being anything like correct steadily fall. Therefore, the volume of the synthesized signal is made to decay linearly, such that after 50ms of missing audio it is reduced to silence. - When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the start of the real speech. If the erasure is small, this smoothes the transition. If the erasure is long, and the synthetic signal has faded to zero, the blending softens the start up of the real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. 3. How do I use it? Before audio is processed, call plc_init() to create an instance of the packet loss concealer. For each received audio packet that is acceptable (i.e. not including those being dropped for being too late) call plc_rx() to record the content of the packet. Note this may modify the packet a little after a period of packet loss, to blend real synthetic data smoothly. When a real packet is not available in time, call plc_fillin() to create a sythetic substitute. That's it! */ /*! Minimum allowed pitch (66 Hz) */ #define PLC_PITCH_MIN(SAMPLE_RATE) ((double)(SAMPLE_RATE) / 66.6) /*! Maximum allowed pitch (200 Hz) */ #define PLC_PITCH_MAX(SAMPLE_RATE) ((SAMPLE_RATE) / 200) /*! Maximum pitch OLA window */ //#define PLC_PITCH_OVERLAP_MAX(SAMPLE_RATE) ((PLC_PITCH_MIN(SAMPLE_RATE)) >> 2) /*! The length over which the AMDF function looks for similarity (20 ms) */ #define CORRELATION_SPAN(SAMPLE_RATE) ((20 * (SAMPLE_RATE)) / 1000) /*! History buffer length. The buffer must also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment. */ //#define PLC_HISTORY_LEN(SAMPLE_RATE) ((CORRELATION_SPAN(SAMPLE_RATE)) + (PLC_PITCH_MIN(SAMPLE_RATE))) namespace audio { typedef struct { /*! Consecutive erased samples */ int missing_samples; /*! Current offset into pitch period */ int pitch_offset; /*! Pitch estimate */ int pitch; /*! Buffer for a cycle of speech */ float *pitchbuf;//[PLC_PITCH_MIN]; /*! History buffer */ short *history;//[PLC_HISTORY_LEN]; /*! Current pointer into the history buffer */ int buf_ptr; } plc_state_t; class PcmConcealer { public: PcmConcealer(); ~PcmConcealer(); void Init(int channels, int bit_depth, int sample_rate); //Process a block of received audio samples. int Receive(short amp[], int frames); //Fill-in a block of missing audio samples. int Fill(short amp[], int frames); void Destroy(); private: int amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames); void save_history(plc_state_t *s, short *buf, int channel_index, int frames); void normalise_history(plc_state_t *s); /** Holds the states of each of the channels **/ std::vector< plc_state_t * > ChannelStates; int plc_pitch_min; int plc_pitch_max; int plc_pitch_overlap_max; int correlation_span; int plc_history_len; int channel_count; int sample_rate; bool Initialized; }; } #endif PcmConcealer.cpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #include "audio/PcmConcealer.hpp" /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */ #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ #if !defined(INT16_MAX) #define INT16_MAX (32767) #define INT16_MIN (-32767-1) #endif #ifdef WIN32 inline double rint(double x) { return floor(x + 0.5); } #endif inline short fsaturate(double damp) { if (damp > 32767.0) return INT16_MAX; if (damp < -32768.0) return INT16_MIN; return (short)rint(damp); } namespace audio { PcmConcealer::PcmConcealer() : Initialized(false) { } PcmConcealer::~PcmConcealer() { Destroy(); } void PcmConcealer::Init(int channels, int bit_depth, int sample_rate) { if(Initialized) return; if(channels <= 0 || bit_depth != 16) return; Initialized = true; channel_count = channels; this->sample_rate = sample_rate; ////////////// double min = PLC_PITCH_MIN(sample_rate); int imin = (int)min; double max = PLC_PITCH_MAX(sample_rate); int imax = (int)max; plc_pitch_min = imin; plc_pitch_max = imax; plc_pitch_overlap_max = (plc_pitch_min >> 2); correlation_span = CORRELATION_SPAN(sample_rate); plc_history_len = correlation_span + plc_pitch_min; ////////////// for(int i = 0; i < channel_count; i ++) { plc_state_t *t = new plc_state_t; memset(t, 0, sizeof(plc_state_t)); t->pitchbuf = new float[plc_pitch_min]; t->history = new short[plc_history_len]; ChannelStates.push_back(t); } } void PcmConcealer::Destroy() { if(!Initialized) return; while(ChannelStates.size()) { plc_state_t *s = ChannelStates.at(0); if(s) { if(s->history) delete s->history; if(s->pitchbuf) delete s->pitchbuf; memset(s, 0, sizeof(plc_state_t)); delete s; } ChannelStates.erase(ChannelStates.begin()); } ChannelStates.clear(); Initialized = false; } //Process a block of received audio samples. int PcmConcealer::Receive(short amp[], int frames) { if(!Initialized) return 0; int j = 0; for(int k = 0; k < ChannelStates.size(); k++) { int i; int overlap_len; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples) { /* Although we have a real signal, we need to smooth it to fit well with the synthetic signal we used for the previous block */ /* The start of the real data is overlapped with the next 1/4 cycle of the synthetic data. */ pitch_overlap = s->pitch >> 2; if (pitch_overlap > frames) pitch_overlap = frames; gain = 1.0 - s->missing_samples * ATTENUATION_INCREMENT; if (gain < 0.0) gain = 0.0; new_step = 1.0/pitch_overlap; old_step = new_step*gain; new_weight = new_step; old_weight = (1.0 - new_step)*gain; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[index]); if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->missing_samples = 0; } save_history(s, amp, j, frames); j++; } return frames; } //Fill-in a block of missing audio samples. int PcmConcealer::Fill(short amp[], int frames) { if(!Initialized) return 0; int j =0; for(int k = 0; k < ChannelStates.size(); k++) { short *tmp = new short[plc_pitch_overlap_max]; int i; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; short *orig_amp; int orig_len; orig_amp = amp; orig_len = frames; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples == 0) { // As the gap in real speech starts we need to assess the last known pitch, //and prepare the synthetic data we will use for fill-in normalise_history(s); s->pitch = amdf_pitch(plc_pitch_min, plc_pitch_max, s->history + plc_history_len - correlation_span - plc_pitch_min, j, correlation_span); // We overlap a 1/4 wavelength pitch_overlap = s->pitch >> 2; // Cook up a single cycle of pitch, using a single of the real signal with 1/4 //cycle OLA'ed to make the ends join up nicely // The first 3/4 of the cycle is a simple copy for (i = 0; i < s->pitch - pitch_overlap; i++) s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]; // The last 1/4 of the cycle is overlapped with the end of the previous cycle new_step = 1.0/pitch_overlap; new_weight = new_step; for ( ; i < s->pitch; i++) { s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]*(1.0 - new_weight) + s->history[plc_history_len - 2*s->pitch + i]*new_weight; new_weight += new_step; } // We should now be ready to fill in the gap with repeated, decaying cycles // of what is in pitchbuf // We need to OLA the first 1/4 wavelength of the synthetic data, to smooth // it into the previous real data. To avoid the need to introduce a delay // in the stream, reverse the last 1/4 wavelength, and OLA with that. gain = 1.0; new_step = 1.0/pitch_overlap; old_step = new_step; new_weight = new_step; old_weight = 1.0 - new_step; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->history[plc_history_len - 1 - i] + new_weight * s->pitchbuf[i]); new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->pitch_offset = i; } else { gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; i = 0; } for ( ; gain > 0.0 && i < frames; i++) { int index = (i * channel_count) + j; amp[index] = s->pitchbuf[s->pitch_offset]*gain; gain -= ATTENUATION_INCREMENT; if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; } for ( ; i < frames; i++) { int index = (i * channel_count) + j; amp[i] = 0; } s->missing_samples += orig_len; save_history(s, amp, j, frames); delete [] tmp; j++; } return frames; } void PcmConcealer::save_history(plc_state_t *s, short *buf, int channel_index, int frames) { if (frames >= plc_history_len) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ //memcpy(s->history, buf + len - plc_history_len, sizeof(short)*plc_history_len); int frames_to_copy = plc_history_len; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + frames - plc_history_len)) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = 0; return; } if (s->buf_ptr + frames > plc_history_len) { /* Wraps around - must break into two sections */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*(plc_history_len - s->buf_ptr)); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = plc_history_len - s->buf_ptr; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } frames -= (plc_history_len - s->buf_ptr); //memcpy(s->history, buf + (plc_history_len - s->buf_ptr), sizeof(short)*len); frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + (plc_history_len - s->buf_ptr))) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = frames; return; } /* Can use just one section */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*len); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } s->buf_ptr += frames; } void PcmConcealer::normalise_history(plc_state_t *s) { short *tmp = new short[plc_history_len]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(short)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(short)*(plc_history_len - s->buf_ptr)); memcpy(s->history + plc_history_len - s->buf_ptr, tmp, sizeof(short)*s->buf_ptr); s->buf_ptr = 0; delete [] tmp; } int PcmConcealer::amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames) { int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < frames; j++) { int index1 = (channel_count * (i+j)) + channel_index; int index2 = (channel_count * j) + channel_index; //std::cout << "Index 1: " << index1 << ", Index 2: " << index2 << std::endl; acc += abs(amp[index1] - amp[index2]); } if (acc < min_acc) { min_acc = acc; pitch = i; } } std::cout << "Pitch: " << pitch << std::endl; return pitch; } } P.S. - I must confess that digital audio is not my forte...

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  • SSL pages under ASP.NET MVC

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    How do I go about using HTTPS for some of the pages in my ASP.NET MVC based site? Steve Sanderson has a pretty good tutorial on how to do this in a DRY way on Preview 4 at: http://blog.codeville.net/2008/08/05/adding-httpsssl-support-to-aspnet-mvc-routing/ Is there a better / updated way with Preview 5?,

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    G'day, I've read both of Steve McConnell's excellent Code Complete books "Code Complete" and "Code Complete 2" and was wondering if people have any other suggestions for commenting code.co My commenting mantra could be summed up by the basic idea of expressing "what the code below cannot say". While enjoying this interesting blog post by Jeff about commenting I was still left wondering "When coding, when do you feel a comment is required?"

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    If I have a plist which is structured like this: Root Array Item 0 Dictionary City String New York People Array Item 0 String Steve Item 1 String Paul Item 2 String Fabio Item 3 String David Item 4 String Penny Item 1 Dictionary City String London People Array Item 0 String Linda Item 1 String Rachel Item 2 String Jessica Item 3 String Lou Item 2 Dictionary City String Barcelona People Array Item 0 String Edward Item 1 String Juan Item 2 String Maria Then what is the most efficient way of getting all the names of the people into one big NSArray?

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    Data.XX.NewValue := Data.XX.SavedValue; Data.XX.OldValue := Data.XX.SavedValue; I need to do the above a large number of times, where XX represents the value in the class. Pretending there were 3 items in the list: Tim, Bob, Steve. Is there any way to do the above for all three people without typing out the above code three times? (Data is a class containing a number of Objects, each type TList, which contain OldValue, NewValue and SavedValue)

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    I am using swfobject to display some Flash content (please don't tell Steve Jobs) and I need to set the background color of the swfobject content to black so it blends nicely with the black page it is on. I tried using the following code but it didn't change anything: <object id="flash" classid="clsid:D27CDB6E-AE6D-11cf-96B8-444553540000" width="900" height="600" bgcolor="#000"> What is the correct way to set the background color to #000000? Thanks.

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  • Separate groups of people based on members

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  • Handing over FaceBook SessionKey from iPhone to PHP

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