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• Transcode MP3 stream (shoutcast/icecast) into HE-AAC stream (for shoutcast/icecast) under linux

  - by Cy

  How can I transcode MP3 stream (shoutcast/icecast) into HE-AAC stream (for shoutcast/icecast) under linux. Note: I've tried with StreamTranscoderV3 unsuccessfully because it's highly broken when it comes to transcoding to AAC+

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• Can mediatomb VLC profile transcode audio as MP3 rather than mpga ?

  - by djangofan

  In the /etc/mediatomb/config.xml, can mediatomb VLC profile transcode audo as MP3 rather than mpga ? My Sony GoogleTV wont render streamed .avi files files with a mpga audio in them. The original files are Divx encoded with 128kbs MP3 audio but mediatomb is transcoding them. How can I change this? Any ideas? Can I turn off the audio and video transcoding somehow? I need some ideas to try. <profile name="vlcprof" enabled="yes" type="external"> <mimetype>video/mpeg</mimetype> <agent command="vlc" arguments="-I dummy %in --sout #transcode{venc=ffmpeg,vcodec=mp2v,vb=4096,fps=25,aenc=ffmpeg,acodec=mpga,ab=192,samplerate=44100,channels=2}:standard{access=file,mux=ps,dst=%out} vlc:quit"/> <buffer size="10485760" chunk-size="131072" fill-size="2621440"/> <accept-url>yes</accept-url> <first-resource>yes</first-resource> </profile> I know that that MP3 encoding support is external to FFmpeg and must be configured appropriately, but I have no idea how to handle that. I would guess I can work around that by somehow telling ffmpeg to not transcode the audio stream? Also, should I create a separate vlcprof entry for video/avi ? Can you create more than one profile for VLC in the config.xml for mediatomb?

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• What extra packages are needed by Amarok to transcode to MP3?

  - by Jon Pawley

  I'm using Amarok 2.6.0, on KDE version 4.9.3, on Kubuntu 12.04. I would like to be able to copy my music onto my MP3 player (in this case, my iPhone 3), but to transcode the tracks as I copy them over. However, when I right-click on the selected track, and choose "Copy to Collection" and select my iPhone, the option to transcode to MP3 is greyed out. What additional packages does Amarok need in order to enable the transcode to MP3 option? Thanks, Jon Oh, the "Amarok DIagnostics" output, from the Help menu gives: Amarok Diagnostics Amarok Version: 2.6.0 KDE Version: 4.9.3 Qt Version: 4.8.2 Phonon Version: 4.6.0 Phonon Backend: GStreamer (4.6.2) PulseAudio: Yes Amarok Scripts: Amarok Script Console 1.0 (stopped) Discogs 1.1b (stopped) Lyricwiki .2 (stopped) Free Music Charts 1.6.0 (stopped) Librivox.org 1.0 (stopped) Cool Streams 1.0 (stopped) BBC 1.1 (stopped) Amarok Plugins: AudioCd Collection (enabled) DAAP Collection (enabled) MTP Collection (enabled) MySQLServer Collection (enabled) MySQLe Collection (enabled) UPnP Collection (enabled) Universal Mass Storage Collection (enabled) iPod, iPad & iPhone Collection (enabled) Ampache (disabled) Jamendo (disabled) Last.fm (enabled) MP3 Music Store (disabled) MP3tunes (disabled) Magnatune Store (disabled) Podcast Directory (enabled) gpodder.net (enabled) Local Files & USB Mass Storage Backend (enabled) NFS Share Backend (enabled) SMB (Windows) Share Backend (enabled)

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• How to adjust and combine multiple lower quality photos into one better using FOSS?

  - by Vi

  I have multiple noisy photos (caputed without tripod) that needs to be adjusted (moved/rotated) and averaged. How it's better to do it in Linux with FOSS console-based programs? Current way is something like: mplayer mf://*.JPG -vo yuv4mpeg:file=qqq.yuv transcode -i qqq.yuv -y null -J stabilize=maxshift=500:fieldsize=100:fieldnum=6:stepsize=50:shakiness=10 transcode -i qqq.yuv -J transform=smoothing=100000:sharpen=0:optzoom=0 -y raw -o www.yuv mplayer www.yuv -vo pnm gm convert -average 0*.ppm q.ppm i.e.: Convert photos to video Apply Transcode's "Stabilize" filter Convert the video back to images Average the images. It works, but bad: photos still not perfectly adjusted and the whole sequence is very slow. What is better way of doing it?

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• How to transcode Windows-1251 to UTF-8?

  - by Ole Jak

  How to transcode Windows-1251 to UTF-8? Will such function do it? function win_to_utf($s) { for($i=0, $m=strlen($s); $i<$m; $i++) { $c=ord($s[$i]); if ($c<=127) {$t.=chr($c); continue; } if ($c>=192 && $c<=207) {$t.=chr(208).chr($c-48); continue; } if ($c>=208 && $c<=239) {$t.=chr(208).chr($c-48); continue; } if ($c>=240 && $c<=255) {$t.=chr(209).chr($c-112); continue; } if ($c==184) { $t.=chr(209).chr(209); continue; }; if ($c==168) { $t.=chr(208).chr(129); continue; }; } return $t; }

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• Skipping video and audio with PS3MediaServer

  - by MaxMackie

  I'm using the latest PS3MediaServer build right from the repos suggested in the Ubuntu Wiki. I'm streaming multiple movies from my server (Ubuntu 10.04 LTS) to my PS3 over wireless. Sometimes, during some movies, the audio and the video will begin skipping. This can last anywhere between 5 and 30 seconds before it goes back to normal. I have a four core i5 processor and 8GB of DDR3 RAM so I don't think my computer is having a hard time keeping up with the transcoding. So this leads me to believe it's either sub-optimal transcoding options from within PS3MS or my network can't handle the heat. Other than the out-of-box configuration, is there any way I can tweak the settings for the application to use my resources more efficiently?

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• Programmatically convert a video to FLV

  - by Raibaz

  i am currently working on a web application that needs to accept video uploaded by users in any format (.avi, .mov, etc.) and convert them to flv for playing in a flash-based player. Since the site is OpenCms-based, the best solution would be a ready-made plugin for OpenCms that allowed to upload and play videos doing the transcode operation in background, but just a set of Java classes to do the transcode would be great and then i could make the uploading form and playback part on my own.

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• Programmatically convert a video to FLV

  - by Raibaz

  Hi all, i am currently working on a web application that needs to accept video uploaded by users in any format (.avi, .mov, etc.) and convert them to flv for playing in a flash-based player. Since the site is OpenCms-based, the best solution would be a ready-made plugin for OpenCms that allowed to upload and play videos doing the transcode operation in background, but just a set of Java classes to do the transcode would be great and then i could make the uploading form and playback part on my own.

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• Convert video files for Maemo (Nokia N900) using ffmpeg/mencoder

  - by Vikrant Chaudhary

  I'm a newbie in video encoding so I'm looking for some expert advice. I'm looking to transcode media files with ffmpeg or mencoder (or something other) on Ubuntu for my Nokia N900 running Maemo. I'd prefer mencoder, because of ffmpeg's crazy dependencies. Video output should be AVC/H.264 (probably hardware accelerated on device). Audio output in AAC (should have preferred Vorbis but not supported natively and requires .mkv which is also not completely supported). Output video should retain the original aspect ratio. Resolution of screen is 800x480 (16:10). (Explanation of why-this-value-is-chosen would be really appreciated). Thanks.

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• Concatenating ogg video files from the command line

  - by Noufal Ibrahim

  Okay. I've got a few ogg files I've created using a desktop recording tool. I've transcoded them using ffmpeg once (mainly to clip out the beginnings and the ends). Now, I have 3 such files which I want to concatenate into a single .ogv file. I tried using oggCat, it crashed with some kind of error (I tried concatenating a file to itself using oggCat and that failed too leading me to believe that my distro is shipping a broken version of the package). Simply cating the files works but I can't seek which is not cool. mencoder run like this mencoder -ovc lavc -oac lavc file1.ogv file2.ogv file3.ogv -o complete.ogv. It transcodes the files into an avi and clips off a little of the 3 videos. So, how do I do this? Update 1: My current workaround is to transcode the 3 files into .mpg using ffmpeg, then cating them together and then transcoding them back into ogv. Update 2: PiTiVi works for this kind of thing but I need something from the command line that I can automate and script.

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• Transcoding audio and video

  - by Lance Fisher

  What is the best way to transcode audio and video to show on the web? I need to do it programmatically. I'd like to do something like YouTube or Google Video where users can upload whatever format they want, and I encode it to flv, mp3, and/or mp4. I could do it on our server, but I would rather use an EC2 instance or even a web service. We have a Windows 2008 server.

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• Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

  - by Ryan Thompson

  To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

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• How to extract a Vorbis stream from a WAVE file?

  - by H.B.

  I would like to move the Vorbis stream into an ogg container but ffmpeg does not seem to recognize the stream. Even though MPlayer gives this output upon playback: Opening audio decoder: [acm] Win32/ACM decoders Loading codec DLL: 'vorbis.acm' Loaded DLL driver vorbis.acm at 10000000 Warning! ACM codec reports srcsize=0 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000-176400) Selected audio codec: [vorbisacm] afm: acm (OggVorbis ACM) ffmpeg: ffmpeg -i Source.wav -acodec copy Target.ogg Input #0, wav, from 'Source.wav': Duration: 00:02:15.17, bitrate: 128 kb/s Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s [ogg @ 00000000003096C0] Unsupported codec id in stream 0 Output #0, ogg, to 'Target.ogg': Metadata: encoder : Lavf53.6.0 Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?) Of course this does not necessarily need to be done via ffmpeg, any method that is workable would be fine... I have cut down one of the files to 512KB: sample.wav (Changed two chunk size fields in the wave header to account for this, the embedded stream is cut "without notice")

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• Is it possible to transcode audio in C# using DirectSound?

  - by Robert Davis

  I want to transcode a lot of audio from its source format to PCM without resampling or messing with the sample size. I figure if Windows Media Player can play the file and it doesn't use a legacy ACM codecs it must be using DirectSound to do so (this is on Windows XP and Windows Server 2k3). So is it possible to access DirectSound from C# and do so? I've tried searching the web but all the examples have been about playback which I have no interest in doing.

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• How to generate a 8 bit per sample wav audio file in VLC

  - by Ahmed safan

  I'm using the following vlc command line to extract first 5 minutes of audio from video file "-I dummy -vvv --no-sout-video --sout-audio --no-sout-rtp-sap --no-sout-standard-sap --ttl=1 --sout-transcode-threads=5 --sout-transcode-high-priority --sout-keep --sout #transcode{acodec=s16l,channels=1,samplerate=8000,ab=64}:std{mux=wav,access=file,dst="c:\dest.wav"} "c:\originalvideo.mpg" --start-time=0 --stop-time=300 vlc://quit"; if ab=64 =64 k bits per second and samples per second=8 k samples then bits per sample=64/8=8 bits per sample but the problem is that the output file always has samples of 16 bits per sample. I know that sample can contain bits from 8 , 16, 24 to 32 bits per sample. i want to get 8 bits per sample file how can this be done ?

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• function not working in production mode

  - by maps

  I am using the rvideo gem to transcode files to a .flv format. class Video < ActiveRecord::Base include AASM aasm_column :status aasm_initial_state :initial aasm_state :initial aasm_state :converting, :exit => :transcode aasm_state :transfering , :exit => :send_s3 aasm_state :completed aasm_state :failed aasm_event :convert do transitions :from => [:initial], :to => :converting end aasm_event :transfer do transitions :from => [:converting], :to => :transfering end aasm_event :complete do transitions :from => [:transfering], :to => :completed end aasm_event :error do transitions :from => [:initial, :converting, :transfering, :completed] end has_attached_file :asset, :path => "uploads/:attachment/:id.:basename.:extension" def flash_path return self.asset.path + '.flv' end def flash_name return File::basename(self.asset.path)# + '.flv' end def flash_url return "#{AWS_HOST}/#{AWS_BUCKET}/#{self.flash_name}" end # transcode file def transcode begin RVideo::Transcoder.logger = logger file = RVideo::Inspector.new(:file => self.asset.path) command = "ffmpeg -i $input_file$ -y -s $resolution$ -ar 44100 -b 64k -r 15 -sameq $output_file$" options = { :input_file => "#{RAILS_ROOT}/#{self.asset.path}", :output_file => "#{RAILS_ROOT}/#{self.flash_path}", :resolution => "320x200" } transcoder = RVideo::Transcoder.new transcoder.execute(command, options) rescue RVideo::TranscoderError => e logger.error "Encountered error transcoding #{self.asset.path}" logger.error e.message end end The input file is added to the asset directory, but I never get an outputted file. On the view page aasm hangs on "converting".

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• Live Screencast under Linux

  - by OmnipotentEntity

  I was having some difficulty with running a Live Screencast under Linux. I've found jtvlc and tried using that, but whenever I use it the stream comes out either blank or lagged with extremely high latency. I have a fast internet connection and a fast computer, but am I perhaps taxing it too much? Any ideas on what I could possibly be doing wrong? # 1. Get an account on http://www.justin.tv/ # 2. Copy streaming key from: http://www.justin.tv/broadcast/adv_other # 2. Install VLC: http://www.videolan.org/vlc/ # 3. Get Win/Mac/Lin Stream Client: \ # http://apiwiki.justin.tv/mediawiki/index.php/Linux_Broadcasting_API # 4. Adjust the vlc parameters to your liking and run VLC like this #!/bin/bash cvlc screen:// --input-slave=pulse:// \ --screen-width 1920 \ --screen-height 1080 \ --screen-fps 5 \ -v input_stream \ --sout='#duplicate{ dst="transcode{ scale=1, venc=x264{ keyint=60 }, vcodec=h264, vb=600, acodec=mp4a, ab=32, channels=2, samplerate=22050 } :rtp{dst=127.0.0.1,port=1234,sdp=file:///tmp/vlc.sdp} "}' \ --sout-transcode-threads=4 & sleep 2 # 5. Run JTVLC to stream like this: ./jtvlc/jtvlc omnipotententity censored /tmp/vlc.sdp # Notes: #- If you want to see what you're about to stream add 'dst=display, ' # before 'dst="transcode[' # More about the VLC parameters: http://wiki.videolan.org/Documentation:Modules/screen

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• ffmpeg add two audio streams to video

  - by Tossin Hausen

  I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

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• QuickTime X incorrect aspect ratio for H.264 video

  - by Adam Robinson

  I'm running Snow Leopard and have a serious issue with QuickTime X. I have a Samsung HMX-H100N/XAA camcorder that records H.264 video in either 720p or 1080i. In either of these resolutions, QuickTime X (and, by extension, all QuickTime-associated applications like FCP, iMovie, etc.) displays an incorrect aspect ratio for all video produced by this camcorder. For example, 720p video is reported as being 1280x720 in the movie inspector (which is normal), but the displayed size is always at an aspect ratio of something like 63:20 (never heard of such a ratio) with sizes like 1700x539. If I open the video in QuickTime 7 player on the same computer, it is displayed correctly. If I process the video through something like MPEG Streamclip to transcode it, it displays correctly. As it stands right now I have to transcode all of my video in order to use it in any iLive (or other QT-based application) unless I want it to look ridiculous. I've tried installing Perian, but that seemed to have no effect.

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• Media Archive System with branches?

  - by Ian McEwen

  In short, how can I get VCS features (revisioning, branching, and deduplication) for a media collection that's far too large for most/all VCS systems? Background I have a 300GB music folder; unfortunately, I only have the hard drive space for this on my desktop system. However, a good portion of my collection is FLAC; therefore, I could theoretically have a space-optimized version in which I transcode all the FLAC to mp3 or some other lossy format, and use only that version on the laptop. However, a portion of my collection isn't FLAC. And that which isn't FLAC shouldn't be transcoded to an equivalent format; it won't have any space savings, which is the point. Moreover, it shouldn't be duplicated: the mp3/ogg portions of the collection should probably be exactly the same files. Thoughts One solution is to have format-specific organization of my music folders, and use some script to transcode the FLAC directory to mp3 or such into another directory. Another is some sort of hack using entirely separate copies and symbolic links for deduplication, or something similar. But these also have a disadvantage of lacking versioning; I'd like to be able to reorganize my music collection, retag things, etc. and save history. This isn't key, but would be awfully nice. I can't see it as entirely unreasonable to set up VCS hooks or something equivalent to keep directory structure synced between two copies, update tags, and transcode FLAC automatically into the space-optimized copy. Basically, the system I really want is a version control system. Two branches: one archival/desktop branch including the FLAC, one space-optimized/laptop branch without it; most VCSes would deal well with whole chunks being the same files well by compressing in a reasonable way (i.e. don't keep two copies of the same data). I could also do a lot of what I talk about above with hooks. But I don't know of any VCS that would deal with a 300GB repository with almost 20k files. Many of them would just not even initialize the whole affair; others would just do it inexpressibly slowly or otherwise badly. checkpoint looks like it's designed for something close (it's at least for media), but wouldn't do deduplication well (and I'm not convinced I'd be able to script it to do things like automatic transcoding and directory-structure syncing). So. Is there anything out there that can do all this, or should I consider it a programming project?

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• Serializing chinese characters with Xerces 2.6

  - by Gianluca

  I have a Xerces (2.6) DOMNode object encoded UTF-8. I use to read its TEXT element like this: CBuffer DomNodeExtended::getText( const DOMNode* node ) const { char* p = XMLString::transcode( node->getNodeValue( ) ); CBuffer xNodeText( p ); delete p; return xNodeText; } Where CBuffer is, well, just a buffer object which is lately persisted as it is in a DB. This works until in the TEXT there are just common ASCII characters. If we have i.e. chinese ones they get lost in the transcode operation. I've googled a lot seeking for a solution. It looks like with Xerces 3, the DOMWriter class should solve the problem. With Xerces 2.6 I'm trying the XMLTranscoder, but no success yet. Could anybody help?

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• Mass audio encoder

  - by bessman

  I have a few thousand FLAC files which I would like to transcode to OGG Vorbis, but I can't find any suitable tools for the job. To name a few I have tried so far and why they are unsuitable: oggenc is single-threaded and would require me to automate it myself, mencoder requires the input to also contain video, and abcde assumes the input is a CD. The ideal tool should be multi-threaded, and support inputing multiple files located in different directories simultaneously. CLI or GUI makes no matter. Does such a tool exist?

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• Convert mp4 video to a format xbox 360 can play

  - by Björn Lindqvist

  Here is a video file my Xbox 360 refuses to play: $ MP4Box -info video.mp4 * Movie Info * Timescale 90000 - Duration 02:18:33.365 Fragmented File no - 2 track(s) File Brand mp42 - version 0 Created: GMT Sat Jul 21 07:08:55 2012 File has root IOD (9 bytes) Scene PL 0xff - Graphics PL 0xff - OD PL 0xff Visual PL: ISO Reserved Profile (0x7f) Audio PL: High Quality Audio Profile @ Level 2 (0x0f) No streams included in root OD iTunes Info: Encoder Software: HandBrake 0.9.6 2012022800 Track # 1 Info - TrackID 1 - TimeScale 90000 - Duration 02:18:33.235 Media Info: Language "Undetermined" - Type "vide:avc1" - 199318 samples Visual Track layout: x=0 y=0 width=1280 height=688 MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 1280 x 688 AVC Info: 1 SPS - 1 PPS - Profile High @ Level 4.1 NAL Unit length bits: 32 Self-synchronized Track # 2 Info - TrackID 2 - TimeScale 48000 - Duration 02:18:33.365 Media Info: Language "English" - Type "soun:mp4a" - 389689 samples MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40 MPEG-4 Audio MPEG-4 Audio AAC LC - 6 Channel(s) - SampleRate 48000 Synchronized on stream 1 $ avconv -i video.mp4 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:33 with gcc 4.6.3 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isomavc1 creation_time : 2012-07-21 07:08:55 encoder : HandBrake 0.9.6 2012022800 Duration: 02:18:33.36, start: 0.000000, bitrate: 2299 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x688, 1973 kb/s, 23.98 fps, 90k tbr, 90k tbn, 180k tbc Metadata: creation_time : 2012-07-21 07:08:55 Stream #0.1(eng): Audio: aac, 48000 Hz, 5.1, s16, 319 kb/s Metadata: creation_time : 2012-07-21 07:08:55 At least one output file must be specified What tool, such as ffmpeg or mencoder, and what magic command line incantation should I use to transcode this file into a format Xbox 360 can play? I want the transcode process to retain as good video quality as possible.

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• VLC desktop streaming

  - by StackedCrooked

  Edit I stopped using VLC and switched to GMax FLV Encoder. It does a much better job IMO. Original post I am sending my desktop (screen) as an H264 video stream to another machine that saves it to a file using the follwoing command lines: Sender of the stream: vlc -I dummy --sout='#transcode{vcodec=h264,vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' Receiver of the stream: vlc -I rc rtp://@:4444 --sout='#std{access=file,mux=ps,dst=/home/user/output.mp4}' --ipv4 This works, but there are a few issues: The file is not playable with most players. VLC is able to playback the file but with some weirdness: = it takes about 10 seconds before the playback actually begins. = seeking doesn't work. Can someone point me in the right direction on how to fix these issues? EDIT: I made a little progress. The initial delay in playback is because the player is waiting for a keyframe. By forcing the sender of the stream to create a new key-frame every 4 seconds I could decrease the delay: :screen-fps=10 --sout='#transcode{vcodec=h264,venc=x264{keyint=40},vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' The seeking problem is not solved however, but I understand it a little better. The RTP stream is saved as a file in its original streaming format, which is normally not playable as a regular video file. VLC manages to play this file, but most other players don't. So I need to convert it to a regular video file. I am currently investigating whether I can do this with ffmpeg if I provide it with an SDP file for the recorded stream. All help is welcome!

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• Xerces C++ SAX Parsing Problem: expected class-name before '{' token

  - by aduric

  I'm trying to run through an example given for the C++ Xerces XML library implementation. I've copied the code exactly, but I'm having trouble compiling it. error: expected class-name before '{' token I've looked around for a solution, and I know that this error can be caused by circular includes or not defining a class before it is used, but as you can see from the code, I only have 2 files: MySAXHandler.hpp and MySAXHandler.cpp. However, the MySAXHandler class is derived from HandlerBase, which is included. MyHandler.hpp #include <xercesc/sax/HandlerBase.hpp> class MySAXHandler : public HandlerBase { public: void startElement(const XMLCh* const, AttributeList&); void fatalError(const SAXParseException&); }; MySAXHandler.cpp #include "MySAXHandler.hpp" #include <iostream> using namespace std; MySAXHandler::MySAXHandler() { } void MySAXHandler::startElement(const XMLCh* const name, AttributeList& attributes) { char* message = XMLString::transcode(name); cout << "I saw element: "<< message << endl; XMLString::release(&message); } void MySAXHandler::fatalError(const SAXParseException& exception) { char* message = XMLString::transcode(exception.getMessage()); cout << "Fatal Error: " << message << " at line: " << exception.getLineNumber() << endl; XMLString::release(&message); } I'm compiling like so: g++ -L/usr/local/lib -lxerces-c -I/usr/local/include -c MySAXHandler.cpp I've looked through the HandlerBase and it is defined, so I don't know why I can't derive a class from it? Do I have to override all the virtual functions in HandlerBase? I'm kinda new to C++. Thanks in advance.

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