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  • Encoding MP3 and adding VBR or Xing headers (with lame or another method)

    - by J. Pablo Fernández
    I'm writing a program that converts wavs to mp3s, so far, by using lame. It's generating a command line more or less like this: "c:\Program Files (x86)\Lame for Audacity\lame.exe" --preset fast medium in.wav out.mp3 The problem I'm having is that no VBR or Xing headers are written to the MP3. How can I make lame.exe write those headers? Should I use another program to write those headers (platform is Windows, .Net 3.5)? Should I use another program for MP3 encoding?

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  • How to get the real, actual duration of an MP3 file (VBR or CBR) server-side

    - by Cummander Checkov
    I used to calculate the duration of MP3 files server-side using ffmpeg - which seemed to work fine. Today i discovered that some of the calculations were wrong. Somehow, for some reason, ffmpeg will miscalculate the duration and it seems to happen with variable bit rate mp3 files only. When testing this locally, i noticed that ffmpeg printed two extra lines in green. Command used: ffmpeg -i song_9747c077aef8.mp3 ffmpeg says: [mp3 @ 0x102052600] max_analyze_duration 5000000 reached at 5015510 [mp3 @ 0x102052600] Estimating duration from bitrate, this may be inaccurate After a nice, warm google session, i found some posts on this, but no solution was found. I then tried to increase the maximum duration: ffmpeg -analyzeduration 999999999 -i song_9747c077aef8.mp3 After this, ffmpeg returned only the second line: [mp3 @ 0x102052600] Estimating duration from bitrate, this may be inaccurate But in either case, the calculated duration was just plain wrong. Comparing it to VLC i noticed that here the duration is correct. After more research i stumbled over mp3info - which i installed and used. mp3info -p "%S" song_9747c077aef8.mp3 mp3info then returned the CORRECT duration, but only as an integer, which i cannot use as i need a more accurate number here. The reason for this was explained in a comment below, by user blahdiblah - mp3info is simply pulling ID3 info from the file and not actually performing any calculations. I also tried using mplayer to retrieve the duration, but just as ffmpeg, mplayer is returning the wrong value. Now i ran out of options. If somebody knows how to get around this, any hints, tips, guides or corrections are welcome! Thank You!

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  • How to generate a VBR video from stream of YUV images?

    - by zitronic
    My hardware (video capture card) gives me the images in YV12 (YUV 420) format and I am trying to generate a video from it. I am using C++ under windows and I would like to generate a mpeg-4 VBR video from that stream but I dont know where I should start... (I need it to be VBR because it is a security camera and there will be a lot of repeated frames) Is there any library that does something like this?

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  • Quality wise, is Windows Media Audio 10 Professional equivalent to WMA?

    - by Louis
    I noticed that for encoding CD rips, Zune is still using WMA 9.2 instead of WMA 10 Pro. On a given file using the highest quality VBR settings looks like this: VBR Quality 98, 44 kHz, stereo 1-pass VBR On the same file if I use WMA 10 Pro, with the same settings, the resulting file is about 20% smaller. Using my ears, I'm unable to tell the difference, but I'm wondering if this was the goal of WMA 10 Pro (to be as good as WMA at a lower bitrate). Is the quality of a WMA 10 Pro file equal to that of a WMA 9.2 file encoded with the same settings?

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  • Sound Juicer doesn't respect Lame's codec settings when ripping CDs

    - by Takkat
    Using Sound Juicer I am able to rip CDs very conveniently. I would like to rip them in about 256 kbit/s variable bitrate. To accomplish this I have defined the settings for mp3 in gnome-audio-profiles-properties as follows: audio/x-raw-int,rate=44100,channels=2 ! lame name=enc mode=0 vbr-quality=0 ! id3v2mux where vbr-quality=0 should give me a variable bitrate averaging 245 kbit/s. The resulting files however always say they are in 128 kbit/s. Is this only a tagging bug or is indeed the bitrate that low? How could I find out?

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  • Why have mp3 files ripped with Lame always have 128 kbit/s irrespect of settings?

    - by Takkat
    Using Sound Juicer I am able to rip Cds very conveniently. I would like to rip them in about 256 kbit/s variable bitrate. To accomplish this I have defined the settings for mp3 in gnome-audio-profiles-properties as follows: audio/x-raw-int,rate=44100,channels=2 ! lame name=enc mode=0 vbr-quality=0 ! id3v2mux where vbr-quality=0 should give me a variable bitrate averaging 245 kbit/s. The resulting files however always say they are in 128 kbit/s. Is this only a tagging bug or is indeed the bitrate that low? How could I find out?

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  • Audio Streaming Latency

    - by killianmcc
    I'm writing a UDP local area network video chat system and have got the video and audio streams working. However I'm experiencing a little latency (about half a second) in the audio and was wondering what codecs would provide the least latency. I'm using NAudio (http://naudio.codeplex.com/) which provides me access to the following codecs for streaming; Speex Narrow Band (VBR) Speex Wide Band (16kHz)(VBR) Speex Ultra Wide Band (32kHz)(VBR) DSP Group TrueSpeech (8.5kbps) GSM 6.10 (13kbps) Microsoft ADPCM (32.8kbps) G.711 a-law (64kbps) G.722 16kHz (64kbps) G.711 mu-law (64kbps) PCM 8kHz 16 bit uncompressed (128kbps) I've tried them out and I'm not noticing much difference. Is there any others that I should download and try to reduce latency? I'm only going to be sending voice over the connection but I'm not really worried about quality or background noises too much. UPDATE I'm sending the audio in blocks like so; waveIn = new WaveIn(); waveIn.BufferMilliseconds = 50; waveIn.DeviceNumber = inputDeviceNumber; waveIn.WaveFormat = codec.RecordFormat; waveIn.DataAvailable += waveIn_DataAvailable; void waveIn_DataAvailable(object sender, WaveInEventArgs e) { if (connected) { byte[] encoded = codec.Encode(e.Buffer, 0, e.BytesRecorded); udpSender.Send(encoded, encoded.Length); } }

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  • Skype est-il assez sécurisé ? Privacy International affirme que non

    Skype est-il assez sécurisé ? Privacy International affirme que non Le puissant groupe Privacy International s'est penché sur les paramètres de sécurisation de Skype, et les juge insuffisants. Le service de VoIP ne protégerait pas assez ses utilisateurs, notamment en affichant leur nom complet dans les listes de contact, avance le groupe. Autre reproche fait au logiciel : il serait facile pour les pirates de lui substituer leur propre version, mais infectée d'un Trojan celle-ci, du fait de l'absence de la protection HTTPS sur la page de téléchargement officielle. Privacy International met également en garde contre le VBR, le codec utilisé pour la compression des flux audios. Ce dernier permettrait à 50%, voire 90% de...

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  • Converting WAV to MP3 on Linux with low bitrates

    - by Olly
    I need to convert WAV files to MP3 files so they can be played on a website. I think that LAME would probably be the best tool. However the WAV files are low bitrate (around 8kbits recorded from a phone) and LAME's website states that it is the "best MP3 encoder at mid-high bitrates and at VBR". Is there is a better encoder for lower bitrates? If so can you define "better"?

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  • ffmpeg XDCAM HD vtag profiles

    - by sebastian
    Hey guys sorry for disturbing you again, but these days I have a lot of work to do with ffmpeg. I found on http://code.google.com/p/ffmbc/wiki/XDCAMHD422Encoding vtag settings for XDCAM HD 422 and used them in my script, everything works fine for HD 422, now I am desperately searching for normal HD vtags and have found only one so far was xd3v for Apple XDCAM HD 1080i50 (35 Mb/s VBR) which I don't need because its interlaced. What I need are the vtags for 1080p24, 1080p25 and 1080p30. Here is the script I have found and adapted a bit: ffmpeg -threads "4" -i "$2" -pix_fmt yuv420p -vtag xdv3 -vcodec mpeg2video -r 25 -flags +ildct+ilme -top 1 -dc 10 -intra_vlc 1 -non_linear_quant 1 -qmin 1 -qmax 3 -lmin '1*QP2LAMBDA' -rc_max_vbv_use 1 -rc_min_vbv_use 1 -b 35000k -minrate 35000k -maxrate 35000k -bufsize 36408333 -bf 2 -aspect 16:9 -acodec pcm_s16be -vf scale=${FC_PARAM_width}:${FC_PARAM_height} "$3" If there are any other mistakes in the script, please correct me :)

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  • Acceptable sound quality: stereo needed for an Android game?

    - by Thomas Calc
    I have various simple short sound effects (damage sound, dying sound, thunderbolt, fanfare, breaking) for a game that is developed for Android currently. I use OGG files: 96kbps VBR, 44.1KHz, 2 channels (that means stereo, right?). I read the other stackexchange topics about "acceptable sound quality", but they're too general, address too many things. My experience is that even with 80kbps, my effects sound OK. But I tested it on a limited number of Android devices (including a Sony Ericsson Xperia Neo and a HTC Desire HD). My questions: For mobile phones and tablets, generally, what parameters are recommended? Won't my 80kbps sounds be bad on a newer device (such as a modern tablet)? I don't hear any difference between stereo and mono (2 channels vs. 1 channel, right?), is there any noticeable difference at all for mobile phones / tablets? (in terms of the player experience) May it worth it at all? I assume that stereo sounds take much more in memory (when they're decoded to PCM), despite of the fact that the compressed OGG size is practically the same. Reacting to Roy T.'s great comment: Actually, I couldn't measure the PCM size (Android decodes OGG internally), but I thought that stereo will take more space than mono when uncompressed After throwing out one of the WAV channels in Audacity, and re-exporting it: The new WAV file size is half than before The OGG file size is practically the same as before The sound effects and game music was recorded by my friend who is an experienced hobby musician/composer, but he knows little about computers & software so he just gave me some high-quality WAV files generated via his hardware.These were stereo, but if I check them in Audacity, both channels appear to be exactly the same.Can I consider them the same (= moving to mono), or might there be some unnoticeable differences to the human eye?

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  • merging video and audio with custom panning

    - by cherouvim
    I have: a video which has mono audio inside a audio (mono) I'd like to merge those two to a single video file containing: video from #1 audio from #1 full left pan + audio #2 full right pan Is this possible in ffmpeg using 1 command? I've tried the following which almost does this but the video/audio gets out of sync: ffmpeg -i video.mp4 -filter_complex "amovie=audio.wav [r] ; [r] amerge" output.mp4 -y I've managed to do it with multiple commands: #1 create right panned audio ffmpeg -i audio.wav -ac 2 -vbr 5 audio-stereo.mp3 -y ffmpeg -i audio-stereo.mp3 -af pan=stereo:c1=c1 audio-right.mp3 -y #2 create left panned video ffmpeg -i video.mp4 -af pan=stereo:c0=c0 video-left.mp4 -y #3 merge the two ffmpeg -i video-left.mp4 -i audio-right.mp3 -filter_complex "amix=inputs=2" video-mixed.mp4 -y It does the job, but is it possible with 1 command?

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  • How to find out when to increase bit rate? (TCP streaming solution)

    - by Kabumbus
    How to find out when to increase bit rate? (TCP streaming solution) We have a stream with "frames". each "frame" has a "timestamp" . frames have bit rate property which is actually there size. We generate frames with our app and stream them one by one on to our TCP server socket. At the same time server post replies so when after each sent frame we try to read from socket we receive which timestamp is currently on server. if timestamp is lover than previous frame we lover bit rate 20%. Such scheme seems to work giving me one way vbr (lowering) but I wonder how to implement increase? I mean we can always try to increase 5% each frame until some limited desired value but each time we have delay will lose real-time feature of our stream... Generally such scheme is for finding out how much of network stream is currently used by other user apps and give picture of how much server is loaded at the same time so we can stream just right amount of data for all to receive it in real time. So what shall I do to add increase to my scheme? So having current bit rate of A I thought we could add +7% for 3 frames and than one -20% and than if all that 3 frames with +7% came in time we could add 14% to A and repeat circle and it would hopefully not be really noticeable if 2nd frame wold come to us with delay... probably this one is too localised because it is a requirement for me to use TCP.

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  • How do I specify the com+ server when registering a vb6 com+ application without using clireg?

    - by user85759
    I've found lots of documentation on how to install com+ components with WiX or an exported msi from dcomcnfg but the problem with these approaches is I can't see where to specify the com+ server. Currently we register the components with clireg and the -s switch which allows us to specify the com+ server like so: clireg32.exe BLEH.VBR -s COMSERVER -t BLEH.TLB -d This is messy to say the least and I've been trying to get this into some automated form of installation that doesn't involve calling a batch file full of clireg32 calls. Currently WiX is the backbone of our packaging automation so a solution with WiX would be awesome. Thanks.

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  • Is there any LAME c++ wraper\simplifier (working on Linux Mac and Win from pure code)?

    - by Ole Jak
    So I want to create simple pcm to mp3 C++ project. I want it to use LAME. I love LAME but It is realy biiig. so I need some kind of OpenSource working from pure code with pure lame code workflow simplifier. So to say I give it File with PCM and DEST file. Call something like LameSimple.ToMP3(file with PCM, File with MP3 , 44100, 16, MP3, VBR); ore such thing in 4 - 5 lines (examples ofcourse should exist) and I have vhat I needed It should be light, simple, powerfool, opensource, crossplatform. Is there any thing like this?!?

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  • Squid external_acl_type Cannot run process

    - by Alex Rezistorman
    I want to restrict uploading for group of the users via squid. So I've choosen to use external_acl_type but after reload of the squid it returns error. WARNING: Cannot run '/usr/local/etc/squid/lists/newupload.sh' process. Permissions of newupload.sh and squid are the same. newupload.sh is executive. How can I solve this problem? Thnx in advance. newupload.sh #!/bin/sh while read line; do set -- $line length=$1 limit=$2 if [ -z "$length" ] || [ "$length" -le "$2" ]; then echo OK else echo ERR fi done Strings from squid.conf external_acl_type request_body protocol=2.5 %{Content-Lenght} /usr/local/etc/squid/lists/newupload.sh acl request_max_size external request_body 5000 http_access allow users request_max_size Squid version squid -v Squid Cache: Version 3.2.13 configure options: '--with-default-user=squid' '--bindir=/usr/local/sbin' '--sbindir=/usr/local/sbin' '--datadir=/usr/local/etc/squid' '--libexecdir=/usr/local/libexec/squid' '--localstatedir=/var' '--sysconfdir=/usr/local/etc/squid' '--with-logdir=/var/log/squid' '--with-pidfile=/var/run/squid/squid.pid' '--with-swapdir=/var/squid/cache/squid' '--enable-auth' '--enable-build-info' '--enable-loadable-modules' '--enable-removal-policies=lru heap' '--disable-epoll' '--disable-linux-netfilter' '--disable-linux-tproxy' '--disable-translation' '--enable-auth-basic=PAM' '--disable-auth-digest' '--enable-external-acl-helpers= kerberos_ldap_group' '--enable-auth-negotiate=kerberos' '--disable-auth-ntlm' '--without-pthreads' '--enable-storeio=diskd ufs' '--enable-disk-io=AIO Blocking DiskDaemon IpcIo Mmapped' '--enable-log-daemon-helpers=file' '--disable-url-rewrite-helpers' '--disable-ipv6' '--disable-snmp' '--disable-htcp' '--disable-forw-via-db' '--disable-cache-digests' '--disable-wccp' '--disable-wccpv2' '--disable-ident-lookups' '--disable-eui' '--disable-ipfw-transparent' '--disable-pf-transparent' '--disable-ipf-transparent' '--disable-follow-x-forwarded-for' '--disable-ecap' '--disable-icap-client' '--disable-esi' '--enable-kqueue' '--with-large-files' '--enable-cachemgr-hostname=proxy.adir.vbr.ua' '--with-filedescriptors=131072' '--disable-auto-locale' '--prefix=/usr/local' '--mandir=/usr/local/man' '--infodir=/usr/local/info/' '--build=amd64-portbld-freebsd8.3' 'build_alias=amd64-portbld-freebsd8.3' 'CC=cc' 'CFLAGS=-O2 -fno-strict-aliasing -frename-registers -fweb -fforce-addr -fmerge-all-constants -maccumulate-outgoing-args -pipe -march=core2 -I/usr/local/include -DLDAP_DEPRECATED' 'LDFLAGS= -L/usr/local/lib' 'CPPFLAGS=-I/usr/local/include' 'CXX=c++' 'CXXFLAGS=-O2 -fno-strict-aliasing -frename-registers -fweb -fforce-addr -fmerge-all-constants -maccumulate-outgoing-args -pipe -march=core2 -I/usr/local/include -DLDAP_DEPRECATED' 'CPP=cpp' --enable-ltdl-convenience Related post: Restrict uploading for groups in squid http://squid-web-proxy-cache.1019090.n4.nabble.com/flexible-managing-of-request-body-max-size-with-squid-2-5-STABLE12-td1022653.html

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  • Remote DLL Registration without access to HKEY_CLASSES_ROOT

    - by mohlsen
    We have a legacy VB6 application that updates itself on startup by pulling down the latest files and registering the COM components. This works for both local (regsvr32) ActiveX COM Components and remote (clireg32) ActiveX COM components registered in COM+ on another machine. New requirements are preventing us from writing to HKEY_LOACL_MACHINE (HKLM) for security reasons, which is what obviously happens by default when calling regsvr32 and clireg32. We have come up with an way to register the local COM componet under HKEY_CURRENT_USER\Software\Classes (HKCU) using the RegOverridePredefKey Windows API method. This works by redirecting the inserts into the registry to the HKCU location. Then when the COM components are instantiated, windows first looks to HKCU before looking for component information in HKLM. This replaces what regsvr32 is doing. The problem we are experiencing at this time is when we attempt to register VBR / TLB using clireg32, this registration process also adds registration keys to HKEY_LOACL_MACHINE. Is there a way to redirect clireg32.exe to register component is HKEY_CURRENT_USER? Are there any other methods that would allow us to register these COM+ components on clients machine with limited security access? Our only solution at this time would be to manually write the registration information to the registry, but that is not ideal and would be a maint issue.

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  • Correlating traditional Windows joystick axes with HID

    - by Wade Williams
    I'm a bit confused on the description of joystick axes and I'm hoping that someone has a link or document which could help clear my confusion. I'm not a Windows guy, so trying to port some traditional Windows gameport code has me a bit confused. We all know about the common first three axes: X Y Z My understanding was that in the gameport-style interface the three other axes are: R U V However, looking in my IOHIDUsageTables (OS X), I see: kHIDUsage_GD_X = 0x30, /* Dynamic Value */ kHIDUsage_GD_Y = 0x31, /* Dynamic Value */ kHIDUsage_GD_Z = 0x32, /* Dynamic Value */ kHIDUsage_GD_Rx = 0x33, /* Dynamic Value */ kHIDUsage_GD_Ry = 0x34, /* Dynamic Value */ kHIDUsage_GD_Rz = 0x35, /* Dynamic Value */ kHIDUsage_GD_Vx = 0x40, /* Dynamic Value */ kHIDUsage_GD_Vy = 0x41, /* Dynamic Value */ kHIDUsage_GD_Vz = 0x42, /* Dynamic Value */ kHIDUsage_GD_Vbrx = 0x43, /* Dynamic Value */ kHIDUsage_GD_Vbry = 0x44, /* Dynamic Value */ kHIDUsage_GD_Vbrz = 0x45, /* Dynamic Value */ kHIDUsage_GD_Vno = 0x46, /* Dynamic Value */ This has me a bit confused due to the three R axis (though that does not appear to be uncommon) and the lack of a U axis. Two questions: 1) Can anyone confirm to what axis the traditional U axis would be? I saw one document describe it as "the axis for rudder pedals" leading me to believe it would be Rz. 2) Can anyone describe in more detail the typical usages of the V and Vbr axes? I understand the descriptions are "vector" and "relative vector,' respectively, but I'm having difficult visualizing what that means in terms of a physical device. All enlightenment and documentation pointers welcome.

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  • Convert mkv/h264 video so it can be played on a "mid-range" Sony Ericsson phone. (using Ubuntu).

    - by Johan
    Hi As a little experiment I thinking of converting some video/movies/tv-series into a format that could be playable on my K850, but to be a little bit more generic in this question let's say "mid range Sony Ericsson" phone since they all more or less behave the same and has the same screen resolution (240 x 320). I am looking for command line based tools (for Ubuntu), since I am thinking about writing a "convert and move" script later if it is successful. A lot of the video I have is encoded in mkv/h264, but since that is not supported by the phone I guess that I need to convert it into some mp4/mpeg4 low quality video. After some googling it seems like a good candidate for the job is ffmpeg, but that seems to be a very versatile tool with a lot of magic tricks. Am I on the right track? And if so how do I use ffmpeg to do this? Thanks Johan Update: After plating a little bit with ffmeg I noticed that it only uses 1 of my 4 cores, so the transcoding takes forever. I found a arg called -threads but that did not change much, maybe I got it wrong. I also found that something like this plays in the phone. ffmpeg -i Mythbusters\ S1D1_1.mkv -threads 4 -t 180 -vcodec mpeg4 -r 15 -s 320x240 Mythbusters\ S1D1_1_mini.mp4 It was possible to use 3gp/h263, but the quality was really useless. ffmpeg -i Mythbusters\ S1D1_1.mkv -t 180 -vcodec h263 -acodec libfaac -s cif Mythbusters\ S1D1_1_cif.3gp And it seems like mp4/h264 is also possible and the result is ok, thanks to this question, this one seem to use more than one core as well so it was a little bit faster for me. ffmpeg -i Mythbusters_S1D1_1.mkv -t 180 -acodec libfaac -ab 60k -s 320x240 -vcodec libx264 -b 500k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -flags2 +mixed_refs -me_method umh -subq 6 -trellis 1 -refs 5 -coder 0 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 500k -maxrate 768k -bufsize 2M -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 13 -threads 0 -f mp4 Mythbusters_S1D1_1_qvga.mp4 Update: I have tried to use HandBrakeCLI and it is no problem creating a new file that seem to be the same as the one created with ffmpeg with something like this. HandBrakeCLI -i Mythbusters_S1D1_1.mkv --size 100 -E faac -B 60 --maxHeight 240 -r 15 -e x264 -o Mythbusters_S1D1_1_hand.mp4 But that one did not play in the phone... I found this in the official manual: If you transfer video clips using another program than Media Go™, we recommend that you select H.264 Baseline profile video, up to QVGA at 30 fps, VBR 384 kbps (max 768 kps) with AAC+ audio at 128 kbps (max 255 kbps), 48 kHz and stereo audio in mp4 file format. So the idea to use H264 seems to be correct.

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