Software :: Mencoder Ignores Audio Bitrate Parameter
Dec 26, 2009
When I convert files with mencoder I get the output with incorrect audio bitrate. Seems that mencoder ignores the bitrate I pass to it. Here's my script:
Code:
#!/bin/bash
vbitrate=800
abitrate=128
#mkdir $2/../$2_rip
unlink frameno.avi 2> /dev/null
mencoder $1 -ovc frameno -oac mp3lame -lameopts abr:br=$abitrate -o frameno.avi
mencoder $1 -oac copy -ovc lavc -lavcopts vcodec=mpeg4:vbitrate=$vbitrate:v4mv:mbd=2:trell:cmp=10:subcmp=3:vpass=1 -ffourcc DX50 -o /dev/null
mencoder $1 -oac copy -ovc lavc -lavcopts vcodec=mpeg4:vbitrate=$vbitrate:v4mv:mbd=2:trell:cmp=10:subcmp=3:vpass=2 -ffourcc DX50 -o $2
unlink frameno.avi 2> /dev/null
unlink divx2pass.log 2> /dev/null
Result file should have 128kbps audio bitrate, but here are the results:
Input file:
02_seminar.avi, 42Mb
video: 432x320 00:05:10 25fps DivX5 1Mbps
audio: 48KHz 00:05:10 Stereo 138Kbps mp3
Output file:
02.avi, 34Mb
video: 432x320 00:05:10 25fps FMP4 802Kbps
audio: 48KHz 00:05:10 Stereo 138Kbps mp3
Why audio bitrate remains unchanged?
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Nov 25, 2010
Mencoder settings... I'm doing a simple transcoding of an avi file from FFmpeg MPEG4 codec to XVID MPEG-4 codec. The problem is that the audio bitrate is changing from 192 kbps to 32 kbps. The audio codec for both files is mp3 (MPEG-1 layer 3). (I'm transcoding the FFmpeg MPEG4 encoded avi because my blu-ray player doesn't see, let alone play it. Any XVID MPEG-4 encoded avi files the blu-ray player sees and plays fine). I ran mencoder trying the four settings below but the audio bitrate still comes out at 32 kbps:
Code:
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=192 -o camping_us.avi
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=687 -o camping_us.avi
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=9146 -o camping_us.avi
mencoder camping_en.avi -ovc xvid -oac mp3lame -xvidencopts bitrate=192000 -o camping_us.avi
How to keep the bitrate at 192 kbps?
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Nov 24, 2009
I really need to burn my 64 kbps uncompressed A-Law PCM file to Audio CD uncompressed as is and make this 8 hour file appear on CD Players. What app does it for me? It must fit on CD! Goldwave makes a-law / u-law wav files BTW. I can do this with linux or windows with any application. tell me how or where to download. I have the wav files already.
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Aug 6, 2010
Or at least that's what I figured, when watching videos from ....., etc., the sound in the left channel turns into distorted high frequency noise, anyone had the same issue? Is there a fix for this? Sound in other applications works just fine.EDIT: Oh, and I'm using the proprietary flash plugin, has worked just fine on my Ubuntu Studio on the exact same computer.
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Nov 22, 2009
I have found that if I change the FPS of a video, the audio is out of sync.
Is there a way in mencoder to have it correct this, and maintain the correct sync?
I don't think it is just a matter of audio/video delay. I have tried many times to correct this via that method, and it doesn't come close. Although, if I encode the video to another format, but with the original fps, the audio is sync'd.
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Mar 7, 2010
I'm trying to convert a video in .ogm format to .avi format so I can stream it to my PS3 via uShare. I'm using Mencoder and the following command to do it (to transcode the audio from vorbis to mp3 and the video from mpeg4 to xvid):
Code: mencoder input.ogm -oac mp3lame -ovc xvid -xvidencopts pass=1 -o output.avi.The only problem is, the video contains 2 audio streams - one Japanese and one English. Needless to say, I need the English one. The above command only seems to transcode the Japanese audio though!Here's what ffmpeg has to say about the input file:
Code:
Input #0, ogg, from 'input.ogm':
Duration: 00:24:16.12, start: 0.000000, bitrate: 1280 kb/s[code]...
what I can do to specify the correct audio stream to encode? Also, how can I keep the bitrate the same as the output.avi seems to default to 96kbps?I'm not sure which audio stream is which but a little trial and error will sort that out in no time.
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Jul 28, 2009
I whipped up a script to convert VOBs to AVIs for storage on ym NFS backend. I'm using mencoder on CentOS5 to do the conversion.For 9 out of 10 videos, this works great. There is the occasional 1 that will not have the correct audio stream however. I've tried -alang, -slang, aid, etc. to no avail. Here is my script (also in case anyone else finds it useful), but can anyone offer any suggestions on how to get the correct audio in my avi? When I open the vob in mplayer, I get English. When I run it through mencoder, I get Spanish.
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Aug 17, 2011
However - is there such a thing as a decent HTML editor like dreamweaver? Komposer is buggy as hell - useless! Bluegriffon, well umm - screen fonts are bizarre, especially in viewing source code - brake down, multicoloured obviously a bug - no deb either, looks like a windows program install (?). This does look really good, but is unusable as I cant see in souce code view without getting a headache! Also, ignores css on links.
Seamonkey - you have to open browser then editor, then open your file. Ignores css totally. Amaya - ignores used fonts unless you re-edit - and ignores css on links. Weird way to select things as well, such as images. There must be at least one decent editor?
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Feb 2, 2011
Code:
mencoder The.Negotiator.mkv -of rawaudio -oac mp3lame -ovc copy -o sound-mencoder.mp3
I got a 195.3MB mp3 file
[code]...
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Jan 18, 2010
My 1st and only grandson turned 1 year old and we have taken about 1 million pics of him. I wanted to make a video from pictures taken in the past year. The first one is an ultra sound picture. The last one was taken this last Saturday. I have 153 12 MPixel pictures in a folder. They are named C001.jpg - C153.jpg (I can rename them if needed).
I had noticed mencoder or ffmpeg can create movies from jpegs. But, I didn't think about the frames per second. Would I need to make like 20 copies of each picture and then make it like 10 FPS? or something like that? Optimally, I would like to have a video where each picture lasts for 8-10 seconds or so and I would be able to add a song as audio. (I think I know how to use mencoder to add the audio).
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Jan 7, 2010
I am trying to use ffmpeg and mencoder to slice a portion of my video out but the audio is always 2 seconds out of sync.
ffmpeg -i bird.avs.MP4 -acodec copy -vcodec copy -ss 00:08:40 -t 00:01:00 test.mp4
mencoder -ss 00:08:40 -endpos 00:01:00 -oac copy -ovc copy bird.avs.MP4 -o test.mp4
mencoder -ss 00:08:40 -endpos 00:01:00 -oac copy -ovc copy -noskip bird.avs.MP4 -o test.mp4
[Code]....
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May 4, 2010
I got the following modprobe scripts modprobe -k -q streams what does the -k parameter mean?. is it exist in older modprobe? I don't see -k parameter in recent modprobe.
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Mar 15, 2011
I'm using mencoder to capture audio from a Encore ENLTV-FM3 video capture device. I have recently noticed that, since one week ago, when the machine was forcibly restarted due to a power outage, all recordings are slightly pitched, they play back slower than they should.
I narrowed down the problem to the following command line:
$ time mencoder -really-quiet -tv driver=v4l2:device=/dev/video1:chanlist=us-cable:audiorate=32000:alsa:adevice=hw.1:input=0:amode=1:normid=11 -endpos 00:10:00 -ovc copy -oac pcm -of rawaudio -o test-32000.wav tv://69
real 9m54.886s
user 0m5.536s
sys 0m1.740s
$ ls -l test-32000.wav
-rw-r--r--@ 1 martin martin 76800000 Mar 15 17:20 test-32000.wav
Somehow, mencode managed to gather precisely 10 minutes worth of raw audio in 9m 55s. That's not physically possible, unless the capture device's A/D converters are "overclocked". I can't think of any other explanation besides hardware failure. Can that be? Could it be that something got burnt during the power outage and now the capture device's internal clock went nuts?
Since the machine's restart, I've also noticed dmesg is flooded with entries like this:
CE: hpet increased min_delta_ns to XXX nsec
Which seem to indicate that the computer's high precision event timer is somehow out of sync. Does this have to do with the audio issue? Can it be that the audio converter's sample rate is linked to the HPET? I'm totally lost here. Has anyone bumped into something similar?
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May 8, 2010
Trying to play a *.mp4 video using mplayer and the picture is jumpy and the sound completely screwed. It say it is guessing the bitrate thus:
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
FAAD: compressed input bitrate missing, assuming 128kbit/s!
AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400)
Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio) decoder)
In fact the bitrate should be 139 or possibly 132. So assume that is what is wrong. Have assiduously coated myself with nourishing smegma and the like but am unable to work out how to set the bitrate in this case.
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Jan 29, 2011
I have the same mp3 on two different distros, Fedora 14 and Ubuntu 10.10, both using gnome. The md5sum of the mp3 is identical on both distros, but on Ubuntu the bitrate is 96kbps and on Fedora it is apparently 257kbps. Discovered this using RhythmBox but also appears in Nautilus browser, under the properties tag.
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Dec 16, 2010
Kernel 2.6.21.5, Slakware 12.0 I have plenty of files like this:
Code:
$ file 23-1.mp3
23-1.mp3: Audio file with ID3 version 23.0 tag, MP3 encoding
$
All I want to know about a file like this is the bitrate is has been created with. Can this data be inside the tag? What's a cli program that lets me know that information?
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Jun 17, 2011
I am wanting to reduce around 9.9gigs of music to 8gigs to fit on my phone. I have done some snooping and noticed a program called Lame. I also noticed a code:
for x in [ `ls -1 *.mp3` ]; do lame --preset 32 $x new32-${x}; done
I was wondering if someone would be able to explain the code and how I go about converting the bitrate to 96?
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Feb 2, 2011
I am trying to find out how I can estimate the IP Layer bitrate I would get from an ethernet link? Say at 100 Mbits/s or 200 Mbits/s (powerline)? Is there a formula that I could use to calculate that?
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Dec 2, 2008
The subject of this post is actually a question e.g. is there a mp3 ripper that allows the bitrate to be modified. I have looked at cdparanioa, which I believe is the foundation of most of the Linux rippers, but it does not allow modification of the bitrate. I realise I could ripper in flacc format but I have an mp3 player as does my grandaughter so I need the mp3 format. Incidentally is there a player(portable) that will play flacc encoded files?
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Dec 3, 2009
I'm new to gStreamer and need to determine the bitrate of a stream. Has anyone here worked with this before and does anyone know how to do it?
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Mar 6, 2011
This may be a minor irritation, but I'm puzzled by it nonetheless. I'm ripping my CDs to MP3 using Sound Juicer with the lamemp3enc plugin and following gstreamer pipeline.
audio/x-raw-int,rate=44100,channels=2 ! lamemp3enc name=enc target=0 quality=0 ! xingmux ! id3v2mux
All was great a couple days ago, but suddenly the files' audio properties show 48 kbps bitrate when it should be well over 200. I even changed the pipeline to use "target=1 bitrate=256" with the same result. The files are being encoded as expected and that is reflected in the file size and the Statistics view in VLC.
I have two computers running Ubuntu 10.10. One has all the latest updates, but the other has not been updated in several days. This problem is happening on the former, but not the latter. gst-inspect lamemp3enc shows both computers have the same version of the encoder.
I thought I'd put it out to the forum here before submitting a bug in launchpad.
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May 3, 2011
I have a large collection of music albums sorted in folders which are named like this: "Zombie Ritual - 2004 - Night Of The Zombie Party" (%{artist} - %{year} - %{album}). I want to rename them so as to be indicative of the bitrate, for example, "Zombie Ritual - 2004 - Night Of The Zombie Party" => "Zombie Ritual - 2004 - Night Of The Zombie Party (@320)". It will be hard to do this manually. I tried to use EasyTag and Kid3 to do this, but they cannot add bitrate to tags.
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Jan 21, 2011
howto to transcode multiple mp3 files to another bitrate? I have a lot of 320bits mp3 which should be converted to 192bits before I can play them on my car stereo.
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Jul 5, 2010
I have 1 router 2 pcs. We had to move one pc away from the router so I bought a wifi card. Everything went fine, the card works and I have internet but the network is incredible slow.
I am using Ubuntu Lucid Lynx 10.04
lspci | grep Wire
Code:
Code:
Then:
Code:
But there are no changes.
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Apr 24, 2011
I've been testing with the SoundConverter software and I want to write a script for it.Most music files on my pc are *.flac. But I want to convert some albums to my mp3 player with a script. Everything works fine. I do this:Code:soundconverter -b -m audio/mpeg -s .mp3 *.flacBut the quality is 128kbps.Is there any way to change the bitrate (in the terminal ofcourse)?And if this is not possible, is there an alternative that copies the tags correct like SoundConverter?
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Jun 13, 2010
I have finally found a native Linux music player that's as light and as simple as Foobar(unfortunately Foobar under Wine is giving me problems so I needed to find a native player). creating custom columns for the player. And since I can't find any DeadBeef forums, I'm hoping people here are familiar with this player.
So, the two main custom columns I'm looking to create are Codec and Bitrate. Codec to show if the tracker is mp3, ogg, flac, etc. And Bitrate to show the bitrate of each track.
Now there's an option in DeadBeef to create columns, but I'm having trouble figuring out how to get the column to show the info that I want. I was hoping that the way you get custom columns to show up in Foobar would work in DeadBeef. But inputting %bitrate% and %codec% into the columns in DeadBeef does not work.
how to get the two columns that I'm wanting to display the info that I want?
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Jun 14, 2011
I am looking for an application to detect mp3 duplicates using the mp3 spectrograph (sound wave matching) or something similar because simply I have many mp3 files that are the same but with different bitrate,ID3 tags,filesize
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Sep 23, 2010
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Apr 11, 2011
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Mar 20, 2010
Today I noticed, that despite a 1kb/s limit upload being set i was seeding at an unlimited bandwidth, it had been running for 3 hours wasting about 300mb.
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