how to increase the volume of ogg and mp3 files using ffmpeg. Some of the podcasts that I listen to are not loud enough, and I have my media player set to the highest volume.
View 1 Replieshow can i merge two files with ffmpeg?
one is : /home/pt/t/pa1.flv
the other is : /home/pt/t/pa2.flv
1 o merge with ffmpeg
ffmpeg -i /home/pt/t/pa1.flv -i /home/pt/t/pa2.flv -vcodec copy -acodec copy /home/pt/t/dd.flv
the problem is: the merged file ( /home/pt/t/dd.flv ) just contain one file--the first one
/home/pt/t/pa1.flv,there is no the second file--/home/pt/t/pa2.flv in the /home/pt/t/dd.flv
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I'm running ClipBucket on my Ubuntu 10.04 LTS server x64.Conversion of AVI to FLV works just fine, however when I try to upload an MPG file, it fails. Apparently I should recompile ffmpeg to allow for MPG > FLV conversions.
View 3 Replies View RelatedSince ffmpeg 6 is out for some time, when will Ubuntu Lucid afford such a new ffmpeg package from its repository?
View 2 Replies View RelatedI used to have Windows on my laptop, until I got the blue screen, so I just installed ubuntu. Well everything was ALOT louder on Windows, and I was wondering if there was a way to increase the maximum volume. I have to turn the volume almost all the way up to watch a movie. On Windows 1/4 volume was plenty. I searched, but alot of the answers I saw were that it was hardware limitations, which cant be the case, since it was a ton louder on Windows
View 5 Replies View Relatedthe system volume on login is max, though my login sound is approximately half. it is not the sound itself, considering when i play it once i'm logged in, its max; just when i log in it seems to be more quiet...
View 3 Replies View RelatedSome audio output from files or streams is too low. It is obvious that hardware is able to play the same sounds but louder but because of the data it just plays it at some low level even at 100% volume. Vlc can generally increase the volume of a file up to 200%. Is there a way to do the same thing VLC does system-wide and if possible for an arbitrary v percentage value.
If there is no application that does this, where should i look into for libs to do it myself or what code should i modify(eg code in the alsamixer)
Note: Asked the same thing on stackoverflow and they directed me here.
I have a problem with my sound. I can't really increase the volume of the sound. I already looked in alsamixer and everything is maximized. With the same headphones in Windows, I can achieve great sound volume. Does someone know why the sound is too low? I started alsaconf and created a new configuration. After that run aumix. Now I have the volume I wanted.
View 2 Replies View RelatedI have centos 5 virtual server running on ESXi (vsphere 4.1).i have to increase disk space. I increased the size of the virtual machine in Vcenter.But i allready have 4 primary partitions. when i run fdisk /dev/sda, i get this :
Device Boot Start End Blocks Id System
/dev/sda1 * 1 13 104391 83 Linux
/dev/sda2 14 3916 31350847+ 8e Linux LVM
/dev/sda3 3917 6527 20972857+ 83 Linux
/dev/sda4 6528 13054 52428127+ 83 Linux
I am running KDE 4.6 in Debian Testing. Is there a way to increase the sound (i.e. more than the standard 100%)? The current settings with my speakers seem a bit too quiet in some cases.
I found a way to do it in PulseAudio, but I don't think Debian's KDE build is compatible.
I converted all of my vinyl to flac format, but when I played them in mplayer, the volume was way lower than the mp3's that I downloaded. So, when I have them all in the playlist, I'll have my flac file at a good volume, then an mp3 comes on and blasts me. Is there a command line program that I can use where I can specify a new volume for the flac files? I wanted a command line program so that I can script this entire process. I do not want to sacrifice the quality of the songs that I encoded (at 96kHz).
View 2 Replies View RelatedI've got my xubuntu 10.10 install just about perfect on a little acer aspire d250, apart from a small sound useability issue: In the interests of simplicity and resource usage I removed pulseaudio. After a bit of fiddling I got it so that my USB soundcard (ProDac) is recognized and automatically set as the default soundcard when plugged in. Any sound applications automatically use the USB sound if it present, no need to around with pulse. The only problem is that my netbook's volume control keys still only control the master volume of the inbuilt soundcard, and have no effect on the usb sound. Does anyone know of a way to change which sound device these keys actually effect? I'd like to write a little script so that when the usb device is detected the keys are remapped.
View 1 Replies View RelatedI'm running Kubuntu 10.10 32 bit on an old DFI KT600AL motherboard based system using the onboard VIA 3058 AC97 audio (because it supports front panel audio connections and none of the add-in PCI soundcards I have do). I have an old Gateway/STB TVPCI TV tuner card (mainly wanted the FM radio part to work) hooked up to the cd audio connector on the motherboard because the digital audio over the pci bus apparently isn't supported for this card (neither is the onboard analog mixer on the tv tuner card, I had to hack a CD-ROM audio cable and solder it to the audio outputs of the tv tuner module on the TV tuner card). When I use the master channel as the master channel (selected in Kmix) then as one would expect it affects the output volume of all other audio playing on the system except that which is being handled by the PCM channel. On Windows the PCM channel was also affected by the "Volume Control" slider such that ALL volume levels were reduced when moving the slider. I'm hoping someone can help me figure out how to make it work like this on Kubuntu.
First off I should warn you that I (like many users of older hardware that is not properly supported under PulseAudio) have uninstalled PulseAudio (because it's garbage) and am using Alsa to manage my audio hardware. I would think that there would be some way to do this using the 'amixer' application to add the PCM channel as a component of the 'Master' channel so that when the volume is turned down using the 'Master' channel control it will affect the PCM channel too (at least the output to the speaker jack, not necessarily the capture or mix volume though), but I'm not really any good at doing things from a terminal window and the options for the amixer command kind of confused me.
I used Ubuntu 10.10 (64 bit) on IMac i7. As far as, I remember, after installing Pulse-Audio Equalizer, for each restart of the system, after login screen, system sets volume to maximum value. What can I do?
View 1 Replies View Relatedsince using 10.04 I have a big problem with my usb headset (freetalk everyman)
1. Problem: I cannot regulate the volume of the phones (output) anymore with gnome-volume-control. By default the volume is set to 100% which is way too loud. When I set it under 100% there is no sound at all. Values over 100% work.
2. Problem: The X server is freezing iregulary when I connect the headset and disconnect it, Magic SysRq works. I checked Xorg.0.log and found out that it recognizes the usb headset as keyboard:
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I am struggling with a problem with the volume control... when i try to adjust the volume with the sliding bar i can not do it. If i put the scrollbar to the 0 position, it is muted. when i move it to anywhere else, i get the full volume. From other programs like vlc, i can adjust the volume with no problem.
View 2 Replies View RelatedMy volume control seems to have only three settings: TOO LOUD, VERY QUIET, OFF. If I adjust the Master volume slider from the Panel, only about the right 1/8th has any effect and if I slide it past about the 95% mark, I get no sound at all. At the far-right end of the volume control, I get total volume. It's basically like the slider only has any impact in the top 5% of it's space. Below 95% of the slide, there is zero volume.
It's so drastic, that one tap down of the volume button on my keyboard kills the sound (because it nudges the slider past that 95% mark). This happens in the Audio settings, etc., system-wide. If I adjust the volume for an individual app from within the app (MPlayer, for example) it behaves exactly as expected (that is, app volume works smoothly, it's just master volume that has this issue). I have run through most of the "obvious" things-- I'm wondering if I broke something by installing the extra KDE packages to get Amarok to work.
I'm running squeeze (last updated today), and everything has been working great. There is only thing that would simplify my life minutely..Anybody know how can I have the volume buttons on my laptop change the "pcm" channel volume rather than "master"? If they could control pcm, then I could adjust the volume coming out of my headphones or my computer speaker (both controlled by pcm, but not master...seems strange to me) with just one click.I tried to find this info online, but all results seem to refer to an older version. The simple "click here, set this" solution no longer is possible.
View 2 Replies View RelatedDebian Squeeze 6.0.0 on a Thinkpad T43, sound volume is not synchronized between the laptop volume buttons and GNOME's Volume Applet. So if I turn the volume all the way down with the physical buttons, the volume applet may still indicate 75%.I did not have this problem in Debian Lenny. Pressing the volume buttons used to show a volume bar on the screen, as did pressing the mute button.
View 8 Replies View RelatedYou want to cut lets say a 15 minute video into 10 minute video, removing the first 5 minutes of it using ffmpeg?s?
View 8 Replies View RelatedI use this command line to transcode media file
Code:
ffmpeg -i in.ts -y -threads 2 -vcodec libxvid -b 300000 -bufsize 8192 -g 300 -s 320x240 -aspect 4:3 -f mp4 -pass 1 out.mp4
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I've created some time-lapse videos from photos, using this command: ffmpeg -i IMG_%03d.JPG -s 1440x1080 -sameq video.MP4
And it worked great. Now I want to join several of these time-lapse videos to make a single, longer video (all the input videos have the exactly same format). I already tried using: cat video1.MP4 video2.MP4 > stitch.MP4
but the output ends up being equal to video1.MP4, I don't want to transcode nor changing any parameter of the video, I just want a end-to-end stitching, as if those videos were on a playlist.
I have so many flac files. I want to have 1 minute samples of the files in mp3 and a bitrate of 64. (First 1 minute of all files) How can I do that with ffmpeg command?
View 5 Replies View RelatedThis is happening on Ubuntu 9.10 serverI'm trying to increase the number of open files allowed for a user. This is for an nginx webserver where the current limit of 1024 is not enough.According to the posts I've found so far, I should be able to put lines into /etc/security/limits.conf like this;
Code:
* soft nofile 4096
* hard nofile 4096
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I use ffmpeg to convert my files, but when I do so, I lose the keyframes metadata from my flv files. The command I use right now is: $/usr/local/bin/ffmpeg -i %s -ab 96k -ar 44100 -b 500k -r 25 -f flv %sHow can I avoid ffmpeg to delete the keyframes info?
View 2 Replies View RelatedI've got a bunch of HD files in .mkv format that I want to convert to a format that the 360 will read (currently just trying .avi files) so I can stream the media to my xbox360. I'd rather avoid transcoding the entire file, so I tried unpacking the .mkv file into audio and video (and subtitle if necessary) and repacking said file into avi using ffmpeg. The repacking is where I am having the problem. Here's the commands I tried at first (it might look familiar):
Code:
mkvextract tracks file.mkv 1:temp_video.avi 2:temp_audio.ogg 3:temp_sub.srt
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy file.avi
However, I keep getting this annoying error:
Code:
Error while opening codec for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height Where stream #0.1 is the output audio stream to the .avi file. I think I have narrowed it down to the fact that the input streams are all 6 channel (5.1 surround sound) which mp2 doesn't support (the ffmpeg audio default) so I did this instead:
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec libmp3lame file.avi
I'm still getting the same error, but when I use
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec libmp3lame -ac 2 file.avi
to change the sound to stereo it works. Problem is I'd like to keep my media in surround sound. Any thoughts on how to get all 6 channels in the new mp3 file? I'm fairly certain that mp3s support 5.1. And after some messing around, I haven't found that my ffmpeg can transcode audio format from 5.1 back to 5.1. So since (I think) the 360 can play .ac3 files I decided to just copy both the video and audio file to the .avi file sans transcoding using
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec copy file.avi
however now about halfway through, I run into this error:
Code:
NULL @ 0x22f3a00]error, non monotone timestamps 62562 >= 62562
av_interleaved_write_frame(): Error while opening file I can't seem to find any solution to this problem anywhere online, does anyone know how to get around this? I'd prefer to use ffmpeg for this if possible. (so I can write a script and just attack an entire directory at once) For what it's worth, I'm using the latest ffmpeg in the ubuntu (karmic) default repositories.
i want to convert a song from ogg format to mp3 using ffmpeg the command i use to do this is
ffmpeg -i /home/john/Music/input.ogg to mp3 output.mp3
i then get then get a message back saying
"unknown format"
how do i correct this ?
The command line to convert to ogg theora
ffmpeg -y -i input.flv -sameq OUTPUT.ogv
runs perfectly on 9.04 but on 10.04 it gives errors:
Code:
ffmpeg -y -i input.flv -sameq OUTPUT.ogv
FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration: --extra-version=4:0.5.1-1ubuntu1 --prefix=/usr --enable-avfilter
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It seems more and more people are encoding with the MKV container on bit torrent these days, and a lot of the shows I'm watching are starting to release almost exclusively with .mkv formatted videos. This is not a problem if I want to watch the shows on my computer but I've become accustomed to watching them on my PlayStation3 using my thumb drive. It seems the offical documentation for the PS3 includes a list of supported codecs [URL], but when I use FFMPEG to convert with libxvid video and aac audio in the MP4 container my PS3 says the output video is not supported. I've also tried most combinations of libxvid, libx264, mpeg4 for -vcodec and aac, libmp3lame for -acodec in several different container formats but nothing seems to work. I have found one option that always works:
Code:
ffmpeg -i inputfile.mkv -sameq -ac 2 outputfile.mpeg
I don't like doing it this way, however, as the output file is twice the size and the audio quality is terrible. If I don't reduce the audio channels to only two using -ac 2 FFMPEG throws an error (apparently MKV audio supports 6 channels). And preserving the video quality in MPEG video using -sameq produces too a large file (and I prefer to keep my files as lossless as possible). Ideally I want to save the files on an external HD I have but if a single episode of a show is 1.5 GB it's not very pratical.
Anyway, the PS3 docs say it supports h264 and xvid with aac audio, but apparently I'm doing something wrong. Has anyone sucessfully used FFMPEG to convert an MKV to MP4 for use on a PS3?
I have a .mkv container with proper video (h264/~6000kbps) and audio (DTS/1500kbps). For some reason I want to have similar quality but with another video and audio codecs. For video it goes well and ffmpeg is very helpful here, no problems. Also, I want DTS to become AC3/320-448kbps/5-6 channels. Commands like this:
Quote:
ffmpeg -i in.dts/mkv -acodec ac3 -ac 5 -ab 448k out.ac3
gives me not proper reordered sound in channels, so I hear center on the right, music on the left, etc.
Is there a way to make that properly with ffmpeg?
Also, I have a question about multi core encoding cause I have a quad processor. Ffmpeg seems not using them in all the way, I get something like 25% of CPU usage while encoding. Command -threads, even using the h264, does not make any extra load of cpu.