Duplicate: audio and video file compressor
I would like to compress a wmv 2mb or larger file to 3gp 250kb file for mobile devices.
any great compressors for video or audio?
I'm a huge fan of ffmpeg. Find out what codec and resolution your mobile device wants. If you're lucky, H.264 will be supported.
You might have some trouble here. WMV is a container, not a codec, so we can't tell specifically the level of compression we're dealing with and what needs to be changed where, but it may be difficult to get such a dramatic reduction in filesize without making huge compromises, like decreasing the resolution of the video by several orders of magnitude. These compromises may be acceptable for mobile viewing, but there's no guarantee you'll be able to get that filesize down, especially if your file is encoded in a modern codec like H.264 or VC-1.
My first piece of advice is to attempt to locate a good wizard-like transcoder, with a nice non-developer interface on it, etc. Video compression is intense work, and the power tools behind it, and the tools that these wizard-like applications use to actually perform their work, are very complex and take lots of practice and tweaking to get right, and are usually restricted to commandlines. If your mobile device's vendor provides these utilities, for instance, you'll be much better off using them.
If you aren't able to locate such a utility, godspeed and spend lots of time with mencoder and ffmpeg's man pages and IRC rooms. It's not difficult per se, it just takes a lot of study and reading to get acceptable output, especially when you're going after the reductions you've mentioned. Good luck.
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I am using the Videogular2 library within my Ionic 3 application. A major feature of the application is the ability to seek to different places within a video.
I noticed that some formats have very quick seek response, while others take seconds to get there, even if the video is in the buffer already - I assume this may depend on the decoding process being used.
What would the best compromise be in order to speed up seek time while still keeping the file size reasonably small so that the video can be streamed from a server?
EDIT
Well, I learned that the fact that my video was recorded in the mov format caused the seek delays. Any transcoding applied to this didn't help because mov is lossy and the damage must have been done already. After screen-capturing the video and encoding it in regular mp4, the seeking happens almost instantaneously.
What would the best compromise be in order to speed up seek time while
still keeping the file size reasonably small so that the video can be
streamed from a server?
Decrease key-frame distance when encoding the video. This will allow for building a full frame quicker with less scanning, depending on codec.
This will increase the file size if using the same quality parameters, so the compromise for this is to reduce quality at the same time.
The actual effect depends on the codec itself, how it builds intermediate frames, and how it is supported/implemented in the browser. This together with the general load/caching-strategy (you can control some of the latter via media source extensions).
I am currently creating a game. My game will use music from an mp3 file that the user sends in in order to make decisions on where to place things, how fast the level moves, etc. I am fairly new at this, I have been reading information about mp3. Currently I have found all the frames in the mp3 file that I am using. I don't really know where to go from here. What I want to do is measure the frequencies of the sound wave of the music at certain times (like every sec) and then based on that frequency, do what I need to for the game. I don't know whether I should decode the mp3, that looks like a lot of work and I don't want to do that if I don't have 2 or if I can just read the bytes in the frame and convert them without decoding anything. I am developing this in c#, using the game engine FlatRedBall. I am not using any libraries. I am also planning on selling this game so I would like to avoid using other people's code if I can avoid it. Please someone help me, I just need a direction to go from here. I know how to parse the header and calculate the framelength, I just don't know the next step in what I want to do...
Convert your music to .ogg format which is free and use free library to play it.
Note: I was going to post this as a comment but it quickly grew too big. :)
Writing your own MP3 enconder/decoder is probably going to take a good ammount of effort; effort which would probably be better spent on your game itself. Therefore, is possible, I would be all means try to use an open source library.
That said, most good MP3 libraries are LGPL/GPL licensed. This means you can use it in a commercial setting, as long as you dynamically link to it. Also the SDL Mixer library, as of version 1.2.12, supports MP3s and is under a more permissive zlib license, but since you mention C# I don't know if stable and up-to-date bindings are available. Also since your project isn't written in SDL to begin with, it might be hard to integrate it.
Also, as #pro_metedor hinted, perhaps using a more open format could help in licensing issues. In general, OGG achieves better compression than MP3, which is a plus for things like download size, bandwidth/resource usage, etc.
Just shop around for a while, and try to be a little flexible. I'm sure you'll find something nice! :)
For a scientific project i need to compress video data. The video however doesn't contain natural video and the quality characteristics of the compression will be different than for natural footage (preservation of hard edges for example is more important than smooth gradients or color correctness).
I'm looking for a library that can be easily integrated in an existing c++ project and that let's me experiment with different video compression algorithms.
Any suggestions?
Look at FFmpeg. It is the the most mature open source tool for video compression and decompression. It comes with a command line tool, and with libraries for codecs and muxers/demuxers that can be statically or dynamically linked.
As satuon already answered, FFmpeg is the go-to solution for all things multimedia. However, I just wanted to suggest an easier path for you than trying to hook your program up to its libraries. It would probably be far easier for you to generate a sequence of raw RGB images within your program, dump each out to disc (perhaps using a ridiculously simple format like PPM), and then use FFmpeg from the command like to compress them into a proper movie.
This workflow might cut down on your prototyping and development time.
As for the specific video codec you will want to use, you have a plethora of options available to you. One of the most important considerations will be: Who needs to be able to play your video and what software will they have available?
How do you programmatically compress a WAV file to another format (PCM, 11,025 KHz sampling rate, etc.)?
I'd look into audacity... I'm pretty sure they don't have a command line utility that can do it, but they may have a library...
Update:
It looks like they use libsndfile, which is released under the LGPL. I for one, would probably just try using that.
Use sox (Sound eXchange : universal sound sample translator) in Linux:
SoX is a command line program that can convert most popular audio files to most other popular audio file formats. It can optionally
change the audio sample data type and apply one or more sound effects to the file during this translation.
If you mean how do you compress the PCM data to a different audio format then there are a variety of libraries you can use to do this, depending on the platform(s) that you want to support. If you just want to change the sample rate of the PCM data then you need a sample rate conversion algorithm instead, which is a completely different problem. Can you be more specific in your requirements?
You're asking about resampling, and more specifically downsampling, not compression. While both processes are lossy (meaning that you will suffer loss of information), downsampling works on raw samples instead of in the frequency domain.
If you are interested in doing compression, then you should look into lame or OGG vorbis libraries; you are no doubt familiar with MP3 and OGG technology, though I have a feeling from your question that you are interested in getting back a PCM file with a lower sampling rate.
In that case, you need a resampling library, of which there are a few possibilites. The most widely known is libsamplerate, which I honestly would not recommend due to quality issues not only within the generated audio files, but also of the stability of the code used in the library itself. The other non-commercial possibility is sox, as a few others have mentioned. Depending on the nature of your program, you can either exec sox as a separate process, or you can call it from your own code by using it as a library. I personally have not tried this approach, but I'm working on a product now where we use sox (for upsampling, actually), and we're quite happy with the results.
The other option is to write your own sample rate conversion library, which can be a significant undertaking, but, if you only are interested in converting with an integer factor (ie, from 44.1kHz to 22kHz, or from 44.1kHz to 11kHz), then it is actually very easy, since you only need to strip out every Nth sample.
In Windows, you can make use of the Audio Compression Manager to convert between files (the acm... functions). You will also need a working knowledge of the WAVEFORMAT structure, and WAV file formats. Unfortunately, to write all this yourself will take some time, which is why it may be a good idea to investigate some of the open source options suggested by others.
I have written a my own open source .NET audio library called NAudio that can convert WAV files from one format to another, making use of the ACM codecs that are installed on your machine. I know you have tagged this question with C++, but if .NET is acceptable then this may save you some time. Have a look at the NAudioDemo project for an example of converting files.
This question has been in my mind for a few years and I never actually found the answer for this.
What I would like to do is extract the actual waveform/PCM of an MP3 file, so that I can play it using the soundcard (of course).
Ideally I would be experimenting some DSP effects.
My first step was to look into LAME, but I didn't find anything relevant about MP3 decoding in a program or stuff like that.
So I'm asking where I could find something like this.
What language should I use? I was thinking C, but maybe there are programming languages out there that would do the job more efficiently.
Thanks!
Guillaume.
The question boils down to: what are you trying to accomplish?
From the description of your question of decoding an MP3 and playing it on the sound card makes it sounds as if you are trying to make a media player.
However, if your intent is to play around with DSP effects, then it sounds like the question is more about processing the sound rather than decoding MP3s. if that's the case, probably looking into writing plug-ins for existing media players (such as Windows Media Player and Winamp) would be easiest path to what you're trying to accomplish.
Frankly, learning to write your own decoder from scratch is not just a programming problem but a mathematical one, so using existing libraries are the way to go. Talking to the operating system or libraries like DirectSound to output audio seems like unnecessary work if anything. I feel that working on plug-ins for existing players would be the way to go, unless your goal is to make your own media player.
If what you really want to accomplish is playing with audio data, then probably decoding an MP3 to uncompressed PCM using any MP3 decoder, then manipulating it in the language of your choice would accomplish your goal of dealing with effects with sound.
The language choice is going to depend on whether you are going to interact directly with MP3 decoding libraries, or whether you can just use raw audio input, which would allow you to use pretty much any language of your choice.
There was a similar question a while back, Getting started with programmatic audio, where I posted an answer on some basic ways to manipulate audio, such as amplification, changing playback speed, and doing some work with FFT.
libmpg123 should do the trick.
I have been using the Windows Media SDK, not for this purpose, but I am pretty sure there are hooks let that let you intercept the audio stream, or convert MP4 to uncompressed WAV. I used C++.
Lots:
http://www.mp3-tech.org/programmer/decoding.html
Pick your poison...
Also, LAME does decode MP3s (check out --decode option), so you might find something interesting in that source.
-Adam
It really depends what platform you are programming on and what you want to do with the code. If you are on Windows you should look at the windows media format sdk or DirectShow. They should both have the ability to decode mp3 files into the raw waveform. On the Mac, I would expect Quicktime to have this same ability. Others have already suggested source for Linux/open source code.
I would recommend looking at Cubase and Wavelab as both will convert MP3 to WAV etc and allow you to play around with the waveform