I have a C++ program that plays a video file from a host. I am playing audio using BASS_StreamCreateURL, but the problem is still there, I used mp4 format changing moov and mdat blocks. I use movavi, it doesn't have web optimization (to replace blocks), it's not very convenient to use other programs. So what other formats can be used with bass
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I'm currently using SFML Audio for my visualizer and it works great but some of the limitations are causing problems. Currently my music library consists of MP3 and M4A files and I'm not keen on having to convert them every time I make additions to the library. I'm also looking for a way to read metadata from MP3 and M4A files so that I can list the title, album, and artist. I would you iTunes' exported playlist to get that info yet it only lists the title of the song. Since this is a visualizer I also need to be able to get at least 8192 samples per channel at the current audio position in order to produce the visualization. And lastly I'm hoping to be able to stream media instead of loading it all at once so that there's not a short pause when switching songs. What are some alternatives that support these features?
Must work with Windows.
C++ Library for loading/playing audio (I can make due with C if need be)
Supports MP3/M4A
Supports Reading metadata (Optionally on alternative library)
Supports getting sample data at the current position in the song (at least 8192 samples in advance)
Supports streaming media instead of loading it all at once to prevent lag on switching songs.
The Kinect OpenNI library uses a custom video file format to store videos that contain rgb+d information. These videos have the extension *.oni. I am unable to find any information or documentation whatsoever on the ONI video format.
I'm looking for a way to convert a conventional rgb video to a *.oni video. The depth channel can be left blank (ie zeroed out). For example purposes, I have a MPEG-4 encoded .mov file with audio and video channels.
There are no restrictions on how this conversion must be made, I just need to convert it somehow! Ie, imagemagick, ffmpeg, mencoder are all ok, as is custom conversion code in C/C++ etc.
So far, all I can find is one C++ conversion utility in the OpenNI sources. From the looks of it, I this converts from one *.oni file to another though. I've also managed to find a C++ script by a phd student that converts images from a academic database into a *.oni file. Unfortunately the code is in spanish, not one of my native languages.
Any help or pointers much appreciated!
EDIT: As my usecase is a little odd, some explanation may be in order. The OpenNI Drivers (in my case I'm using the excellent Kinect for Matlab library) allow you to specify a *.oni file when creating the Kinect context. This allows you to emulate having a real Kinect attached that is receiving video data - useful when you're testing / developing code (you don't need to have the Kinect attached to do this). In my particular case, we will be using a Kinect in the production environment (process control in a factory environment), but during development all I have is a video file :) Hence wanting to convert to a *.oni file. We aren't using the Depth channel at the moment, hence not caring about it.
I don't have a complete answer for you, but take a look at the NiRecordRaw and NiRecordSynthetic examples in OpenNI/Samples. They demonstrate how to create an ONI with arbitrary or modified data. See how MockDepthGenerator is used in NiRecordSynthetic -- in your case you will need MockImageGenerator.
For more details you may want to ask in the openni-dev google group.
Did you look into this command and its associated documentation
NiConvertXToONI --
NiConvertXToONI opens any recording, takes every node within it, and records it to a new ONI recording. It receives both the input file and the output file from the command line.
I am working on a C++ project with openCV. It is a simple web cam application with basic features like capturing pictures and videos. I have already been able to save video (w/o audio). Since openCV doesnot support audio processing, I was wondering if there is any way I can record audio separately in a different file and later combine those together to get one video file.
While searching on the internet, I did hear something about using ffmpeg with openCV. But I just cant figure out how to do it exactly.....
Can you guys help me? I would be very grateful... Thankyou!
P.S. I have used openCV and QT (for GUI)
As you said, opencv doesn't by itself deal with audio. However once you get a separate audio and video file, you can combine them using a technique called muxing. There are many many ways to do this. I use VirtualDub for most of my muxing needs, although it is windows only (not sure if that's a problem). I know ffmpeg is also capable of muxing (via the command line interface), I can't recall what the command is. There's also mplayer and a multitude of other programs out there to do this.
as far as i know openCV is good for video/image processing. To support audio processing, you can use other libraries e.g. PortAudio or C-sound.
How to write mp3 frames (not full mp3 files with ID3 etc) from PCM data?
I have something like PCM data (for ex 100mb) I want to create an array of mp3 frames from that data. How to perform such operation? (for ex with lame or any other opensource encoder)
What do I need:
Open Source Libs for encoding.
Tutorials and blog articles on How to do it, about etc.
You should be able to use LAME. It has a -t command line switch that turns off the INFO header in the output (otherwise present in frame 0). If that still leaves too much bookkeeping data, you should be able to write a separate tool to strip that away.
You are already on the right track: use LAME external executable, or any other shell-invoked encoder.
To build MP frames, were your layer of interest is 3, is not easy to do from scratch. There are compression steps, Fast-fourier transforms followed by quantization, which are of complex and tediously long explanation. The amount of work required for a developer to build it from scratch is very big.
There are programmatic C and C++ MP encoding libs, but you will be either asked for fees, be left with very limited support, or have very limited interfacing options.
Go LAME, study their wiki.
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.