What is the algorithm to convert an mp3 file to a wav file?
It's non-trivial to say the least. You could get an overview of the algorithm here:
http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.26.5956&rep=rep1&type=pdf
But I would suggest using a library for your programming needs, for example:
http://www.ffmpeg.org/
As for the mp3 file, I can offer you mpg123 .net wrapper that is extremely easy to use - you will get the sample for reading the mp3, extracting PCM information from it (this is main part of WAV file - PCM data for the sound).
Please, go here:
http://sourceforge.net/projects/mpg123net/
and download sample code from here:
http://sourceforge.net/projects/mpg123net/files/
Ping me if you need more info/help on the subject.
As aac files are concerned, there is faad project here:
http://www.audiocoding.com/faad2.html
that enables you to do the same with the aac file. If you need .net wrapper, I'm about to put it on sourceforge also.
Related
Currently, I am implementing a new feature of my software using the Libav API. This is the requirement: to merge a list of audio files (MP3 and WAV) and create a unique
audio file (MP3) as output. Note: The challenge is not about concatenating files, but merging them. When the output sound is played, all the input audio content must sound at the same time, as when you merge several files in a video editor.
I was researching about Libav audio streams, and I am just guessing that my requirement is related to the "channels" concept, I mean, that there is possible to include several audios in the stream, using one channel per audio or something like that. I was hoping to find more information about this topic, but FFmpeg/Libav documentation is actually scarce.
Right now, I am able to merge several audio streams to a video stream successfully and I can create a playable MP4 file. My problem is that players like MPlayer/VLC only reproduce the first audio stream with the video, the other two audio streams are ignored.
I was looking at the set of examples included in the FFmpeg source code, but there is nothing specifically related to my requirement, so I would appreciate any
source code reference or algorithm explanation about how to merge several audio files into one using libav. Thanks.
Update:
The ffmpeg command to merge several audio files requires de filter flag "amix", like in this example:
ffmpeg -i 1.mp3 -i 2.mp3 -i 3.mp3 -filter_complex amix=inputs=3:duration=first result.mp3
All the syntax related to this option is described in the FFmpeg Documentation
Checking the FFmpeg source code, it seems the amix feature implementation is included in the file af_amix.c
I am not 100% sure, but it seems the general algorithm is described in the function:
static int activate(AVFilterContext *ctx)
Do you know how to merge several audio files using command line ffmpeg? It would help you if you first understand how to do it with the ffmpeg command then reverse engineer how it achieves it. It's all about how to constrct a filtergraph and pass data through it.
As for examples, check out examples/filter_audio.c and examples/filtering_audio.c
This C example gets two WAV audio files and merges them to generate a new WAV file using ffmpeg-4.4 API. Tip: The key of the process is to use these filters: abuffer, amix and abuffersink.
https://github.com/xtingray/audio_mixer/
Although it doesn't support MP3 format as the output, it gives you the basics to understand how to implement your own requirements. I hope it can be handy for anyone looking for references about this specific topic.
I am trying to write a code in c/c++ (objective c) to parse the audio and video data from mp4 file.
I know that data in mp4 file contains under the mp4 atom but not sure how i can parse out the audio and video data separately.
Thanks in advance for any help.
Mp4 format is fairly complicated. I suggest you use a library. But if you can't use a library, or just wan to learn the format, Than you must parse about a dozen boxes or atoms under the root moov box. The information from there can be used to find frames within the mdat atom. The full specifications is numbered ISO/IEC 14496-12 You should be able to find a copy online.
I'm wondering if it's possible to draw an audio channel of a video or audio file as an image using ffmpeg, or if there's another tool that would do it on Win2k8 x64. I'm doing this as part of an encoding process after a user uploads a video or audio file.
I'm using ColdFusion 10 to handle the upload and calling cfexecute to run ffmpeg.
I need the image to look something like this (without the horizontal lines):
You can do this programmatically very easily.
Study the basics of FFmpeg. I suggest you to compile this sample. It explains how to open a video/audio, identify the streams and loop over the packets.
Once you have the data packet (in this case you are interested only in the audio packets). You will decode it (line 87 of this document) and obtain the raw data of an audio. It's the waveform itself (the analogue "bitmap" for an audio).
You could also study this sample. This second example is how to write a video/audio file. You don't want to write any video, but with this sample you can easily understand how the audio raw data packet works, if you see the functions get_audio_frame() and write_audio_frame().
You need to have some knowledge about creating a bitmap. Any platform has an easy way to do that.
So, the answer for you: YES, IT IS POSSIBLE TO DO THIS WITH FFMPEG! But you have to code a little bit in order to get what you want...
UPDATE:
Sorry, there are ALSO built-in features for this:
You could use those filters... or
showspectrum, showwaves, avectorscope
Here are some examples on how to use it: FFmpeg Filters - 12.22 showwaves.
I have a program that captures and stores H.264 encoded video as well as audio into a proprietary format file. I need to be able to export that video and audio to an mp4 file. I prefer C# but will use C++ if necessary. Any suggestions?
To produce MPEG-4 Part 14 .MP4 file you need a multiplexer. There is a choice of multiplexers out there:
FFmpeg (libavformat)
DirectShow filters (free and open source from GDCL, commercial)
Windows 7+ Media Foundation file sink
API and complexity might vary because some of multiplexers are expected to be a part of pipeline, they are not completely standalone classes. You might want to check respective samples (and license agreements, perhaps, too) to see what is best for you.
Take a look at libmp4v2. Fairly straightforward to use..
http://code.google.com/p/mp4v2/
How to decode mp3 audio using ffmpeg (using the API)? If not complicated - example code?
PS I went to open the file, find the audio channels. then I do not understand what to do ...
It's nice to go all the way through the tutorials but this part of the tutorial deals with decoding audio (although I am currently having a problem with it as avcodec_decode_audio3() is the updated version of avcodec_decode_audio2()).
Hope it helps,
Infinitifizz