I am using moviepy to generate MP4 files from sets of shorter clips, each with their own audio. The problem is that the resulting MP4 often has a very high dynamic range from one clip to the next and I would like to apply audio compression to make it easier on the ears. In Google I can only find results about audio information compression, but not about audio compression from the audio engineering perspective.
I would like to know if there is some way of doing this with moviepy, or with some other library. I have no issue with invoking (non interactive) command line utilities either.
Thank you.
Related
I am pretty new with processing audio file. '
I want to build a web app that can take audio file and turn the into visualization for user like this https://github.com/CrowdCurio/audio-annotator
Right now I want to research on visualize audio datas. Original data that was stored in S3 come in two form .ts and .flac. That's why I want to ask if there's any visualization tool which can directly use .ts or .flac audio file.
Because right now the solution I think of will be first convert them into .wav or .mp3, so most visualization tool can process them, but .wav file is really storage-wasting as far as I know.
So if you know any approach or tool to do this. Please let me know!
Audio visualization requires audio data. Your compressed audio isn't audible until decoded. Therefore, you must decode them to PCM before visualizing.
This doesn't require that you store the files as WAV, but you'll at least have to decode them on-the-fly.
How to get file information like sampling rate, bit rate etc of .raw audio files using terminal in linux? Soxi works for .wav files but it isn't working for .raw.
If your life depended on discovering an answer you could make some assumption to tease apart the unknowns ... however there is no automated way since the missing header would give you the easy answers ...
The audio analysis tool called audacity allows you to open up a RAW file, make some guesses and play the track
http://www.audacityteam.org
In audacity goto File -> Import -> Raw Data...
Above settings are typical for audio ripped from a CD ... toy with trying stereo vs mono for starters.
Those picklist widgets give you wiggle room to discover the format of your PCM audio given that the source audio is something when properly rendered is recognizable ... would be harder if the actual audio was noise
However if you need a programmatic method then rolling your own solution to ask those same questions which appear in above window is possible ... is that what you need or will audacity work for you ? We can go down the road of writing code to play off the unknowns mentioned in #Frank Lauterwald's comment
To kick start discovering this information programmatically, if the binary raw audio is 16 bit then each audio sample (point on the audio curve) will consume two bytes of your PCM file. For mono audio then the following two bytes would be your next sample, however if its stereo then these two following bytes would be the sample from the other channel. If more than two channels then just repeat. Typical audio is little endian. Sampling rate is important when rendering the audio, not when programmatically parsing raw bytes. One approach would be to create an output file with a WAV header followed by your source PCM data. Populate the header with answers from your guesswork. This way you could listen to this output file to help confirm your guesses.
Here is a sample 500k mono PCM audio file signed 16 bit which can be imported into audacity or used as input to rolling your own identification code
The_Constructus_Corporation_Long_Street-ycexQvMy03k_excerpt_mono.pcm
I am trying to merge two different AAC audio files and a H264 video file to form a single TS file using C++ code. I have been successful in it. So now my TS file possess the following order. First, video part from the video file, then audio part from the first audio file and then audio part from the second audio file and then again the video part and it goes on the same way. On hearing the resulting file, I recognized the presence of the different audio files with the video.The problem is that the resulting audio ain't that much cleared. Distortions can be recognized making it unclear to hear. Also note that the resulting audio seems slow as compared to the original.Can anyone guide me in getting off those distortions and procuring the exact replica of my original files ?
Thanks,
Ashish.
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/
The MPEG-4 file format allows multiple streams to be present in a file.
This is useful for videos containing audio in multiple languages. In the case of such a video, the audio streams are synchronized to the video.
Is it possible to create a MPEG-4 file the contains desynchronized audio streams, i.e. the audio track are played on after another?
I want to design a MPEG-4 file that contains a music album, so it is crucial that the tracks are played one after another by media players such as VLC.
When I use MP4Box (from the GPAC framework) the resulting file is recognised by VLC as having synchronized audio streams. Which box of the MPEG-4 file format is responsible for this? Or how can I tell VLC that these audio streams are not synchronized?
Thanks in advance!
I can think of two ways you could do that, and both would be somewhat problematic.
You could concatenate all the audio streams into one audio track in the MP4 file. This won't be ideal, for some obvious reasons. For one thing, it's not exactly what you were asking for.
You could also just store the tracks as synchronized audio streams, but set the timing information in such a way that the first sample of the second track won't start playing until the first track finished playing, etc.
I'm not aware of any tools that can do this, but the file format will support such a scheme. Since it's an unusual way to store audio in an MP4 file, I would expect players to have problems with this, too.
Concatenating all streams would work and the individual tracks can be addressed by adding chapters. It works at least with VLC.
MP4Box -new -cat track1.m4a -cat track2.m4a -chap chapters.txt album.m4a
The chapters.txt would look something like this:
CHAPTER1=00:00:00.00
CHAPTER1NAME=Track 1
CHAPTER2=00:03:40.00
CHAPTER2NAME=Track 2
But this is only a hack.
The solution I'm looking for should preserve the tracks as individual streams.