I am new to audio file processing. I want to limit the bandwidth of some audio files. The problem is that all the solutions that I find online suggest to limit the sample rate. Is there a way to limit the bandwidth of an audio file, without changing the sample rate?
If audio file is not compressed format (WAV,AIFF,AU), sampling rate is bandwidth (44.1Khz = 44100 Bytes/sec)
but compressed format, Bitrate is bandwidth, re-encode it for changing bandwidth.
Related
I have a video which has 3 audio streams in the file. The first one is English and the other ones are in different languages. How can I get rid of these audio streams without losing the quality of the video and the English stream.
I think ffmpeg should be used but I don't know how to do it.
Video
Bit rate mode: Variable
Overall bit rate: 38.6 Mb/s
Chroma subsampling: 4:2:0
Audio
Format: DTS-HD
Compression mode: Lossless
Is there any limit on how many audio tracks can be muxed within a mp4 container format? If yes, what is the maximum number of audio tracks I can package in MP4 container?
does HLS has any maximum number audio track support?
Many thanks in advance.
There is no practical limit to the number of audio tracks in an MP4 container.
The same is true for HLS.
In both cases though, expect to run into random player incompatibility if you do something strange. If you're creating more than a few alternate tracks, test thoroughly.
I have similar problem to this: Azure Media Services encoded mp4 file size is 10x the original I have a 500MB mp4 file. After encoding with 'H264 Multiple Bitrate 720p' file size is 11.5 GB. Processing costed a lot too. It's no problem with 1 file, but I have to be prepared to share about 100 1GB mp4 files. It has to cost so much? Maybe I didn't know about something? I'd like to share files with AES encryption.
There are a lot of dependencies on what your final output file will be.
1) You have a 500MB mp4 file. Is it SD, HD, 720P? 1080p? Resolution you started at will matter. If you had SD originally, the 720p preset will scale your video up and use more data.
2) The bitrate you started out at matters. If you have such a small file, it is likely encoded at a very low bitrate. Using the 720p multi bitrate preset will also throw more bits at the file. Basically creating bits that are not necessary for your source- as encoding cannot make things look better.
3) You started with a single bitrate - the preset you are using is generating Multiple bitrates (several MP4 files at different bitrates and resolutions.) The combination of starting with a small low bitrate file, and blowing it up to larger resolutions and bitrates, multiplied by 6 or more files tends to add a lot of data to the output.
The solution to your problem is to use "custom' presets. You don't have to use the built in presets that we define. You can modify them to suit your needs.
I recommend downloading the Azure Media Services Explorer tool at http://aka.ms/amse and using that to modify and submit your own custom JSON presets that match your output requirements better.
So I have 2 audio flac files converted from mp4 files. Both are 31 seconds long but one is of 1MB and the other one comes out to be of 4MB. Well, I am using ffmpeg with 8000 sample rate in exactly the same manner. Can anyone explain why this could be happening ?
Is there any particular way in which the mp4 source file has to be coded ? or any other pointers please ?
Thanks already,
asmi
Flac files are compressed using lossless compression so the output file size depends on how well that compression works on a particular file. So even for input with the same duration you would expect the output size to vary.
It is only if you were producing uncompressed output (such as a Wav file) that you would expect the sizes to be the same.
I use ffmpeg to convert videos from one format to another.
Is bitrate the only parameter which decides the output size of a video/audio file?
Yes, bitrate is essentially what will control the file size (for a given playback duration). It is the number of bits used to represent each second of material.
However, there are some subtleties, e.g. :
a video file encoded at a certain video bitrate probably contains a separate audio stream, with a separately-specified bitrate
most file formats will contain some metadata that won't be counted towards the basic video stream bitrate
sometimes the algorithm will not actually aim to achieve the specified bitrate - for example, using the CRF factor. http://trac.ffmpeg.org/wiki/x264EncodingGuide explains how two-pass would be preferred if targeting a specific file size.
So you may want to do a little experimenting with a particular set of options for a particular file format.
Bitrate describes the quality of an audio or video file.
For example, an MP3 audio file that is compressed at 192 Kbps will have a greater dynamic range and may sound slightly more clear than the same audio file compressed at 128 Kbps. This is because more bits are used to represent the audio data for each second of playback.
Similarly, a video file that is compressed at 3000 Kbps will look better than the same file compressed at 1000 Kbps. Just like the quality of an image is measured in resolution, the quality of an audio or video file is measured by the bitrate.