guys!
I've been asked at work to prepare a large audio library for tests, which includes different files of HE-AACv1 & HE-AACv2. To be specific, all sample rates from 8000 up to 48000 and bit rates from 8 bit to 32 bit. But I've been digging the internet for a while and can't find the exact information. Is it even possible to make HE-AAC with 8KHz sample rate? And are there any tools that allow to directly set this specific values (bit rate and sample rate)? I've tried adobe audition, audacity and a bunch of something I found on google, but neither gives me opportunities that I need, or it's just me being dumb.
Please, if you know the way to create a HE-AACv1 and v2 from for example mp3 or wav, tell me how to do it, I would be very grateful.
Thx!
Adobe Audition can definitely create all WAVs that you need, look under "Edit->Convert sample type". You should first create your wav files, and then use some encoder to transcode to AAC.
Related
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 have built a source client using Portaudio and LAME which streams the microphone input to an Icecast server to be listened to online via the HTML5 tag. I have managed to (supposedly) get the quality of the stream to MP3 320kbps at 44.1kHz and am looking for a way to confirm this using tests and or benchmarks.
I have an indication that these stats are somewhat correct from looking at stream inspectors in software such as iTunes and VLC, but I am looking to get a more in-depth data set.
What I basically want is to be able to test how much of the original file is being lost over the stream and if or how much the quality changes depending on environmental conditions of the broadcaster or streamer.
Does anyone know of any tools, frameworks to get some hard numbers or representations of this data?
If VLC tells you the stream is 320kbit CBR, then it is.
It sounds like what you're looking for is a comparison of the actual audio content. This is highly subjective. MP3 is built to use features of how our hearing works to save bandwidth. For example, quiet sounds are masked by loud sounds. High frequencies are harder to hear and are simply rolled off.
You can compare the spectral analysis between the original PCM-sampled waveform and the MP3 decoded waveform, but this doesn't tell you how humans interpret that sound. For that, you would have to survey humans.
I would like to create a utility in either PHP or Perl to convert an audio file created by the Nortel's Callpilot voice mail system into a wave file. The problem is that the format, which has the .vbk file extension, is unknown to virtually any audio player. To date, I have not found one that will play a .vbk file. I've looked at audio file conversion libraries in CPAN and tried many of them, they don't recognize the file. I was not successful with PHP's audio formats manipulation either. Nortel does provide a converter, however, it does not suite my needs. I would like to have this run via cron on a CentOS system. I don't know how to reverse engineer this format. There seems to be just scraps of info on this format on the web. This page indicates that it is "based on the H.232 format":
https://www.odesk.com/o/jobs/job/Reverse-Engineer-Nortel-VBK-Audio-Format_~~f501f11679f3f6bb/
I know this is a very old thread, but I've recently been looking into converting Nortel's vbk format as well. Importing the vbk files into Audacity with raw data option, Encoding: U-Law, Byte order: little-endian, Channels: 1 Channel (Mono), Sample rate: 8000 Hz. Not sure if they have multiple formats for their vbk files, but mine were from a BCM50 phone system.
Well, this is the joy of closed proprietary systems. But there is a chance they could play nice. Try to contact Callpilot and see if they'll give you the format specs. It's worth a shot.
As for reverse engineering, you need to be able to generate known content. Like a constant tone at 60Hz for exactly 1 second. Then at 50Hz. Then at 10 seconds. Compare them. Isolate the data from the metadata. There is going to be compression involved, so try a handful of common compression schemes, maybe research into Nortel's practices will probably tell you more. If you can feed that into a player and get a tone back out, you're on your way.
There's probably more informed and structured ways to go about reverse engineering, but from my experience it's a lot of trial and error.
I searched many questions - but no one seems to be giving simplest, most uniform approach, hence please do not close as duplicate.
My requirement is simple: I have quiz app.
I want to include:
background music that plays continually - probably more than one
audio.
I need occassional sounds played at specific events - they
are very short in duration. Maybe 4-5 in number.
What sound format do I use? [aac etc]
How do I produce it? (optionally, get it from internet, if free)
What is the best approach to incorporate it? [audioplayback, openal etc)
Forgive me if this is quite stupid, but I am going very generic here and can't seem to find it.
Thanks for the help!
For sound format, use AAC or uncompressed 16-bit little endian in a CAF container (avoid mp3 since it's difficult to make it loop cleanly). You can convert using the command line tool 'afconvert':
Compressed:
afconvert -f caff -d aac sourcefile.wav destfile.caf
Uncompressed 16-bit:
afconvert -f caff -d LEI16 sourcefile.wav destfile.caf
For production, either record it yourself (using an audio program such as Audacity), get a professional to do it, or buy royalty free sounds/music.
To incorporate it, use AVAudioPlayer for music and OpenAL for sounds. OpenAL is difficult to use and doesn't decode compressed audio on its own, so you may want to use an audio library such as https://github.com/kstenerud/ObjectAL-for-iPhone
i am not that big on programming and i need to sample an audio file that is recorded via a mobile, i understand that the mobile device records the voice, and there are sample rates and each sample has a 8bit or 16 bit sample size. i need 8 bit sample size in 8khz. the programming environment is lua but that doesnt matter really the basic deal is sampling. Any help would be appreciated.
You need to either set your recording parameters at the time that you record the sound sample to 8 bits # 8 kHz, or, if you can't, or don't have any control over the actual recording process, then you will need to resample the original recording to get it into the format that you need. You can do this off-line with a tool such as SoX, or you can write code that does this on-the-fly. If you give more detail in your question then I can add more specific detail in my answer.
I don't know about lua, but as suggested by Paul R, you can always post-process your audio files using SoX or ffmpeg. On linux, just install ffmpeg :
sudo apt-get install ffmpeg
Then :
ffmpeg -i trenet.wav -ar 8000 trenet-8k.wav
This resamples your audio at 8000 Hz. I did not find how to deal with the bits/sample parameter but you can always tune the bitrate. For documentation :
man ffmpeg