My iOS project requires to retrieve some audio data (i.e. frequency , decibel) from an audio file.
By using AudioKit framework, I can get those data from the microphone by use AKFrequencyTracker, however, I am struggling on how to get the frequency straight away from the audio file without playing it. Because I need those data to plot some graphs (i.e. frequency vs. time, etc)
PS: I'm saving the record as an m4a format at the moment. (the format is optional)
Thanks in advance
You can use Accelerate framework FFT API's to get the frequency information from an audio file.
Here is a useful library to understand vDSP API usage.
https://github.com/tomer8007/real-time-audio-fft
Related
I need to extract waveform data asynchronously from a file after having applied some equalization. Using the AVAudioEngine I encounter in random silences at the beginning of the samples (it's more of a delay of the real data but unfortunately such delay can't be detected for instance with a longer number of samples).
I asked directly to Apple using the code support tickets but they don't have a solution. Looks like AVAudioEngine offline rendering mode is bugged.
Looking for third party solutions, I came across AudioKit which seems to be using AVAudioEngine. So, assuming AudioKit can extract asynchronously a waveform after EQ is applied, would it do it bypassing AVAudioEngine offline mode?
Is there a known way to convert an AVAudioRecorder object into an AKNode object?
My use case for this is that I have an application that is pulling an audio stream from a custom piece of bluetooth hardware. I've already written all the handlers for this, and the output of the hardware ends up as an AVAudioRecorder.
I'd like to make use of all the nicer visualisation of audio that AK offers - specifically the plotting of amplitude on a graph in my view as it is recorded, but to get it to work, it appears that I need to turn the AVAudioRecorder into an AKNode.
Is there an easy way to do this without going back through all the code that interfaces with the hardware and replacing it to use AKNode from the start?
I have gone through the documentation of AK and it doesn't seem possible at this time to use an existing AVAudioRecorder as a source node.
Thanks!
I don't believe so. AVAudioPlayer is also unavailable, but we do have AVAudioPlayerNode. There is not a corresponding AVAudioRecorderNode.
Statement of Problem:
I have a collection of sound effects in my app stored as.m4a files (AAC format, 48 KHz, 16-bit) that I want to play at a variety of speeds and pitches, without having to pre-generate all the variants as separate files.
Although the .rate property of an AVAudioPlayer object can alter playback speed, it always maintains the original pitch, which is not what I want. Instead, I simply want to play the sound sample faster or slower and have the pitch go up or down to match — just like speeding up or slowing down an old-fashioned reel-to-reel tape recorder. In other words, I need some way to essentially alter the audio sample rate by amounts like +2 semitones (12% faster), –5 semitones (33% slower), +12 semitones (2x faster), etc.
Question:
Is there some way fetch the Linear PCM audio data from an AVAudioPlayer object, apply sample rate conversion using a different iOS framework, and stuff the resulting audio data into a new AVAudioPlayer object, which can then be played normally?
Possible avenues:
I was reading up on AudioConverterConvertComplexBuffer. In particular kAudioConverterSampleRateConverterComplexity_Mastering, and kAudioConverterQuality_Max, and AudioConverterFillComplexBuffer() caught my eye. So it looks possible with this audio conversion framework. Is this an avenue I should explore further?
Requirements:
I actually don't need playback to begin instantly. If sample rate conversion incurs a slight delay, that's fine. All of my samples are 4 seconds or less, so I would imagine that any on-the-fly resampling would occur quickly, on the order of 1/10 second or less. (More than 1/2 would be too much, though.)
I'd really rather not get into heavyweight stuff like OpenAL or Core Audio if there is a simpler way to do this using a conversion framework provided by iOS. However, if there is a simple solution to this problem using OpenAL or Core Audio, I'd be happy to consider that. By "simple" I mean something that can be implemented in 50–100 lines of code and doesn't require starting up additional threads to feed data to the a sound device. I'd rather just have everything taken care of automatically — which is why I'm willing to convert the audio clip prior to playing.
I want to avoid any third-party libraries here, because this isn't rocket science and I know it must be possible with native iOS frameworks somehow.
Again, I need to adjust the pitch and playback rate together, not separately. So if playback is slowed down 2x, a human voice would become very deep and slow-spoken. And if playback is sped up 2–3x, a human voice would sound like a fast-talking chipmunk. In other words, I absolutely do not want to alter the pitch while keeping the audio duration the same, because that operation results in an undesirably "tinny" sound when bending the pitch upward more than a couple semitones. I just want to speed the whole thing up and have the pitch go up as a natural side-effect, just like old-fashioned tape recorders used to do.
Needs to work in iOS 6 and up, although iOS 5 support would be a nice bonus.
The forum link Jack Wu mentions has one suggestion, which involves overriding the AIFF header data directly. This may work, but you will need to have AIFF files since it relies on a specific range of the AIFF header to write into. This also needs to be done before you create the AVAudioPlayer, which means that you can't modify the pitch once it is running.
If you are willing to go to the AudioUnits route, a complete simple solution is probably ~200 lines (note that this assumes the code style that has one function take up to 7 lines with one parameter on each line). There is an Varispeed AudioUnit, which does exactly what you want by locking pitch to rate. You would basically need to look at the API, docs and some sample AudioUnit code to get familiar and then:
create/init the audio graph and stream format (~100 lines)
create and add to the graph a RemoteIO AudioUnit (kAudioUnitSubType_RemoteIO) (this outputs to the speaker)
create and add a varispeed unit, and connect the output of the varispeed unit (kAudioUnitSubType_Varispeed) to the input of the RemoteIO Unit
create and add to the graph a AudioFilePlayer (kAudioUnitSubType_AudioFilePlayer) unit to read the file and connect it to the varispeed unit
start the graph to begin playback
when you want to change the pitch, do it via AudioUnitSetParameter, and the pitch and playback rate change will take effect while playing
Note that there is a TimePitch audio unit which allows independent control of pitch and rate, as well.
For iOS 7, you'd want to look at AVPlayerItem's time-pitch algorithm (audioTimePitchAlgorithm) called AVAudioTimePitchAlgorithmVarispeed. Unfortunately this feature is not available on early systems.
guys.
I'm working on some audio services on iOS.
I trying to search any examples or tutorials about
how audio service or stream can read a existing audio file than
process something like filter, than write another file.
Is there any body who can help me?
Dirac3LE (by Stephan M. Bernsee) is a great library for this job.
There are examples and manual included in the download.
It is particulary inteded for time and pitch manipulation
but in your case you'll be interested in its EAFRead and EAFWrite
classes.
If you want to get familiar with the lower level library that you can also use for microphone input/sound output, and that you can get raw samples into and out of, I would suggest taking a look at Audio Queue Services.
I used it in my side project to get audio from the microphone, and I also wrote some code you might find useful to do fast vectorized, FFT based FIR filtering on input audio. You can find the code here https://github.com/jamescarlson/FreeAPRS
What I'm trying to do is exactly as the title says, decode multiple compressed audio streams/files - it will be extracted from a modified MP4 file - and do EQ on them in realtime simultaneously.
I have read through most of Apple's docs.
I have tried AudioQueues, but I won't be able to do equalization, as once the compressed audio goes in, it doesn't come out ... so I can't manipulate it.
Audio Units don't seem to have any components to handle decompression of AAC and MP3 - if I'm right it's converter only handles converting from one LPCM format to another.
I have been trying to work out a solution on and off for about a month and a half now.
I'm now thinking, use a 3rd party decoder (god help me; I haven't a clue how to use those, the source code is greek; oh and any recommendations? :x), then feed the decoded-to LPCM into AudioQueues doing EQ at the callback.
Maybe I'm missing something here. Suggestions? :(
I'm still trying to figure out Core Audio for my own needs, but from what I can understand, you want to use Extended Audio File Services which handles reading and compression for you, producing PCM data you can then hand off to a buffer. The MixerHost sample project provides an example of using ExtAudioFileOpenURL to do this.