My aim is code a project which records human sound and changes it (with effects).
e.g : a person will record its sound over microphone (speak for a while) and than the program makes its like a baby sound.
This shall run effectively and fast (while recording the altering operation must run, too)
What is the optimum way to do it ?
Thanks
If you're looking for either XNA or DirectX to do this for you, I'm pretty sure you're going to be out of luck (I don't have much experience with DirectSound; maybe somebody can correct me). What it sounds like you want to do is realtime digital signal processing, which means that you're either going to need to write your own code to manipulate the raw waveform, or find somebody else who's already written the code for you.
If you don't have experience writing this sort of thing, it's probably best to use somebody else's signal processing library, because this sort of thing can quickly get complicated. Since you're developing for the PC, you're in luck; you can use any library you like using P/Invoke. You might try out some of the solutions suggested here and here.
MSDN has some info about the Audio namespace from XNA, and the audio recording introduced in version 4:
Working with Microphones
Recording Audio from a Microphone
Keep in mind that recorded data is returned in PCM format.
Related
I'm trying to put together an open source library that allows iOS devices to play files with unsupported containers, as long as the track formats/codecs are supported. e.g.: a Matroska video (MKV) file with an H264 video track and an AAC audio track. I'm making an app that surely could use that functionality and I bet there are many more out there that would benefit from it. Any help you can give (by commenting here or—even better— collaborating with me) is much appreciated. This is where I'm at so far:
I did a bit of research trying to find out how players like AVPlayerHD or Infuse can play non-standard containers and still have hardware acceleration. It seems like they transcode small chunks of the whole video file and play those in sequence instead.
It's a good solution. But if you want to throw that video to an Apple TV, things don't work as planned since the video is actually a bunch of smaller chunks being played as a playlist. This site has way more info, but at its core streaming to Apple TV is essentially a progressive download of the MP4/MPV file being played.
I'm thinking a sort of streaming proxy is the way to go. For the playing side of things, I've been investigating AVSampleBufferDisplayLayer (more info here) as a way of playing the video track. I haven't gotten to audio yet. Things get interesting when you think about the AirPlay side of things: by having a "container proxy", we can make any file look like it has the right container without the file size implications of transcoding.
It seems like GStreamer might be a good starting point for the proxy. I need to read up on it; I've never used it before. Does this approach sound like a good one for a library that could be used for App Store apps?
Thanks!
Finally got some extra time to go over GStreamer. Especially this article about how it is already updated to use the hardware decoding provided by iOS 8. So no need to develop this; GStreamer seems to be the answer.
Thanks!
The 'chucked' solution is no longer necessary in iOS 8. You should simply set up a video decode session and pass in NALUs.
https://developer.apple.com/videos/wwdc/2014/#513
I have sound, video source file and I have to verify my program which open and play this file is work correctly.
I don't know how to verify file like this!
I think i should capture (sound/video) and then compare it to source file.
Till this time, I've searched on the internet but didn't get any solution.
This is going to be a real challange for you, I personally have never done this but hopefully I can provide you with some help to set you on your way...
First you need to know that robotframework is run on python so anything you will need to be in python or have python bindings so asking there may be a good start.
In terms of capturing sound I believe it would be eaiser to use a program with a api you can use, I found a document here of someone doing this, as to whether this is still correct I am not sure:
http://www.nektra.com/files/DirectSound_Capture_With_Deviare.pdf
For video capture try looking here:
https://www.youtube.com/watch?v=j344j34JBRs
Next would be stripping the video, seperating the audio and video frames and comparing them seperatly. For this you are going to need a video editor, audio comparison library and a tool for comparing images.
In terms of how this would work I dont know as I have never done this...
Why do you need to do this tho, is there not a better way of doing this? Does you application make the video? In which case could just doing some checks on frames, length, file size suffice? You need to provide for information.
This is a bit long for a comment but this answer is incomplete.
Let me know how you get on?
Hope this question makes some sense, I'm completely lost....
In my proto-app I'm recording micro input and saving it, and so far no problems at all.
I now need to access the buffer while I'm recording it in order to pass chunks of data to another class (written in C, not by me) that will do some analysis.
I spent the whole day browsing and reading, and looks like I need use Audio Queues in order to access the buffer.
The problem is that the syntax is C, and I don't understand it at all :)
So my questions are:
1) Is there any other way to achieve what I'm looking for? I don't need in-depth explanation, just some hints and I will browse my way through :) I'm asking because I'm not 100% sure that Audio Queues are the only way to go
2) Any good tutorial or example about Audio Queues? The aurioTouch tutorial by Apple wasn't very useful (again, I don't know C). I could bypass my problems in C by following a good tutorial that a noob like me can understand
Thanks a lot, and for any help you could offer.
Good question.
You can use code written by other people like:
Novocaine - pretty straightforward. (but there are some bugs, at least in older version I used ~ 6 months ago. Something with mono and stereo.)
Momu - quite a good thing in C++ (you need to use .mm extension for you files)
Those will save you time if you want some low level audio programming. Some basic skills in C still required though. Check out this guy. His explanations and enthusiasm are excellent.
With all mentioned above you can be ready in a 1-2 days of work carrying away good skills in C.
EDIT
Basically, everywhere you work with low-level audio you deal with a C array of numbers (represented like float *audioBuffer;) called audio samples. You cycle through it in a loop, do some operations, copy it, send somewhere, analyze.
To copy it you have to allocate space for it. Actual byte size of the buffer can be calculated like this: numberOfSamples*sizeof(type).
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
I'm developing a virtual instrument app for iOS and am trying to implement a recording function so that the app can record and playback the music the user makes with the instrument. I'm currently using the CocosDenshion sound engine (with a few of my own hacks involving fades etc) which is based on OpenAL. From my research on the net it seems I have two options:
Keep a record of the user's inputs (ie. which notes were played at what volume) so that the app can recreate the sound (but this cannot be shared/emailed).
Hack my own low-level sound engine using AudioUnits & specifically RemoteIO so that I manually mix all the sounds and populate the final output buffer by hand and hence can save said buffer to a file. This will be able to be shared by email etc.
I have implemented a RemoteIO callback for rendering the output buffer in the hope that it would give me previously played data in the buffer but alas the buffer is always all 00.
So my question is: is there an easier way to sniff/listen to what my app is sending to the speakers than my option 2 above?
Thanks in advance for your help!
I think you should use remoteIO, I had a similar project several months ago and wanted to avoid remoteIO and audio units as much as possible, but in the end, after I wrote tons of code and read lots of documentations from third party libraries (including cocosdenshion) I end up using audio units anyway. More than that, it's not that hard to set up and work with. If you however look for a library to do most of the work for you, you should look for one written a top of core audio not open al.
You might want to take a look at the AudioCopy framework. It does a lot of what you seem to be looking for, and will save you from potentially reinventing some wheels.