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How to write C++ audio processing applications?

By : Ralitsa Ivanova
Date : October 14 2020, 01:41 PM
Does that help Checkout this link which will give you some information on the available (commercial and open source) audio editing softwares.
Some interesting open source audio editing tools which are written in c++,
code :

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Is there a suitable replacement for C++, when I would like to write video processing applications?

By : jasontsect
Date : March 29 2020, 07:55 AM
wish help you to fix your issue I can't speak to Lisp, but both Erlang and Haskell are capable of the performance necessary for video processing. Achieving that performance is likely to be more difficult than with C++ because there are fewer existing libraries in the domain, so you'll have to implement more yourself. Which means you'll have to be capable of writing high-performance code yourself. In Haskell I expect this would require a significant investment of time (6 months minimum) to become proficient.
Which language you choose should depend a great deal upon the goals of the project. If it's a hobby project, or you want to learn a lot about processing algorithms (and therefore don't mind having to do a lot of low-level coding yourself), there's nothing wrong with using an out-of-mainstream language. Haskell has bindings to a lot of things you would probably want to use eventually, such as a wrapper for GLSL.

Simplest way to capture raw audio from audio input for real time processing on a mac

By : user3084043
Date : March 29 2020, 07:55 AM
I hope this helps you . Try the AudioQueue Framework for this. You mainly have to perform 3 steps:
setup an audio format how to sample the incoming analog audio start a new recording AudioQueue with AudioQueueNewInput() Register a callback routine which handles the incoming audio data packages
code :
static void HandleAudioCallback (void                               *aqData,
                                 AudioQueueRef                      inAQ,
                                 AudioQueueBufferRef                inBuffer, 
                                 const AudioTimeStamp               *inStartTime, 
                                 UInt32                             inNumPackets, 
                                 const AudioStreamPacketDescription *inPacketDesc) {
    // Here you examine your audio data

static void StartRecording() {
    // now let's start the recording
    AudioQueueNewInput (&aqData.mDataFormat,  // The sampling format how to record
                        HandleAudioCallback,  // Your callback routine
                        &aqData,              // e.g. AudioStreamBasicDescription
                        &aqData.mQueue);      // Your fresh created AudioQueue

How can I port audio of specific applications audio on Windows?

By : Fumy
Date : March 29 2020, 07:55 AM
wish of those help I personally use a combo of Virtual Audio Cable and Ableton Live 9 to capture audio from all kinds of things that play through my speakers. I will gladly explain how I do it. (I am using Windows 7.)
Here's how:

OSX Audio Hijack style audio recording from other applications (cocoa)

By : user3777423
Date : March 29 2020, 07:55 AM
should help you out I think you should take a look at the Jack OS X source. It has a CoreAudio user space driver that allows you to send any application's sound output to Jack OS X and support for audio plugins.

AVAssetWriter real-time processing audio from file and audio from AVCaptureSession

By : kusha
Date : March 29 2020, 07:55 AM
wish of those help I discovered that the reason for this is:
I correctly use the Presentation Time Stamp of the incoming capture session data (I use the PTS of the video data at the moment) to begin a writer session (startSessionAtSourceTime), and that meant that the timestamp of the audio data read from file had the wrong timestamp - outwith the time range that was dictated to the AVAssetWriter session. So I had to further process the data from the audio file, changing its timing information by using CMSampleBufferCreateCopyWithNewTiming.
code :
CMTime bufferDuration = CMSampleBufferGetOutputDuration(nextBuffer);
CMSampleBufferRef timeAdjustedBuffer;
CMSampleTimingInfo timingInfo;
timingInfo.duration = bufferDuration;
timingInfo.presentationTimeStamp = _presentationTimeUsedToStartSession;
timingInfo.decodeTimeStamp = kCMTimeInvalid;

CMSampleBufferCreateCopyWithNewTiming(kCFAllocatorDefault, nextBuffer, 1, &timingInfo, &timeAdjustedBuffer);
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