Try to nAudio read the wave file wile you record it - real-time

as I wrote in the title I need to read the wave file, from an external application, while it is recording it. I noticed that until I use StopRecording () and Dispose ()
the wave file remains locked and the external application tells me the file is locked for reading
Do you have any suggestions on how I can do it?
Thank you
Mimmo
I should read the file as if it were some kind of real time

That's really hard to do successfully. One trick I've used in the past is to create a large blank WAV file and have one process overwrite it in advance of the process that is reading.
But generally I'd recommend using something like a BufferedWaveProvider to supply audio for playback in a streaming scenario.

Related

Download Result of Source Destination (Web Audio API)

I'm building a tool to edit audio with the Web Audio API.
Here is where I'm stuck:
...
source
.connect(gainNode)
.connect(analyser)
.connect(analyser2)
.connect(audioCtx.destination);
};
What I've written (which ends with the code above) successfully allows the user to upload a file, apply effects, and listen to it on play(). How would I then allow the user to click a button to export the results to a WAV file?
I've tried several methods online that have not worked for my use case.
Please let me know if more code is needed. Thank you for taking a look!
If you want a WAV file, I think you have to do that yourself. WAV files are quite simple. In this case, you'll need to add either a ScriptProcessorNode or AudioWorkletNode just before the destination to capture all the audio and convert it to a WAV file that can be downloaded.
If a compressed file is ok, you can look into MediaRecorder to save the data for you.
I ended up solving this by writing an entirely different script to download the file using OfflineAudioContext.
My original script plays the audio with effects, and the second script downloads it with the same effects. Now to figure out why there is latency on the effects while using OfflineAudioContext.

how to record the voip call using sipsorcery sdk?

I am using sample programs provided by sipsorcery:
https://github.com/sipsorcery/sipsorcery/tree/master/sipsorcery-softphonev2
What I want to record the call or record the part of one side spoken text, process it, then generate the answer test and speak it back.
What I need right now to process the spoken text. I wanted to record the parts of call and save them to a wav file and generate text from it. but it seems to me that I am doing wrong. I am not able to generate the correct wav file using the provided method of sipsorcery SDK.
I have tried to follow the example on this forum as well, but it didn't work
https://markheath.net/post/how-to-record-and-play-audio-at-same
I expect that this should work using a small temporary wave file at each time the user speaks a sentence and response back again playing back the processed response file.
Any guidance how can I achieve this sense of interception and processing of the call?
Thanks,
Vivek
This example should be pretty close to what you need. It plays the audio (only ulaw support) via the default speaker using NAudio. To record it should be a matter of switching from using NAudio playback to saving to a wav file.

How to change aurioTouch project to write recordings to .caff or .wav file

I'm investigating aurioTouch2 sample code.
I wanna change smth to record audio from microphone and to write these data to .caff or .wav file. So later I will be able to play this audio with AVAudioPlayer.
My idea is to use AVAudioRecorder in applicationDidFinishLaunching, but may be it's not the best solution or may be even it's not possible by this way (I've not tried to use AVAudioRecorder, because I think that most likely it's not the best idea).
I've tried to write byte data in
void FFTBufferManager::GrabAudioData(AudioBufferList *inBL)
{
memcpy(mAudioBuffer+mAudioBufferCurrentIndex, inBL->mBuffers[0].mData, bytesToCopy);
// after this I copy to my own buffer and collect all the data during 30 seconds
}
So as you see, I copy to my own buffer and collect all the data during 30 seconds. Than I write data to .caff file. That doesn't work.
Update:
Or may be the way of playing binary data of audio (not encoded to .waf or .caff audio) exist?
There's a few different ways to do this. One approach is to use ExtAudioFileWriteAsync.
Apple provide a sample project that demonstrates how to do this. It's RecordAudioToFile.
Have a look at some other posts as well, e.g. Can anybody help me in recording iPhone output sound through Audio Unit
I am not sure if I got you correctly but you can't just write the buffer to disk. CAF and WAV files need their own headers and encoding of the audio data. The AVRecorder class does this for you. There are plenty of questions on this topic. See for example this question.

Record audio, add effects, then save result to a audio file

I am having trouble doing what the title said. My goal is to be able to add any desired effects to your recording, save the modified audio, then send that to a server.
I have searched the fourms and came across these threads:
viewtopic.php?f=7&t=13029&p=45362&hilit=saving#p45362
viewtopic.php?f=7&t=12660&p=44586&hilit=saving#p44586
viewtopic.php?f=7&t=13178&p=45746&hilit=saving#p45746
After reading those, I see it is possible to save the modified audio, but can it only be saved as a wav? Like I said after it is saved it will be sent to a server, so size is a big deal and wavs are relatively big compared to other formats. Ignoring that fact, I tried to implement FMOD_OUTPUTTYPE_WAVWRITER and I cannot get that to work; are there any good examples of using it? I looked though the examples in the library but I didn't see any..
But the basic structure of the app is to record, turn some switches off and on to see what filters you want, preview it, then press a button "Save" that will save it. What would this save function consist of?
Any help appreciated, thanks.
Using FMOD_OUTPUTTYPE_WAVWRITER is fairly straight forward, you set the type via System::setOutput, specify the output file via System::init extradriverdata. The extradriverdata should be an absolute path to a writable area of the device such as the documents directory. After you have finished playing, call System::release and the file will be complete.
The other option for recording wave data with effects is by creating a custom DSP and connecting it to the channel playing the recorded data. You will then get regular callbacks giving you float data that you must write out to disk yourself. You can find examples of DSPs and writing wav files in the dsp_custom and recordtodisk examples respectively.
Finally note that FMOD doesn't come with the facility to write compressed audio to disk, you will need another API to achieve this goal.
You can save as an AAC file via the ExtAudioFile API.

Artifact on playing a audio file on a phone from a Asterisk server

I have an Asterisk SIP server. When I playback an audio file (.ulaw file, compressed using ulaw) I hear a noticeable click (or sound artifact) before the playback begins. This "click" is not in the actual audio file and happens at the start of every Playback command in the ael script. Should I be using a different format, is this a codec issue, how do I resolve this issue?
Here are some of my files:
http://kscserver.com/hello.zip
http://kscserver.com/thankyou.zip
Without looking at the file, it's hard to say, but if the first sample of the file starts at some value other than 0, you may get a click (since the output will go from 0 to N in one sample - a broad noise impulse). If you don't know a sample starts "clean" it can make sense to ramp it in volume-wise, or search the uncompressed data for a zero-crossing and start there.