samplerate | small wrapper around libsamplerate

I made a program that's supposed to record people in my household talking for 1 minute. I think my code has successfully (though messily) been able to save the *.wav file and classifying the recording on gender. The male recordings are supposed to be saved in the male_voices folder and the female recording are supposed to be saved in the female_voices folder.

My question is: I have searched and couldn't seem to find a way to save these recordings to a specific file path. As you can see I tried using

I am currently trying to make a .wav file that will play sos in morse.

The way I went about this is: I have a byte array that contains one wave of a beep. I then repeated that until I had the desired length. After that I inserted those bytes into a new array and put bytes containing 00 (in hexadecimal) to separate the beeps.

If I add 1 beep to a WAVE file, it creates the file correctly (i.e. I get a beep of the desired length). Here is a picture of the waves zoomed in (I opened the file in Audacity): And here is a picture of the entire wave part:

The problem now is that when I add a second beep, the second one becomes completely distorted: So this is what the entire file looks like now:

If I add another beep, it will be the correct beep again, If I add yet another beep it's going to be distorted again, etc. So basically, every other wave is distorted.

Does anyone know why this happens?

Here is a link to a .txt file I generated containing the the audio data of the wave file I created: byteTest19.txt

And here is a lint to a .txt file that I generated using file format.info that is a hexadecimal representation of the bytes in the .wav file I generated containing 5 beeps (with two of them, the even beeps being distorted): test3.txt

You can tell when a new beep starts because it is preceded by a lot of 00's.

As far as I can see, the bytes of the second beep does not differ from the first one, which is why I am asking this question.

If anyone knows why this happens, please help me. If you need more information, don't hesitate to ask. I hope I explained well what I'm doing, if not, that's my bad.

EDIT Here is my code:

The following Azure Resource Manager Template fails validation, Visual Studio Code indicates:

1. Template validation failed: Invalid property identifier character: {. Path 'resources[6].identity', line 311, position 8.
2. Missing member name. arm-template (syntax) [311, 9]
3. The object is unclosed, '}' expected. arm-template -syntax) [319, 10]
4. Missing required property "xmlCfg" arm-template (schema) [51, 13]
5. Missing required property "properties" arm-template (schema) [258, 9]

All.

I found this script which is rather promising for what I am trying to do.
It works great with the hardcoded XML script.
However, I need to parse the file itself from an external URL.

Thank You.

Contents of the example file.

I'm using AVFoundation capture session to output audio buffers through AVCaptureAudioDataOutput. The capture session is using the default application audio session. (ie. captureSession.usesApplicationAudioSession = true). I don't alter the audio session in any way.

The strange behavior is that the capture session returns audio buffers captured at a different frequency than the default audio session's sample rate.

Specifically:

I am currently working on a project for which I am trying to use Deepspeech on a raspberry pi while using microphone audio, but I keep getting an Invalid Sample rate error. Using pyAudio I create a stream which uses the sample rate the model wants, which is 16000, but the microphone I am using has a sample rate of 44100. When running the python script no rate conversion is done and the microphones sample rate and the expected sample rate of the model produce an Invalid Sample Rate error.

The microphone info is listed like this by pyaudio:

I want to make Siren sound which's frequency changing 960Hz and 770Hz every 0.65sec. (in 8sec Wav file) But I have no idea how to build function as I write above. I tried to use 'for(...=0; ... < 0.65; ...++)' every period. But y[0] and y[1] are function, so I'm confused. My final goal is to make siren wav sound, which come from right side to left side.

To say to the point, I want to know how to make frequency changeable 960Hz and 770Hz every 0.65 sec. I'll be thankful to you if you give me advice to achieve my final goal.

As I'm not good at English, if you're hard to understand my Question, plz comment me.

I would like to create an acoustic fingerprint of a FLAC or MP3 file using the chromaprint library in Go. I've been playing around with the following two Go libraries:

• gochroma, Go bindings for chromaprint
• fingerprint

Using the following code, a fingerprint of a "raw audio data stream" can be created (where reader is of type io.Reader):

I'm downloading via ffmpeg a mp3 hosted on some data source: