pcm | Processor Counter Monitor | Performance Testing library

I have been using FFmpeg Android for a music app I'm working on. I built a custom audio engine from stratch with C++ and FFmpeg and it works amazing and it fulfilled all my needs. However, Due to FFmpeg being Lgpl lisence, it seems to me after some researching it is not possible to use a lgpl lisence due to app stores policy. Im not a lawyer or have the money to hire a lawyer for a commercial advise. So I am thinking to replace ffmpeg with another audio decoder, processor library. I am planning to feed the custom decoded data to audio devices through Apples core audio library.

Here are my needs:

• Need to decode ogg files
• Need to encode pcm data as aac file
• Need to add post process FX to decoded data such as low pass filter etc

So what I am asking for is an answer to one of the following:

• Could FFmpeg really not be used in app store due to lgpl static linking issues? (I looked at the most famous apps that use FFmpeg on Android, all of them does not use FFmpeg on IOS)
• If I were to use another library for FFmpeg what is the best alternative to work with? Did anyone actually had experienced the same situation that I am in?

I also tried using AudioKit but it has a critical problem that does not meet with my requirement so I dropped it.

I am looking for advice here. Thanks!

I am using oboe library to make a music app. There I produce music by writing PCM float values to the given pointer. I rarely have underruns which I can hearwhich. I also verify this with the following oboe APIs:

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:

Is there a way to get the total number of pcm samples inside an ogg file containing opus audio using libogg and libopus?

Thanks,

My discord bot currently has a local recording feature that creates a file every time someone speaks. When the bot disconnects from the voice call (from !stoprecording), I need to manually run node merge.js in my terminal to merge all those files into 1 main PCM file. From there, I need to run another FFMPEG terminal command to convert the merged PCM file into an MP3. That's fine on it's own, but is was curious if there was any way I could automate that? Is there a way to execute terminal commands in the actual code itself. So that every time !stoprecording is ran, it would merge the file and convert to mp3 using FFMPEG?

I'm trying to figure out how to play a raw 8-bit (signed) waveform using Java, and struggling because it seems like every API example I find assumes media playback from a file. Here's what I have so far (shortened for clarity):

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 am trying to read correctly a WAVE file, PCM, mono, 16 bits (2 bytes per sample). I have managed to read the header. The problem is reading (writing) the data part.

As far as I understand the 16-bit samples in the data chunk are little-endian, and "split" into two chunks of 8 bits each. So for me a way to read the correct data should be:

1. Read file and put chunks into two differentint8_t variables (or a std::vector..)
2. In some way "join" these two variables to make a int16_t and being able to process it.

The problem is I have no idea on how to deal with the little-endianness and the fact that these samples aren't unsigned, so I can't use the << operator.

This is one of the test I've done, without success:

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.