Other PCM Audio compression And decompression Paid

[Addo]

i am looking for Audio Compression / decompression encoding that effectively reduce the size of the AudioStreamer Packet

i am using a sample rate of 32000 and encoding of 16 bit

the current internet usage per second is 36 kbps /s download an upload

i am looking for a compression method that can reduce this size to 3 or 6 kpbs /s

when client side send that packet compress it and when its recived decompress it with its original size

the module is needed to be compatible with delphi and b4x

i will pay 500 USD for a version that works with delphi for compression / decompression and for b4a as well

 

[Midimaster]

There is no way to compress 1:10 without loosing quality. So you have to think about, what you want to send. Music? human Voice? Mono? HiFi necessary? This will decide the sample rate, bit-depth and transfer rate the algorith you should use.

MP3 is still the best solution. You can scale the Transferrate as you want. You should find a library that encodes/decodes MP3 from/to streams

 

[Addo]

I am sending a mono pcm format to send an audio directly from microphone to the server.

I already used Alaw / ulaw by @OliverA it compressed pretty fine but not as i needed it..
Thats why i make this offer to find a suitable compression as i need

 

[Midimaster]

some ideas:

A-Law and U-Law are already compress algos for the communication sector like GSM6.1 or GSM6.6. Have a look at them.

I still do not know, what source you are recording and how important is the quality of the signal. You could try to reduce sampling rate to 8000Hz. This will cost some high frequencies, but still good for communication. Perhaps this will shrink the rate to 9kbps.

Check also the quality if you use MP3 with the minimum feature "8kbps".

What is your app doing when the speaker says nothing for a moment? You could add a SILENCE recognition

are you interested in reducing the peak amout of kbps or the average amout of kbps? How important is real time performance for you? You could add a "bootleneck" in your stream and delay transportation. This will reduce bandwidth and the end the listener will hear all perfect, but with a delay of some seconds.

 

[moster67]

Maybe Opus, which has now become a standard, can help?
https://en.m.wikipedia.org/wiki/Opus_(audio_format)

I will send you a PM if you are interested.