codec preference optimization

[krishna mohan]

Hello Arsen,

My ASR expects 16bit,16khz wav data from unimrcp.

I am following below procedure for codec. 

Freeswitch:

Source->[LPCM/8000/1]->Bridge->[LPCM/8000/1]->Encoder->[PCMU/8000/1]->Sink

UniMRCP:

[PCMU/8000]->ulawtolinear(g711)->resampling from 8kHz to 16kHz

ASR:

16khz wav data

1) Can it be optimized further? 

2) Sometimes transcription for the data transferred from freeswitch to ASR via unimrcp doesnt seem perfect.

Do you feel anything wrong in my codec conversions?

[Arsen Chaloyan]

Hello Krishna,

There is nothing wrong in your media flow. Needless to say, though, that 16kHz audio data can be streamed directly from FreeSWITCH, of course, if the incoming call has the same capabilities.

If you encounter problems, then I'd suggest to first make a network capture and analyze RTP streams received from FS. The source of the problem can also be in your resampler.

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Arsen Chaloyan
Author of UniMRCP
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[krishna mohan]

I think incoming call doesnt have 16khz support.

For resampling i am using sox. Do you have any other suggestion?

Also i have one more doubt regarding log captured in unimrcp server.

Source->[PCMU/8000/1]->Decoder-> 
[LPCM/8000/1]->Bridge->[LPCM/8000/1]->Sink 

End output  LPCM/8000/1 is 16bit linear PCM or 8 bit linear PCM?

Thank you.

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[Arsen Chaloyan]

> End output  LPCM/8000/1 is 16bit linear PCM or 8 bit linear PCM?

16-bit linear PCM

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[krishna mohan]

Hello Arsen,

I am using soxr library for resampling from 8kHz to 16kHz.

Do you suggest any other resampler?

On Saturday, 10 August 2019 03:11:48 UTC+5:30, Arsen Chaloyan wrote:

Hello Krishna,

There is nothing wrong in your media flow. Needless to say, though, that 16kHz audio data can be streamed directly from FreeSWITCH, of course, if the incoming call has the same capabilities.

If you encounter problems, then I'd suggest to first make a network capture and analyze RTP streams received from FS. The source of the problem can also be in your resampler.

On Thu, Jul 25, 2019 at 2:36 AM krishna mohan <krishna...@gmail.com> wrote:

Hello Arsen,

My ASR expects 16bit,16khz wav data from unimrcp.

I am following below procedure for codec. 

Freeswitch:

Source->[LPCM/8000/1]->Bridge->[LPCM/8000/1]->Encoder->[PCMU/8000/1]->Sink

UniMRCP:

[PCMU/8000]->ulawtolinear(g711)->resampling from 8kHz to 16kHz

ASR:

16khz wav data

1) Can it be optimized further? 

2) Sometimes transcription for the data transferred from freeswitch to ASR via unimrcp doesnt seem perfect.

Do you feel anything wrong in my codec conversions?

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[Arsen Chaloyan]

I've never used sox programmatically, but as a command-line utility only. There are many alternate re-samplers available. I am not in a position to suggest a particular one.

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