[DrBoomBotz]

I here folks saying that bbm voice sound quality is very good. What codec is bb using?

[diegonei]

We have no idea. I don't think BlackBerry has ever disclosed that info.

But it is as clear as it gets. Beats Skype if you ask me.

[DrBoomBotz]

a wire capture would be educational but I don't have a bb10 device.
I do have a playbook. I wonder if a bbm voice call to a playbook would also use an hd codec. If you would like to make a test pm me?

[bloodyrek]

Actually from experience when i last used it, i forgot i was using BBM Voice.

Top notch quality from Blackberry.

[DrBoomBotz]

does that mean that you thought you were jist making a normal call?

[DrBoomBotz]

Actually from experience when i last used it, i forgot i was using BBM Voice.

Top notch quality from Blackberry.

Your not saying you thought the other person was in the same room?

Sent from my Galaxy Nexus using Tapatalk

[diegonei]

a wire capture would be educational but I don't have a bb10 device.
I do have a playbook. I wonder if a bbm voice call to a playbook would also use an hd codec. If you would like to make a test pm me?

If you have ever used the PlayBook to make a voice-only call, then you know how the quality is: That good.

[DrBoomBotz]

I have only done one video chat for less than a minute in a loud room. So I don't know actually. I wanted to get a wire trace to see what goes on behind the curtain.

Sent from my Galaxy Nexus using Tapatalk

[castano22]

It's really good better than calling normally same goes with bbm voice. I like it way more than Skype

Posted via CB10

[DrBoomBotz]

So I made a voice call playbook to playbook and capured the results.

First off subjectively the call sounded fine to me, better than toll quality I think.
I noticed that the playbook works great as a speaker phone.

Analysis shows
stun is used (to blackberry infrastructure)
signalling is tls encrypted (to blackberry infrastructure)

media goes directly from playbook to playbook.
rtp payload is 114(dynamic)
payload size is 129 bytes 183 on the wire (does not appear to be dynamic).
interval is 20ms
rtcp sender report interval is 1 second

In summary I still don't know what codec the playbook uses and can only guess it is the same codec that bbm voice uses.

[wout000]

Skype provides 'MONO audio' only through it's SILK codec which is a proprietary codec they created back in 2009.
They are gearing up for a change of codec to OPUS as it utilizes a heavily modified portion of SILK and CELT, effectively getting ultra-wide band without the delays. They are a partner in creating the OPUS codec so it will be provided by Skype in the foreseeable future. Last word is that they have already incorporated the codec in their systems but haven't released a client yet that takes advantage of it.

BBM Voice is 'STEREO audio' and operates on frequencies in the wide-band providing a more crisp and clear sound. The downside of this is that they use more computing power/bandwidth, something they can afford to do as they already have the infrastructure in place and BIS users are going down, freeing up more power.

On the type of codec being used, I can't tell you as BlackBerry never revealed any of that information. Most of the speculation points towards the AMR-WB codec being used as this would provide for significant clear and crisp voice calls.

I'm hoping they will eventually switch to the OPUS codec as it is a more robust codec that supports every range of the bitrate spectrum meaning a drop in bandwidth would be picked up on the fly without any noticeable differences. As an added bonus, it has between 0-20 ms delay time compared to 80 ms delay for the AMR-WB codec which only operates at the middle level of bandwitdth frequency.
As an added bonus, the OPUS codec is an all-rounder. It can be used for storage (music files) and it's better than MP3, VOIP (BBMV), Low Latency speech (Voicenotes, calls) and it's an internet standard, under active development all royalty free.

Some info about OPUS:
Opus (audio codec) - Wikipedia, the free encyclopedia
Unfortunately the big companies want their own...Why OPUS isn't the defacto standard
If you want to see some technical information and hear some comparison samples:

Comparison at 28:00
Going from low to high bitrate without glitches: 30:00

[DrBoomBotz]

Is the stereo audio sent as interleaved samples?
Funny thing about the super wide band codecs is that some people perceive the quality to be worse than toll quality. I too would like to see opus get some traction.

[wout000]

Is the stereo audio sent as interleaved samples?

I was going to give start an elaborate speech, but I realized I couldn't have said it better than this:

A pulse code modulated (PCM) signal consists of a stream of samples; if the type is monaural, each sample will belong to one channel. If there is more than one channel, the channels will almost always be interleaved. In the case of stereo data: left sample, right sample, left, right, etc.
Source: Interleaved sample audio - Glossary - Federal Agencies Digitization Guidelines Initiative

So yes, it's being sent as interleaved samples.

Funny thing about the super wide band codecs is that some people perceive the quality to be worse than toll quality. I too would like to see opus get some traction.

Those people are wrong.
Some super wide bands give you access to the full audio spectrum range up to 20 khz which is the maximum a human ear can detect. So it really is as if the person is standing next to you.

The standard used on cellphones now is 4khz, the clear audio from BBM Voice is bumped up to 7 khz. Imagine what 20 khz will do