#89 [Manufacturer driver] Echo on Xperia X10

[Anonymous]

Originally created by: mcampbel...@gmail.com
Originally owned by: r3gis...@gmail.com

This is an intermittent problem that occurs sometimes when dialing an extension that is not registered. 

csipsimple hangs on the 'Confirmed' dialing screen.  This seems to hang audio in the phone as well as hanging the dialer.  I have to restart the phone to restore functionality.

What version of the product are you using? On what operating system?
Sony Ericsson X10 (and X10 mini), Android 1.6, Csipsimple 12

Please provide any additional information below.
I have previously provided logs via email, but wanted to file an issue report to track it.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Yes, thanks, I forgot to open the issue here.

For reference, exactly same issue on Samsung Galaxy i7500.
I can reproduce this issue for each outgoing call with the X10 of a colleague.

Owner: r3gis.3R
Status: Accepted

[Anonymous]

Originally posted by: r3gis...@gmail.com

Sounds that there is the same issue on Galaxy S.

Summary: No audio with Xperia X10, Galaxy S i9000, Galaxy i7500

[Anonymous]

Originally posted by: r3gis...@gmail.com

Issue 93 has been merged into this issue.

[Anonymous]

Originally posted by: winfa...@gmail.com

I did verify My Galaxy S uses codec "GSM @8000" from (i)nfo function. I used csipsimple that version is 'build 0.00-12-04'.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Well a new version is available but... in fact there were already a easy solution in all previous versions to solve this issue :

Some devices (X10, galaxy i7500 and i9000) audio driver is not really stable with 16kHz audio streams. Setting Frequency (in Option > Media) to 8KHz will solve audio stack deadlocks.

To be absolutely sure everything is good now, a new version (that also ensure a better audio routing).

https://code.google.com/p/csipsimple/downloads/detail?name=CSipSimple_0.00-12-05.apk.

Install it. Ensure that Options > Media > Frequency is set to 8kHz.
It's now the default value but if a previous version is installed 16kHz can still appear.

This analyze confirms that in some configuration there is no problem (when the codec in use force 8kHz audio stream - which can be different from 8kHz data stream).

Just a last point : on Sony Ericson has a conceptual point of view of the MICRO source. It appears that micro stream include audio playback line !
I played all the day long with an X10 (great thanks goes to a colleague who lent it). Thanks to these test I understood that 16kHz audio pledge the CPU in this implementation. Unfortunately I didn't find a work around to prevent the micro source to be mixed with audio output. The result is that you will ear clearly what your remote contact say....
But your remote contact will become crazy with it's own echo.

So question to the X10 owner, did you find a android sip application without this problem (remote user ear his voice)?

If no, we probably should open a bug for Sony Ericson devs. They will probably more focus this point if they implement 2.1 since changes in the 7 API clearly make a distinction on the mic source (mixed or not to output voice).

Status: Started

[Anonymous]

Originally posted by: winfa...@gmail.com

I tested some calls on csipsimple(0.00-12-05) using Galaxy S.

Caller used Galaxy S and callee used Bria(by counterpath) and Acrobits.
The first I did test call through OpenSIPS and second I tested through Asterisk.
I could verify the result is different.

On the OpenSIPS callee could never hear my voice, but on the Asterisk callee could hear some sound delayed and choppy. I don't know why.

You can verify detailed information in my zip file I attached
that contains three debug logs and two wave files.

I wish it is solved. If you need more information to solve I'll give you.
Thanks for your work.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Well many thanks for theses logs.

I will search more about the error that could explain the fact mic doesn't work well.
For reference :
ERROR/(2160): AFCCreateReSampler: avAFCInfo->bUsed[0] inSampleRate[44100] outSampleRate[8000] nChannel[2] outbitDepth[16]

There is only one google result for this one... on stackoverflow but no answer :(
Seems to be specific to Galaxy S.

On your side, can you try to do the same tests disabling both echo cancellation and voice auto detection.
(Uncheck Menu>Option>Media>Echo cancellation & Voice audio detection).
Another interesting test could be to enable PCMU and PCMA codecs and disable GSM codec: Menu>Option>Media>Codecs long clic on codec to enable/disable it.
According to provided traces, you have probably already disabled GSM but as your asterisk only support GSM, PCMU and PCMA, pjsip fall back to GSM to allow the call.

Thx again for your tests.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Another test you could do is changing media quality from 4 (the current default value) to 3.

On my HTC magic I get a choppy sound with quality 4 using GSM codec. But setting media quality to 3, disabling VAD and disabling echo cancellation make things going really better (from something that you can't ear to something pretty clear).

The other solution I have is to use PCMA instead of GSM.

Seems to be a little bit tricky to find out what are the good params and on what device :).
Probably we should try to start writing a wiki page.
We should probably also define what should be default values for audio parameters. Technically not a problem to define different params for different phones.

[Anonymous]

Originally posted by: winfa...@gmail.com

Thanks for your response.
I tested according to the above, but it's still the same as before.

The first I set both echo cancellation and VAD disabled.
Second, I set the available codecs that
1) enabled GSM, ulaw, alaw.
    => csipsimple (GSM 8KHz) <---------------> Bria (G711u 8KHz)
    => csipsimple (GSM 8KHz) <---------------> Acrobits (ulaw)
2) enabled just ualw and alaw. It was the same as 1).
    => csipsimple (GSM 8KHz) <---------------> Bria (G711u 8KHz)
    => csipsimple (GSM 8KHz) <---------------> Acrobits (ulaw)

If you want logs I'll give you. Thanks.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Did you try with Media quality set to 3 instead of 4? On my HTC Sapphire it's the only way I have to get GSM codec working properly.

There is a problem with the codec disabling feature. That was already evoked in another issue. I'll have a closer look on it too.

[Anonymous]

Originally posted by: winfa...@gmail.com

Yes, I did test on conditions that set quality to 3. The result is the same.

So I tried to test the 'fring' on my phone under the same conditions. It works well.
But I could find the same problem reports on the internet. It seems to be fixed recently. Now you can find it. But I could not find how to solve it.

[Anonymous]

Originally posted by: jar...@gmail.com

I have version 0.00-12 and Galaxy S. I am using custom SIP. Incoming voice works perfect but outgoing voice is useless. It is robotics, it buzz and it cutting words.

I can make some tests if you are going to fix this problem. It seems that this problems appear also with other sip app. Only application which works with Galaxy now is Fring but its voice quality is very poor. I think that CSipSimple has so far best voice quality if you can fix that outgoing voice issue.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Ok sounds to be an issue with the codec (in incoming call codec is chosen by caller and in outgoing csipsimple choose the codec and probably a codec that require a high cpu usage).
Today, I'll build a new version that should improve CPU usage. And maybe it will be intersting to test with it.
Besides, as it is probably an issue with the codec that is used, it would be interesting to deactivate some codecs such as ilbc or speex for example that seems to use a lot of extra cpu to encode/decode things. (This could be done with the old release and the new one)
I'll notify the thread today when the new release is available.

[Anonymous]

Originally posted by: jar...@gmail.com

Ok, I will test new version when it appear.

(Just to make sure, I mean above situation that I am caller so in that situation my voice is bad and what I hear is very good)

In my situation my SIP provide accepts these codecs:
- G711
- G729A
- G729AB
- G723
- G723A
- ILBC

Of course G729 would be the best. With symbian I have got excellent voice quality with iLBC.

I think Fring uses iLBC but that is terrible even if I compare Frings symbian and android versions. So far other applications in Android uses G711 (pcma, pcmu) without good result.

[Anonymous]

Originally posted by: ggpa...@gmail.com

I have the same issue, on my Galaxy S I am able to listen to the incomming call but the other party is listening garbage.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Ok, can also be an issue with micro stream too so. I know that Galaxy S produce a warning when I create the stream but I didn't succeed in finding why. And as I have no Galaxy S to test on, I can't debug it more precisely. For reference a link on stack overflow about this issue : http://stackoverflow.com/questions/3314760/what-the-error-message-afccreateresampler-avafcinfo-bused0-in-samplerate441/3316467 (as you can see, this is still an open issue !).

Well this said, maybe worth to try the latest build that consume less cpu. But I'm not confident on the fact it will solve the issue since really match the problem described on stack overflow and really seems to be linked to a bad audio driver from Samsung (maybe as it was reported to me, the latest rom from samsung for galaxy S solves things).
https://code.google.com/p/csipsimple/downloads/detail?name=CSipSimple_0.00-12-13.apk

[Anonymous]

Originally posted by: r3gis...@gmail.com

Just as reminder if using a version from the android market you must uninstall before installing the latest one.

[Anonymous]

Originally posted by: jar...@gmail.com

I tested that fast. I will test it better on evening with variant codecs and wlan and 3g. Do you want logs and how I should set logs option?

with that new version my voice is muted now so anybode cannot hear me. Also there appear issue with whole application. It autostart itselves and when I tried to exit, it autostart again. So finally it thinks that it is on but i cannot make calls.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Did you uninstall previously the older version before installing that one?

For logs, there is an option in Settings > UI and log level (at the bottom) Set it to 4. You can send me your logs if uninstalling/reinstalling (and maybe, why not, rebooting too - maybe it keeps the old native library in cache).

[Anonymous]

Originally posted by: jar...@gmail.com

Yes I uninstall and reinstall. Were I can found those logs? Which dir those are?

I tested know more with wlan but my voice doesn't go through.

I think that uninstall and reinstall again and make test again. Do you have any suggestion for settings? I tried iLBC, PCMU and PCMA codecs. But how should I set frequency and those others quality settings?

[Anonymous]

Originally posted by: r3gis...@gmail.com

Logs are available on android using a tool : logcat. There is a lot of application available on the market that will allow you to share these logs (for example search alogcat).

As for frequency and quality settings 8Khz and quality of 4 should be good enough but on such a device we should be able to reach quality of 5 and frequency of 16kHz (that's what I use on my nexus one).
For codecs, disable ilbc there is issues with this one. It consume a lot of cpu (and besides I'm not sure if there is not a bug / cpu leak in it). I advise also to disable speex (as it consume a lot of cpu too). If you are using wifi PCMU/A should be ok.

Last point, if the device fall in this awful state that I don't know yet how to solve (android-devs didn't reply to my answer on the android-dev group)... but in that case, on some devices and for an unknown reason, the entiere audio stack of the device is frozen. It means that if you enter this state the only way I found to restore a good state and be able to call again is to reboot the phone. (I observed this problem only on galaxy spica i7500 & sony x10 for now). If you want to quickly test if you are in this case, try to play music using the standard music application... if nothing happens, let me know I'm interesting in listing all devices for which we are able to reproduce this issue....

[Anonymous]

Originally posted by: jar...@gmail.com

I make some tests using only PCMA and wifi. At first notice that PCMU doesn't work. I cannot even call if only PCMU is selected.

Second notice is that I have to you stun-server when I am using wifi because I cannot hear anything when I am calling. This is strange because other sip-application doesn't need that.

So with PCMA, wifi, 8kHz and quality 4 and still not got my voice through. I don't be sure should I send debug log or what but here is a debug log.

I can test that music player feature later when I got another phone.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Ok for pcmu (to say right I never really tested ulaw separately... even not sure that one of my sip provider support it).

For stun server that strange. With my config I never need neither stun nor turn nor ice to get it working. Just simple direct configuration works properly. Is there special firewalls, network architecture in front of your wifi access point?

Thanks for the log, it well confirms the fact that nothing is get from the mic. Can you get some logs a little bit before these one (when call is established). You can directly send it to me by mail if you don't want your personal infos to be displayed on the website.

So, you confirmed that issue with galaxy S is different from the one with i7500 and X10. That's a good news in one side but also a bad one since it will be more complicated for me to reproduce it and debug it. Maybe if you can get the log that appear before it could help me to ask on forums or samsung (why not). I'm particulary interested in something like that : "AFCCreateReSampler: avAFCInfo " that was the first thing I noticed with the log of another galaxy S owner.

P.S. : I will probably split this issue into one for galaxy s and one for X10/i7500. P.S.2: Any tests with x10/i7500 with the last version? (excepted the fact there is a big echo on x10 that's another problem due to sony implementation.... if only manufacturer has respected a good implementation of their audio drivers.... grrrr)

[Anonymous]

Originally posted by: jar...@gmail.com

There should not be any special thing on my wifi. Just NAT and all sip calls has worked without STUN.

Yes, I can send the log directly to you. That logcat is just quite tricky :) even when I tried to save the log or send it directly I feel that it doesn't take all log-data within.

What log level do you want? Verbose, Debug...? Is your email your username and gmail?

Has somebody before noticed Samsung of those audio driver problem? Could they fix it until Android 2.2?

[Anonymous]

Originally posted by: r3gis...@gmail.com

Verbose log level if possible. My email address is r3gis.3R at gmail dot com.

That's really strange for the need of stun. I'll check that in your log too.

Other devs has noticed the problem : follow the link on stack overflow on my comment 17, it was asked by somebody else and matches exactly the issue we have on csipsimple - I added a comment to be notified of the progress on this issue but unfortunately nothing new. Someone who own the galaxy S with the firmware XXJM2 get it work and said me that the firmware XXJF3 is known to be buggy.
(He had other issues related to network too... maybe something special with network on galaxy S too :/ )

I don't know what is the version of the firmware you have. Can you provide me the info with your mail (you'll find the info in Android Settings>About the phone >Build number or Firmware version if any).