Audio not working with Samsung GT-S5300 (Galaxy Pocket)

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What steps will reproduce the problem?
1. Place an audio call between A using GT-S5300 and B using a different device 
(e.g. GT-I9000, GT-I9100,...).

What is the expected output? What do you see instead?
Playback audio quality should be ok on both sides. On the side B the incoming 
audio is not working instead. No words can be understood.

What version of the product are you using? On what operating system?
Imsdroid v2.532.758 on Samsung GT-S5300 running Android 2.3.6 

Original issue reported on code.google.com by massimo....@gmail.com on 21 Jan 2013 at 4:23

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Please attach network trace and device logs using latest svn revision

Original comment by boss...@yahoo.fr on 22 Jan 2013 at 4:58

• Changed state: Accepted

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Please find attached the Logcat taken on the GT-S5300 and the wireshark trace 
using tha latest svn of imsdroid.

From logcat looks like the AudioRecord cannot get a buffer. You can also try to 
listen to the resulting audio of the conversation using wireshark tools for 
Telephony, since we used PCMA during the conversation.

The overall CPU load during the conversation was about 20%-25% with imsdroid 
using about 15%.

Thanks in advance.

P.S. The issue is probably quite the same discussed here: 
https://groups.google.com/forum/#!topic/doubango/nKogleJbGms

Original comment by massimo....@gmail.com on 23 Jan 2013 at 10:53

Attachments:

• call-from-i9000-to-s5300-with-aec-enabled.pcap-log.txt
• call-from-i9000-to-s5300-with-aec-enabled.pcap

[GoogleCodeExporter]

1)
You logs show that the ScreenAV is loaded and unloaded several times (onStart, 
onResume, onDestroy...). Could you put a breakpoint to see why this happen?

2)
Increase the "AUDIO_BUFFER_FACTOR" 
(http://code.google.com/p/imsdroid/source/browse/branches/2.0/android-ngn-stack/
src/org/doubango/ngn/media/NgnProxyAudioProducer.java?r=515#44) and check if 
you still have the warnings ("AudioRecord: obtainBuffer timed out").

3)
If increasing the "AUDIO_BUFFER_FACTOR" do not fix the issue, try to enable 
openSL-ES instead of AudioRecord: 
http://code.google.com/p/imsdroid/source/browse/branches/2.0/android-ngn-stack/s
rc/org/doubango/ngn/NgnApplication.java?r=526#561

Original comment by boss...@yahoo.fr on 28 Jan 2013 at 7:09

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Thanks for the hints, here the results for the tests you suggested:

1)
I probably was playing with the phone during the tests causing contineously 
screen orientation switches. That's why the ScreenAV was loaded/unloaded 
several times. Sorry. :-)

2)
Increasing the "AUDIO_BUFFER_FACTOR" seems to not affect the "obtainBuffer 
timed out" error. Instead changing the audio codec from PCMA to G.722 caused 
the error to disappear, even if the audio quality was still the same on the 
other side (no word can be understood).

3)
Enabling the openSL-ES caused several other issues and the call was terrible on 
both sides (see logs in attach).

What I noted is that, no matter of the audio codec selected, I always get this 
error on the s5300:

E/tinyWRAP(4610): ***ERROR: function: "tdav_session_audio_get()" 
E/tinyWRAP(4610): file: "src/audio/tdav_session_audio.c" 
E/tinyWRAP(4610): line: "299" 
E/tinyWRAP(4610): MSG: No codec for the producer

Is this ok?

Thanks in advance.

Original comment by massimo....@gmail.com on 28 Jan 2013 at 4:41

Attachments:

• call-from-i9000-to-s5300-with-aec-openSL-ES-enabled.pcap-log.txt

[GoogleCodeExporter]

There is something wrong with the audio driver but still don't understand why 
it fails.
1) Try to disable all audio procession features (AEC, AGC, NS...).
2) Have you tried with any other SIP open source client (e.g. Linphone or 
SipDroid)?

Original comment by boss...@yahoo.fr on 29 Jan 2013 at 11:19

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Disabling the AEC gives exactly the same results reported here: 
https://groups.google.com/forum/#!topic/doubango/nKogleJbGms

"After disabled the AEC when make call from s5360 to any sip client call works 
perfect with all codecs, but when receive call to s5360, call work one way, 
s5360 can hear perfect but another party hear some crashes without voice and 
log appear error in buffer".

Regarding using other SIP OSS clients:
1. SipDroid on S5300 <-> Imsdroid on i9100
On i9100 can just catch few words sent from the S5300, no matter what codec is 
in use.
On S5300 can hear perfectly.

2. Linphone on S5300 <-> Imsdroid on i9100
On i9100 can hear perfectly every word sent from S5300, no matter what codec is 
in use.
On S5300 can't hear anything (weird).

3. CSipSimple on S5300 <-> Imsdroid on i9100
Both ways the audio is fine, no matter what codec is in use.

Please let me know if you need the logs/traces of each test.

Original comment by massimo....@gmail.com on 8 Feb 2013 at 4:25

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I have done tests on GT-S5360 and all is working as expected: Sound is clear on 
both sides.
I'm:
- calling from an iPhone5 (iDoubs) to GT-S5360
- audio codec is PCMA
- media is p2p (no proxy)
- both devices use Doubango r816

Original comment by boss...@yahoo.fr on 12 Feb 2013 at 1:21

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Original comment by boss...@yahoo.fr on 18 Mar 2013 at 1:20

• Changed state: Started

[GoogleCodeExporter]

wat moet je doen als je je telefoon hebt beveiligt en de code niet meer weet 

Original comment by yenma...@gmail.com on 1 Apr 2013 at 8:41

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i have the same issue, it occur when two imsdroid make audio call, there is 
audio in first 30 seconds, after 30 seconds or more, this is one side audio 
issue. 

Original comment by openser@yeah.net on 19 Aug 2014 at 2:58