john990 / sipdroid

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Audio Buffer over flow issue in AudioRecord #62

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago
Hi,
 Iam using the latest sipdroid code . I am having Audio Buffer overflow in
audiorecord , and the audio RTP layer is tearing down , bringing down the
whole call. Can you please let me know what could be the issue. I am using
G.711 codec.

Thanks,
Anu

Original issue reported on code.google.com by anugagga...@gmail.com on 27 Jun 2009 at 9:24

GoogleCodeExporter commented 8 years ago
[deleted comment]
GoogleCodeExporter commented 8 years ago
Thanks for the reply. No after the buffer overflow, my call gets disconnected. 
Iam
using G1. I tried increasing the buffer size in AudioRecord , it didnt 
help.This is
what I see in the logcat. 

Anu

06-28 16:19:04.864: WARN/AudioFlinger(35): AudioRecordThread: buffer overflow
06-28 16:19:06.114: DEBUG/dalvikvm(111): GC freed 2309 objects / 138512 bytes 
in 320ms
06-28 16:19:10.604: DEBUG/dalvikvm(1772): GC freed 3708 objects / 579832 bytes 
in 158ms
06-28 16:19:10.704: INFO/System.out(1772): AudioLauncher: halting java audio..

Original comment by anugagga...@gmail.com on 28 Jun 2009 at 9:09

GoogleCodeExporter commented 8 years ago
Do your calls work for a while? (before the buffer overflow)

Original comment by pmerl...@googlemail.com on 28 Jun 2009 at 10:37

GoogleCodeExporter commented 8 years ago
It works for around 20-30 seconds or so. 

Original comment by anugagga...@gmail.com on 28 Jun 2009 at 10:40

GoogleCodeExporter commented 8 years ago
[deleted comment]
GoogleCodeExporter commented 8 years ago
Buffer overflows in AudioRecord are normal. We have already begun reducing them 
but I 
don't think they have anything to do with your calls getting disconnect. There 
is a 
timeout of 22 seconds no RTP received the call gets hung up. Do you hear the 
other end?

Original comment by pmerl...@googlemail.com on 28 Jun 2009 at 11:04

GoogleCodeExporter commented 8 years ago
Thanks a lot. It helped. I was using a send-only stream , but didnt set 
recv-only to
false. So that was the reason it was timing out on RtpReceiveStreamer. Thanks a 
lot ,
now I see my audio line stay up.

Anu

Original comment by anugagga...@gmail.com on 28 Jun 2009 at 11:25

GoogleCodeExporter commented 8 years ago
Just a question, how do you set two devices such that one is send-only and the 
other
is receive-only? What things are to be added/modified/removed in the two sets of
codes to make a test like this without the 22 second time out being triggered?

Original comment by Silversp...@gmail.com on 29 Jun 2009 at 2:12

GoogleCodeExporter commented 8 years ago
hello sir,

I m using Android G1.

i also have a same problem.
i can hear other-end but other-end user can't get my voice.
i got same error/warning as below...

07-02 12:26:52.997: WARN/AudioFlinger(554): AudioRecordThread: buffer overflow

please help me.

Original comment by trushs...@gmail.com on 2 Jul 2009 at 7:26

GoogleCodeExporter commented 8 years ago
and if you'll have any solution please provide me..

thanks in advance...

Original comment by trushs...@gmail.com on 2 Jul 2009 at 7:28

GoogleCodeExporter commented 8 years ago
hello sir,

I'm a G1 user.

i also have the same problem.
there are many nosie,and bad voice,when i goto the logcat,there are many logs 
like below:
04-12 17:22:02.102: WARN/AudioFlinger(51): AudioRecordThread: buffer overflow

what can i do? is something wrong with my phone or the AudioRecord class??

Original comment by dzxsk1...@gmail.com on 12 Apr 2010 at 9:36

GoogleCodeExporter commented 8 years ago
hi,
I konw one reason:
if you "new MediaRecorder" >= twice and first MediaRecorder not do" = null", 
then must happen :"buffer overflow".

Hope useful for you ^_^

Original comment by wnqwang on 25 Sep 2010 at 9:05