What steps will reproduce the problem?
1. Set up accounts on CSipSimple connected to Asterisk server
2. Make various adjustments to audio / media configuration
3. Get a wide variety of wacky audio routing or quality problems
What is the expected output? What do you see instead?
Expect good quality audio in speaker, and on Bluetooth, instead I get some of
the following problems.
Best Case:
Echo Cancellation on, Echo method Speex, noise reduction on, 8khz sampling, IO
Queue on, using Speex 8khz codec, all other media menu settings off (expert
mode)
Under Audio troubleshooting: Use Mode audio API, Focus Audio, Setup audio
before init, mic source = VOICE_COMMUNICATION (or VOICE_CALL seems to make no
difference), audio mode IN_COMMUNICATION, Restart stream when change routing=
yes, audio mode = openSL-es
No bluetooth headset paired or connected
With the above settings, the audio will be of acceptable but not great quality
(ie. only 8khz sampling). Delay is significant. In particular, if I call myself
and hold the second phone to my ear, what I say into CSipSimple will be heard
almost immediately on the second phone, but what I say into the second phone
will be heard 1.5 to 3 seconds later (ie. delay on receiving / speaker on
CSipSimple). There is however no echo.
If instead 16kHz sampling is enabled, then remote end will hear audio and then
ear-splitting static that grows and decays after a few seconds. This static can
be significantly reduced by enabling echo cancellation as above, but can not be
fully eliminated. Audio quality at 16KHz sampling seems degraded as well. My
impression is maybe the hardware echo canceler doesn't work at 16khz and is
amplifying the higher frequency noise spuriously, but it's just a guess.
With Bluetooth enabled using my Samsung HM3350, audio reception will be routed
to bluetooth earpiece, but microphone will still be the handset. Turning
bluetooth "on" from the call screen results in NO AUDIO anywhere. However,
there have been some combinations of settings which have allowed me to use
bluetooth some times, perhaps different modes, or I don't know what.
Various other configurations, including "Use WebRTC implementation" have caused
other strange audio routing, such as normal audio routing when placing a call,
but audio routed to speakerphone when receiving a call...
In general, the audio is screwy on this phone. I would be happy to respond to
your requests for audio debugging if you suggest combinations of settings you
think would be interesting to try.
Desirable behavior:
With some appropriate combination of settings, 16khz audio in high quality with
no echo either on handset, speakerphone, or bluetooth, all enabled at the
appropriate times as indicated on screen (ie. when pressing bluetooth button or
when pressing speakerphone button). If not 16khz on this phone due to hardware
limitation, then at least 8khz in high quality with appropriate routing.
audio should never split between bluetooth speaker and handset mic, or sending
audio to speakerphone when speakerphone not selected etc.
What version of the product are you using? On what device / operating
system?
CSipSimple 1.02.03 r2457 (Market version)
HTC Desire 510 handset, stock ROM: 1.42.652.1, Android version 4.4.2, HTC
Sense v 6.0, kernel 3.4.0-ga9a2990
Please provide any additional information below.
Original issue reported on code.google.com by dlake...@gmail.com on 2 Mar 2015 at 11:06
Original issue reported on code.google.com by
dlake...@gmail.com
on 2 Mar 2015 at 11:06