Hi Bill,
You saved me again!
I have clean audio in both directions now.
The latest milestone I need to implement is the inband DTMF.
But let me put it in a separate message as it may be better visible
Best Regards
Dimitar
From: Bill Gardner
Sent: Saturday, April 4, 2015 01:45
To: pjsip at lists.pjsip.org
Subject: Re: TDM sound device to the pjmedia, how?
Hi Dimitar,
I'm pretty sure that if you return an error from the create_stream, pjsua will try with different sampling rates until it finds one that works, and will insert sample rate conversion between the 16kHz internal bridge and the sound device. But it's also easy to force everything to run at 8 kHz by putting this in your config_site.h:
#define PJSUA_DEFAULT_CLOCK_RATE 8000
Regards,
Bill
On 4/3/2015 6:13 PM, dpn at switchvoice.com wrote:
Hi Bill,
I have printed the params sent by the create_stream. So you are right
clock_rate=16000,
channel_count=1
samples_per_frame=320
bits_per_sample = 16
So should I interpolate my data to provide 320 samples each 20ms tick?
Is there a way to change this pjsip behavior?
Thanks
Dimitar
From: Bill Gardner
Sent: Friday, April 3, 2015 22:57
To: pjsip at lists.pjsip.org
Subject: Re: TDM sound device to the pjmedia, how?
Hi Dimitar,
You have to be prepared for any parameters in create_stream, the caller can override the defaults you provide. It's likely pjsip is running at 16 kHz and 20 msec frame size, hence the 320 samples per frame.
Bill
On 4/3/2015 3:19 PM, dpn at switchvoice.com wrote:
Hi Bill, Hi Gents
I have implemented my audio devices as Bill pointed me out.
I even get some scrappy one direction audio already between my analog phone and a sip client.
In my audio device in tdm_factory_default_param() I set
clock_rate = 8000
channel_count = 1
samples_per_frame = 160
bits_per_sample = 16
Then in tdm_factory_create_stream() however I get param structure with samples_per_frame set to 320 instead,
and this value seems not affected by the settings I have in tdm_factory_default_param().
How should I set my frame size in pjmedia?
Thanks.
Dimitar
From: dpn@xxxxxxxxxxxxxxx
Sent: Saturday, March 28, 2015 00:21
To: pjsip at lists.pjsip.org
Subject: Re: TDM sound device to the pjmedia, how?
Hi Bill,
Thanks, I think I see the global picture now.
Dimitar
From: Bill Gardner
Sent: Friday, March 27, 2015 21:20
To: pjsip at lists.pjsip.org
Subject: Re: TDM sound device to the pjmedia, how?
Hi Dimitar,
pjsip runs all media processing from the audio device callback, so it is important that the callbacks happen periodically at the frame rate, typically 10 or 20 msec. pjsip can tolerate some burstiness in the callbacks, say two reads followed by two writes, but long pauses will cause big problems. A typical implementation starts a high priority thread with a loop that polls the hardware device and makes the callbacks when data is to be transferred.
Regards,
Bill
On 3/27/2015 2:38 PM, dpn at switchvoice.com wrote:
Hi Bill,
Thank you for the explanation.
The only conceptual think I am not clear for my pjsua application now is
if I can block in my audio device until my hardware has or requires audio data.
Best Regards
Dimitar
From: Bill Gardner
Sent: Friday, March 27, 2015 15:56
To: pjsip at lists.pjsip.org
Subject: Re: TDM sound device to the pjmedia, how?
Hi Dimitar,
You want to write a new audiodev following pjmedia/include/pjmedia-audiodev/audiodev_imp.h,
then set up a macro to enable/disable this device, such as PJMEDIA_AUDIO_DEV_HAS_TDM,
and then add the hooks in pjmedia\src\pjmedia-audiodev\audiodev.c so pjmedia will use this device,
and then enable the device in config_site.h and disable the others.
You can then enable/disable the hardware echo cancellation by using the pjmedia capability
PJMEDIA_AUD_DEV_CAP_EC, then pjmedia can use the built-in AEC.
Regards,
Bill
On 3/27/2015 8:30 AM, dpn at switchvoice.com wrote:
Hi All,
I?ve ported pjproject-1.16 on an embedded platform based on Mediatek MT 7620 CPU running OpenWRT.
(Version 1.16 is rather old but I don?t need video and it seems it covers my needs. )
On this platform there is SLIC chip supporting FXS interface to an analog phone.
The SLIC has TDM interface to the CPU and I have completed the TDM driver (running in the Linux kernel space)
which works fine with Asterisk.
Now I am trying to test pjsua (or some other basic pjsip user agent? ) so I would like to integrate my driver to pjmedia.
I?ve checked the very clean pjsip documentation but still I am not fully clear how to approach the project.
I think using PortAudio or ALSA will overcomplicate my experiment.
My TDM driver expose ioctl interface which I can use from my pjsua application from the user space.
I think the best would be to derive the application clock from my tdm driver.
As far as I understand I should use ?Sound Device Port Framework? so eventually the application hooks
two callbacks rec_cb and play_cb which are called when my TDM driver has frame of data or needs frame of data.
I am looking at /pjproject-1.16/pjmedia/src/pjmedia-audiodev/null_dev.c which looks as the simplest starting point I could hack.
However sound devices in pjmedia provide lot of stuff which I don?t think I need.
For example acoustic echo cancelation. I don?t need that.
I rather use line echo canceler in my TDM driver which is more important I think for my softphone example.
From the other hand the ?Media Ports Framework? looks much cleaner than ?Sound Device Port Framework?.
For example /pjproject-1.16/pjmedia/src/pjmedia/null_port.c looks much simpler than null_dev.c.
But as far as understand using this framework the application calls the functions get_frame and put_frame so what I could do
is to block the execution of those functions (using linux system call select() for example)
and return when I get/send the frame data from/to my driver. This way I will get the timings provided from my driver to the pjsua.
However I am not sure how this will fit with the SIP part of my test softphone.
Anyone willing to clarify if I am on or fully out of track?
Thanks
Dimitar
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