Call for Symbian S60 testers for APS-direct
Hi All,
As we finish the current iteration of Symbian S60 implementation, most
importantly the implementation of APS-direct, we need all of you to
participate in testing it. This is from experience we know mobile
devices are very tricky and can behave differently from one firmware
to another.
So we need as many Symbian S60 3rd Edition phone as possible.
To do this we need your IMEI and a few other details, this is because
the requirements of Symbian Signed, we need to 'burn' your IMEI into
our test application.
Don't reply with your details to the public list, instead fill in this
form: http://l.teluu.com/symbiantesting
We can't promise everyone will be accepted, because there is a hard
limit to the number of IMEIs we can burn, and we'll prioritize
variation of devices, rather than first come first serve.
If you can get your friends family neighbours to join in that will be
even better.
If you any questions, you can contact me.
Thank you all!
--
Perry Ismangil
Ok, there was some issue with the link I gave, it should be sorted now.
There is an alternative link:
https://spreadsheets.google.com/viewform?key=p73LYJuTbrW9RhzjVhPSMbQ
Let me know if it stops you again.
2009/2/12 Perry Ismangil perry@teluu.com:
Hi All,
As we finish the current iteration of Symbian S60 implementation, most
importantly the implementation of APS-direct, we need all of you to
participate in testing it. This is from experience we know mobile
devices are very tricky and can behave differently from one firmware
to another.
So we need as many Symbian S60 3rd Edition phone as possible.
To do this we need your IMEI and a few other details, this is because
the requirements of Symbian Signed, we need to 'burn' your IMEI into
our test application.
Don't reply with your details to the public list, instead fill in this
form: http://l.teluu.com/symbiantesting
We can't promise everyone will be accepted, because there is a hard
limit to the number of IMEIs we can burn, and we'll prioritize
variation of devices, rather than first come first serve.
If you can get your friends family neighbours to join in that will be
even better.
If you any questions, you can contact me.
Thank you all!
--
Perry Ismangil
--
Perry Ismangil
Managing Partner
Teluu - Communicate Everywhere
E perry@teluu.com
twitter.com/ismangil
T +44 114 299 8883
F +44 870 974 9023
W http://www.teluu.com
23 Langdon Street, Sheffield S11 8BH, United Kingdom
Registered in England as Teluu LLP, no. OC323977
Hello,
I already started testing code as I need them soon.
Device N95
OS: 9.2
SDK: FP1
It did compile for me
as instructed. But fails at in PJMEDIA conference.c line 1701 in get_frame
function
pj_assert(frame->size ==
conf->samples_per_frame * conf->bits_per_sample / 8);
It fails in the above logic and the application quits.
I also found few problems in switchin codec but yet to figure out
whether default using the APS or not. seems like default is not, only when
I switch the codec.
Thanks
Rawshan
Ok, there was some issue with the link I
gave, it should be sorted now.
There is an
alternative link:
Let me know if it stops you again.
2009/2/12 Perry Ismangil perry@teluu.com:
Hi All,
As we finish the current iteration of
Symbian S60 implementation, most
importantly the
implementation of APS-direct, we need all of you to
participate in testing it. This is from experience we know mobile
devices are very tricky and can behave differently from one
firmware
to another.
So we
need as many Symbian S60 3rd Edition phone as possible.
To do this we need your IMEI and a few other details, this is
because
the requirements of Symbian Signed, we need to
'burn' your IMEI into
our test application.
Don't reply with your details to the public list, instead
fill in this
because there is a hard
limit to the number of IMEIs we can
burn, and we'll prioritize
variation of devices, rather
than first come first serve.
If you can get
your friends family neighbours to join in that will be
even
better.
If you any questions, you can
contact me.
Thank you all!
--
Perry Ismangil
--
Perry Ismangil
Managing Partner
Teluu - Communicate Everywhere
E perry@teluu.com
twitter.com/ismangil
T +44 114 299 8883
F +44 870 974 9023
W
23 Langdon Street,
Sheffield S11 8BH, United Kingdom
Registered in England as Teluu LLP, no. OC323977
Visit our blog:
pjsip mailing list
Hi,
We're not quite ready yet, still undergoing internal testing.
This an draft of the how to use it:
https://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
Again it's not quite finished, some details might be missing.
If you want to test the binary version, don't forget to add your phone
to https://spreadsheets.google.com/viewform?key=p73LYJuTbrW9RhzjVhPSMbQ
On Fri, Feb 13, 2009 at 22:19, iajdani@provati.com wrote:
Hello,
I already started testing code as I need them soon.
Device N95
OS: 9.2
SDK: FP1
It did compile for me as instructed. But fails at in PJMEDIA conference.c
line 1701 in get_frame function
pj_assert(frame->size == conf->samples_per_frame * conf->bits_per_sample /
8);
It fails in the above logic and the application quits.
I also found few problems in switchin codec but yet to figure out whether
default using the APS or not. seems like default is not, only when I switch
the codec.
Thanks
Rawshan
Ok, there was some issue with the link I gave, it should be sorted now.
There is an alternative link:
https://spreadsheets.google.com/viewform?key=p73LYJuTbrW9RhzjVhPSMbQ
Let me know if it stops you again.
2009/2/12 Perry Ismangil perry@teluu.com:
Hi All,
As we finish the current iteration of Symbian S60 implementation, most
importantly the implementation of APS-direct, we need all of you to
participate in testing it. This is from experience we know mobile
devices are very tricky and can behave differently from one firmware
to another.
So we need as many Symbian S60 3rd Edition phone as possible.
To do this we need your IMEI and a few other details, this is because
the requirements of Symbian Signed, we need to 'burn' your IMEI into
our test application.
Don't reply with your details to the public list, instead fill in this
form: http://l.teluu.com/symbiantesting
We can't promise everyone will be accepted, because there is a hard
limit to the number of IMEIs we can burn, and we'll prioritize
variation of devices, rather than first come first serve.
If you can get your friends family neighbours to join in that will be
even better.
If you any questions, you can contact me.
Thank you all!
--
Perry Ismangil
--
Perry Ismangil
Managing Partner
Teluu - Communicate Everywhere
E perry@teluu.com
twitter.com/ismangil
T +44 114 299 8883
F +44 870 974 9023
W http://www.teluu.com
23 Langdon Street, Sheffield S11 8BH, United Kingdom
Registered in England as Teluu LLP, no. OC323977
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
Perry Ismangil
I am trying to play ringtone to the caller at 180 response. What I
wrote in on_call_state callback is given below. The code just execute fine
but no ringtone in my earpiece. I am now using APS-DIRECT.
case PJSIP_INV_STATE_EARLY:
if (g_call_id ==
PJSUA_INVALID_ID)
g_call_id = call_id;
if (g_cb.on_call_connected) {
static wchar_t
msg[256];
g_cb.on_call_ringing(msg);
/Ring Tone Code**/
#define SAMPLES_PER_FRAME 64
#define ON_DURATION 100
#define OFF_DURATION 100
pj_pool_t *ring_pool;
pjmedia_port *ring_port;
pj_status_t status;
ring_pool = pjsua_pool_create("krypt_ring",
4000, 4000);
status = pjmedia_tonegen_create(ring_pool,
8000, 1, SAMPLES_PER_FRAME, 16, 0,
&ring_port);
if (status != PJ_SUCCESS)
return;
{
pjmedia_tone_desc tones[3];
tones[0].freq1 = 200;
tones[0].freq2 = 0;
tones[0].on_msec = ON_DURATION;
tones[0].off_msec = OFF_DURATION;
tones[1].freq1 = 400;
tones[1].freq2 = 0;
tones[1].on_msec = ON_DURATION;
tones[1].off_msec = OFF_DURATION;
tones[2].freq1 = 800;
tones[2].freq2 = 0;
tones[2].on_msec = ON_DURATION;
tones[2].off_msec = OFF_DURATION;
status = pjmedia_tonegen_play(ring_port, 3,
tones, 0);
}
pjmedia_port_destroy(ring_port);
pj_pool_release( ring_pool );
/end
ringtone/
}
I think I am missing pjmedia_port here.
What is the API to
check the currently active pjmedia_port ??? The sound port is already
active in symbian_ua I suppose. So I need to grab that pjmedia instead of
creating new one????????
Rawshan
I am trying to play ringtone
to the caller at 180 response. What I
wrote in on_call_state
callback is given below. The code just execute fine
but no
ringtone in my earpiece. I am now using APS-DIRECT.
case PJSIP_INV_STATE_EARLY:
if (g_call_id ==
PJSUA_INVALID_ID)
g_call_id = call_id;
if
(g_cb.on_call_connected) {
static wchar_t
msg[256];
g_cb.on_call_ringing(msg);
/Ring Tone Code**/
#define SAMPLES_PER_FRAME 64
#define ON_DURATION 100
#define OFF_DURATION 100
pj_pool_t *ring_pool;
pjmedia_port
*ring_port;
pj_status_t status;
ring_pool = pjsua_pool_create("krypt_ring",
4000,
4000);
status =
pjmedia_tonegen_create(ring_pool,
8000, 1, SAMPLES_PER_FRAME,
16, 0,
&ring_port);
if (status != PJ_SUCCESS)
return;
{
pjmedia_tone_desc tones[3];
tones[0].freq1 =
200;
tones[0].freq2 = 0;
tones[0].on_msec =
ON_DURATION;
tones[0].off_msec = OFF_DURATION;
tones[1].freq1 = 400;
tones[1].freq2 = 0;
tones[1].on_msec = ON_DURATION;
tones[1].off_msec =
OFF_DURATION;
tones[2].freq1 = 800;
tones[2].freq2 = 0;
tones[2].on_msec = ON_DURATION;
tones[2].off_msec = OFF_DURATION;
status = pjmedia_tonegen_play(ring_port, 3,
tones, 0);
}
pjmedia_port_destroy(ring_port);
pj_pool_release( ring_pool );
/*****************************end
ringtone*****************************/
mailing list
On Sun, Feb 15, 2009 at 8:18 PM, iajdani@provati.com wrote:
I am trying to play ringtone to the caller at 180 response. What I wrote in
on_call_state callback is given below. The code just execute fine but no
ringtone in my earpiece. I am now using APS-DIRECT.
First of all, the aps-direct branch is supposed to be an internal branch and
we're not quite finished with it so expect few rough edges here and there.
Even the API is not quite finalized yet, so it's really not ready to be used
for anything. But I do appreciate, and surprised at the same time, with the
level of interests that this has generated.
Answering your question. When using APS-Direct, you always need to remember
that the whole point of having APS-Direct is to enable codec compression in
sound device (APS/VAS), hence to let encoded frames flowing end-to-end from
the microphone down to the socket and vice versa.
The sample on_stream_created() snippet shows how to open the sound device in
codec format according to the codec being used by the call. So since the
sound device is opened in codec mode, you can't no longer feed it with PCM
frames (e.g. the tone generator).
To use the tone generator (or any other pjmedia features that works on PCM
frames such as WAV files), you will need to open the sound device in PCM
mode to play the ring tones, then when you want to communicate with the
call/stream, you will need to close the sound device, and re-open it using
the codec that is used by the call.
The on_stream_created() snippet currently doesn't show how to do this, so
you need to do it yourself.
cheers
Benny
Thanks Benny for your help. It was really insightfull. The good news is
the application initially initialized as PCM mode. I think I know how to
open it up in PCM mode as well. I successfully created the ringtone and
could play it as well. But the bad news is, once ringtone is played I am
unable to disconnect the sound port. The on_stream_created callback suppose to disconnect
that automatically and well it does, but I loose the calles sound. after
that my phone looses sound but i can hear sound in the other side.
The way i did that. I opened up a media port for ring tone. then i
assigned that to conference port with pjsua_conf_add_port. then i connect
the conference id with master conference port id 0. now when i play the
ringtone i can hear.
I think somehow I am missing something
here.
#define SAMPLES_PER_FRAME 64
#define
ON_DURATION 1500
#define
OFF_DURATION 2500
status = pjmedia_tonegen_create2(app_pool, NULL
,8000, 1, SAMPLES_PER_FRAME, 16, 0, &ring_port);
if (status != PJ_SUCCESS)
return;
status = pjsua_conf_add_port(
app_pool,ring_port,&c_id);
pjsua_conf_connect(c_id,0);
if (status != PJ_SUCCESS) {
PJ_LOG(1,(THIS_FILE, "connecting ring port failed to device,
status=%d", status));
return;
}
{
pjmedia_tone_desc tones[1];
tones[0].freq1 = 400;
tones[0].freq2 = 0;
tones[0].on_msec = ON_DURATION;
tones[0].off_msec = OFF_DURATION;
status = pjmedia_tonegen_play(ring_port, 1,
tones,1);
/********for deinitialization/
if (ring_port){
pjmedia_tonegen_stop(ring_port);
pjsua_conf_disconnect(0,c_id);
pjsua_conf_remove_port(c_id);
pjmedia_port_destroy(ring_port);
ring_port=NULL;
return PJ_TRUE;
}
On Sun, Feb 15, 2009 at 8:18 PM, iajdani@provati.com
wrote:
I am trying to play ringtone to the
caller at 180 response. What I wrote
in
on_call_state callback is given below. The code just execute fine but no
ringtone in my earpiece. I am now using APS-DIRECT.
First of all, the aps-direct branch is
supposed to be an internal branch
and
we're not
quite finished with it so expect few rough edges here and there.
Even the API is not quite finalized yet, so it's really not ready
to be
used
for anything. But I do appreciate, and
surprised at the same time, with
the
level of
interests that this has generated.
Answering your
question. When using APS-Direct, you always need to
remember
that the whole point of having APS-Direct is to enable codec
compression
in
sound device (APS/VAS), hence to let
encoded frames flowing end-to-end
from
the
microphone down to the socket and vice versa.
The
sample on_stream_created() snippet shows how to open the sound device
in
codec format according to the codec being used by the
call. So since the
sound device is opened in codec mode, you
can't no longer feed it with PCM
frames (e.g. the tone
generator).
To use the tone generator (or any other
pjmedia features that works on PCM
frames such as WAV files),
you will need to open the sound device in PCM
mode to play the
ring tones, then when you want to communicate with the
call/stream, you will need to close the sound device, and re-open it using
the codec that is used by the call.
The
on_stream_created() snippet currently doesn't show how to do this, so
you need to do it yourself.
cheers
Benny
mailing list
On Sun, Feb 15, 2009 at 8:18 PM, iajdani@provati.com
wrote: > >> I am trying to play ringtone to the caller at 180
response. What I wrote >> in >> on_call_state callback is
given below. The code just execute fine but no >> ringtone in my
earpiece. I am now using APS-DIRECT. >> > > First of all, the
aps-direct branch is supposed to be an internal branch > and > we're
not quite finished with it so expect few rough edges here and there. >
Even the API is not quite finalized yet, so it's really not ready to be
used > for anything. But I do appreciate, and surprised at the
same time, with > the > level of interests that this has generated.
Answering your question. When using APS-Direct, you always need
to > remember > that the whole point of having APS-Direct is to
enable codec compression > in > sound device (APS/VAS), hence to let
encoded frames flowing end-to-end > from > the microphone down to
the socket and vice versa. > > The sample on_stream_created()
snippet shows how to open the sound device > in > codec format
according to the codec being used by the call. So since the > sound
device is opened in codec mode, you can't no longer feed it with PCM >
frames (e.g. the tone generator). > > To use the tone generator (or
any other pjmedia features that works on PCM > frames such as WAV
files), you will need to open the sound device in PCM > mode to play
the ring tones, then when you want to communicate with the >
call/stream, you will need to close the sound device, and re-open it using
the codec that is used by the call. > > The on_stream_created()
snippet currently doesn't show how to do this, so > you need to do it
yourself. > > cheers > Benny >
_______________________________________________ > Visit our blog:
http://blog.pjsip.org > > pjsip mailing list >
pjsip@lists.pjsip.org >
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
Dear Rawshan,
Did you find solution to this issue? Any clue?
Regards,
Manoj
-----Original Message-----
From: pjsip-bounces@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org]On Behalf Of iajdani@provati.com
Sent: Tuesday, February 17, 2009 1:58 AM
To: pjsip list
Subject: Re: [pjsip] Need little help here --- Ringtone with APS
Thanks Benny for your help. It was really insightfull. The good news is
the application initially initialized as PCM mode. I think I know how to
open it up in PCM mode as well. I successfully created the ringtone and
could play it as well. But the bad news is, once ringtone is played I am
unable to disconnect the sound port. The on_stream_created callback suppose
to disconnect that automatically and well it does, but I loose the calles
sound. after that my phone looses sound but i can hear sound in the other
side.
The way i did that. I opened up a media port for ring tone. then i
assigned that to conference port with pjsua_conf_add_port. then i connect
the conference id with master conference port id 0. now when i play the
ringtone i can hear.
I think somehow I am missing something here.
#define SAMPLES_PER_FRAME 64
#define ON_DURATION 1500
#define OFF_DURATION 2500
status = pjmedia_tonegen_create2(app_pool, NULL ,8000, 1,
SAMPLES_PER_FRAME, 16, 0, &ring_port);
if (status != PJ_SUCCESS)
return;
status = pjsua_conf_add_port( app_pool,ring_port,&c_id);
pjsua_conf_connect(c_id,0);
if (status != PJ_SUCCESS) {
PJ_LOG(1,(THIS_FILE, "connecting ring port failed to device,
status=%d", status));
return;
}
{
pjmedia_tone_desc tones[1];
tones[0].freq1 = 400;
tones[0].freq2 = 0;
tones[0].on_msec = ON_DURATION;
tones[0].off_msec = OFF_DURATION;
status = pjmedia_tonegen_play(ring_port, 1, tones,1);
/********for deinitialization/
if (ring_port){
pjmedia_tonegen_stop(ring_port);
pjsua_conf_disconnect(0,c_id);
pjsua_conf_remove_port(c_id);
pjmedia_port_destroy(ring_port);
ring_port=NULL;
return PJ_TRUE;
}
On Sun, Feb 15, 2009 at 8:18 PM, iajdani@provati.com wrote:
I am trying to play ringtone to the caller at 180 response. What I
wrote
in
on_call_state callback is given below. The code just execute fine but
no
ringtone in my earpiece. I am now using APS-DIRECT.
First of all, the aps-direct branch is supposed to be an internal branch
and
we're not quite finished with it so expect few rough edges here and
there.
Even the API is not quite finalized yet, so it's really not ready to be
used
for anything. But I do appreciate, and surprised at the same time, with
the
level of interests that this has generated.
Answering your question. When using APS-Direct, you always need to
remember
that the whole point of having APS-Direct is to enable codec compression
in
sound device (APS/VAS), hence to let encoded frames flowing end-to-end
from
the microphone down to the socket and vice versa.
The sample on_stream_created() snippet shows how to open the sound
device
in
codec format according to the codec being used by the call. So since the
sound device is opened in codec mode, you can't no longer feed it with
PCM
frames (e.g. the tone generator).
To use the tone generator (or any other pjmedia features that works on
PCM
frames such as WAV files), you will need to open the sound device in PCM
mode to play the ring tones, then when you want to communicate with the
call/stream, you will need to close the sound device, and re-open it
using
the codec that is used by the call.
The on_stream_created() snippet currently doesn't show how to do this,
so
you need to do it yourself.
cheers
Benny
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
On Sun, Feb 15, 2009 at 8:18 PM, iajdani@provati.com wrote: > >> I am
trying to play ringtone to the caller at 180 response. What I wrote >> in >>
on_call_state callback is given below. The code just execute fine but no >>
ringtone in my earpiece. I am now using APS-DIRECT. >> > > First of all, the
aps-direct branch is supposed to be an internal branch > and > we're not
quite finished with it so expect few rough edges here and there. > Even the
API is not quite finalized yet, so it's really not ready to be > used > for
anything. But I do appreciate, and surprised at the same time, with > the >
level of interests that this has generated. > > Answering your question.
When using APS-Direct, you always need to > remember > that the whole point
of having APS-Direct is to enable codec compression > in > sound device
(APS/VAS), hence to let encoded frames flowing end-to-end > from > the
microphone down to the socket and vice versa. > > The sample
on_stream_created() snippet shows how to open the sound device > in > codec
format according to the codec being used by the call. So since the > sound
device is opened in codec mode, you can't no longer feed it with PCM >
frames (e.g. the tone generator). > > To use the tone generator (or any
other pjmedia features that works on PCM > frames such as WAV files), you
will need to open the sound device in PCM > mode to play the ring tones,
then when you want to communicate with the > call/stream, you will need to
close the sound device, and re-open it using > the codec that is used by the
call. > > The on_stream_created() snippet currently doesn't show how to do
this, so > you need to do it yourself. > > cheers > Benny >
_______________________________________________ > Visit our blog:
http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org >
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >