Minimum latency achievable aside
Hi all,
You're my hero if you decide to answer even 10% of my question.
My question regards latency in applications using the PJSUA library. I am
currently running my application on either a machine with Debian or a
Raspberry PI board with Raspbian OS using the ALSA audio backend.
I was wondering, in your experience is it possible to achieve overall
latencies < 50ms with the Pjsua library?
I found that the main limit in this is lowering the latency introduced by
audio devices, so I was wondering if you could help me understand, if there
is a limit, why it so high (in my experience I was not able to go below 42
ms for capture and 42ms for playback, not even by setting the parameter to
lower values in the pjsua_media_config object passed during library
initialization with pjsua_init()) compared to other applications that use
audio devices (e.g. any type of DAWs allows latencies below 20ms).
Is there some design requirement to have large buffers at some point in the
processing?
Is there anything that can be done by modifying the audio backend?
Can I achieve <50ms latencies by using the pjmedia library directly and not
going through pjsua?
Any help will be highly appreciated, Thanks
Gianni Massi
Hi Gianni,
I don't know what minimum latencies you can get with ALSA on PI, you'll
just have to experiment. DAWs typically use ASIO format which is
designed for low latency operation, I have no idea how hard it would be
to implement ASIO or an alternative to ALSA on PI. As you already know
most of the pjsua latency is in the audio device buffering. There is
also a wsola buffer which holds recorded samples from the microphone
awaiting processing by the bridge, this buffer is needed to compensate
for differing playback and record sampling rates. And, there is the
jitter buffer which is obviously essential to compensate for network
delivery timing, missing packets, etc. There isn't any latency in the
media path which isn't essential, more latency allows more variation in
the behavior of the audio device or network. You can tweak the
parameters to decrease latency at the risk of getting getting poor audio
quality when the audio device or network doesn't behave optimally.
I can't see there would be an advantage in using just the pjmedia
library instead of pjsua.
Cheers,
Bill
On 5/26/2016 9:44 AM, Gianni Massi wrote:
Hi all,
You’re my hero if you decide to answer even 10% of my question.
My question regards latency in applications using the PJSUA library. I
am currently running my application on either a machine with Debian or
a Raspberry PI board with Raspbian OS using the ALSA audio backend.
I was wondering, in your experience is it possible to achieve overall
latencies < 50ms with the Pjsua library?
I found that the main limit in this is lowering the latency introduced
by audio devices, so I was wondering if you could help me understand,
if there is a limit, why it so high (in my experience I was not able
to go below 42 ms for capture and 42ms for playback, not even by
setting the parameter to lower values in the pjsua_media_config object
passed during library initialization with pjsua_init()) compared to
other applications that use audio devices (e.g. any type of DAWs
allows latencies below 20ms).
Is there some design requirement to have large buffers at some point
in the processing?
Is there anything that can be done by modifying the audio backend?
Can I achieve <50ms latencies by using the pjmedia library directly
and not going through pjsua?
Any help will be highly appreciated, Thanks
/Gianni Massi/
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Bill,
Thank you for answering my question in a such a detailed and clear way.
Using ASIO to achieve low latencies makes sense, Ill need to look into it,
dont know if its possible with PJSIP, Ill need to find out. Thanks for
pointing me in the right direction!
Have a nice day,
Gianni
Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Bill Gardner
Inviato: giovedì 26 maggio 2016 18:50
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] Minimum latency achievable aside
Hi Gianni,
I don't know what minimum latencies you can get with ALSA on PI, you'll just
have to experiment. DAWs typically use ASIO format which is designed for low
latency operation, I have no idea how hard it would be to implement ASIO or
an alternative to ALSA on PI. As you already know most of the pjsua latency
is in the audio device buffering. There is also a wsola buffer which holds
recorded samples from the microphone awaiting processing by the bridge, this
buffer is needed to compensate for differing playback and record sampling
rates. And, there is the jitter buffer which is obviously essential to
compensate for network delivery timing, missing packets, etc. There isn't
any latency in the media path which isn't essential, more latency allows
more variation in the behavior of the audio device or network. You can tweak
the parameters to decrease latency at the risk of getting getting poor audio
quality when the audio device or network doesn't behave optimally.
I can't see there would be an advantage in using just the pjmedia library
instead of pjsua.
Cheers,
Bill
On 5/26/2016 9:44 AM, Gianni Massi wrote:
Hi all,
Youre my hero if you decide to answer even 10% of my question.
My question regards latency in applications using the PJSUA library. I am
currently running my application on either a machine with Debian or a
Raspberry PI board with Raspbian OS using the ALSA audio backend.
I was wondering, in your experience is it possible to achieve overall
latencies < 50ms with the Pjsua library?
I found that the main limit in this is lowering the latency introduced by
audio devices, so I was wondering if you could help me understand, if there
is a limit, why it so high (in my experience I was not able to go below 42
ms for capture and 42ms for playback, not even by setting the parameter to
lower values in the pjsua_media_config object passed during library
initialization with pjsua_init()) compared to other applications that use
audio devices (e.g. any type of DAWs allows latencies below 20ms).
Is there some design requirement to have large buffers at some point in the
processing?
Is there anything that can be done by modifying the audio backend?
Can I achieve <50ms latencies by using the pjmedia library directly and not
going through pjsua?
Any help will be highly appreciated, Thanks
Gianni Massi
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Follow up question if youd be so kind to indulge in my ignorance!
While experimenting to find out the best tradeoff between audio quality and
latency, I discovered I was unable to get to the point where quality is
affected because it seems that Pjsua wont allow me to go below 42 ms for
either play/cap latency even if I set the values to 15ms in
pjsua_media_config (I can see it from the information printed out when a
device is opened with log-level set to 6).
Do you think there is something that can be done here? It all ends in the
ALSA lib method in alsa_dev.c that gets the nearest possible buffer size
(snd_pcm_hw_params_set_buffer_size_near) as far as I can tell. Do you have
any suggestions regarding this? Any other setting that I might change in
order to get smaller values returned from this function? My understanding of
Alsa is very limited, any pointer in the right direction would be of great
help.
Also Im achieving much better results by compiling the library with the
audio switchboard enabled (#define PJMEDIA_CONF_USE_SWITCH_BOARD 1 in
config_site.h) but I still experience the same limit while experimenting.
Any suggestions?
Thanks,
Gianni
Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Bill Gardner
Inviato: giovedì 26 maggio 2016 18:50
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] Minimum latency achievable aside
Hi Gianni,
I don't know what minimum latencies you can get with ALSA on PI, you'll just
have to experiment. DAWs typically use ASIO format which is designed for low
latency operation, I have no idea how hard it would be to implement ASIO or
an alternative to ALSA on PI. As you already know most of the pjsua latency
is in the audio device buffering. There is also a wsola buffer which holds
recorded samples from the microphone awaiting processing by the bridge, this
buffer is needed to compensate for differing playback and record sampling
rates. And, there is the jitter buffer which is obviously essential to
compensate for network delivery timing, missing packets, etc. There isn't
any latency in the media path which isn't essential, more latency allows
more variation in the behavior of the audio device or network. You can tweak
the parameters to decrease latency at the risk of getting getting poor audio
quality when the audio device or network doesn't behave optimally.
I can't see there would be an advantage in using just the pjmedia library
instead of pjsua.
Cheers,
Bill
On 5/26/2016 9:44 AM, Gianni Massi wrote:
Hi all,
Youre my hero if you decide to answer even 10% of my question.
My question regards latency in applications using the PJSUA library. I am
currently running my application on either a machine with Debian or a
Raspberry PI board with Raspbian OS using the ALSA audio backend.
I was wondering, in your experience is it possible to achieve overall
latencies < 50ms with the Pjsua library?
I found that the main limit in this is lowering the latency introduced by
audio devices, so I was wondering if you could help me understand, if there
is a limit, why it so high (in my experience I was not able to go below 42
ms for capture and 42ms for playback, not even by setting the parameter to
lower values in the pjsua_media_config object passed during library
initialization with pjsua_init()) compared to other applications that use
audio devices (e.g. any type of DAWs allows latencies below 20ms).
Is there some design requirement to have large buffers at some point in the
processing?
Is there anything that can be done by modifying the audio backend?
Can I achieve <50ms latencies by using the pjmedia library directly and not
going through pjsua?
Any help will be highly appreciated, Thanks
Gianni Massi
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org
pjsip mailing list
mailto:pjsip@lists.pjsip.org pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Gianni,
You should take a look at the alsa_dev.c code. There is logic to set the
buffer size based upon the param. It then calls
snd_pcm_hw_params_set_buffer_size_near to ask ALSA to set it to
something "near" the desired param. You should step through that logic
or insert logging to figure out how the latency is getting set.
Regards,
Bill
On 5/27/2016 6:15 AM, Gianni Massi wrote:
Follow up question if you’d be so kind to indulge in my ignorance!
While experimenting to find out the best tradeoff between audio
quality and latency, I discovered I was unable to get to the point
where quality is affected because it seems that Pjsua won’t allow me
to go below 42 ms for either play/cap latency even if I set the values
to 15ms in pjsua_media_config (I can see it from the information
printed out when a device is opened with log-level set to 6).
Do you think there is something that can be done here? It all ends in
the ALSA lib method in alsa_dev.c that gets the nearest possible
buffer size (snd_pcm_hw_params_set_buffer_size_near) as far as I can
tell. Do you have any suggestions regarding this? Any other setting
that I might change in order to get smaller values returned from this
function? My understanding of Alsa is very limited, any pointer in the
right direction would be of great help.
Also I’m achieving much better results by compiling the library with
the audio switchboard enabled (“#define PJMEDIA_CONF_USE_SWITCH_BOARD
1” in config_site.h) but I still experience the same limit while
experimenting.
Any suggestions?
Thanks,
Gianni
*Da:*pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Bill
Gardner
Inviato: giovedì 26 maggio 2016 18:50
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] Minimum latency achievable aside
Hi Gianni,
I don't know what minimum latencies you can get with ALSA on PI,
you'll just have to experiment. DAWs typically use ASIO format which
is designed for low latency operation, I have no idea how hard it
would be to implement ASIO or an alternative to ALSA on PI. As you
already know most of the pjsua latency is in the audio device
buffering. There is also a wsola buffer which holds recorded samples
from the microphone awaiting processing by the bridge, this buffer is
needed to compensate for differing playback and record sampling rates.
And, there is the jitter buffer which is obviously essential to
compensate for network delivery timing, missing packets, etc. There
isn't any latency in the media path which isn't essential, more
latency allows more variation in the behavior of the audio device or
network. You can tweak the parameters to decrease latency at the risk
of getting getting poor audio quality when the audio device or network
doesn't behave optimally.
I can't see there would be an advantage in using just the pjmedia
library instead of pjsua.
Cheers,
Bill
On 5/26/2016 9:44 AM, Gianni Massi wrote:
Hi all,
You’re my hero if you decide to answer even 10% of my question.
My question regards latency in applications using the PJSUA
library. I am currently running my application on either a machine
with Debian or a Raspberry PI board with Raspbian OS using the
ALSA audio backend.
I was wondering, in your experience is it possible to achieve
overall latencies < 50ms with the Pjsua library?
I found that the main limit in this is lowering the latency
introduced by audio devices, so I was wondering if you could help
me understand, if there is a limit, why it so high (in my
experience I was not able to go below 42 ms for capture and 42ms
for playback, not even by setting the parameter to lower values in
the pjsua_media_config object passed during library initialization
with pjsua_init()) compared to other applications that use audio
devices (e.g. any type of DAWs allows latencies below 20ms).
Is there some design requirement to have large buffers at some
point in the processing?
Is there anything that can be done by modifying the audio backend?
Can I achieve <50ms latencies by using the pjmedia library
directly and not going through pjsua?
Any help will be highly appreciated, Thanks
/Gianni Massi/
_______________________________________________
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org <mailto: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
Thanks for pointing me in the right direction and indulging in my ignorance.
If I make any meaningful step in this direction I will write about it here,
but the latency I achieved by switching to the switchboards might be enough
for my needs, Ill know for sure when I do some test.
Thanks for your help Bill, have a nice weekend!
Gianni
Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Bill Gardner
Inviato: venerdì 27 maggio 2016 16:45
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] R: Minimum latency achievable aside
Hi Gianni,
You should take a look at the alsa_dev.c code. There is logic to set the
buffer size based upon the param. It then calls
snd_pcm_hw_params_set_buffer_size_near to ask ALSA to set it to something
"near" the desired param. You should step through that logic or insert
logging to figure out how the latency is getting set.
Regards,
Bill
On 5/27/2016 6:15 AM, Gianni Massi wrote:
Follow up question if youd be so kind to indulge in my ignorance!
While experimenting to find out the best tradeoff between audio quality and
latency, I discovered I was unable to get to the point where quality is
affected because it seems that Pjsua wont allow me to go below 42 ms for
either play/cap latency even if I set the values to 15ms in
pjsua_media_config (I can see it from the information printed out when a
device is opened with log-level set to 6).
Do you think there is something that can be done here? It all ends in the
ALSA lib method in alsa_dev.c that gets the nearest possible buffer size
(snd_pcm_hw_params_set_buffer_size_near) as far as I can tell. Do you have
any suggestions regarding this? Any other setting that I might change in
order to get smaller values returned from this function? My understanding of
Alsa is very limited, any pointer in the right direction would be of great
help.
Also Im achieving much better results by compiling the library with the
audio switchboard enabled (#define PJMEDIA_CONF_USE_SWITCH_BOARD 1 in
config_site.h) but I still experience the same limit while experimenting.
Any suggestions?
Thanks,
Gianni
Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Bill Gardner
Inviato: giovedì 26 maggio 2016 18:50
A: pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] Minimum latency achievable aside
Hi Gianni,
I don't know what minimum latencies you can get with ALSA on PI, you'll just
have to experiment. DAWs typically use ASIO format which is designed for low
latency operation, I have no idea how hard it would be to implement ASIO or
an alternative to ALSA on PI. As you already know most of the pjsua latency
is in the audio device buffering. There is also a wsola buffer which holds
recorded samples from the microphone awaiting processing by the bridge, this
buffer is needed to compensate for differing playback and record sampling
rates. And, there is the jitter buffer which is obviously essential to
compensate for network delivery timing, missing packets, etc. There isn't
any latency in the media path which isn't essential, more latency allows
more variation in the behavior of the audio device or network. You can tweak
the parameters to decrease latency at the risk of getting getting poor audio
quality when the audio device or network doesn't behave optimally.
I can't see there would be an advantage in using just the pjmedia library
instead of pjsua.
Cheers,
Bill
On 5/26/2016 9:44 AM, Gianni Massi wrote:
Hi all,
Youre my hero if you decide to answer even 10% of my question.
My question regards latency in applications using the PJSUA library. I am
currently running my application on either a machine with Debian or a
Raspberry PI board with Raspbian OS using the ALSA audio backend.
I was wondering, in your experience is it possible to achieve overall
latencies < 50ms with the Pjsua library?
I found that the main limit in this is lowering the latency introduced by
audio devices, so I was wondering if you could help me understand, if there
is a limit, why it so high (in my experience I was not able to go below 42
ms for capture and 42ms for playback, not even by setting the parameter to
lower values in the pjsua_media_config object passed during library
initialization with pjsua_init()) compared to other applications that use
audio devices (e.g. any type of DAWs allows latencies below 20ms).
Is there some design requirement to have large buffers at some point in the
processing?
Is there anything that can be done by modifying the audio backend?
Can I achieve <50ms latencies by using the pjmedia library directly and not
going through pjsua?
Any help will be highly appreciated, Thanks
Gianni Massi
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org
pjsip mailing list
mailto:pjsip@lists.pjsip.org pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/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 mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org