Portaudio Improvements
I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006 It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can). To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" /> you will then be able to use the new features fully. Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know. ~Mitch
Mitch, I will work on continuing FSComm for a thick client multi-platform solution but one of the big show stoppers for me was the lack of AEC on mod_pa. I tried the preprocessors embedded on the core by Tony that are using the speexdsp but I got no luck. The main tests I made were on a mac using the speakerphone. Do you have any experience with this type of technology? It seems to me that the only AEC available is the one implemented on speexdsp but I am completely new to this. Regards, João Mesquita On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper@gmail.com>wrote:
I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006
It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can).
To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" />
you will then be able to use the new features fully.
Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know.
~Mitch
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I did look into AEC a bit, although one of the best options for me looked like the DirectX AEC support, but obviously this would be windows only. There is oslec for linux which tries to do AEC at a driver level and is in the kernel. The speex AEC processing does not look very complex ( http://www.speex.org/docs/manual/speex-manual/node7.html using speex_echo_playback/speex_echo_capture). Of course easier said then done, and does require passing it the audio in the structure it expects. One of my main concerns is I do not also know about how good the speex AEC processing is for it to be worthwhile and would be curious if it has been found to give good results in voip settings (as I only saw to the contrary). Finally, it probably hasn't been too high on my list due to the fact that I have been using my client with a headset so the need is not really there. ~Mitch 2011/1/21 João Mesquita <jmesquita@freeswitch.org>
Mitch, I will work on continuing FSComm for a thick client multi-platform solution but one of the big show stoppers for me was the lack of AEC on mod_pa.
I tried the preprocessors embedded on the core by Tony that are using the speexdsp but I got no luck. The main tests I made were on a mac using the speakerphone. Do you have any experience with this type of technology? It seems to me that the only AEC available is the one implemented on speexdsp but I am completely new to this.
Regards, João Mesquita
I also would like to contribute more on FSComm when I have more free time. I actually "stoled" some code from FSComm to a client I was working on. We use a headset so AEC is not a big deal for now. But I still would like to see AEC work for this. Actually someone reported ECHO even with a headset I haven't confirm. Is it possible to get OSLEC work on Windows/Mac ? In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this? Thanks. On Sat, Jan 22, 2011 at 2:22 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
I did look into AEC a bit, although one of the best options for me looked like the DirectX AEC support, but obviously this would be windows only. There is oslec for linux which tries to do AEC at a driver level and is in the kernel. The speex AEC processing does not look very complex (http://www.speex.org/docs/manual/speex-manual/node7.html using speex_echo_playback/speex_echo_capture). Of course easier said then done, and does require passing it the audio in the structure it expects. One of my main concerns is I do not also know about how good the speex AEC processing is for it to be worthwhile and would be curious if it has been found to give good results in voip settings (as I only saw to the contrary).
Finally, it probably hasn't been too high on my list due to the fact that I have been using my client with a headset so the need is not really there.
~Mitch
2011/1/21 João Mesquita <jmesquita@freeswitch.org>
Mitch, I will work on continuing FSComm for a thick client multi-platform solution but one of the big show stoppers for me was the lack of AEC on mod_pa. I tried the preprocessors embedded on the core by Tony that are using the speexdsp but I got no luck. The main tests I made were on a mac using the speakerphone. Do you have any experience with this type of technology? It seems to me that the only AEC available is the one implemented on speexdsp but I am completely new to this. Regards, João Mesquita
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On 01/22/2011 09:44 AM, Seven Du wrote:
I also would like to contribute more on FSComm when I have more free time. I actually "stoled" some code from FSComm to a client I was working on. We use a headset so AEC is not a big deal for now. But I still would like to see AEC work for this. Actually someone reported ECHO even with a headset I haven't confirm. Is it possible to get OSLEC work on Windows/Mac ?
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this? Echo is a big problem with headsets, especially those with the mic up near the earpiece, and a tube to the mouth. Its also a problem in traditional telephone handsets. A good IP phone handset should have an acoustic echo canceller cleaning up its mic signal. A lot (even many expensive ones) don't, and produce considerable echo if you turn up the handset volume for the hearing impaired.
Steve
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc). So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:) ~Mitch On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature. On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
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Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output. ~Mitch On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote:
Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature.
On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
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HiYou can try PortMixer for volume control that is intended to work along with PortAudio I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach Paulo From: mitch.capper@gmail.com Date: Sat, 22 Jan 2011 21:05:22 -0800 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output. ~Mitch On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote: Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature. On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating.
Adding gain is pretty simple as you are for the most part just multiplying
up the samples I believe so would not be a major item to add to portaudio, I
however decided not to go this route for two reasons. 1) In Windows each
application has its own volume setting (Vista and higher), and apps are
starting to tie into this than internal volume controls. The downside is it
doesn't distort audio/have a gain option. This is when I thought about
allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually
has its own built in volume control settings with set_audio_level.
set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a
good job at digital volume control.
Avoiding modifying the audio stream itself could also result in less
distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as
it may be the simplest method. Adding it to portaudio would work but one
of the nice advantages of set_audio_level is the fact you can also do it per
channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the
sound device volume. I tried to control sound devices in QT but
haven't find a way. Ideas on this?
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Wow, I didn't even know that somebody is working on PA2. Paulo, could you share some more information about that module? Why would somebody use it instead of the current one? Advantages/disadvantages? Is the source code available somewhere? Do you have a roadmap for the module? :) Regards, Tamas On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini@hotmail.com>wrote:
Hi You can try PortMixer for volume control that is intended to work along with PortAudio
I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach
Paulo
------------------------------ From: mitch.capper@gmail.com Date: Sat, 22 Jan 2011 21:05:22 -0800 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements
Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output.
~Mitch
On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote:
Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature.
On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
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Paulo, What is the current status of PA with non-blocking I/O? A little while ago it was not moving forward and pablio was the only way out of it. What are your plans? Maybe some of us can help? Regards, João Mesquita On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Wow, I didn't even know that somebody is working on PA2. Paulo, could you share some more information about that module? Why would somebody use it instead of the current one? Advantages/disadvantages? Is the source code available somewhere? Do you have a roadmap for the module? :)
Regards, Tamas
On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini@hotmail.com>wrote:
Hi You can try PortMixer for volume control that is intended to work along with PortAudio
I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach
Paulo
------------------------------ From: mitch.capper@gmail.com Date: Sat, 22 Jan 2011 21:05:22 -0800 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements
Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output.
~Mitch
On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote:
Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature.
On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
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It would be very cool to be able to send audio to portaudio from another program using something like PulseAudio. For instance I am running a freeswitch box with no soundcard but i want to be able to send audio from a program such as MPD to freeswitch. Could something like this be possible in PA2? 2011/1/28 João Mesquita <jmesquita@freeswitch.org>
Paulo,
What is the current status of PA with non-blocking I/O? A little while ago it was not moving forward and pablio was the only way out of it. What are your plans? Maybe some of us can help?
Regards, João Mesquita
On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot@gmail.com>wrote:
Wow, I didn't even know that somebody is working on PA2. Paulo, could you share some more information about that module? Why would somebody use it instead of the current one? Advantages/disadvantages? Is the source code available somewhere? Do you have a roadmap for the module? :)
Regards, Tamas
On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini@hotmail.com>wrote:
Hi You can try PortMixer for volume control that is intended to work along with PortAudio
I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach
Paulo
------------------------------ From: mitch.capper@gmail.com Date: Sat, 22 Jan 2011 21:05:22 -0800 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements
Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output.
~Mitch
On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote:
Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature.
On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
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HiCode is quite experimental but I intend to release it asap (probably within a few weeks)Let's see how it goes RegardsPaulo Date: Fri, 28 Jan 2011 08:48:51 -0500 From: vipkilla@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements It would be very cool to be able to send audio to portaudio from another program using something like PulseAudio. For instance I am running a freeswitch box with no soundcard but i want to be able to send audio from a program such as MPD to freeswitch. Could something like this be possible in PA2? 2011/1/28 João Mesquita <jmesquita@freeswitch.org> Paulo, What is the current status of PA with non-blocking I/O? A little while ago it was not moving forward and pablio was the only way out of it. What are your plans? Maybe some of us can help? Regards,João Mesquita On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot@gmail.com> wrote: Wow, I didn't even know that somebody is working on PA2. Paulo, could you share some more information about that module? Why would somebody use it instead of the current one? Advantages/disadvantages? Is the source code available somewhere? Do you have a roadmap for the module? :) Regards, Tamas On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini@hotmail.com> wrote: HiYou can try PortMixer for volume control that is intended to work along with PortAudio I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach Paulo From: mitch.capper@gmail.com Date: Sat, 22 Jan 2011 21:05:22 -0800 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output. ~Mitch On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote: Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature. On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating.
Adding gain is pretty simple as you are for the most part just multiplying
up the samples I believe so would not be a major item to add to portaudio, I
however decided not to go this route for two reasons. 1) In Windows each
application has its own volume setting (Vista and higher), and apps are
starting to tie into this than internal volume controls. The downside is it
doesn't distort audio/have a gain option. This is when I thought about
allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually
has its own built in volume control settings with set_audio_level.
set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a
good job at digital volume control.
Avoiding modifying the audio stream itself could also result in less
distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as
it may be the simplest method. Adding it to portaudio would work but one
of the nice advantages of set_audio_level is the fact you can also do it per
channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the
sound device volume. I tried to control sound devices in QT but
haven't find a way. Ideas on this?
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It would have been best to share it all along and let it be in contrib to get testers and contributors. There are many developers here who can give guidance etc. That's why we do this in open source. On Fri, Jan 28, 2011 at 8:30 AM, Paulo Vicentini <vizentini@hotmail.com> wrote:
Hi Code is quite experimental but I intend to release it asap (probably within a few weeks) Let's see how it goes Regards Paulo ________________________________ Date: Fri, 28 Jan 2011 08:48:51 -0500 From: vipkilla@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements
It would be very cool to be able to send audio to portaudio from another program using something like PulseAudio. For instance I am running a freeswitch box with no soundcard but i want to be able to send audio from a program such as MPD to freeswitch. Could something like this be possible in PA2?
2011/1/28 João Mesquita <jmesquita@freeswitch.org>
Paulo, What is the current status of PA with non-blocking I/O? A little while ago it was not moving forward and pablio was the only way out of it. What are your plans? Maybe some of us can help? Regards, João Mesquita
On Fri, Jan 28, 2011 at 9:54 AM, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Wow, I didn't even know that somebody is working on PA2. Paulo, could you share some more information about that module? Why would somebody use it instead of the current one? Advantages/disadvantages? Is the source code available somewhere? Do you have a roadmap for the module? :)
Regards, Tamas
On Fri, Jan 28, 2011 at 12:51 PM, Paulo Vicentini <vizentini@hotmail.com> wrote:
Hi You can try PortMixer for volume control that is intended to work along with PortAudio I am developing/testing "mod_portaudio2" that does not rely on pablio, maybe things will be better with this approach
Paulo ________________________________ From: mitch.capper@gmail.com Date: Sat, 22 Jan 2011 21:05:22 -0800 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Portaudio Improvements
Ah well viewing volume levels had not been something I had looked into. It wouldn't be too hard but the question is how do you constantly report that (FS event every second? every half second? port audio commands to return that value frequently?). If your client is windows however again the native windows volume handler for apps does also keep track of their current output.
~Mitch
On Sat, Jan 22, 2011 at 5:26 PM, Seven Du <dujinfang@gmail.com> wrote:
Thanks Mitch, I actually had tried the equivalent uuid_audio and looks like it's the only way to do it for now. We had wanted to get energy levels dynamically to show a live indication but we given up that feature.
On Sun, Jan 23, 2011 at 2:52 AM, Mitch Capper <mitch.capper@gmail.com> wrote:
It is true PA lacks volume control, it had been something I was debating. Adding gain is pretty simple as you are for the most part just multiplying up the samples I believe so would not be a major item to add to portaudio, I however decided not to go this route for two reasons. 1) In Windows each application has its own volume setting (Vista and higher), and apps are starting to tie into this than internal volume controls. The downside is it doesn't distort audio/have a gain option. This is when I thought about allowing for a fixed gain amount added to portaudio. 2) Freeswitch actually has its own built in volume control settings with set_audio_level. set_audio_level allows for 9 settings (4 lower 4 higher) and seems to do a good job at digital volume control. Avoiding modifying the audio stream itself could also result in less distortion when using actual volume controls (headsets, etc).
So if you want volume control I would say take a look at set_audio_level as it may be the simplest method. Adding it to portaudio would work but one of the nice advantages of set_audio_level is the fact you can also do it per channel rather than globally:)
~Mitch
On Fri, Jan 21, 2011 at 5:44 PM, Seven Du <dujinfang@gmail.com> wrote:
In addition to AEC. I also found PA lacks the ability to change the sound device volume. I tried to control sound devices in QT but haven't find a way. Ideas on this?
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On 01/22/2011 02:22 AM, Mitch Capper wrote:
I did look into AEC a bit, although one of the best options for me looked like the DirectX AEC support, but obviously this would be windows only. There is oslec for linux which tries to do AEC at a driver level and is in the kernel. The speex AEC processing does not look very complex (http://www.speex.org/docs/manual/speex-manual/node7.html using speex_echo_playback/speex_echo_capture). Of course easier said then done, and does require passing it the audio in the structure it expects. One of my main concerns is I do not also know about how good the speex AEC processing is for it to be worthwhile and would be curious if it has been found to give good results in voip settings (as I only saw to the contrary).
Finally, it probably hasn't been too high on my list due to the fact that I have been using my client with a headset so the need is not really there. OSLEC is not an AEC. It is a line echo canceller, and very much optimised for that role. It would be useless for acoustic echoes.
The AEC built into Windows and the speex echo canceller have a problem all echo cancellation is now suffering on modern PCs. Its a problem that keeps coming up on the speex mailing list. Like most echo cancellers they assume the mic and speaker sampling rates match. It turns out that with an increasing number of sound cards this is not the case. The sample rates are very similar, but they are not locked, and drift in relation to each other. This wrecks the performance of the echo canceller, and is not an easy problem to work around. That said, there are products, like skype, which *appear* to do a good job of echo cancellation. If you play around with skype, though, you'll notice it isn't actually echo cancelling. It seems to be adaptively juggling gains to give the appearance of clean duplex communication, and the result is fairly pleasing. I think it may be using a similar approach to the DSP Group duplex speakerphone chips from the early 90s. They were never quite as good as the kind of full AEC seakerphones I was developing around the same time. They did, however, give results far better than traditional simplex speakerphones using a DSP too underpowered to perform full echo cancellation. Steve
Steve, I was actually hoping you would tackle this thread, thank you! :-) I am not a DSP guy and to be honest, I don't think I am much of anything guy, but I did some shallow study on AEC algos... It seems that the delay between the recording and the playback can also degrade the AEC performance considerably. I personally thought that the use of pablio and the lack o lower latency audio path on mod_pa would help make the speex AEC be completely useless... I am not talking about bad performance, I am talking about doing nothing with the filters and making it even worse if you tweak too much... Now, I think that before we pursue a solution, we have to find the problem and that's the question I was going to ask you. Can you point us towards the right direction? Skype is _very_ useable on a Mac and "most" windows machines I have used. On the mac they almost never fail... Are we able to achieve that type of quality? If we are able to, in which direction should we start going towards? I honestly thought that replacing mod_pa completely was the solution at first (and creating some abstraction layer to plug in os dependent drivers such as coreaudio for mac or dx for windows directly into FS, thus reducing this delay of buffers such as pablio) but now I am almost convinced that it was never the problem. If we can decide that before anything else, I think we could make some progress. Thank you, João Mesquita On Sat, Jan 22, 2011 at 1:17 AM, Steve Underwood <steveu@coppice.org> wrote:
On 01/22/2011 02:22 AM, Mitch Capper wrote:
I did look into AEC a bit, although one of the best options for me looked like the DirectX AEC support, but obviously this would be windows only. There is oslec for linux which tries to do AEC at a driver level and is in the kernel. The speex AEC processing does not look very complex (http://www.speex.org/docs/manual/speex-manual/node7.html using speex_echo_playback/speex_echo_capture). Of course easier said then done, and does require passing it the audio in the structure it expects. One of my main concerns is I do not also know about how good the speex AEC processing is for it to be worthwhile and would be curious if it has been found to give good results in voip settings (as I only saw to the contrary).
Finally, it probably hasn't been too high on my list due to the fact that I have been using my client with a headset so the need is not really there. OSLEC is not an AEC. It is a line echo canceller, and very much optimised for that role. It would be useless for acoustic echoes.
The AEC built into Windows and the speex echo canceller have a problem all echo cancellation is now suffering on modern PCs. Its a problem that keeps coming up on the speex mailing list. Like most echo cancellers they assume the mic and speaker sampling rates match. It turns out that with an increasing number of sound cards this is not the case. The sample rates are very similar, but they are not locked, and drift in relation to each other. This wrecks the performance of the echo canceller, and is not an easy problem to work around.
That said, there are products, like skype, which *appear* to do a good job of echo cancellation. If you play around with skype, though, you'll notice it isn't actually echo cancelling. It seems to be adaptively juggling gains to give the appearance of clean duplex communication, and the result is fairly pleasing. I think it may be using a similar approach to the DSP Group duplex speakerphone chips from the early 90s. They were never quite as good as the kind of full AEC seakerphones I was developing around the same time. They did, however, give results far better than traditional simplex speakerphones using a DSP too underpowered to perform full echo cancellation.
Steve
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And one side not for the ppl who have been following this thread and might be interested on FSComm. This subject was precisely the reason why I decided to put FSComm a bit on hold. The idea is not to just make a multi-platform softphone but rather to make something that behaves better then what we currently have available. I really DON'T want to have another XLite experience (no offense) where we all use it because there's really no good alternative. Not having a decent AEC really caused me the impression that the main objective will fail dramatically... Comments on this could be interesting from anyone willing to provide their point of view. Regards, João Mesquita 2011/1/22 João Mesquita <jmesquita@freeswitch.org>
Steve, I was actually hoping you would tackle this thread, thank you! :-)
I am not a DSP guy and to be honest, I don't think I am much of anything guy, but I did some shallow study on AEC algos... It seems that the delay between the recording and the playback can also degrade the AEC performance considerably. I personally thought that the use of pablio and the lack o lower latency audio path on mod_pa would help make the speex AEC be completely useless...
I am not talking about bad performance, I am talking about doing nothing with the filters and making it even worse if you tweak too much... Now, I think that before we pursue a solution, we have to find the problem and that's the question I was going to ask you.
Can you point us towards the right direction? Skype is _very_ useable on a Mac and "most" windows machines I have used. On the mac they almost never fail... Are we able to achieve that type of quality? If we are able to, in which direction should we start going towards?
I honestly thought that replacing mod_pa completely was the solution at first (and creating some abstraction layer to plug in os dependent drivers such as coreaudio for mac or dx for windows directly into FS, thus reducing this delay of buffers such as pablio) but now I am almost convinced that it was never the problem. If we can decide that before anything else, I think we could make some progress.
Thank you, João Mesquita
On Sat, Jan 22, 2011 at 1:17 AM, Steve Underwood <steveu@coppice.org>wrote:
On 01/22/2011 02:22 AM, Mitch Capper wrote:
I did look into AEC a bit, although one of the best options for me looked like the DirectX AEC support, but obviously this would be windows only. There is oslec for linux which tries to do AEC at a driver level and is in the kernel. The speex AEC processing does not look very complex (http://www.speex.org/docs/manual/speex-manual/node7.html using speex_echo_playback/speex_echo_capture). Of course easier said then done, and does require passing it the audio in the structure it expects. One of my main concerns is I do not also know about how good the speex AEC processing is for it to be worthwhile and would be curious if it has been found to give good results in voip settings (as I only saw to the contrary).
Finally, it probably hasn't been too high on my list due to the fact that I have been using my client with a headset so the need is not really there. OSLEC is not an AEC. It is a line echo canceller, and very much optimised for that role. It would be useless for acoustic echoes.
The AEC built into Windows and the speex echo canceller have a problem all echo cancellation is now suffering on modern PCs. Its a problem that keeps coming up on the speex mailing list. Like most echo cancellers they assume the mic and speaker sampling rates match. It turns out that with an increasing number of sound cards this is not the case. The sample rates are very similar, but they are not locked, and drift in relation to each other. This wrecks the performance of the echo canceller, and is not an easy problem to work around.
That said, there are products, like skype, which *appear* to do a good job of echo cancellation. If you play around with skype, though, you'll notice it isn't actually echo cancelling. It seems to be adaptively juggling gains to give the appearance of clean duplex communication, and the result is fairly pleasing. I think it may be using a similar approach to the DSP Group duplex speakerphone chips from the early 90s. They were never quite as good as the kind of full AEC seakerphones I was developing around the same time. They did, however, give results far better than traditional simplex speakerphones using a DSP too underpowered to perform full echo cancellation.
Steve
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Mitch, I will work on continuing FSComm for a thick client multi-platform solution but one of the big show stoppers for me was the lack of AEC on mod_pa. I tried the preprocessors embedded on the core by Tony that are using the speexdsp but I got no luck. The main tests I made were on a mac using the speakerphone. Do you have any experience with this type of technology? It seems to me that the only AEC available is the one implemented on speexdsp but I am completely new to this. Regards, João Mesquita On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper@gmail.com>wrote:
I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006
It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can).
To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" />
you will then be able to use the new features fully.
Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know.
~Mitch
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Sorry, Gmail went daft on me. João Mesquita On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper@gmail.com>wrote:
I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006
It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can).
To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" />
you will then be able to use the new features fully.
Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know.
~Mitch
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I wanted to make note that these changes were committed to trunk today, you can now use it without patching. ~Mitch On Fri, Jan 21, 2011 at 9:00 AM, Mitch Capper <mitch.capper@gmail.com>wrote:
I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006
It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can).
To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" />
you will then be able to use the new features fully.
Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know.
~Mitch
participants (8)
-
Anthony Minessale
-
João Mesquita
-
Mitch Capper
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Paulo Vicentini
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Seven Du
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Steve Underwood
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Tamas Jalsovszky
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vip killa