Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
Hi, On Tue, 2009-07-14 at 21:45 +0300, George Kiagiadakis wrote: > 2009/7/13 Olivier Crête : > > On Sun, 2009-07-12 at 21:19 +0300, George Kiagiadakis wrote: > >> > 2009/7/10 Olivier Crête : > >> I did some more investigation. It seems that in calls between my > >> client and empathy (or with my client on both sides) over jabber, the > >> codec that is used is "SIREN". On the other hand, calls with empathy > >> on both sides use the "speex" codec, which sounds much more > >> reasonable, and calls between my client and google talk use the "PCMU" > >> codec. I also tried to do some sip calls between my client and ekiga > >> and it works quite well, also using the "PCMU" codec. > > > > You definitely want to set some codec preferences, the current state of > > my codec recommendations are on this bug along with example code: > > > > http://bugzilla.gnome.org/show_bug.cgi?id=588252 > > Thank you very much. This seems to work. I used your codec preferences > file as it is, but I had to additionally put the SIREN codec in the > list of disabled codecs to make it work. After that it fell back to > PCMU, which works. However, I am wondering why it doesn't prefer > SPEEX. Looking at the logs it seems that it doesn't even try: > ... > This happens with *all* calls I make, no matter what is on the other > side. As a result, the PCMU codec is always used. Speex is installed > though, as empathy makes use of that codec on the same computer. Any > idea? I don't see speex in there, try running it with GST_DEBUG=fsrtp*:5 and see if there is something in the log ? Btw, you can reach me faster in #farsight on freenode -- Olivier Crête olivier.cr...@collabora.co.uk signature.asc Description: This is a digitally signed message part -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge___ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
2009/7/13 Youness Alaoui : > George Kiagiadakis wrote: >> So, if I understand it right, my problem is that the SIREN codec does >> not work... I still don't understand though why this codec is used >> with my client only. What determines the codec that is used and why >> empathy to empathy calls use speex? > > Hi.. interesting! Well, SIREN is the proprietary audio codec used by > MSN. You have it installed on your system (comes with gst-plugins-bad) > so it takes it, you can give a different priority to your codecs by > using the fs_set_codec_preferences API (iirc). Anyways, SIREN is not > broken, it works quite good (it's used by aMSN with farsight2 and it > works nicely), the sound can become muted or contain noise if the volume > is too high, so maybe try to lower the volume and see if it fixes it. Nope, that doesn't work. SIREN seems completely broken to me. I have a volume element connected right before the FsConference and no matter in what volume level I set it, SIREN still produces the same noise. It sounds like an analog clock where the seconds pointer moves on each second producing a short noise. That's all I hear. No sound from the microphone. -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
2009/7/13 Olivier Crête : > Hi, > > On Sun, 2009-07-12 at 21:19 +0300, George Kiagiadakis wrote: >> > 2009/7/10 Olivier Crête : >> I did some more investigation. It seems that in calls between my >> client and empathy (or with my client on both sides) over jabber, the >> codec that is used is "SIREN". On the other hand, calls with empathy >> on both sides use the "speex" codec, which sounds much more >> reasonable, and calls between my client and google talk use the "PCMU" >> codec. I also tried to do some sip calls between my client and ekiga >> and it works quite well, also using the "PCMU" codec. > > You definitely want to set some codec preferences, the current state of > my codec recommendations are on this bug along with example code: > > http://bugzilla.gnome.org/show_bug.cgi?id=588252 Thank you very much. This seems to work. I used your codec preferences file as it is, but I had to additionally put the SIREN codec in the list of disabled codecs to make it work. After that it fell back to PCMU, which works. However, I am wondering why it doesn't prefer SPEEX. Looking at the logs it seems that it doesn't even try: (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) _tf_stream_try_sending_codecs: 0: audio PCMU clock:8000 channels:0 (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) _tf_stream_try_sending_codecs: 8: audio PCMA clock:8000 channels:0 (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) _tf_stream_try_sending_codecs: 3: audio GSM clock:8000 channels:0 (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) _tf_stream_try_sending_codecs: 100: audio telephone-event clock:8000 channels:0 events=0-15 (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) fs_codecs_to_tp: adding codec PCMU [0] (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) fs_codecs_to_tp: adding codec PCMA [8] (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) fs_codecs_to_tp: adding codec GSM [3] (:31558): tp-fs-DEBUG: stream 1 0x17a8800 (audio) fs_codecs_to_tp: adding codec telephone-event [100] This happens with *all* calls I make, no matter what is on the other side. As a result, the PCMU codec is always used. Speex is installed though, as empathy makes use of that codec on the same computer. Any idea? Best regards, George -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
Hi, On Sun, 2009-07-12 at 21:19 +0300, George Kiagiadakis wrote: > > 2009/7/10 Olivier Crête : > I did some more investigation. It seems that in calls between my > client and empathy (or with my client on both sides) over jabber, the > codec that is used is "SIREN". On the other hand, calls with empathy > on both sides use the "speex" codec, which sounds much more > reasonable, and calls between my client and google talk use the "PCMU" > codec. I also tried to do some sip calls between my client and ekiga > and it works quite well, also using the "PCMU" codec. You definitely want to set some codec preferences, the current state of my codec recommendations are on this bug along with example code: http://bugzilla.gnome.org/show_bug.cgi?id=588252 -- Olivier Crête olivier.cr...@collabora.co.uk Collabora Ltd signature.asc Description: This is a digitally signed message part -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge___ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 George Kiagiadakis wrote: >> 2009/7/10 Olivier Crête : >>> This is strange, all that FsRtpConference does to audio is to >>> encode/decode it. Which codec is used? > > Hi, > I did some more investigation. It seems that in calls between my > client and empathy (or with my client on both sides) over jabber, the > codec that is used is "SIREN". On the other hand, calls with empathy > on both sides use the "speex" codec, which sounds much more > reasonable, and calls between my client and google talk use the "PCMU" > codec. I also tried to do some sip calls between my client and ekiga > and it works quite well, also using the "PCMU" codec. > > So, if I understand it right, my problem is that the SIREN codec does > not work... I still don't understand though why this codec is used > with my client only. What determines the codec that is used and why > empathy to empathy calls use speex? > > Best regards, > George > > PS: Also, for the first problem I mentioned (that calls between my > client and gtalk don't last much because audio is cut from my client), > it seems that this is a gstreamer bug with alsasrc. I can reproduce it > with "gst-launch alsasrc ! alsasink" and it works fine when I use > "osssrc". I am going to ask the gstreamer guys about it. Sorry for > bugging you about that. > Hi.. interesting! Well, SIREN is the proprietary audio codec used by MSN. You have it installed on your system (comes with gst-plugins-bad) so it takes it, you can give a different priority to your codecs by using the fs_set_codec_preferences API (iirc). Anyways, SIREN is not broken, it works quite good (it's used by aMSN with farsight2 and it works nicely), the sound can become muted or contain noise if the volume is too high, so maybe try to lower the volume and see if it fixes it. Hope that helps. KaKaRoTo > -- > Enter the BlackBerry Developer Challenge > This is your chance to win up to $100,000 in prizes! For a limited time, > vendors submitting new applications to BlackBerry App World(TM) will have > the opportunity to enter the BlackBerry Developer Challenge. See full prize > details at: http://p.sf.net/sfu/Challenge > ___ > Farsight-devel mailing list > Farsight-devel@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/farsight-devel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkpadcYACgkQqvxLNLvrsxw6MQCfZzizCI/tJgqT3SL9MRORmQHM FxkAnj4iXkBoeUfLstWlgNpHllfbMXz5 =76lI -END PGP SIGNATURE- -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
> 2009/7/10 Olivier Crête : >> This is strange, all that FsRtpConference does to audio is to >> encode/decode it. Which codec is used? Hi, I did some more investigation. It seems that in calls between my client and empathy (or with my client on both sides) over jabber, the codec that is used is "SIREN". On the other hand, calls with empathy on both sides use the "speex" codec, which sounds much more reasonable, and calls between my client and google talk use the "PCMU" codec. I also tried to do some sip calls between my client and ekiga and it works quite well, also using the "PCMU" codec. So, if I understand it right, my problem is that the SIREN codec does not work... I still don't understand though why this codec is used with my client only. What determines the codec that is used and why empathy to empathy calls use speex? Best regards, George PS: Also, for the first problem I mentioned (that calls between my client and gtalk don't last much because audio is cut from my client), it seems that this is a gstreamer bug with alsasrc. I can reproduce it with "gst-launch alsasrc ! alsasink" and it works fine when I use "osssrc". I am going to ask the gstreamer guys about it. Sorry for bugging you about that. -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
2009/7/10 Olivier Crête :
> On Fri, 2009-07-10 at 00:39 +0300, George Kiagiadakis wrote:
>> First, I have tried doing calls between my client (on the desktop) and google
>> talk (on the laptop). This scenario mostly works, I get correct audio on both
>> sides, but after a while (~30 seconds or so) the audio stream from my client
>> stops being delivered to gtalk and instead I get a continuous noise (always
>> the same) on gtalk.
>>
>> Second, I have tried doing calls between my client and empathy. In this case,
>> although they connect fine, I don't get any audio on either side, instead I
>> get
>> a "tak tak tak" noise with exactly 1 Hz frequency. This noise seems to be
>> coming from FsConference itself. I'll try to explain what makes me believe
>> that: I have a gstreamer pipeline that looks like this:
>>
>> audiosrc -> volume -> FsConference -> liveadder -> audioresample -> volume ->
>> audiosink
>>
>> (audiosrc and audiosink can be anything. I have tried so far autoaudio{src,
>> sink}, alsa{src,sink}, oss{src, sink} and audiotestsrc.)
>>
>> Empathy has a similar one that looks like this:
>>
>> gconfaudiosrc -> volume -> level -> FsConference -> liveadder ->
>> audioresample
>> -> volume -> gconfaudiosink
>>
>> The only practical difference is the "level" element, which apparently feeds
>> a
>> small visualization widget on the empathy window that shows the level of the
>> audio coming from the microphone.
>>
>> Both sides receive this "tak" noise. Now, if I mute one side by setting the
>> volume on the "volume" element of the input to 0, the other side stops
>> getting
>> the noise. However, increasing the volume doesn't make any significant
>> difference. An important detail is that the visualization widget on empathy
>> reacts to the microphone and is not synchronized with the noise that I get on
>> the other side. So, that leads me to the conclusion that the FsConference
>> element somehow gets correct audio from the input and transforms it into this
>> noise. Thus, if it gets no audio, there is no noise.
>>
>> The same exactly happens when I am trying to make calls with my client on
>> both
>> sides. However, calls with empathy on both sides work fine. This makes me
>> wonder what can be wrong with my client. Both my client and empathy are very
>> similar applications, both using telepathy, telepathy-farsight, gstreamer and
>> even (almost) the same pipeline.
>>
>> Does anybody have any clue? What can be wrong?
>
> This is strange, all that FsRtpConference does to audio is to
> encode/decode it. Which codec is used?
I have no idea actually. I am attaching the console log of my client
from a call between my client and empathy. I hope it makes sense. Is
there anything else I could do to get more debugging information?
Best regards,
George
kcall-farsight-log
Description: Binary data
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Re: [Farsight-devel] Strange issues with telepathy-farsight and jingle
On Fri, 2009-07-10 at 00:39 +0300, George Kiagiadakis wrote:
> First, I have tried doing calls between my client (on the desktop) and google
> talk (on the laptop). This scenario mostly works, I get correct audio on both
> sides, but after a while (~30 seconds or so) the audio stream from my client
> stops being delivered to gtalk and instead I get a continuous noise (always
> the same) on gtalk.
>
> Second, I have tried doing calls between my client and empathy. In this case,
> although they connect fine, I don't get any audio on either side, instead I
> get
> a "tak tak tak" noise with exactly 1 Hz frequency. This noise seems to be
> coming from FsConference itself. I'll try to explain what makes me believe
> that: I have a gstreamer pipeline that looks like this:
>
> audiosrc -> volume -> FsConference -> liveadder -> audioresample -> volume ->
> audiosink
>
> (audiosrc and audiosink can be anything. I have tried so far autoaudio{src,
> sink}, alsa{src,sink}, oss{src, sink} and audiotestsrc.)
>
> Empathy has a similar one that looks like this:
>
> gconfaudiosrc -> volume -> level -> FsConference -> liveadder ->
> audioresample
> -> volume -> gconfaudiosink
>
> The only practical difference is the "level" element, which apparently feeds
> a
> small visualization widget on the empathy window that shows the level of the
> audio coming from the microphone.
>
> Both sides receive this "tak" noise. Now, if I mute one side by setting the
> volume on the "volume" element of the input to 0, the other side stops
> getting
> the noise. However, increasing the volume doesn't make any significant
> difference. An important detail is that the visualization widget on empathy
> reacts to the microphone and is not synchronized with the noise that I get on
> the other side. So, that leads me to the conclusion that the FsConference
> element somehow gets correct audio from the input and transforms it into this
> noise. Thus, if it gets no audio, there is no noise.
>
> The same exactly happens when I am trying to make calls with my client on
> both
> sides. However, calls with empathy on both sides work fine. This makes me
> wonder what can be wrong with my client. Both my client and empathy are very
> similar applications, both using telepathy, telepathy-farsight, gstreamer and
> even (almost) the same pipeline.
>
> Does anybody have any clue? What can be wrong?
This is strange, all that FsRtpConference does to audio is to
encode/decode it. Which codec is used?
--
Olivier Crête
olivier.cr...@collabora.co.uk
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