I'm running vdr 1.3.21 and bitstreamout-0.70
I have tried with my MoBo's integrated soundcard ( via82xx ) and with a C- Media ( cmipci ) + latest stable alsa. Same problems with both setups.
I'm getting a A/V sync delay thats very hard to get correct by using the 'delay/livedelay' parameters. Does the delay correction actually work? I'm only asking because I know there has been major work in the AC3/DD functionality in vdr lately.
I have also problem running recorded material and bitstreamout. Actually vdr crashes without any warnings when I try to start a recording. Maybe it has to do with that I'm using ttextsubs-plugin? If I switch to a channel with 'live- DD' and use the DD-stream in the Audio setup and then start playback of the same recording, the recording starts to play - without sound.
i.e
current channel PCM (1) -> vdr crash current channel DD (33) -> recorded material play without sound
The audio meny is very intermittent when using bitstreamout. By this I mean that the OSD dpid selection does not show up all the time. Has this been corrected in 1.3.22
As before my test setup is connected to a Sony STR-DB830 which I know is marked as failure in the DD compability list. But with bitstreamout my reciever this far actually works with DD2.0 material.
On Thu, Mar 10, 2005 at 12:36:49PM +0200, Per Mellander wrote:
I'm running vdr 1.3.21 and bitstreamout-0.70
I have tried with my MoBo's integrated soundcard ( via82xx ) and with a C- Media ( cmipci ) + latest stable alsa. Same problems with both setups.
I'm getting a A/V sync delay thats very hard to get correct by using the 'delay/livedelay' parameters. Does the delay correction actually work? I'm only asking because I know there has been major work in the AC3/DD functionality in vdr lately.
I have also problem running recorded material and bitstreamout. Actually vdr crashes without any warnings when I try to start a recording. Maybe it has to do with that I'm using ttextsubs-plugin? If I switch to a channel with 'live- DD' and use the DD-stream in the Audio setup and then start playback of the same recording, the recording starts to play - without sound.
i.e
current channel PCM (1) -> vdr crash
Which CPU AMD64 or i686 aka 32bit?
current channel DD (33) -> recorded material play without sound
The audio meny is very intermittent when using bitstreamout. By this I mean that the OSD dpid selection does not show up all the time. Has this been corrected in 1.3.22
Yep, it should work with 1.3.22. At least I've never seen a crash with 1.3.22. The delay problem with MP2 is know but it is not that easy to fix. On the other hand: using the direct way of connecting the S/P-DIF out on J2 of the DVB FF to the S/P-DIF in of the sound card works flawless, e.g. is in sync.
Werner
"Having a smoking section in a restaurant is like having a peeing section in a swimming pool." -- Edward Burr
Werner,
This is one of the best taglines I've seen recently! And the best thing, it's 100% TRUE! (I hope that the anti-smoking laws should be more strict everywhere in the world!)
René
On Thu, 10 Mar 2005 12:03:17 +0100, Dr. Werner Fink wrote
current channel PCM (1) -> vdr crash
Which CPU AMD64 or i686 aka 32bit?
32bit
current channel DD (33) -> recorded material play without sound
The audio meny is very intermittent when using bitstreamout. By this I
mean
that the OSD dpid selection does not show up all the time. Has this been corrected in 1.3.22
Yep, it should work with 1.3.22. At least I've never seen a crash with 1.3.22. The delay problem with MP2 is know but it is not that easy to fix. On the other hand: using the direct way of connecting the S/P-DIF out on J2 of the DVB FF to the S/P-DIF in of the sound card works flawless, e.g. is in sync.
Then I'll try with direct connecting and 1.3.22. I have one FF with J2, and another with 'true' coax output ( aka. Nexus ). Can I loop the Nexus thru soundcard aswell?
/Mel
Hi,
On Thu, Mar 10, Dr. Werner Fink wrote:
Yep, it should work with 1.3.22. At least I've never seen a crash with 1.3.22. The delay problem with MP2 is know but it is not that easy to fix. On the other hand: using the direct way of connecting the S/P-DIF out on J2 of the DVB FF to the S/P-DIF in of the sound card works flawless, e.g. is in sync.
it is in sync, but the pcm headers are wrong with ac3 stream. Bye the way, may it be possibles to receive the data via S/P-DIF in into the soundcard and then read the data via a userspace programm to redirect it to S/P-DIF out.
Then the data should be fine for all amplifiers, or I'am total wrong ?
Le jeudi 10 mars 2005 à 13:53 +0200, Rene Hertell a écrit :
"Having a smoking section in a restaurant is like having a peeing section in a swimming pool." -- Edward Burr
Werner,
This is one of the best taglines I've seen recently! And the best thing, it's 100% TRUE! (I hope that the anti-smoking laws should be more strict everywhere in the world!)
An Uzi can help... Oh look you have holes in your lungs, smoking here is dangerous for your health =:-D
Tony
On Thu, Mar 10, 2005 at 01:18:37PM +0100, Dieter Bloms wrote:
Hi,
On Thu, Mar 10, Dr. Werner Fink wrote:
Yep, it should work with 1.3.22. At least I've never seen a crash with 1.3.22. The delay problem with MP2 is know but it is not that easy to fix. On the other hand: using the direct way of connecting the S/P-DIF out on J2 of the DVB FF to the S/P-DIF in of the sound card works flawless, e.g. is in sync.
it is in sync, but the pcm headers are wrong with ac3 stream.
What does wrong mean? IMHO the headers which bitstreamout produce should be correct accordingly to IEC 61937. This works with AC3, DTS and even with MP2 Audio. Only for the later one my AV receiver has no auto detection which requires switching my AV receiver into MP2 Audio mode by hand.
Bye the way, may it be possibles to receive the data via S/P-DIF in into the soundcard and then read the data via a userspace programm to redirect it to S/P-DIF out.
This is what a sound card should do with loop through, that means redirect the data from its S/P-DIF in to its S/P-DIF out _without_ modifying (e.g. _NO_ resampling, clipping or so whatever sound `washing' which kill the pure compressed data).
Then the data should be fine for all amplifiers, or I'am total wrong ?
No, this is correct.
Werner
On Thu, Mar 10, 2005 at 03:55:11PM +0200, Per Mellander wrote:
On Thu, 10 Mar 2005 12:03:17 +0100, Dr. Werner Fink wrote
current channel PCM (1) -> vdr crash
Which CPU AMD64 or i686 aka 32bit?
32bit
current channel DD (33) -> recorded material play without sound
The audio meny is very intermittent when using bitstreamout. By this I
mean
that the OSD dpid selection does not show up all the time. Has this been corrected in 1.3.22
Yep, it should work with 1.3.22. At least I've never seen a crash with 1.3.22. The delay problem with MP2 is know but it is not that easy to fix. On the other hand: using the direct way of connecting the S/P-DIF out on J2 of the DVB FF to the S/P-DIF in of the sound card works flawless, e.g. is in sync.
Then I'll try with direct connecting and 1.3.22. I have one FF with J2, and another with 'true' coax output ( aka. Nexus ). Can I loop the Nexus thru soundcard aswell?
Depends on your sound card, if it has a S/P-DIF in connector on the out side of its slot sheet metal and is able to handle the S/P-DIF out of the Nexus, it should work.
You have to play with amixer to get a corredt setting to loop through the data from the S/P-DIF in connector to the S/P-DIF out connector. See the mute scripts in the mute directory for some examples. But note: These _are_ examples and be aware the the sound card never modifies the data streams for AC3 (no resampling, no clipping, no washing of so what ever kind).
Werner
On Thu, 10 Mar 2005 14:13:32 +0100, Dr. Werner Fink wrote
Depends on your sound card, if it has a S/P-DIF in connector on the out side of its slot sheet metal and is able to handle the S/P-DIF out of the Nexus, it should work.
You have to play with amixer to get a corredt setting to loop through the data from the S/P-DIF in connector to the S/P-DIF out connector. See the mute scripts in the mute directory for some examples. But note: These _are_ examples
It's a CMI8738 based card, Innoax Audio Extreme 5.1 with both coax and toslink in/out capabilities.
alsamixer has a settings for 'Loop mode'.
The Coax and Toslink connectors are mounted on a separate slot connected to the soundcard via flat cable connector. There is some buffering components on the extra card with the connectors.
and be aware the the sound card never modifies the data streams for AC3 (no resampling, no clipping, no washing of so what ever kind).
Not even setting the 'non-audio'-bit ;)
/Mel
On Thu, Mar 10, 2005 at 05:05:27PM +0200, Per Mellander wrote:
On Thu, 10 Mar 2005 14:13:32 +0100, Dr. Werner Fink wrote
Depends on your sound card, if it has a S/P-DIF in connector on the out side of its slot sheet metal and is able to handle the S/P-DIF out of the Nexus, it should work.
You have to play with amixer to get a corredt setting to loop through the data from the S/P-DIF in connector to the S/P-DIF out connector. See the mute scripts in the mute directory for some examples. But note: These _are_ examples
It's a CMI8738 based card, Innoax Audio Extreme 5.1 with both coax and toslink in/out capabilities.
alsamixer has a settings for 'Loop mode'.
The you should go with mute/cmi8738.sh :^)
The Coax and Toslink connectors are mounted on a separate slot connected to the soundcard via flat cable connector. There is some buffering components on the extra card with the connectors.
and be aware the the sound card never modifies the data streams for AC3 (no resampling, no clipping, no washing of so what ever kind).
Not even setting the 'non-audio'-bit ;)
This is done by the plugin its self.
Werner
On Thu, 10 Mar 2005 14:36:04 +0100, Dr. Werner Fink wrote
Not even setting the 'non-audio'-bit ;)
This is done by the plugin its self.
Just for clearification. As I understand it when bitstreamout plugin is used in 'loop mode', the only thing thats done by the soundcard is to make sure the headers in SPDIF stream are correct?
/Mel
On Thu, Mar 10, 2005 at 06:05:44PM +0200, Per Mellander wrote:
On Thu, 10 Mar 2005 14:36:04 +0100, Dr. Werner Fink wrote
Not even setting the 'non-audio'-bit ;)
This is done by the plugin its self.
Just for clearification. As I understand it when bitstreamout plugin is used in 'loop mode', the only thing thats done by the soundcard is to make sure the headers in SPDIF stream are correct?
You may read the manual page ... you can give the mute script as an option to the plugin. Then the plugin uses this to mute/unmute and enable loop or not loop mode depending on your configuration in the plugins menu entry.
Werner
Tested all night and as said earlier, by using 'loop mode' the A/V sync problem is solved.
My problem still apply - DD/AC3 does not work! Reciever detects both DD2.0 and DD5.1 but as before there is no sound.
I double checked by disconnecting SPDIF cable from DVB card to CMI8738 card to make sure the loop mode was 'active'. There must be something 'fishy' in my setup, or maybe my reciever will never work with this setup. ( Which is strange because it works very well with my standalone DVD player. )
@Werner:
Any diagnostic test ideas?
I might have missed if there is any tool in your plugin to natively send DD/AC3 from the CMI8738. I will search for an easy way to test what happens if I send DD/AC3 from CMI8738 to reciever. ( ie not a DVB/VDR stream. )
/Mel
Hi,
On Thu, Mar 10, Dr. Werner Fink wrote:
it is in sync, but the pcm headers are wrong with ac3 stream.
What does wrong mean? IMHO the headers which bitstreamout produce should be correct accordingly to IEC 61937. This works with AC3, DTS and even with MP2 Audio. Only for the later one my AV receiver has no auto detection which requires switching my AV receiver into MP2 Audio mode by hand.
The nonaudio bit is not set and my sony amplifier stumble on it.
This is what a sound card should do with loop through, that means redirect the data from its S/P-DIF in to its S/P-DIF out _without_ modifying (e.g. _NO_ resampling, clipping or so whatever sound `washing' which kill the pure compressed data).
I know, but I want the soundcard to modifying the pcm data, so that the nonaudio bit is set. My amplifier works great With your bitstreamout plugin, but I can't get the sound in sync. And with loopthrough the nonaudiobit isn't set :( I use the soundcard "C-Media Electronics Inc CM8738 (rev 10)".
On Fri, Mar 11, 2005 at 08:33:17AM +0100, Dieter Bloms wrote:
Hi,
On Thu, Mar 10, Dr. Werner Fink wrote:
it is in sync, but the pcm headers are wrong with ac3 stream.
What does wrong mean? IMHO the headers which bitstreamout produce should be correct accordingly to IEC 61937. This works with AC3, DTS and even with MP2 Audio. Only for the later one my AV receiver has no auto detection which requires switching my AV receiver into MP2 Audio mode by hand.
The nonaudio bit is not set and my sony amplifier stumble on it.
Then your sound card does not support the nonaudio bit or you've set the Z680 option to yes, taht means skip setting of the nonaudio bit. You may try to set the Z680 option to `no'.
This is what a sound card should do with loop through, that means redirect the data from its S/P-DIF in to its S/P-DIF out _without_ modifying (e.g. _NO_ resampling, clipping or so whatever sound `washing' which kill the pure compressed data).
I know, but I want the soundcard to modifying the pcm data, so that the nonaudio bit is set.
Sorry but the nonaudio bit has _nothing_ to do with the PCM stream its self. The nonaudio bit is set in the S/P-DIF data 32bit data word stream which is used to transport the 16bit PCM data word stream. Within this S/P-DIF 32bit data words there is a sectionn for a PCM 16bit word and a preamble, a parity check bit, some more bits, and the so called status bit. Within 192 S/P-DIF 32bit data words for each channel you have 192 status bits. This 192 32bit data words starts with a special preample in the first 32bit data word. Now the second bit of the 192 status bits is the nonaudio bit. If it is raised the S/P-DIF 32bit data stream is marked as nonlinear.
My amplifier works great With your bitstreamout plugin, but I can't get the sound in sync. And with loopthrough the nonaudiobit isn't set :( I use the soundcard "C-Media Electronics Inc CM8738 (rev 10)".
You can not set the nonaudio bit in loop through mode. AFAIK the ALSA driver does not support that.
Werner
On Fri, Mar 11, 2005 at 10:59:46AM +0200, Per Mellander wrote:
Tested all night and as said earlier, by using 'loop mode' the A/V sync problem is solved.
My problem still apply - DD/AC3 does not work! Reciever detects both DD2.0 and DD5.1 but as before there is no sound.
I double checked by disconnecting SPDIF cable from DVB card to CMI8738 card to make sure the loop mode was 'active'. There must be something 'fishy' in my setup, or maybe my reciever will never work with this setup. ( Which is strange because it works very well with my standalone DVD player. )
@Werner:
Any diagnostic test ideas?
Then your sound card does not support the nonaudio bit or you've set the Z680 option to yes, that means skip setting of the nonaudio bit. You may try to set the Z680 option to `no'.
I might have missed if there is any tool in your plugin to natively send DD/AC3 from the CMI8738. I will search for an easy way to test what happens if I send DD/AC3 from CMI8738 to reciever. ( ie not a DVB/VDR stream. )
Natively send DD/AC3 would cause a very noisy environment. There are two method to transport DD/AC3 over S/P-DIF aka IEC (60)958 ... one is a 32bit method and the other is the 16bit embbeding method into a PCM stream accordingly to IEC 61937.
For the first solution you've to generate the S/P-DIF stream its self. In other words you need access and docs of the HW of a programmable sound card (chip).
The second solution is more flexible: it should be usable with any sound card, and if this sound card is able to support nonlinear S/P-DIF streams which are used to transport the nonlinear PCM data most receivers out there should work.
-> http://www.epanorama.net/documents/audio/spdif.html
Werner
On Fri, 11 Mar 2005 12:15:44 +0100, Dr. Werner Fink wrote
My amplifier works great With your bitstreamout plugin, but I can't get the sound in sync. And with loopthrough the nonaudiobit isn't set :( I use the soundcard "C-Media Electronics Inc CM8738 (rev 10)".
You can not set the nonaudio bit in loop through mode. AFAIK the ALSA driver does not support that.
Why loop at all then? What's the difference between connecting DVB card SPDIF output directly to reciever and looping via CMI8738 card ??
Does the soundcard do anything else than act as a 'dumb' connector? Pretty expensive extension ;D
On Fri, Mar 11, 2005 at 03:37:05PM +0200, Per Mellander wrote:
On Fri, 11 Mar 2005 12:15:44 +0100, Dr. Werner Fink wrote
My amplifier works great With your bitstreamout plugin, but I can't get the sound in sync. And with loopthrough the nonaudiobit isn't set :( I use the soundcard "C-Media Electronics Inc CM8738 (rev 10)".
You can not set the nonaudio bit in loop through mode. AFAIK the ALSA driver does not support that.
Why loop at all then? What's the difference between connecting DVB card SPDIF output directly to reciever and looping via CMI8738 card ??
You can switch between MP2 and AC3 audio on channel change. You can mute/unmute even if digital connected.
Does the soundcard do anything else than act as a 'dumb' connector? Pretty expensive extension ;D
CM8738 are extremly inexpensive ... OK beside the huge winning marges of some resellers.
Expensive sound cards cross the 100 Euro/Dollar boundary and really good sound cards can be used in slave mode. That is that the sound card does not use its own quart as clock reference but the incomming clock given by the S/P-DIF in ... which is very usefull for e.g. LiveTV.
Werner
On Fri, 11 Mar 2005 14:59:37 +0100, Dr. Werner Fink wrote
On Fri, Mar 11, 2005 at 03:37:05PM +0200, Per Mellander wrote:
On Fri, 11 Mar 2005 12:15:44 +0100, Dr. Werner Fink wrote
My amplifier works great With your bitstreamout plugin, but I can't
get
the sound in sync. And with loopthrough the nonaudiobit isn't set :( I use the soundcard "C-Media Electronics Inc CM8738 (rev 10)".
You can not set the nonaudio bit in loop through mode. AFAIK the ALSA driver does not support that.
Why loop at all then? What's the difference between connecting DVB card
SPDIF
output directly to reciever and looping via CMI8738 card ??
You can switch between MP2 and AC3 audio on channel change. You can mute/unmute even if digital connected.
Does the soundcard do anything else than act as a 'dumb' connector?
Pretty
expensive extension ;D
CM8738 are extremly inexpensive ... OK beside the huge winning marges of some resellers.
Expensive sound cards cross the 100 Euro/Dollar boundary and really good sound cards can be used in slave mode. That is that the sound card does not use its own quart as clock reference but the incomming clock given by the S/P-DIF in ... which is very usefull for e.g. LiveTV.
Werner
Point taken ;) Thanks for your efforts in explaining the subject.
Have a nice weekend!
/Mel
-
Dieter Bloms -
Dr. Werner Fink -
Per Mellander -
Rene Hertell -
tony