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rongage
Joined: 10 Jun 2008
Posts: 5
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 Posted: Sun Jul 27, 2008 8:54 pm Post subject: Problems with darkice - no audio |
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I am trying to get darkice to send audio and it's not going too well. The stream is being set up on the icecast server without problem but no audio is coming out.
I *know* that the audio config on the source machine is correct - I can get MuSE to run and the audio does come up (though with significant "issues" - skipping, dropouts - probable MuSE issues). Running DarkICE from the same computer connecting to the same Icecast server from the same source (/dev/dsp) gives no audio output whatsoever.
The Icecast server is 2.3.2, the DarkIce version is 0.19.
HELP!!!!!
Ron |
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karlH
Code Warrior
Joined: 13 Jun 2005
Posts: 5476
Location: UK
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| Posted: Mon Jul 28, 2008 12:02 am Post subject: |
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check what the mixer is set to capture.
karl. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Wed Oct 08, 2008 12:44 pm Post subject: |
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Hello Karl,
| Quote: |
| check what the mixer is set to capture. |
I have the problem that, if I set mixer to "capture mux", the streaming of metadata works but I have no sound.
If I set mixer to "capture" I get the error message:
| Code: |
| EROR input-alsa/alsa_read snd_pcm_readi failed: Input/output error |
I am using Icecast with Ices2.
Do You have any recommendation?
Thaks in advance,
LowTower. |
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karlH
Code Warrior
Joined: 13 Jun 2005
Posts: 5476
Location: UK
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| Posted: Wed Oct 08, 2008 1:35 pm Post subject: |
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Usually with the alsa setups I've seen, you have a capture slider (which needs to be raised a little, start about 30%) and capture/record also needs to be enabled on Mix or at least one of the inputs.
karl. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Wed Oct 08, 2008 1:56 pm Post subject: |
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Hello karl,
thanks for Your quick response.
I have tried to raise the capture slider only a little and the stream does not work.
Even with the slider at zero but "capture" enabled the stream doesn't work.
Any further suggestions?
Cheers,
LowTower.
P.S.: I have the following sound card: 82801G (ICH7 Family) |
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karlH
Code Warrior
Joined: 13 Jun 2005
Posts: 5476
Location: UK
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| Posted: Wed Oct 08, 2008 2:17 pm Post subject: |
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capture and mix controls have both capture enabled? check with arecord, save a sample wave file, make sure that part works. There may be quirks with some devices/drivers that I'm not aware of, sometimes other settings could affect record so check with arecord, make sure that captures the audio you want. If it doesn't then the issues is driver/device related, if it does then it's something at the application end (ices2 etc).
karl. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Wed Oct 08, 2008 2:42 pm Post subject: |
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Hello again karl,
I am afraid You must guide me somehow!
I ran arecord -l and arecord -L with the following result:
| Quote: |
**** List of CAPTURE Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
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| Quote: |
default:CARD=Intel
HDA Intel, STAC92xx Analog
Default Audio Device
front:CARD=Intel,DEV=0
HDA Intel, STAC92xx Analog
Front speakers
surround40:CARD=Intel,DEV=0
HDA Intel, STAC92xx Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Intel,DEV=0
HDA Intel, STAC92xx Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Intel,DEV=0
HDA Intel, STAC92xx Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Intel,DEV=0
HDA Intel, STAC92xx Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Intel,DEV=0
HDA Intel, STAC92xx Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Intel,DEV=0
HDA Intel, STAC92xx Digital
IEC958 (S/PDIF) Digital Audio Output
hdmi:CARD=Intel,DEV=0
HDA Intel
HDMI Audio Output
null
Discard all samples (playback) or generate zero samples (capture |
Then I ran
| Code: |
| arecord -t wav -D default > test.wav |
with the following result:
| Quote: |
| Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono |
and an empty test.wav file.
But I guess recording the stdin must result in an empty file. Tried different names for the -D option (e.g. front) but then I got an error.
Something like:
| Quote: |
Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono
arecord: set_params:954: Sample format non available
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What did I wrong?
Cheers,
LowTower.
--
ices.xml:
| Quote: |
<?xml version="1.0" ?>
<ices>
<!-- run in background -->
<background>0</background>
<!-- where logs go. DONT TOUCH PLEASE! -->
<logpath>/home/MUSIK/ICECAST/ICES/logs</logpath>
<logfile>ices.log</logfile>
<logsize>2048</logsize>
<!-- 1=error, 2=warn, 3=infoa ,4=debug -->
<loglevel>2</loglevel>
<!-- logfile is ignored if this is set to 1 -->
<consolelog>0</consolelog>
<!-- optional filename to write process id to. DONT TOUCH PLEASE! -->
<pidfile>/home/MUSIK/ICECAST/ICES/dump/ices.pid</pidfile>
<stream>
<!-- metadata used for stream listing -->
<metadata>
<name>MyIceCast</name>
<genre>private</genre>
<description>My very first IceCast stream</description>
<url>http://icecast</url>
</metadata>
<!-- Input module.
This example uses the 'oss' module. It takes input from the
OSS audio device (e.g. line-in), and processes it for live
encoding. -->
<input>
<module>alsa</module>
<param name="rate">44100</param>
<param name="channels">2</param>
<param name="device">hw:0,0</param>
<!-- Read metadata (from stdin by default, or -->
<!-- filename defined below (if the latter, only on SIGUSR1) -->
<param name="metadata">1</param>
<param name="metadatafilename">/home/MUSIK/ICECAST/ICES/metadata/ices.metadata</param>
</input>
<!-- Stream instance.
You may have one or more instances here. This allows you to
send the same input data to one or more servers (or to different
mountpoints on the same server). Each of them can have different
parameters. This is primarily useful for a) relaying to multiple
independent servers, and b) encoding/reencoding to multiple
bitrates.
If one instance fails (for example, the associated server goes
down, etc), the others will continue to function correctly.
This example defines a single instance doing live encoding at
low bitrate. -->
<instance>
<!-- Server details.
You define hostname and port for the server here, along
with the source password and mountpoint. -->
<hostname>localhost</hostname>
<port>8000</port>
<password>source</password>
<mount>/icecast.ogg</mount>
<yp>1</yp> <!-- allow stream to be advertised on YP, default 0 -->
<!-- Live encoding/reencoding:
channels and samplerate currently MUST match the channels
and samplerate given in the parameters to the oss input
module above or the remsaple/downmix section below. -->
<encode>
<quality>4</quality>
<samplerate>44100</samplerate>
<channels>2</channels>
</encode>
<!-- stereo->mono downmixing, enabled by setting this to 1 -->
<downmix>0</downmix>
<!-- resampling.
Set to the frequency (in Hz) you wish to resample to, -->
<resample>
<in-rate>48000</in-rate>
<out-rate>48000</out-rate>
</resample>
</instance>
</stream>
</ices>
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karlH
Code Warrior
Joined: 13 Jun 2005
Posts: 5476
Location: UK
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| Posted: Wed Oct 08, 2008 5:57 pm Post subject: |
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something like
arecord -f cd -d 20 sample.wav
should give you 20secs of audio, about 3.5Meg. If it's <1k then the driver/device is having have trouble. You could try at 48klhz instead
karl. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Wed Oct 08, 2008 6:20 pm Post subject: |
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Hello karl,
thanks again!
I tried Your suggested
| Code: |
| arecord -f cd -d 20 sample.wav |
and get a file with 44B size and this to the shell:
| Quote: |
| Recording WAVE 'sample.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo |
With 48kHz, i.e.
| Code: |
| arecord -f dat -d 20 sample.wav |
I get the same small file and:
| Quote: |
| Recording WAVE 'sample.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo |
Don't I have to specify the device with the -D option??
Cheers and thanks for Your patience and the effort You are taking,
LowTower. |
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karlH
Code Warrior
Joined: 13 Jun 2005
Posts: 5476
Location: UK
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| Posted: Wed Oct 08, 2008 7:39 pm Post subject: |
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the 44 bytes will be the wave header, so you are not getting any data being recorded. So the problem at the moment is purely at the alsa driver end. If arecord doesn't work then apps like ices won't work either. You'll have to search for reports on your particular driver/device with regard to alsa. You may find that some other setting is required in the mixer or maybe even a module parameter needs to be set.
karl. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Thu Oct 09, 2008 5:35 am Post subject: |
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Hello karl,
a lot of thanks so far!
Well, I have the "Intel 82801G (ICH7 Family) High Definition Audio Controller" and I never had problems with it.
But now I think that some time ago (maybe before upgrading to openSuse 11.0) I was able to adjust the volume not only with "master" in kmix but also with "pcm".
I have "master", "pcm", "capture mux" and "digital" on the output tab in kmix.
But now, nothing happens when using the "pcm" slider.
In the openSuse Yast sound card configuration diaog, I can test the configuration with a sound and set the volume. And then also the "pcm" works! But when closing the dialog, then it is not!
I am google-ing about this.
But thanks a lot so far!
Cheers,
LowTower. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Fri Oct 10, 2008 5:24 am Post subject: |
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Hello karl,
me again !
I tried it yesterday with an openSUSE live CD and it worked.
I had full control over master and pcm and was able to adust the volume with both sliders.
I am going to check the configuration files (it's KDE4 on the live CD, but in general they should be similar).
Checked
| Code: |
| /etc/modprobe.d/sound |
and
and they are the same.
Do You have any suggestions what else I should look for.
Cheers,
LowTower. |
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lowtower1974
Joined: 08 Oct 2008
Posts: 7
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| Posted: Fri Oct 10, 2008 10:11 am Post subject: |
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Hello karl,
I was able to run ices and icecast with a playlist-file.
That means, the config in general is working.
As my soundcard has its problems and I am going to lose my nerves, I try to workaround.
----
What I really want to do is NOT streaming what is sent to the soundcard, but what is being played by my music player, e.g. amarok.
(I am afraid it' the same)
So, my question is:
Is there any way to create a "dynamic" playlist that is being created "on-the-fly" of the content of the amarok playlist.
Could that be possible? That means, that the playlist for icecast must be updated continuously.
Is there any chance to do so, maybe with scripting and cronjob, or whatever.
Cheers,
LowTower. |
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