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[B H de C]

Hi guys,

Newbie with JRiver I decided to give a it a try on Linux. My choice was for OpenSuse for the convenience of Yast and the fact I know this distro pretty well.

I bought a test fan less machine to be the basis for my new music server project. I went for a Shuttle XS35V2, dual core Atom at 1.8G with 4G RAM and 250G Samsung SSD.
My DAC is a Cambridge Audio DACMagic over USB.

I used these little machines with great success before.

I installed the OS from a USB stick in 10 minutes, update the OS for the next 20. I VNC into that box so got that working, and connect to that machine using tightvnc from a windoz 7 laptop. I used the iceWM window manager in my VNC display to have the vnc super light weight.

Now for JRiver install, I added the 32bits libraries like the nice post from another OpenSuse Installation thread (Thanks to him) along with rpm-build and alien, deb and libuui.
I also added the alsafirmware package and alsatools with Yast. I added audio group to my current user (and a reboot) and tested my audio with aplay -L and aplay -l and speaker-test -c2. All worked great.

Then I installed MC by transforming the .deb MC package to a rpm using Alien, then zipper install -f <rpm>, and all worked great.
I use the web interface to play my music and it worked fine, with the exception of the Internet Radio I couldn't get to work and some random crash and the functionality not working/finished but I understand it's a beta version.

The only problem is I can't play higher than 44k rate, but I think this is not a JRIver issue but a Kernel one.

Overall it was working and sounded great. I was playing music and ftp'ing some flacs at the same time and didn't even skip once. Great performance for a 10 watts silent machine.

Now I wanted to create the same system on a Samsung SSD 750G. Here I go, I install the new drive and re-install all like above and push my 400G flac collection to that machine over ftp.

BUT (of course there is one):

Now JRiver does not work. I have a blank white screen and nothing more. I can move it, so MC is not frozen or dead, But I can't see the UI. Seems like I am missing a X package.

I tried many thing but can't get to see the JRiver UI. I let the white blank screen running overnight to see but same result.

Here is my library dependency on MC:

ldd mediacenter19
        linux-gate.so.1 (0xf7725000)
        libcryptlib.so => /usr/lib/jriver/Media Center 19/libcryptlib.so (0xf73a4000)
        libboost_regex.so.1.54.0 => /usr/lib/jriver/Media Center 19/libboost_regex.so.1.54.0 (0xf72f1000)
        libX11.so.6 => /usr/lib/libX11.so.6 (0xf71a1000)
        libuuid.so.1 => /usr/lib/libuuid.so.1 (0xf719b000)
        librt.so.1 => /lib/librt.so.1 (0xf7191000)
        libpthread.so.0 => /lib/libpthread.so.0 (0xf7174000)
        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0xf7087000)
        libm.so.6 => /lib/libm.so.6 (0xf7041000)
        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0xf7024000)
        libc.so.6 => /lib/libc.so.6 (0xf6e73000)
        libdl.so.2 => /lib/libdl.so.2 (0xf6e6d000)
        libxcb.so.1 => /usr/lib/libxcb.so.1 (0xf6e4a000)
        /lib/ld-linux.so.2 (0xf7726000)
        libXau.so.6 => /usr/lib/libXau.so.6 (0xf6e46000)

I tried root user thinking it was a right issue, but got the same result.

Any ideas guys?

Thanks
B.

Note: I managed to get MC to work outside vnc. So my blank window is a vnc problem, and not a MC.

[bob]

Your vnc issue could be the color depth settings in vnc.
As far as the 44.1k issue goes, in the audio options, choose a hardware device and you'll get all of the sample rates the audio device supports.
It will be something like front:.....

[B H de C]

Hi Bob,

SOLVED !

Thank you!! After investigating all day, you put me in the right direction.

Yes indeed it was as simple as the color depth. I used 32 on the server side and that was the root cause of the problem.
I now used -depth 24 when starting the VNC server and all worked nicely.

MC updated my 300+ Gig in a minute and that was it. All good.

Do the Internet Radio work on Linux? I was unable to make it work.

For the 48k limitation, I use:
iec958:CARD=Set,DEV=0
    C-Media USB Headphone Set, USB Audio
    IEC958 (S/PDIF) Digital Audio Output

I have to dig more but I thought only Kernel 3.13 have this limitation removed and I am using the standard kernel Opensuse 3.11.10-7.

I bought a license and now hours of playing awaits me 

Thank you.

B.

[bob]

The internet radio won't work until the browsing is functional. It's on the list!
Good to hear you got it going.

[B H de C]

For the MC developers,

another recurring crash is happening when doing Audio Analysis.

It is random, and not relevant to the number of files already processed. The messages says:

[54580.172713] mediacenter19[9555]: segfault at 0 ip 00000000f50b61f2 sp 00000000ee9fbd10 error 4 in libin_FLAC.so[f4f31000+27e000]

or

[40326.336600] mediacenter19[20679]: segfault at 10 ip 00000000f6978d9e sp 00000000e7cff010 error 4 in libJRTools.so[f690e000+435000]

I have 10000+ files to processed and that will take a long time as it crashes often.

I hope it helps,

B.

[bob]

For the MC developers,

another recurring crash is happening when doing Audio Analysis.

It is random, and not relevant to the number of files already processed. The messages says:

[54580.172713] mediacenter19[9555]: segfault at 0 ip 00000000f50b61f2 sp 00000000ee9fbd10 error 4 in libin_FLAC.so[f4f31000+27e000]

or

[40326.336600] mediacenter19[20679]: segfault at 10 ip 00000000f6978d9e sp 00000000e7cff010 error 4 in libJRTools.so[f690e000+435000]

I have 10000+ files to processed and that will take a long time as it crashes often.

I hope it helps,

B.

This very likely a bad flac file. You might be able to see it in the log if logging is on.

[B H de C]

This very likely a bad flac file. You might be able to see it in the log if logging is on.

Hi Bob,
I was wondering about a log file and maybe core files. I activated it. Hopefully this is the case.

Also I see this in the log:

CAutoConfigureAudioOutput::GetOutputFormat: Start
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: Using specified output format
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: 44.1kHz 16bit 2ch
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: Finish (0 ms)
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: Start
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: Using specified output format
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: 44.1kHz 64bit 2ch
0934226: 3988761408: Playback: CAutoConfigureAudioOutput::GetOutputFormat: Finish (0 ms)

Does this mean that the flac is re-sampled to the output format specified? If this is the case I don't want that.
I thought it was only to set Volume information in the tag for playback but the content of the file remains intact.

Please could you let me know what is happening in the file data content.

Thank you.
B.

[bob]

Two buttons to the right of the repeat button is the dsp studio button.
It will be blue while playing if bit perfect.
Hovering over that while playing will tell you whats going on.
You can click on that DSP studio button and turn off output format to prevent resampling.
In tools->options->audio->advanced, there is an option called "auto configure output settings on playback error" which you could set to no.

Note that if your music won't play after that, the device you've chosen doesn't support the sample rate of your file.

[B H de C]

Two buttons to the right of the repeat button is the dsp studio button.
It will be blue while playing if bit perfect.
Hovering over that while playing will tell you whats going on.
You can click on that DSP studio button and turn off output format to prevent resampling.
In tools->options->audio->advanced, there is an option called "auto configure output settings on playback error" which you could set to no.

Note that if your music won't play after that, the device you've chosen doesn't support the sample rate of your file.

Thank you Bob.
I don't yes have a grip on all the MC features, and they are many.

I confirm, the file sample rate is untouched by the Audio Analyzer function.

I still can't play higher than 48K but that's another story.

Thanks again,
B.

[bob]

Anything that uses the system mixer will very likely be fixed to one sample rate.
So, under audio options, alsa you want to avoid
default
pulse
sysdefault
devices unless you need to be able to have multiple audio streams playing at the same time in which case you must use one of the fixed rate devices listed above.
Search the linux forum for "alsacap" which is a small executable you can run that will tell you the range of sample rates your device can handle.

[Awesome Donkey]

I'm wondering if a function like alsacap could be added within MC itself, for troubleshooting purposes of course.

[bob]

I'm wondering if a function like alsacap could be added within MC itself, for troubleshooting purposes of course.

Good suggestion!

[B H de C]

Anything that uses the system mixer will very likely be fixed to one sample rate.
So, under audio options, alsa you want to avoid
default
pulse
sysdefault
devices unless you need to be able to have multiple audio streams playing at the same time in which case you must use one of the fixed rate devices listed above.
Search the linux forum for "alsacap" which is a small executable you can run that will tell you the range of sample rates your device can handle.

Thank you for the tip, I didn't know alsacap.

It gave me:
Card 0, ID `Intel', name `HDA Intel'
  Device 0, ID `92HD81B1X5 Analog', name `92HD81B1X5 Analog', 1 subdevices (1 available)
    2 channels, sampling rate 44100..192000 Hz
    Sample formats: S16_LE, S32_LE
      Subdevice 0, name `subdevice #0'
Card 1, ID `Set', name `C-Media USB Headphone Set'
  Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available)
    2 channels, sampling rate 44100..48000 Hz
    Sample formats: S16_LE
      Subdevice 0, name `subdevice #0'

So that is why I can get higher than 48khz.

This does not make sense as my DAC is listed for up to 96k. I think the limitation is somewhere else.

But at least MC behaves correctly.

Thanks
B.

[B H de C]

Thank you for the tip, I didn't know alsacap.

It gave me:
Card 0, ID `Intel', name `HDA Intel'
  Device 0, ID `92HD81B1X5 Analog', name `92HD81B1X5 Analog', 1 subdevices (1 available)
    2 channels, sampling rate 44100..192000 Hz
    Sample formats: S16_LE, S32_LE
      Subdevice 0, name `subdevice #0'
Card 1, ID `Set', name `C-Media USB Headphone Set'
  Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available)
    2 channels, sampling rate 44100..48000 Hz
    Sample formats: S16_LE
      Subdevice 0, name `subdevice #0'

So that is why I can get higher than 48khz.

This does not make sense as my DAC is listed for up to 96k. I think the limitation is somewhere else.

But at least MC behaves correctly.

Thanks
B.

I think I have a USB limitation here as described in this http://www.edn.com/design/consumer/4376143/1/Fundamentals-of-USB-Audio
My little server does not have a USB3 port.
How do you guys manage to have sample rate > 48k over regular USB?

[InflatableMouse]

I think I have a USB limitation here as described in this http://www.edn.com/design/consumer/4376143/1/Fundamentals-of-USB-Audio
My little server does not have a USB3 port.
How do you guys manage to have sample rate > 48k over regular USB?

USB has more than enough bandwidth for a stereo audio stream. My DAC is USB and plays 32-bit 384Khz and DSD128 (1 bit/5.4Mhz).

I didn't fully read the article but I think you might have misunderstood something. In bold, it said in AND out, that's bi-directional. Under normal playing circumstances this is not the case and you could (according to that article) play 42 channels at 48kHz or 10 at 192.

A single isochronous transfer can carry 1024 bytes, and can carry at most 256 samples (at 24/32 bits). This means that a single isochronous endpoint can transfer 42 channels at 48 kHz, or 10 channels at 192 kHz (assuming that High Speed USB is used - Full Speed USB cannot carry more than a single stereo IN and OUT pair at 48 kHz).

The issue you're having I think is because you're playing to a mixer device. Check the device output in MC and make sure you've selected a device that starts with "FRONT:". If you don't have that, we need to do some troubleshooting  . You could also try a search on the Linux board for alsa, its come up a few times before.

[B H de C]

USB has more than enough bandwidth for a stereo audio stream. My DAC is USB and plays 32-bit 384Khz and DSD128 (1 bit/5.4Mhz).

I didn't fully read the article but I think you might have misunderstood something. In bold, it said in AND out, that's bi-directional. Under normal playing circumstances this is not the case and you could (according to that article) play 42 channels at 48kHz or 10 at 192.

The issue you're having I think is because you're playing to a mixer device. Check the device output in MC and make sure you've selected a device that starts with "FRONT:". If you don't have that, we need to do some troubleshooting  . You could also try a search on the Linux board for alsa, its come up a few times before.

Thank you. You are correct and I know from your experience it is doable.

I will be trying soon a simple sound card with optical out and see. My limitation is my DAC I think.

Thanks again.
B.

[B H de C]

Hi guys,

I received my new DAC, a McIntosh D100. I plugged it to my OpenSuse 13.1 machine and I now see:

Card 1, ID `Audio', name `McIntosh Labs USB Audio'
  Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available)
    2 channels, sampling rate 32000..192000 Hz
    Sample formats: S16_LE, S32_LE, S24_3LE
      Subdevice 0, name `subdevice #0'

I used speaker-test to push 192k 32bits and the McIntosh displayed it correctly. It worked nicely.

Now I am having tons of issues with MC.
When I select the device front: or iec958, it does play a song ONCE, then never again. I get freeze, crash, or nothing happens at all.

Here is when nothing works anymore:

I play a 96k 24bits songs, no problem (Metallica Enter Sadman).
Then I played a 48k 16bits (Ally And The Fiddle): nothing. It stops immediately.

In MC logs I see:
0410994: 3994016576: Playback: CMJWaveFeeder::Thread: Preparing to feed data
0410995: 3994016576: Playback: CMJWaveFeeder::Thread: Memory playback: 1; Maximum play buffer bytes: 311440179; Grow by bytes: 33554432; System available bytes: 389300224
0410995: 3994016576: Playback: CMJWaveFeeder::Thread: Running feeder loop
0410998: 3994016576: Playback: CPlayerZone::OnNewStream: Start
0410998: 3994016576: Playback: CPlayerZone::OnNewStream: Finish (0 ms)


in the /var/log/messages I see:
2014-03-22T09:58:08.245853-04:00 linux-htpc kernel: [54148.528644] ALSA clock.c:391 current rate 96000 is different from the runtime rate 44100
2014-03-22T09:58:08.248867-04:00 linux-htpc kernel: [54148.531279] ALSA clock.c:391 current rate 96000 is different from the runtime rate 44100

I don't know what's happening but it does not make sense: I can play a high bit rate songs with no problem,but not a lower.

And then, that's it. I can't play any other songs, regardless of the sample rate. MC is dead in the water.

I have to exit and start again MC for playing a new song.

Has anybody experienced such problem?

Thank you,
B.

[bob]

The MC logs should also indicate what parameters it used to open the alsa device.
The only references I could find about this online pointed towards a firmware error.
The new ALSA audio interface code in MC is very careful about how it opens the device, setting all of the parameters and querying for support of the desired sample rate before using it.

[B H de C]

The MC logs should also indicate what parameters it used to open the alsa device.
The only references I could find about this online pointed towards a firmware error.
The new ALSA audio interface code in MC is very careful about how it opens the device, setting all of the parameters and querying for support of the desired sample rate before using it.

Hi Bob,

I tried to catch the logs when the track plays, and when it does not. Here are the differences.
I hope it helps,
Thanks
B.

TRACK NOT PLAYING:

0026950: 4081056576: Playback: CALSAPlugin::OpenALSA: Start
0026969: 4081056576: Playback: CALSAPlugin::OpenALSA: Opened audio device front:CARD=Audio,DEV=0
0026969: 4081056576: Playback: CALSAPlugin::OpenALSA: Set sample rate 96000
0026969: 4081056576: Playback: CALSAPlugin::OpenALSA: Using device at bitdepth S32_LE
0026969: 4081056576: Playback: CALSAPlugin::OpenALSA: Buffer size is 48000 uFrames
0026969: 4081056576: Playback: CALSAPlugin::OpenALSA: Period size is 9600 uFrames
0026972: 4081056576: Playback: CALSAPlugin::OpenALSA: Hardware cannot resume
0026989: 4081056576: Playback: CALSAPlugin::OpenALSA: Finish (38 ms)
0026994: 4057983808: Playback: CWaveOutBin::CreateBin: Finish (48 ms)
0026994: 4057983808: Playback: CMJWaveFeeder::Thread: Preparing to feed data
0026995: 3829578560: Playback: CWaveOutBinPlayThread::Thread: Start
0026997: 3821185856: Playback: CPlayerZoneDisplayInfoLoadImageThread::Thread: Start
0026998: 3821185856: Import: JRAnalyzer::Open: Start
0026998: 3821185856: Import: JRAnalyzer::AddFile: Start
0026999: 3821185856: Import: JRAnalyzer::AddFile: Filename: /home/bob/Music/Metallica/Metallica/Metallica - Metallica - 1 - Enter Sandman.flac
0026999: 4090698560: Reader: TranscodingReader::TranscodingReader (settings): Start
0026999: 4057983808: Playback: CMJWaveFeeder::Thread: Memory playback: 1; Maximum play buffer bytes: 268435456; Grow by bytes: 134217728; System available bytes: 120872960
0026999: 4057983808: Playback: CMJWaveFeeder::Thread: Running feeder loop
0027000: 4090698560: Reader: TranscodingReader::TranscodingReader (settings): Finish (0 ms)


TRACK PLAYING:

0026094: 4115258176: Playback: CALSAPlugin::OpenALSA: Start
0026109: 4115258176: Playback: CALSAPlugin::OpenALSA: Opened audio device front:CARD=Audio,DEV=0
0026110: 4115258176: Playback: CALSAPlugin::OpenALSA: Set sample rate 96000
0026110: 4115258176: Playback: CALSAPlugin::OpenALSA: Using device at bitdepth S32_LE
0026110: 4115258176: Playback: CALSAPlugin::OpenALSA: Buffer size is 48000 uFrames
0026110: 4115258176: Playback: CALSAPlugin::OpenALSA: Period size is 9600 uFrames
0026111: 4115258176: Playback: CALSAPlugin::OpenALSA: Hardware cannot resume
0026112: 4115258176: Playback: CALSAPlugin::OpenALSA: Finish (18 ms)
0026114: 3904875328: Playback: CWaveOutBinPlayThread::Thread: Start
0026114: 4081056576: Playback: CWaveOutBin::CreateBin: Finish (27 ms)
0026114: 4081056576: Playback: CMJWaveFeeder::Thread: Preparing to feed data
0026115: 3869403968: Playback: CPlayerZoneDisplayInfoLoadImageThread::Thread: Start
0026115: 3869403968: Import: JRAnalyzer::Open: Start
0026115: 3869403968: Import: JRAnalyzer::AddFile: Start
0026115: 4081056576: Playback: CMJWaveFeeder::Thread: Memory playback: 1; Maximum play buffer bytes: 268435456; Grow by bytes: 134217728; System available bytes: 107749376
0026115: 4081056576: Playback: CMJWaveFeeder::Thread: Running feeder loop
0026115: 3869403968: Import: JRAnalyzer::AddFile: Filename: /home/bob/Music/Metallica/Metallica/Metallica - Metallica - 1 - Enter Sandman.flac
0026116: 3869403968: Import: JRAnalyzer::AddFileMJ: Start
0026116: 3869403968: Import: JRAnalyzer::AddFileMJ: Finish (0 ms)
0026116: 3869403968: Import: JRAnalyzer::AddFile: Finish (0 ms)
0026116: 3869403968: Import: JRAnalyzer::Open: Finish (1 ms)
0026117: 3869403968: General: CFileInfo::CFileInfo: Start
0026117: 3869403968: General: CFileInfo::CFileInfo: Finish (0 ms)

[bob]

In Options->Audio Settings->Track change
change from gapless to standard gapped .1 sec
and try again.

[B H de C]

In Options->Audio Settings->Track change
change from gapless to standard gapped .1 sec
and try again.

Hi Bob.
Standard gapped .1sec set (I had it to 0.7s).

Same issue. Change of sampling makes the next song not play, then nothing plays.

TRACK PLAYING:
0035520: 3955174208: Playback: CALSAPlugin::OpenALSA: Start
0035539: 3955174208: Playback: CALSAPlugin::OpenALSA: Opened audio device front:CARD=Audio,DEV=0
0035539: 3955174208: Playback: CALSAPlugin::OpenALSA: Set sample rate 96000
0035539: 3955174208: Playback: CALSAPlugin::OpenALSA: Using device at bitdepth S32_LE
0035539: 3955174208: Playback: CALSAPlugin::OpenALSA: Buffer size is 48000 uFrames
0035539: 3955174208: Playback: CALSAPlugin::OpenALSA: Period size is 9600 uFrames
0035540: 3955174208: Playback: CALSAPlugin::OpenALSA: Hardware cannot resume
0035541: 3955174208: Playback: CALSAPlugin::OpenALSA: Finish (20 ms)
0035544: 3892312896: Playback: CWaveOutBin::CreateBin: Finish (30 ms)
0035544: 3892312896: Playback: CMJWaveFeeder::Thread: Preparing to feed data
0035545: 3850349376: Playback: CWaveOutBinPlayThread::Thread: Start
0035545: 3892312896: Playback: CMJWaveFeeder::Thread: Memory playback: 1; Maximum play buffer bytes: 345191219; Grow by bytes: 134217728; System available bytes: 431489024
0035545: 3892312896: Playback: CMJWaveFeeder::Thread: Running feeder loop
0035546: 3841956672: Playback: CPlayerZoneDisplayInfoLoadImageThread::Thread: Start
0035549: 3841956672: Import: JRAnalyzer::Open: Start
0035549: 3841956672: Import: JRAnalyzer::AddFile: Start
0035549: 3841956672: Import: JRAnalyzer::AddFile: Filename: /home/bob/Music/Metallica/Metallica/Metallica - Metallica - 1 - Enter Sandman.flac
0035549: 3841956672: Import: JRAnalyzer::AddFileMJ: Start
0035549: 3841956672: Import: JRAnalyzer::AddFileMJ: Finish (0 ms)
0035549: 3841956672: Import: JRAnalyzer::AddFile: Finish (0 ms)
0035549: 3841956672: Import: JRAnalyzer::Open: Finish (0 ms)
0035550: 3841956672: General: CFileInfo::CFileInfo: Start
0035550: 3892312896: Playback: CPlayerZone::OnNewStream: Start
0035550: 3841956672: General: CFileInfo::CFileInfo: Finish (0 ms)


SECOND TRACK NOT PLAYING:
0063935: 3955174208: Playback: CALSAPlugin::OpenALSA: Start
0063937: 3955174208: Playback: CALSAPlugin::OpenALSA: Opened audio device front:CARD=Audio,DEV=0
0063937: 3955174208: Playback: CALSAPlugin::OpenALSA: Set sample rate 44100
0063937: 3955174208: Playback: CALSAPlugin::OpenALSA: Using device at bitdepth S32_LE
0063937: 3955174208: Playback: CALSAPlugin::OpenALSA: Buffer size is 22050 uFrames
0063937: 3955174208: Playback: CALSAPlugin::OpenALSA: Period size is 4410 uFrames
0063940: 3955174208: Playback: CALSAPlugin::OpenALSA: Hardware cannot resume
0063957: 3955174208: Playback: CALSAPlugin::OpenALSA: Finish (21 ms)
0063959: 3875527488: Playback: CWaveOutBin::CreateBin: Finish (55 ms)
0063959: 3850349376: Playback: CWaveOutBinPlayThread::Thread: Start
0063959: 3875527488: Playback: CMJWaveFeeder::Thread: Preparing to feed data
0063960: 3875527488: Playback: CMJWaveFeeder::Thread: Memory playback: 1; Maximum play buffer bytes: 295999897; Grow by bytes: 33554432; System available bytes: 369999872
0063960: 3875527488: Playback: CMJWaveFeeder::Thread: Running feeder loop
0063968: 3875527488: Playback: CPlayerZone::OnNewStream: Start
0063968: 3875527488: Playback: CPlayerZone::OnNewStream: Finish (0 ms)

[bob]

Most likely the device.

[B H de C]

Most likely the device.

By device you mean the Unix device, or the physical DAC?

[bob]

The DAC

[B H de C]

The DAC

The DAC? Hum a McIntosh DAC. I don't think so.
I can play all my 96/32 no problem and it works very nicely.

I think more an Alsa/USB problem. I am going to change my USB link, and also convert USB to SPDIF Coax and Optical.
I will post my findings.

I can also try on Windows, but I am trying to avoid it as best as possible.

Thanks
B.

[InflatableMouse]

There used to be an issue with an older version that stopped playback when sample rate changed with the next song in the playlist.

That was solved, but it seems you're having something similar.

It's also not unthinkable that your DAC or driver behaves differently, causing MC to stop playback.

What you could try as a workaround is using DSP studio to to resample everything to 96 or 192kHz. Does everything continue to play that way?

[bob]

..
What you could try as a workaround is using DSP studio to to resample everything to 96 or 192kHz. Does everything continue to play that way?

That's a good test. You could also try another audio device, onboard for example if you have one.
You might need to remove pulseaudio if it's installed, I've heard a few people say it can interfere with playback.
The change to standard gapped forces a re-open on each track played back. It's pretty hard to see how it could fail then unless there is a DAC firmware problem, some kind of USB problem or a system problem (driver or pulseaudio).

[B H de C]

That's a good test. You could also try another audio device, onboard for example if you have one.
You might need to remove pulseaudio if it's installed, I've heard a few people say it can interfere with playback.
The change to standard gapped forces a re-open on each track played back. It's pretty hard to see how it could fail then unless there is a DAC firmware problem, some kind of USB problem or a system problem (driver or pulseaudio).

Thanks InflatableMouse and Bob. I didn't think about it. Also I don't use pulseaudio.

I did force MC to re-sample all to 96k and all tracks are playing with no problem. So we're progressing.

[bob]

Thanks InflatableMouse and Bob. I didn't think about it. Also I don't use pulseaudio.

I did force MC to re-sample all to 96k and all tracks are playing with no problem. So we're progressing.

Even if you don't use pulse, if it's installed it can cause trouble.

[B H de C]

Even if you don't use pulse, if it's installed it can cause trouble.

Ok, I checked and pulseaudio is not even installed.

[B H de C]

Hi guys,

I do not give up easily, so I tried a couple of things:

1) I recompiled and upgraded my kernel to the latest stable 13.4 version on my OpenSuse 13.1 distro. Worked like a charm.

Now the behavior is a little bit different and better in a way.

I still can't play Flacs with different sample rate at once. The second play does not work. But if I press the stop button, I hang a little bit for a couple of seconds, then if I play again the track it works. I don't have to kill MC and restart.

When I play the different flac rate I get a message:
linux-htpc kernel: [ 7476.413271] ALSA sound/usb/clock.c:391 current rate 96000 is different from the runtime rate 44100

When I stop the play, I get:
linux-htpc kernel: [ 7576.789042] ALSA sound/core/pcm_native.c:1544 playback drain error (DMA or IRQ trouble?)

I don't think the second message is an error, but rather the consequence of the first log.

2) I tried a CM6631A USB to SPDIF coaxial optical board. No luck. I could see it with alsacap and play through with MC but I had no output signal whatsoever.
I think it was DOA.

That's all for my progress. I was using the 19.0.124 version of MC.

Thanks,

B.

[bob]

Hi guys,

I do not give up easily, so I tried a couple of things:

1) I recompiled and upgraded my kernel to the latest stable 13.4 version on my OpenSuse 13.1 distro. Worked like a charm.

Now the behavior is a little bit different and better in a way.

I still can't play Flacs with different sample rate at once. The second play does not work. But if I press the stop button, I hang a little bit for a couple of seconds, then if I play again the track it works. I don't have to kill MC and restart.

When I play the different flac rate I get a message:
linux-htpc kernel: [ 7476.413271] ALSA sound/usb/clock.c:391 current rate 96000 is different from the runtime rate 44100

When I stop the play, I get:
linux-htpc kernel: [ 7576.789042] ALSA sound/core/pcm_native.c:1544 playback drain error (DMA or IRQ trouble?)

I don't think the second message is an error, but rather the consequence of the first log.

2) I tried a CM6631A USB to SPDIF coaxial optical board. No luck. I could see it with alsacap and play through with MC but I had no output signal whatsoever.
I think it was DOA.

That's all for my progress. I was using the 19.0.124 version of MC.

Thanks,

B.

It still seems to me like something else is using the device.
From a root terminal window

lsof | grep alsa

If you get an error make sure lsof is installed.

Also note MC does NOT set system volume. You may have to run the system mixer and turn up the volume on the respective device to max.

[B H de C]

Hi Bob,

lsof | grep alsa gives me no result.

I did another test suggested by someone on the alsa community.

I played 2 files with different sample rates using aplay. It worked no problem.

The transition was perfect and the DAC displayed the new rate.

If aplay uses the same alsa lib then MC then there is a problem in MC in handling specific case like mine?

Thanks
B.

[bob]

Hi Bob,

lsof | grep alsa gives me no result.

I did another test suggested by someone on the alsa community.

I played 2 files with different sample rates using aplay. It worked no problem.

The transition was perfect and the DAC displayed the new rate.

If aplay uses the same alsa lib then MC then there is a problem in MC in handling specific case like mine?

Thanks
B.

One difference I can see is that you are using 64 bit libasound when you use aplay from the command line.
Were you able to duplicate the problem using a different output device?

[B H de C]

One difference I can see is that you are using 64 bit libasound when you use aplay from the command line.
Were you able to duplicate the problem using a different output device?

Hi Bob,
You mean a physical device? I can try I have the onboard crappy intel analog.

I also started mpd ALSA only mode with a nice frontend and it's working fine.
B.