XBMC over Jack (Revisited)

I updated my ubuntu to karmic (9.10) lately, which works great, though my hacked version of XBMC cannot compile, after some hacks, it compiles fine, though gave me segment fault when starting. So I decided to take some time to install the current version of XBMC and apply my jack patch on it.

This time I decide to use the version in PPA's karmic release instead the svn version (the previous one on svn was not very stable, crash now and then), first add xbmc and nvidia-vdpau repositories (xbmc depends on nvidia-190-libvdpau-dev, though I am still using 185 since the 190 version crash with tv-out, might goes to another post)

sudo add-apt-repository ppa:team-xbmc/ppa
sudo add-apt-repository ppa:nvidia-vdpau/ppa

You still need to add the source part into apt source list, (add-apt-repository can get the key for you, which is convenient, wish it can support an option to add the source part too). Add this line to /etc/apt/sources.list.d/team-xbmc-ppa-karmic.list

deb-src http://ppa.launchpad.net/team-xbmc/ppa/ubuntu karmic main

Then install the needed libraries for building xbmc, and download the source codes.

sudo apt-get build-dep xbmc
apt-get source xbmc
sudo apt-get install libbio2jack0-dev libjack-dev libjack0.100.0-dev libjackasyn-dev

You should have the source codes ready after a while. As a comments in my previews post said, the audio interface was changed a little bit in the new xbmc source, though fortunately it's not a big change, so the previous patch can works just fine after some minor changes. get the patch from this address.
* https://sites.google.com/site/yjpark/downloads (xbmc-9.11~beta1-jack.patch)

patch -p0 < xbmc-9.11~beta1-jack.patch

Then do the normal configure, before make, please add "-ljack" into Makefile, (search the line LIBS=..., add it to the end of the line will be fine), after that make should work.

One important thing to note, to make the audio matching video, you probably should change the delay constant in xbmc/cores/AudioRenderers/JackDirectSound.cpp.

float CJackDirectSound::GetDelay()
{
Update();

return m_timePerPacket * (float)m_packetsSent + 0.4;
}

Just play an correct video file, adjust audio delay to make it sync, then add the value you used into the end of the return line will be ok (ahead means plus)

Note: C++ library for Jack is from Jack_CPP (http://x37v.info/jack_cpp/doc/index.html) (by Alex Norman, with pretty good documetation and examples, thanks again), I included the needed files in the patch, though suggest you to look at the documents on his site for more information.

What's next

Currently this is not working if enabling real time support in jackd, probably will do some research on this part.

Also plan to find a way to deal with the ugly delay part, ideally can calculate the delay, if too hard, probably will retrieve the value from config, then won't need rebuild for it.

And make the library detecting with the proper way, both in code and also in configure process (not a c/cpp programmer, so might need some time), then no manual change needed.

If I can finish all these, then will try to commit the patch to xbmc.

XBMC over JACK

I have a XBox, which I use it watching video mostly, by installing XBMC (http://xbmc.org), it's the best media player IMO, though the very aged CPU cannot take H.264 or X.264 decode, it can only support MPEG2 HD, so after a while, I found out that it's not as useful as before.

While now the XBMC is available on most platform: Windows, OS/X (Plex is better IMO, which is based on XBMC, http://www.plexapp.com/), and Linux.

XBMC works fine on ubuntu through ALSA, though I am using JACK (http://jackaudio.org/) for better sound quality, so I have to stop JACK before start XBMC which is quite annoying.

After some research on the web, I decide to implement JACK audio bridge for XBMC, which is quite easy, here is how I did it:

First, get the source code

* http://xbmc.org/development/svn/

XBMC already has a flexible structure to support different audio interfaces, under linuxport/XBMC/xbmc/cores/AudioRenderers, there are a bunch of supported interfaces: ALSA, PulseAudio, Windows Direct sound.

Seconds, get the library

JACK comes with C library, it's asynchronus and need some time to learn it, did some research, found out there is a good C++ library at http://x37v.info/jack_cpp/doc/index.html (by Alex Norman, with pretty good documetation and examples, thanks a lot)

Now, link them together

To make it simple, I just put all the files from jackcpp under AudioRenderers, then write a simple wrapper to let them talk with each other.

There is a NullDirectSound.cpp there, which is a very good example to learn how to write a new interface.

In AudioRendererFactory.cpp, a quick hacky way to use the JACK interface:

--- AudioRendererFactory.cpp    (revision 19572)
+++ AudioRendererFactory.cpp    (working copy)
@@ -22,6 +22,7 @@
#include "stdafx.h"
#include "AudioRendererFactory.h"
#include "NullDirectSound.h"
+#include "JackDirectSound.h"

#ifdef HAS_PULSEAUDIO
#include "PulseAudioDirectSound.h"
@@ -59,10 +60,14 @@
{
 IAudioRenderer* audioSink = NULL;

+//For Jack
+  audioSink = new CJackDirectSound();
+  ReturnOnValidInitialize();
+

Then create JackDirectSound.cpp and JackDirectSound.h (I just copied from NullDirectSound.cpp and NullDirectSound.h)

if you see the diff from JackDirectSound.cpp and NullDirectSound.cpp, you will find out most of them are same except for the change of name, the only logic I added are these lines (full version below):

In Initialize():

 jackBuffer = new JackCpp::BlockingAudioIO("XBMC.Jack", iChannels, iChannels);
 jackBuffer->start();
 for(int i = 0; i <>connectToPhysical(i,i);
 }
   m_uiChannels = iChannels;

I am using blocking interface here, since it's very simple to use.

In Deinitialize():

 if (jackBuffer) {
   for(int i = 0; i <>disconnectOutPort(i);
   }
   jackBuffer->close();
   //TODO: Cannot delete jackBuffer, otherwise will crash.
   //delete jackBuffer;
 }
 jackBuffer = 0;

And in AddPackets():

 CLog::Log(LOGERROR,"Jack.AddPackets() len=%d, add=%d", len, add);

 if (jackBuffer){
   short* pSamples = (short*)data;
   for (int i=0; i< j =" 0;">write(j, (float) pSamples[i*m_uiChannels + j] / 32768.0);
     }
   }
 }

The logic to convert the data chunks here cause me some trouble, had to do some search and experiments before get the sound right, but before that some result sound were quite interesting.

The last thing in GetDelay():

return m_timePerPacket * (float)m_packetsSent + 0.325 + 0.075;

To be honest, I don't know how to calculate the correct delay value here, this value here was the delay I found out on my machine, just try to play some movie, then try to match the audio to the video, then it's done. (I know, it's very hacky and not the right way, but since the constant value seems to solve my problem perfectly, don't feel pressure to dig into it anymore :) )

Put everything together and make it

Now every thing is ready, you can set it up and make it.

I suppose you have enough knowledge about how to compile XBMC on your system, this page http://xbmc.org/wiki/?title=Installing_XBMC_for_Linux has all the information you need.

Modify linuxport/XBMC/xbmc/cores/AudioRenderers/Makefile.in to include the new files (both jack_cpp and my codes)

--- Makefile.in    (revision 19572)
+++ Makefile.in    (working copy)
@@ -4,11 +4,17 @@

ifeq ($(findstring osx,$(ARCH)), osx)
SRCS = \
+    jackaudioio.cpp \
+    jackblockingaudioio.cpp \
+    JackDirectSound.cpp \
   NullDirectSound.cpp \
   AudioRendererFactory.cpp \
   PortaudioDirectSound.cpp
else
SRCS = \
+    jackaudioio.cpp \
+    jackblockingaudioio.cpp \
+    JackDirectSound.cpp \
   NullDirectSound.cpp \
   AudioRendererFactory.cpp \
   ALSADirectSound.cpp \

And also add jack lib into XBMC/Makefile, search for "-lasound" add "-ljack" the the same line. (Again, this is my hacky and improper way)

Now you are ready to compile it and enjoy the two fantastic software working together!

Download the Patch:

xbmc-jack.tgz

Better Audio Under Linux with TerraTec DMX6Fire (ICE1712 based)

Some background

I've been using Linux as main OS for more than two years, here is my experience of the audio on linux so far.

At the early stage of my linux usage, I spent some time to figure out how to use sound on linux, setup alsa, and use optical output (s/pdif) for better quality(the poor built-in sound card in the Desktop can produce pretty clean sound with this, but you need some device can accept optical in, I used an old minidisc player for that).

After switch to a Thinkpad t60 as my main development machine, I don't have optical output anymore, then I noticed that the sound quality was pretty poor with the default sound output, I didn't dig into that, just use my cell phone for music listening in work.

I have a windows PC at home for a long time, the biggest reason is I have a good sound card on it (TerraTec DMX6Fire, cost me quite some money years before), and I have some musical software on windows. After not doing anything with the software for a few years, and feeling more and more comfortable with Linux, I decide to reinstall it with Ubuntu too, then leads to this article.

The sound quality is ... POOL

Ubuntu install pretty smooth on this machine, find all the hardware including the sound card, reboot correctly, and sound works, with a pretty bad poor quality, and the digital output(Both optical and coaxial) on the card stop working anymore, the sound quality from the analog out is terrible, very not clear, it's impossible to enjoy any music on this pretty cool sound card with the default configuration at all!

Since this is Linux and the hardware is already been recognized, I think it's just something related to the configuration, so I started to google around and try to fix this, cost me about two days so far, here is what I've learned from this.

* http://alsa.opensrc.org/index.php/Ice1712

How to make it better

The biggest reason to cause the messy sound is the configuration of the card, since it's a sort of professional class card, it's not as simple as most consumer card, need some setup to make it working in the right way, there is a graphics tool for ICE1712 based cards. it's name is envy24control in package alsa-tools-gui , for my card, I am using "envy24control -p 8 -w 15" to start it, "-p 8" means use 8 pcm outputs, "-w 15" to make the window wider enough to show everything without a scrollbar.

The most important here is to make the pcm channels as stereo, the cards has 10 mono channels, by default players will play sound to the first two channels as a stereo one, but if you didn't adjust the digital mixer correctly, you will hear mix-down mono sound which is pretty bad. so you should mute the right part of pcm output 1 and mute the left part of pcm output 2.

If you have experience of using a hardware mixer, then this is much easier to understand, by allowing left/right control over each channel, you can adjust the pan value(left/right position) for it, but for stereo channels, you must set the mixer as stereo by pairing two mono channels.

Then I can have rather clean stereo again, but the sound quality is still not very nice, it's much worse than using foobar2000 on windows with ASIO output with the same card. and the digital output is not working correctly yet. so I still have a lots works to do.

How to get digital and make the sound quality even better

Using envy24control to direct digital output from s/pdif out (on Patchbay/Router tab), can use iec958 as alsa output, but you need to make sure the internal clock is same with the sound you're playing, otherwise it will be too fast or too slow. you can use envy24control's Hardware Settings to change it.

Note: Be careful about the combinations with Professional/Consumer and different clock rate, I found out that Professional+44100 and Consumer+48000 works best, other combinations will cause problems. (Noise on coaxial, or wrong clock display on my amplifier). My amplifier doesn't support higher clock rate, so I don't know how that works.

There is a good post about ALSA's sample rate converter in dmix, and how to improve it, I followed it, feels that the sound is a bit better, but not very sure about the actual effect

* http://www.hydrogenaudio.org/forums/index.php?showtopic=47591

Then I decide to give JACK a try, according to it's website at http://jackaudio.org/

JACK is a low-latency audio server, written for POSIX conformant operating systems such as GNU/Linux and Apple's OS X. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server (ie. as a "plugin").

JACK was designed from the ground up for professional audio work, and its design focuses on two key areas: synchronous execution of all clients, and low latency operation.

Sounds pretty good to me, and I found some guy saying that it's sound quality is pretty good on the web, after several hours of using it, I will say, it's a very cool and very good stuff, I will stick using it for a quite long time I think. The sound quality over JACK is noticeable better, make me very satisfied, and seems with JACK, can do low latency audio/midi/synth stuffs under Linux too (you can install some good musical software with package ubuntustudio-audio and ubuntustudio-audio-plugins), I haven't tried them yet, but some of them seems pretty nice.

I will leave the JACK related stuffs in another post, includes some experiments about JACK, how to use ALSA over JACK.