August 25th, 2008 by Multimedia Mike
I have been studying the FFmpeg regressions test infrastructure ever so carefully so I can decide the best way to convert all of its functionality and test coverage into the FATE system as individual test specs. I have some ideas but I never know when I will have both the time and the motivation to implement them. In the interim, I am retiring the master ‘make test’ full regression test spec. In studying the regression architecture, it’s immediately obvious that ‘make test’ invokes these 3 sub-regression suites, which each have their own test specs now:
That should help until I break down the tests even further. Oh, and I also added a test spec for ‘ffmpeg -h’. Hey, FFmpeg got into a state once where it segfaulted on help. We don’t want to let that go uncaught again.
Posted in FATE Server | 5 Comments »
August 24th, 2008 by Multimedia Mike
For his FFmpeg Summer of Code project, Jai Menon successfully completed an encoder for the Apple Lossless Audio Codec (ALAC). As you can see, Jai’s encoder is competitive with Apple’s iTunes, as well as dBpoweramp, another program that implements an ALAC encoder:
Thanks to Benjamin Larsson for setting up the graph. We’re rooting for the cyan bar to be the lowest one of each bar quartet. The code is already in the mainline tree.
The 13 samples I used for this test come from the OpenMusic CD and showcase a reasonable diversity of genres. The raw rips are available here.
Posted in Open Source Multimedia | 4 Comments »
August 23rd, 2008 by Multimedia Mike
The MOOBEX project was really the first real exposure I had to the Amarok application. For the uninitiated, Amarok is a rather full-featured KDE-based music playing application and has been around since 2003. Back when I used Linux more exclusively, I stuck to xine directly for all of my media playback needs.
How was my first impression of Amarok? Well… it’s hard to say. The program was different on each distribution that I tried. In OpenSUSE, I was thoroughly frustrated when I first tried to play a file through Amarok, or rather, when I selected an audio file through the file manager and just let the system do its thing. Amarok popped up and just sort of stared at me. Okay, I recognize that I’m throwing a bunch of formats toward Amarok that the distro has administratively determined to be “legally unsafe”. But I should still get some obvious feedback to that effect. Oh, wait, look way down there in the status bar:
When I launched an audio file on another KDE-based distro installation (Mandriva), I expected the same behavior from Amarok. To my surprise, the program threw a far more brazen error. Ubuntu has Amarok version 22.214.171.124. Mandriva sports 1.4.8. The Amarok version in Mandriva exhibited the superior UI characteristic. I understand the source of the unsupported codecs. But I wonder about the dialog discrepancy. I am not sure if this dialog behavior is a version specific thing in Amarok, or a configuration option in Amarok, or an dialog characterstic configurable in the hosting KDE libraries. It’s a minor matter, to be sure, but it still makes me wonder how individual distributions would be able to devalue something like the Adobe Flash Player if it were open source and allowed distros to make “choice” changes to the functionality.
I’m sure that the Amarok developers worked very hard to produce the best music player they could. And it probably hurts to see their fruits of their labor mangled by various distributions to the point of practical worthlessness. I know the feeling based on my work on xine.
But here’s a huge annoyance with Amarok which I suspect is a core behavior: stop should mean STOP! When I did get a file to play and then tried to stop playback, the audio kept right on going long enough to make me think that something was wrong. Eventually, the audio started to gently fade out. This is not reasonable behavior. The expected action for a stop button is an immediate halt.
Posted in Open Source Multimedia | 1 Comment »
August 22nd, 2008 by Multimedia Mike
It certainly has been a long journey for native Advanced Audio Coding (AAC) in FFmpeg. It started with a Google Summer of Code project back in FFmpeg’s inaugural FFmpeg SoC season (2006). It went unfinished. Since then, many people have endeavored to fix it up to the point where it can be included into the mainline. But it was Robert Swain who persevered toward the end goal. And now look:
$ ffmpeg -formats
D V 4xm 4X Movie
D V D 8bps QuickTime 8BPS video
D A 8svx_exp 8SVX exponential
D A 8svx_fib 8SVX fibonacci
D A aac Advanced Audio Coding
Robert profiled the new AAC decoder to be significantly faster than the libfaad, the prevailing AAC decoding solution in the open source community. Further optimization work is ongoing, as is support for more advanced coding modes. Currently, the decoder only deals with low complexity (AAC-LC), the most common variant you are likely to encounter.
And of course, thanks also to Robert for creating more FATE work for me. I can’t avoid the problem of testing perceptual audio decoders for much longer.
Posted in FATE Server, Open Source Multimedia | 5 Comments »