[livesupport-dev] helix, gstreamer and v1.0 release
  • Hi,

    I've been keeping close eye on LiveSupport development lately and
    waiting for 1.0 prerelease to put it on test run.
    I'm managing technical stuff for a low budget regional FM station and
    I've been looking around for Open Source radio automation package for a
    long time as commercial packages costing around 1000-2000USD are no
    satisfactory - they all have some pretty annoying software bugs or are
    simply badly designed - and I would prefer to run radio on linux or
    macosx not on windoze.

    As you developers are discussing helix or gstreamer player inclusion
    into v1.0 I would like to give my opinion on it.

    I understand that the release date is quite a bit overdue and therefor
    there is pressure to get the thing out - but I still think making quick
    release with helixplayer would be bit pointless also for the following
    reasons:

    1) As gstreamer matures it will replace helix anyway in next release -
    right? I would consider player in Studio playout module as one of the
    Core components and I don't see point to start testing helixrealease
    when I would have start test run again when gstreamer release comes out.

    2) If with helix there is no cue support (aka multiple sound outputs)
    then it's no good for real life FM radio anyway. At least I don't see a
    workaround for this...

    3) Helix will support only mp3, no ogg - we are small FM station but we
    never saw mp3 as a true option for our music database because of bad mp3
    compression quality on desirable bitrates. MP3 especially distorts high
    frequencies - while that may not matter directly for FM broadcasting
    (limited frequncy bandwith anyway) I still would like to have music
    database which sounds as original material and is not crippled anyhow
    for other purposes in future. Ogg compression sounds much better. And
    storage space is cheap nowdays - so even uncompressed wav would be ok.
    Changing music database format is not an easy job if you have around
    5000 titles - so converting to mp3 and then later to something else is
    pointless.

    The biggest showstopper for us would be missing cue support. We could do
    test run with mp3 but in long term it's not an good choice to have music
    database as mp3-s.

    So we better wait another 1-2 weeks for gstreamer getting more mature in
    livesupport than starting livesupport discovery with helix now.

    Best Regards,
    --

    ----------------------------------------------
    Andres Toomsalu, andres@active.ee
    juhataja - general manager, O
  • 10 Comments sorted by
  • hi andres,

    thanks for your input. first of all: good to know someone will test the
    beast. now we know where you live (!) and will ping you to see what you
    make of it Wink. seriously, we will need feedback on the release from people
    in the field.

    therefore especially your take on the cue function is valid. what use is
    the 1.0 if it contains a showstopper for professional use?

    >I understand that the release date is quite a bit overdue and therefor
    >there is pressure to get the thing out

    yes. fortunately and unfortunately there is a time pressure on the 1.0
    release. in the best scenario such pressure leads to boiling down what is
    working and packaging this. in the worst scenario, as we are seeing right
    here, the pressure urges fundamental decisions to be made for the better or
    the worse, i nour case gstreamer vs. helix.

    > So we better wait another 1-2 weeks for gstreamer getting more mature in
    livesupport than starting livesupport discovery with helix now.

    or, if things work out, have the 1.0 rc with gstreamer waiting for you by
    the end of the week. it's down to akos and ferenc now to make the decision.
    we all would be happier with gstreamer, as this is the future for livesupport.

    let's see what the c++ wizzards boil for us...

    Micz Flor - micz@mi.cz

    content and media development http://mi.cz
    ------------------------------------------------------------------
    http://www.campware.org -- http://www.redall.de -- http://suemi.de
    ------------------------------------------------------------------

    ------------------------------------------
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  • Micz Flor wrote:
    > thanks for your input. first of all: good to know someone will test the
    > beast. now we know where you live (!) and will ping you to see what you
    > make of it Wink. seriously, we will need feedback on the release from
    > people in the field.

    yes indeed Smile

    > let's see what the c++ wizzards boil for us...

    I made it compile on Mandrake as well. (BTW, ls-dev is sooooo sloooow,
    I'm so happy we're migrating to one big dev server soon...)

    So I'll work on making the scheduler use gstreamer, while Ferenc can
    work on the GUI issues.


    Akos

    ------------------------------------------
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  • At 14:59 30.06.2005, you wrote:
    >So I'll work on making the scheduler use gstreamer, while Ferenc can
    >work on the GUI issues.

    have you spoken to him?

    Micz Flor - micz@mi.cz

    content and media development http://mi.cz
    ------------------------------------------------------------------
    http://www.campware.org -- http://www.redall.de -- http://suemi.de
    ------------------------------------------------------------------

    ------------------------------------------
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  • Micz Flor wrote:
    > have you spoken to him?

    yes. the problem is, that he's upgraded his machine, which in return now
    doesn't want to boot Sad

    ------------------------------------------
    Posted to Phorum via PhorumMail
  • We have around 3-4 weeks for figuring out which automation system we
    need to adopt - LiveSupport, Rivendell or lastly BSIUSA Simian (the only
    commercial one that sort of works but with MS access DB and is build on
    other MS technologys which is horror in administrative viewpoint). So
    once something enough complete for testing comes out from LiveSupport
    we'll spend some 1-2 weeks on testing that througly and try to help with
    squashing bugs. If LiveSupport figures out for our needs then we might
    get some contract resources from our clients and throw some working
    hours behind LiveSupport for bug squashing and further development of
    new features.


    Rivendell system and its AirPLay module is getting pretty close to
    being stable enough but the overall system lacks some functionality -
    most importantly the open source music selector/scheduler/playlist
    creator is not part of the system. While it is possible to use
    commercial selector we would prefer open-source one on long term. Well,
    it's always possible to code our own...

    So we thought we will wait for LS 1.0 preview and then see which
    platform makes more sense to stick with. I'm not very familiar with LS
    yet (wanted to wait 1.0 preview before getting to know it better) but
    features like additional web frontend, music database/archive handling,
    nice GUI design and playlist mix editing support seemed appealing. Is LS
    going to have an music selector functionality (aka playlist composer
    based on clocks) or that needs to be developed additionally - that I
    didn't get clearly from initial docs.

    Our test and main production platform is Gentoo, so we could start
    testing LS with gstreamer even if you still have Debian compile problems
    with it.

    Cheers,

    --
    ----------------------------------------------
    Andres Toomsalu, andres@active.ee
    juhataja - general manager, O









  • Hi Andreas,

     

    Thank you for your
    constructive live input, ecpecially  because FM radio station
    professional area is far smaller under linux enviroment, 

     and every
    experience with these radio software solution enviroment, as long with
    players and used codecs, is welcome here.Smile)




     I myself as a
    user , am confronted with different hearing results under linux
    enviroment.  And I am talking here about "the first look" impression,
    and not some hi-fi esoteric.

     

    ....we are small FM station but we never saw mp3 as a true option
    for our music database because of bad mp3 compression quality on
    desirable bitrates.

    MP3 especially distorts high frequencies





    Can you
    tell me which programs and encoders you use to get mp3 compression for
    the test - and also what compression rate did you tested?

     






    Best regards

     

    Robert Klajn

     

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  • Hi Robert,

    I haven't used mp3 compression lately anymore but 2-3 years ago I recall
    having not so great sound quality when doing 128 or 192 kbps mp3
    encoding with Xing or Fraunhofer codec. There was always a chance that
    some songs high frequencies were getting distorted, so we chose MS ADPCM
    instead as we were using BSIUSA Wavestation automation system at that
    time. These days ogg wasn't mainstream yet...and still there is no ogg
    support in radio automation - not taking account LiveSupport ofcourse.

    Secondly I think it would be wise to have audio database as lossless as
    it can be or as original sounding as it can be (in compression wise).
    For FM broadcasting this is not needed but there is a lot of work going
    into good audio database (setting cues, mixes, other info) and maybe one
    day you want to use it for something else too where maybe more quality
    is needed (or maybe less quality output - but for conversion source you
    maybe need good quality to make conversion to succeed fully). So if your
    audio database is low rate mp3 and some songs are distorted - all the
    great work might be more or less wasted future wise.

    Nowdays if I have to choose mp3 or ogg - well that is no brainer:
    1) no royalties on ogg codec
    2) ogg is well supported in OSS software while mp3 support is not so
    great... ok there is LAME still, but I don't know is it legal overall
    3) I think ogg sounds better on lower bitrates

    Only thing that mp3 is more widely known still in commercial
    applications - but WMA and other proprietary formats will override that
    soon anyway.
    So if I have to compress I would choose ogg but I even would prefer to
    have audio database in some lossless or near lossless format.

    Cheers,

    Robert Klajn wrote:

    > Hi Andreas,
    > Thank you for your constructive live input, ecpecially because FM
    > radio station professional area is far smaller under linux enviroment,
    > and every experience with these radio software solution enviroment, as
    > long with players and used codecs, is welcome here.Smile)
    >
    > I myself as a user , am confronted with different hearing results
    > under linux enviroment. And I am talking here about "the first look"
    > impression, and not some hi-fi esoteric.
    >
    > /....we are small FM station but we never saw mp3 as a true option for
    > our music database because of bad mp3 compression quality on desirable
    > bitrates.
    > MP3 especially distorts high frequencies/
    >
    > Can you tell me which programs and encoders you use to get mp3
    > compression for the test - and also what compression rate did you tested?
    >
    >
    >
    > Best regards
    > Robert Klajn
    > ///////////// /////////////
    >
    > ------------------------------------------------------------------------
    >
    > *Robert Klajn *
    >
    > * klajn@b92.net
    >
    > ------------------------------------------------------------------------
    >


    --

    Andres Toomsalu, andres@active.ee
    juhataja - general manager, OÜ Active Systems
    Lille 4-205, Pärnu 80041, phone +372 44 70 595
    GSM +372 56 496 124, IM: frame@jabber.org
    http://www.active.ee

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  • jxn sly wrote:
    > don't forget about ogg's flexibility features -- the ability to easily
    > split the track (with ogg metadata maintained), a broad container format
    > similar accross flac/speex/theora, and (my personal favorite) the
    > ability to downcode (is that the word? move to a lower bitrate without

    bitrate peeling

    > completely reencoding from scratch, to preserve audio quality) tracks
    > recorded at higher bitrates when you decide your priority is diskspace

    while this feature of ogg vorbis has been hyped for several years now,
    I've never seen _any_ implementation that actually makes it possible...
    it would be nice though, especially for streaming...


    Akos

    ------------------------------------------
    Posted to Phorum via PhorumMail
  • ------=_Part_2833_3231969.1120513958399
    Content-Type: text/plain; charset=ISO-8859-1
    Content-Transfer-Encoding: quoted-printable
    Content-Disposition: inline

    On 7/2/05, =C1kos Mar=F3y wrote:
    >=20
    > jxn sly wrote:
    > > don't forget about ogg's flexibility features -- the ability to easily
    > > split the track (with ogg metadata maintained), a broad container forma=
    t
    > > similar accross flac/speex/theora, and (my personal favorite) the
    > > ability to downcode (is that the word? move to a lower bitrate without
    >=20
    > bitrate peeling
    >=20
    > > completely reencoding from scratch, to preserve audio quality) tracks
    > > recorded at higher bitrates when you decide your priority is diskspace
    >=20
    > while this feature of ogg vorbis has been hyped for several years now,
    > I've never seen _any_ implementation that actually makes it possible...
    > it would be nice though, especially for streaming...
    >=20
    >=20
    > Akos
    >=20
    Interesting...I had used a script on my old machine that implimented bitrat=
    e=20
    peeling and seemed to work quite well, but I cannot seem to find it anymore=
    .=20
    I found quite a few bitrate-peeling mini-scripts out there, looking for the=
    =20
    old one I had used, but the audio quality is horrible (i.e., something's no=
    t=20
    working right).

    ------=_Part_2833_3231969.1120513958399
    Content-Type: text/html; charset=ISO-8859-1
    Content-Transfer-Encoding: quoted-printable
    Content-Disposition: inline



    On 7/2/05, rname">=C1kos Mar=F3y <darkeye@=
    tyrell.hu
    > wrote:
    order-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddin=
    g-left: 1ex;">
    jxn sly wrote:
    > don't forget about ogg's flexibility features -- the=
    ability to easily
    > split the track (with ogg metadata maintained), =
    a broad container format
    > similar accross flac/speex/theora, and (my=
    personal favorite) the

    > ability to downcode (is that the word? move to a lower bitrate wit=
    hout

    bitrate peeling

    > completely reencoding from scratch,=
    to preserve audio quality) tracks
    > recorded at higher bitrates when=
    you decide your priority is diskspace


    while this feature of ogg vorbis has been hyped for several years n=
    ow,
    I've never seen _any_ implementation that actually makes it possible=
    ...
    it would be nice though, especially for streaming...



    Akos
    Interesting...I
    had used a script on my old machine that implimented bitrate peeling
    and seemed to work quite well, but I cannot seem to find it
    anymore.  I found quite a few bitrate-peeling mini-scripts out
    there, looking for the old one I had used, but the audio quality is
    horrible (i.e., something's not working right).


    ------=_Part_2833_3231969.1120513958399--

    ------------------------------------------
    Posted to Phorum via PhorumMail