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yorkie98Participant
[quote author=”thagurt”]I have a new PC so I’ve installed on my breakaway and edcast, it all seems to work, but when I stream the playback sounds "delayed" like pitch down 10%. also standalone with edcast i have the same problem. i’ve installed it like the how-to on this website made by leif.
when i use another streamtool like opticodec i don’t have that problem. i’ve also tried other encoders in edcast like aac . but this doesn’t solved my problem?Can anyone help me????[/quote]
Check that you have the correct sample rate selected on the output config screen, if your streaming at 44100 but the L/R output is at 48000 then this may cause an issue, might not but give it a look.
I’ve certainly never came across this problem with edcast (DSP or standalone) and I’m very likely to have overlooked a samplerate or two..yorkie98Participant[quote author=”jameslawson”]I’ve already tried this and dosent seem to want to work because it wants sam broadcaster to be the output, I’m using the setup that uses the live link for the dsp section.[/quote]
Ah, sorry I have no experience of this method of useage.
yorkie98ParticipantAs an interim fix, maybe set the output to a dummy device or an unused pipeline if it absolutely won’t allow you to disable.
yorkie98Participant[quote author=”Leif”]Jesse’s answer regarding 128k mp3 files is spot on. Low bitrate MP3 files absolutely do not belong in a broadcast environment. Quality starts at the source is (ideally) maintained through the chain. If at any point quality is lost, it can never be regained later in the chain.
I use 320k MP3 or flac, and I’m a "home user", not a radio station. It makes no sense that a radio station, whose sole business is audio, would use less than that — especially not with todays cost of storage media.
///Leif[/quote]
The fact is that there are stations out there who are run by people who think they know better and even worse, have no concept of audio quality. I have heard stations playing audio which as absolutely awful but the even more frustrating thing is that 99% of the listeners do not seem to notice, nor care about the quality.
I think it come down to the fact that in this day and age, many are used to the sound of low bitrate audio and have become unaware of the difference between the source audio and a poor copy.
Speak to most station managers nowadays and they will tell you that audio quality is well down the list of priorities, money is their "sole business" audio is merely part of their product. Engineers may think differently but nowadays many of these lack the kind of passion and commitment that used to taken for granted.yorkie98Participant[quote author=”didac”]Thanks!
I know others software but are more expensive, for example AudioTX is a good choice but you need this software on 2 machines, one for transmit and one for receive.
Thanks to all![/quote]
Yes, there is plenty of software out there for direct linking, either over a private network or the internet, some of which is quite reasonably priced and I use this kind of software in all cases for private links. I only use Winamp where a public webstream is being used as the source.
yorkie98ParticipantDidac,
With regards to your enquiry as a more stable stream player than Winamp, there is not much else out there, there are other media players, such as VLC etc, but they all have their problems. Winamp is really the best of a bad bunch but to be honest it’s a domestic stream player, not designed for 24/7 broadcast use, and most webstreams would not be reliable enough for continuous broadcast use either.
The best method I have come up with for using Winamp is to add your stream address several times to the playlist, save the playlist and put this in the startup directory. Then switch on loop play and shuffle play in Winamp. If the stream underruns or gets disconnected, it will keep trying and trying to reconnect, the rest is down to the reliablilty of the stream itself. There are occasional circumstances where Winamp will freeze so make sure you can get remote acces to the machine as you may need to press stop and start every now and then.
At this time, I do not know of any better software or methods for using a webstream as a feed. If there are any better methods, I’d love to know.Yorkie.
yorkie98ParticipantThis may have been missed…check that if there is an internal stereo generator in the transmitter this is disabled or bypassed.
yorkie98ParticipantYou’re right, the batch processor would either need to be integral to BBP or, a separate program but which requires breakaway to be installed to function. that should alleviate any piracy worries? No point making it part of the STL package as the batch tool is only of use to process the audio and this is done in BBP..
Otherwise, these files could be captured via the pipeline and recorded into Audition or similar.
Maybe your clipper/encoder idea is less hassle.yorkie98Participant[quote author=”Leif”]Thank you!
Only thing is, how would we handle the fallback playout if running as an MPX-link?
I suppose I could put an extremely optimized breakaway-processor with the low cpu clipper and stereo generator, to handle the fallback 🙂.///Leif[/quote]
I was thinking that, If running in stereo mode, fallback would be normal stereo files, maybe pre-processed by a clever little batch processing tool (whoops, I’ve just created you another job..) which encodes files using the breakaway processor and saves them with whichever codec the user desires, although it would be advisable to save in the same format the STL is usually used in..?
OR in the case of MPX, ready processed PCM or FLAC files using the same clever batch tool…Your own suggestion would work also, I think in the case of fallback audio, the end user should expect the audio to not neccesarily be as good as the main audio but just as long as it does not allow over-deviation.
It sounds like you are really starting to get into this idea?
yorkie98ParticipantGood work, so it’s looking do-able. As you rightly state, it’s unlikely a DSL line could sustain even a 1mbps stream constantly but it’s not my intention, I’m looking at this to work over a 5.8ghz 54mbps wireless point to point link so even in poor signal condidtions, 3-6mbps is acheivable and no problems with sustainability.
I’d suggest having the program work in two modes:
1. Stereo audio with options to use ultra-low bandwidth codecs like AAC+, medium bandwidth (MP3/WMA) right up to FLAC and PCM for lossless audio OR
2. Mpx with the flexibility to have high bandwidth, high SNR or lower bandwidth with the slightly lower SNR, even the 10 bit looked okay at a push.
I should imagine that there would need to be software downsampling/upsampling as many soundcards jump from 96 to 192khz and only do 16, 24 or 32 bit (my Emu 0404 does 96, 176.4 and 192 but not 128!).
Finally, not forgetting the fallback playout if the link is lost..This would make your program industry leading as all the current ones have limitations either in the codecs available, lack of fallback, lack of sound device selection or inflexible network settings and overall reliability. There is currently no program which I would call complete. Obviously i’m considering this as a standalone product as opposed to a part of breakaway (although it would work as a DSP plugin too..) as this would allow it to be marketed to a wider client base so also may bring those in looking for an STL solution and introduce them to breakaway. You could sell the STL at a reduced rate to Breakaway licence holders and also bundle a demo version of breakaway with the STL software. What a great cross-marketing opportunity!
This is your chance to lead the next generation of software STL programs and also compete with the current generation.
I’m sure this is a challenge but I’m also sure you are up to it. If this program is made to the standard of BBP, it will be excellent.
yorkie98ParticipantThanks Leif, understood about the 2 instance Licence. If building a system this way in future will make sure the client orders a 2 instance license instead of 2 single licences.
yorkie98ParticipantHi Leif, its a good point but again, I was not planning to run this link over the internet, just over a private, direct TCP/IP connection.
[quote author=”Leif”]Lack of bandwidth is only part of the problem.
No hardware supports MPX streaming, so you’d still need a computer at the transmitter site to receive the stream.
///Leif[/quote]
Not exacltly true, these units from Elettronika sends MPX digitally over a 2mbps E1 connection.
http://www.broadcastwarehouse.com/elett … 06/product
I’m guessing probably using FLAC as the compression.. It is a software version of this which I aspire to as E1 connections are a specific protocol and not available in many countries.
There are a dozen or more programs and hardware solutions which can do this job over an internet connection or a direct tcp/ip connection. Most of these solutions use compression of some form although a minority do also support direct PCM but often with limited success.
There would be no point in making such a program unless it can bring something new to the party.
I can really only think of a couple of features which could be considered as a breakthrough in STL features.Firstly, I’d like to see an STL program (and a streaming program for thst matter) which uses FLAC, that would be nice, a lossless IP STL, the best commercially available product which is 100% reliable I have found so far uses a 320k WMA stream but this is still not lossless. A step up to FLAC would definately open my wallet.
The reason for wishing for mpx STL reception at the remote end is again to do with minimisation of investment in equipment (and software) which will be lost in the event of theft or communications authority raids. A PC running STL software, Breakaway, Airomate and remote control software will have cost $400+ alone not including the hardware costs would be lost and all would need to be replaced.
If there were a program capable of delivering either, the MPX over FLAC OR the L/R Lossless (no overshoots) then whis would help minimise costs as the Breakaway etc would be at the studio end and the only software running would be the STL program on the remote (TX) PC (which would need to priced in such a way that the TX is most of the cost and the RX is low cost and easily relicencable (maybe locked to the licence code in the TX program?)).
This would also allow for much lower-end (less costly) pcs to left at the TX site as the processing is handled at the studio end.yorkie98ParticipantHi Leif,
I do quite often work at sites where there are multiple TXs at one site. As some of my clients are pirate broadcasters, with multiple stations broadcasting from the same site, it’s all about minimising the equipment at the transmitter site whilst keeping the engineering standards as high as possible. That’s why software like yours is a godsend but if these tweaks can be possible, even more so.. If a raid happens, why lose two PCs when one PC could be doing all the work?
Hope this makes sense..yorkie98ParticipantCan I add my entirely selfishly impatient support for this product, especially if it can acheive results such as the MPX over E1 connection units out there (or maybe even better), but with TCP/IP.
I am oftem using 54mbps TCP/IP links on 5.8ghz (which still acheive 6mbps with low signal condidtions) and using all this potential to carry a mere 320Kbps audio stream.
What a dream it would be to carry a full mpx signal over one of these especially as raw PCM or maybe FLAC for those with bandwidth limits! Only one channel @192 khz would be needed (approx 3mbps) but if two separate streams could be implemented and both come out of seperate channels at the other end (like with my ongoing discussion about this..), this would add loads of value too. So in theory, this could carry two uncompressed mpxs over approx 6mbps link or approx 3mbps with FLAC. If this was the kind of idea you were working on, I’d be very excited by this.
Please make my dream a reality soon Leif.PS, for good measure, throw in a fallback audio playback engine which plays out pre-processed PCM or FLAC mpx files if the link goes down. I’m a great ideas man.. If only I knew how to write the code.. over to you Leif.. 🙂
yorkie98Participant[quote author=”Leif”]Hi Yorkie!
There are professional sound cards where this would be possible, where L and R could show up as different devices. However, most sound cards do not support this.
///Leif[/quote]
Would it be possible to add this as a feature in BBP? If a device cannot be individually addressed, maybe if the current 1/2 channel selection on the main output (which seems to provide no function) can be set to L or R or both. I know it’s not strictly possible but here’s how I trick Airomate into giving me 2 seperate outputs from 1 card.
Instance 1 of airomate has audio from station 1 going in, the output is set to the appropriate soundcard (eg Audigy 4). I then switch the syncronisation channel ON (to channel 1) but then turn the volume down to 0. This then gives me the mpx for station 1 on channel 2 and nothing at all on channel 1.
I bring in the audio for station 2 into instance 2 and again turn on the sync channel but this time select channel 2 as the sync channel and again turn it down to 0.
Hey presto! each channel of my soundcard is carrying a completely different mpx signal which is then in turn fed into the two transmitters.So, to make this possible, you simply need to make BBP output the mpx from instance 1 on the selected channel only and digital silence on the other just like airomate can be tricked into doing. If this is reversed on the other instance, this will work. I’m sure this would not be hard to acheive. It would certainly help in reducing PC costs.
My other, but less preferred method would be to let BBP do just the processing, and feed the L/R outputs into airomate and let it do the stereo encoding (it does the RDS anyway) but I’m sure the encoder in Airomate is inferior to your own (Sorry Arjen!).
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