Common video dimensions and bit rates for dynamic streaming?

I'm going to be converting my videos to flv and am trying to decide what to use for video dimensions and bit rates. Some of my users have slow computers and connections so I'm thinking 150 on the low end.
Is there a common practice? What has worked well for you in the past?

Hello MrWizzer
I am not sure what FMS version you are currently using.
Well, if you look at the sampe vod folder that ships with FMS4.5, it has files encoded @ 150kbps, 500 kbps, 700 kbps, 1000 kbps, 1500 kbps.
I am sure that all of these files stream perfectly fine given the correct bandwidth environment.
Its totally depend upon the FMS hosting service provider depending upon the user base of a particular provider.
You need to judge what kind of BW is available to your viewers and put the files encoded at appropriate bitrates.
You may also look at http://www.adobe.com/devnet/flashmediaserver/articles/beginning-fms45-pt06.html for refrences.
Regards,
Shiraz Anwar

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Video time and currentTime unsynced for dynamic stream

Using a DynamicStreamingResource with four mp4 rtmp streams is causing a problem.
Sometimes when the streams is auto switched, the switch is seamless but the mediaplayer.currentTime starts reporting the wrong time (mostly about 50 seconds before the "actual time"). After scrubbing the currentTime seems to be returning a correct value again.
When playing the streams seperatly the problem does not occur, so it seems that it's got something to do with the swithing.
Is it a bug in OSMF? In Flash? Or does it have something to do with the keyframes in video files?

I am getting this exact same behaviour all of a sudden with MBR streams served out of Akamai. If I move the streams to Amazon Cloudfront they work fine. I think this is an issue with FMS 4.5 as Cloudfront is still on FMS3.5 but Akamai has upgraded.
I can provide test streams. I did some detailed player logging, see below:
player init
loading release
release loaded
playing ad
ad complete
playing ad
ad complete
adding the stream: mp4:videolink/Bones_E804_low_16x9.mp4 @ 625
adding the stream: mp4:videolink/Bones_E804_lowest_16x9.mp4 @ 429
adding the stream: mp4:videolink/Bones_E804_medium_16x9.mp4 @ 820
adding the stream: mp4:videolink/Bones_E804_high_16x9.mp4 @ 1210
conviva title: Bones - S08E04 The Tiger in the Tale
starting main content
2-Second Heartbeat: 0
stream switching? false, current stream index: 0, current time is: 0
2-Second Heartbeat: 0
2-Second Heartbeat: 0.901
2-Second Heartbeat: 2.894
2-Second Heartbeat: 4.903
stream switching? true, current stream index: 0, current time is: 4.997
2-Second Heartbeat: 6.899
2-Second Heartbeat: 8.901
2-Second Heartbeat: 10.902
stream switching? false, current stream index: 3, current time is: 12.679
2-Second Heartbeat: 12.896
2-Second Heartbeat: 14.898
2-Second Heartbeat: 16.904
2-Second Heartbeat: 18.9
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pause content - 341.084 secs
playing ad
172-Second Heartbeat: 341.084
ad complete
playing ad
2-Second Heartbeat: 341.084
82-Second Heartbeat: 341.084
ad complete
playing ad
2-Second Heartbeat: 341.084
162-Second Heartbeat: 341.084
ad complete
playing ad
102-Second Heartbeat: 341.084
ad complete
resuming content - 341.084 secs
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2-Second Heartbeat: 771.865
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2-Second Heartbeat: 837.871
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2-Second Heartbeat: 843.872
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2-Second Heartbeat: 847.875
2-Second Heartbeat: 849.864
2-Second Heartbeat: 851.848
2-Second Heartbeat: 853.856
2-Second Heartbeat: 855.853
2-Second Heartbeat: 857.85
2-Second Heartbeat: 859.837
pause content - 861.179 secs
playing ad
72-Second Heartbeat: 861.179
stream switching? true, current stream index: 3, current time is: 861.179
112-Second Heartbeat: 861.179
ad complete
playing ad
2-Second Heartbeat: 861.179
ad complete
playing ad
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142-Second Heartbeat: 861.179
ad complete
playing ad
2-Second Heartbeat: 861.179
32-Second Heartbeat: 861.179
2-Second Heartbeat: 861.179
132-Second Heartbeat: 861.179
ad complete
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2-Second Heartbeat: 864.423
stream switching? true, current stream index: 3, current time is: 865.101
stream switching? true, current stream index: 3, current time is: 865.926
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2-Second Heartbeat: 868.423
stream switching? false, current stream index: 0, current time is: 920.908 // things come "back in sync" here, resulting in "skipped" content
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joewherefast wrote:
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Itunes plays purchased HD videos fine, but constantly stalls while trying to play home created videos converted to the same file format and bit rate.
System specs
Win 7pro x64
ATI 5830
Phenom x6 1055t
8gigs ram
1tb 7200rpm sata hdd
500 gig 5400 rpm sata hdd

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jcbQuadrosSegundo is an JComboBox.
Thanks !!

Skype is generally on autopilot. I believe 1920x1080 (1080p) would be the max resolution as far as I can recall. If the Internet connection isn't good enough at both ends, it will drop down to the resolution the call can handle. The max resolution of the mobile device would be dependent on the hardware and the Skype client. The webcam resolution compatibility is sometimes dependent on the version Skype. Some people achieve better compatibility with the desktop version 6.1 vs the current 6.3. If you want to monitor the resolution you can view the call technical info. It will display the current resolution of both feeds and other call stats.
This external site has some good info on it, though a bit dated, it should provide some insight. I'm not sure if any of the mobile resolutions have been updated greater than 320x240.
http://voiceontheweb.biz/referenceguide/skype/skype-video-calling/

Bit Depth and Bit Rate

I have a pre recorded mp3 VO. I placed it into a track bed in GB. Clients wants a compressed audio file with bit depth: 16 bit and bitrate: 128kps max, but recommends 96kbps. If I need to adjust the bit depth and bite rate, can I do it in GB? and if so, where? Thanks for any help.

Please be aware that Bit Depth and Bit Rate are two completely different things!
They belong to a group of buzz words that belong to Digital Audio and that is the field we are dealing with when using GarageBand or any other DAW. Some of those terms pop up even in iTunes.
Digital Audio
To better understand what they are and what they mean, here is a little background information.
Whenever dealing with Digital Audio, you have to be aware of two steps, that convert an analog audio signal into a digital audio signal. These magic black boxes are called ADC (Analog Digital Converter) and “on the way back”, DAC (Digital Analog Converter).
Step One: Sampling
The analog audio (in the form of an electric signal like from an electric guitar) is represented by a waveform. The electric signal (voltage) changes up and down in a specific form that represents the “sound” of the audio signal. While the audio signal is “playing”, the converter measure the voltage every now and then. These are like “snapshots” or samples, taken at a specific time. These specific time intervals are determined by a “Rate”, it tells you how often per seconds something happens. The unit is Hertz [Hz] defined as “how often per seconds” or “1/s”. A Sample Rate of 48kHz means that the converter takes 48,000 Samples per second.
Step Two: Quantize (or digitize)
All these Samples are still analog, for example, 1.6Volt, -0.3Volt, etc. But this analog value now has to be converted into a digital form of 1s and 0s.This is done similar to quantizing a note in GarageBand. The value (i.e. the note) cannot have any position, it has to be placed on a grid with specific values (i.e. 1/16 notes). The converter does a similar thing. It provides a grid of available numbers that the original measured Sample has to be rounded to (like when a note get shifted in GarageBand by the quantize command). This grid, the amount of available numbers, is called the Bit Depth. Other terms like Resolution or Sample Size are also used. A Bit Depth of 16bit allows for 65,535 possible values.
So the two parameters that describe the quality of an Digital Audio Signal are the Sample Rate (“how often”) and the Bit Depth (“how fine of a resolution”). The very simplified rule of thumb is, the higher the Sample Rate, the higher the possible frequency, and the higher the Bit Depth, the higher the possible dynamic.
Uncompressed Digital Audio vs. Compressed Digital Audio
So far I haven’t mentioned the “Bit Rate” yet. There is a simple formula that describes the Bit Rate as the product of Sampel Rate and Bit Depth: Sample Rate * Bit Depth = Bit Rate. However, Bit Depth and how it is used (and often misused and misunderstood) has to do with Compressed Digital Audio.
Compressed Digital Audio
First of all, this has nothing to do with a compressor plugin that you use in GarageBand. When talking about compressed digital audio, we talk about data compression. This is a special form how to encode data to make the size of the data set smaller. This is the fascinating field of “perceptual coding” that uses psychoacoustic models to achieve that data compression. Some smart scientists found out that you can throw away some data in a digital audio signal and you wouldn’t even notice it, the audio would still sound the same (or almost the same). This is similar to a movie set. If you shoot a scene on a street, then you only need the facade of the buildings and not necessary the whole building.
Although the Sample Rate is also a parameter of uncompressed digital audio, the Bit Depth is not. Instead, here is the Bit Rate used. The Bit Rate tells the encoder the maximum amount of bits it can produce per second. This determines how much data it has to throw away in order to stay inside that limit. An mp3 file (which is a compressed audio format) with a Bit Rate of 128kbit/s delivers a decent audio quality. Raising the Bit Rate to 256bit/s would increase the sound quality. AAC (which is technically an mp4 format) uses a better encoding algorithm. If this encoder is set to 128kbit/s, it produces a better audio quality because it is smarter to know which bits to throw away and which one to keep.
Conclusion
Whenever you are dealing with uncompressed audio (aiff, wav), the two quality parameters are Sample Rate [kHz] and Bit Depth [bit] (aka Resolution, aka Bit Size)
Whenever you are dealing with compressed audio (mp3, AAC), the two quality parameters are Sample Rate [kHz] and Bit Rate [kbit/s]
If you look at the Export Dialog Window in GarageBand, you can see that the Quality popup menu is different for mp3/AAC and AIFF. Hopefully you will now understand why.
Hope that helps
Edgar Rothermich
http://DingDingMusic.com/Manuals/
'I may receive some form of compensation, financial or otherwise, from my recommendation or link.'

Suggestions for bit rate for books on cd

Hi
I'm trying to clear some room on my ipod. Thus far I've been importing everything at 192. I'm hoping to reimport some of the books on cd that I've put on and would like some suggestions as to how low of a bit rate I can use without affecting sound quality too much (I tried 96 and really couldn't tell the difference).
Thanks
Evelyn

Hi, Evelyn.
For spoken word recordings, 32 kbps is generally considered an acceptable setting.
And, for what it's worth, that's the Bit Rate for Audiobooks purchased from the iTunes Store.
EDIT: You may also want to have a look at these suggestions: Link

Best average bit rate & max bit rate for DVDs playing on laptops...

I encode and burn many DVDs for use in regular DVD players....usually in the half hour range and have been encoding with Compressor 3 using these specs:
6.6 average bit rate
7.8 max bit rate
Those same DVDs, which play flawlessly in the DVD players, can occasionally hiccup when played from a laptop. Should I change the max bit rate from 7.8 to 7.7. or lower? Is that the biggest part of the equasion that may affect a laptop's ability to play without choking?
Many thanks,
John

Thanks for all your help, David S, but the point is
that I have tried many different bit rates, from 4.0
to 7.0, and I don't have time to waste all day trying
different bit rates for an 8-minute project.
My main question is: Why would it work fine on all
of my other short projects, including the 8-minute
one, and then when I cut out 20 seconds of that same
project, it starts screwing up!?!
I am going to try trashing my preferences. It is
starting to seem like a corrupt pref file to me.
It turned out to be a corrupt pref file, as I thought. Thanks for your many, many questions, David S.

How do I reduce the bit rate for existing songs

Some of the songs are too large. How can I reduce the file size of selected songs? Thanks.

Hi,
You can do it by converting your file either with a lower bit rate or using an another compression algorithm. Be aware taugh that by doing so, you will loss audio quality. The best way is to restart, if possible, from your original files with your new importing settings.
To change your importing settings, go in the iTunes preferences, in the Advanced tab and then in the Importing tab and set the compressing algortihm and bit rate. AAC encoder is the algorith that will gives you the better compressing ratio.
Then, you you can either chose to reimport from your original CD (iTunes will ask you to replace the previously imported files) or select from the advanced menu the converting selection.

Common video dimensions and bit rates for dynamic streaming?

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