Old 9th June 2004, 22:35   #1
Stephan Graller
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Audio quality low in NSV stream

Dear friends!

First I would like to state out what great work you all did! Thanx alot many times!

The problem I have is the audio quality of MP3 within a NSV live stream from a video camera and audio from a stereo mixer via line-in. Independent of the video codec used, audio quality at e.g. 24kbit/s (either CBR or ARB as average bitrate) is less then acceptable, and even more the same audio signal sent as pure audio stream with Winamp DSP Encoder at 16kbit/s is even better in quality. It seems to me as if the NSV Live Capture tool addresses lame_enc.dll in a way that makes it perform this poor.

I want to stream voice in mono. So the audio stream should be quite acceptable at 16-24 kbit/s.

Anybody any similar experiences or hints or whatever?

Thanx.
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Old 9th June 2004, 22:43   #2
Inedible Bulk
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ARB?

Hint, don't use mp3, use AAC, much better at low quality. Kthx.

(20kbps is what i've found to be the good medium with AAC)

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Old 9th June 2004, 22:49   #3
ken52787
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Well, if you are gonna stick with the MP3, the problem is with the sample rate. In the main NSV config window, in the upper right corner is something that says Audio Resampling. Make sure it is on Mono and something like 22050. In the MP3 options, make sure it is also Mono. The audio quality should be better.

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Old 9th June 2004, 22:58   #4
Stephan Graller
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Thank you for quick replies.
I always set to Mono and 11025 sampling rate in NSV, because only with these settings I was able to choose lower bitrates than 32kbit/s! Otherwise the NSV Live Capture tool simply crashed. I have never tried AAC, does it perform good at 16kbit/s also?
While streaming pure audio with dsp_sam_encoder I was even able to decrease to 8kbit/s mono with 11025 sampling rate with still understandable voice quality streamed via satellite uplink ;-)
I need to make the complete NSV stream as small as possible for two reasons: available upstream bandwidth at the sender location and client download bandwidth at regions where broadband internet connections are not widely spread.
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Old 9th June 2004, 23:51   #5
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AAC offers greater audio fidelity and clarity at lower encoding speeds vs. MP3 format at the same encoding speed.

Scenic Television is your ambient window to the world - - www.ScenicTelevision.com
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Old 10th June 2004, 05:26   #6
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it was discussed before, get the latest LAME package(http://sourceforge.net/projects/lame/)
to make a latest lame_enc.dll, then you can use broader range of parameters.

i often use 16k sampling rate, with 16kbps bitrate, the quality is good enough.

check the channel "Falundafa.org -- Truthfulness,Compassion,Forbearance"
as an example.

Quote:
Originally posted by Stephan Graller
Thank you for quick replies.
I always set to Mono and 11025 sampling rate in NSV, because only with these settings I was able to choose lower bitrates than 32kbit/s! Otherwise the NSV Live Capture tool simply crashed. I have never tried AAC, does it perform good at 16kbit/s also?
While streaming pure audio with dsp_sam_encoder I was even able to decrease to 8kbit/s mono with 11025 sampling rate with still understandable voice quality streamed via satellite uplink ;-)
I need to make the complete NSV stream as small as possible for two reasons: available upstream bandwidth at the sender location and client download bandwidth at regions where broadband internet connections are not widely spread.
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Old 10th June 2004, 19:27   #7
Stephan Graller
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OK, I got almost the latest lame_enc.dll and it did not get better.
How do I get AAC into NSV tools, it only offers PCM or MP3?
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Old 10th June 2004, 19:34   #8
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Go to the Winamp 5 folder. Copy the enc_acc.dll out of the plug-ins folder, and rename it to nsv_coder_aac.dll and put it in the NSV Tools folder.

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Old 10th June 2004, 19:55   #9
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Quote:
Originally posted by Stephan Graller
OK, I got almost the latest lame_enc.dll and it did not get better.
How do I get AAC into NSV tools, it only offers PCM or MP3?
oh, you need to check the option "lowpass filter when downsampling"
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Old 10th June 2004, 20:29   #10
Stephan Graller
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OK, I tried to copy enc_aac.dll and rename it, but nsvacp constantly kept crashing with it. Now i did another thing: I copied enc_lame.dll from Winamp5 to NSVtools and guess what happened? Now I had new options for MP3 and the sound also seems much better.
But I still saw NSVcap crashing when I set a false sampling rate, like 22100 or such. But when I set 22050, everything is OK.

But now I have another question: is there any possibility to save the stream that I send?
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Old 10th June 2004, 21:14   #11
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When using AAC, keep the resample rate at 44100. The codec resamples internally and you can't control it. That will stop the errors.

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Old 10th June 2004, 21:20   #12
Stephan Graller
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Yeah, it is working now with AAC. And what a difference there is to MP3!!! Thanx all!
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Old 13th June 2004, 02:46   #13
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Hi everyone:
Quote:
Originally posted by ken52787
Go to the Winamp 5 folder. Copy the enc_acc.dll out of the plug-ins folder, and rename it to nsv_coder_aac.dll and put it in the NSV Tools folder.
Actually, it's best if you download Ken's NSV GUI program and get it from there.

Cheers for now

Jeeper One
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Old 13th June 2004, 13:16   #14
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Quote:
Originally posted by Jeeper One
Hi everyone:Actually, it's best if you download Ken's NSV GUI program and get it from there.

Cheers for now

Jeeper One
If you get my program, you just have that step done for you. Nothing special.

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