Solution for Removing Audio Glitches in a RTP Audio Source Filter
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1.Test the Timestamp of the large files.
- it is also having some strange values like Start Time : 40000 and End Time :0
2.Set only start Time and be the stop time always 0.3.Dump the RTP Packet within the file.( No Problem with audio data)4.Implement the IAMPushSource interface.
I tested the large MP2 file in GraphEdit :
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If I inserted the dummy filter before the MPEG1 Audio Decoder filter, I got the following :
Start Time : 28114700 Stop Time : 23998200 Diff : 4116500Another time I got it as follows:
Start Time :31588108 Stop Time :28454648
Differnece :3133460
If I inserted the Dummy Filter after the MPEg1 Audio Decoder, I got the timestamp values as follows:
Grabber Sample size is : 18432 Start Time :0 Stop Time :0
Grabber Sample size is : 18432 Start Time :960000 Stop Time :0
Grabber Sample size is : 18432 Start Time : 1920000 Stop Time : 0
For the RTP Source Filter I got the following:
----------------------------------------------- 1.If I inserted the dummy filter before the MPEG1 Audio Decoder filter, I got the following :
Start Time Stop Time 2520000 2520000 2520000 2835000 2835000 3150000 2.If I inserted the Dummy Filter after the MPEG1 Audio Decoder, I got the timestamp values as follows:
Grabber Sample Size is : 18432 Start Time : 2520000 Stop Time : 0
Start Time : 2835000 Stop Time : 0 Start Time : 3150000 Stop Time : 0
Start Time : 3465000 Stop Time : 0 Start Time : 3780000 Stop Time : 0
Nothing above is works.
Solution:
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I am sending mp2 audio data with 48Hz frequency. At the RTP Audio Source Filter, I modified the frequency as 22.05 ( 22050 samples per second)..
Samples per second is the speed controller of a filter graph. it determines the speed of the filter.
48000(48 KHz) filter will run fast.
44100 (44.1KHz) filter will be some extent slow compare to the 48000.
22050( 22.05 KHz ) is very slow.
we are rendering the audio half of the transmitted audio's speed.
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