Hi all,
After much research and thinking(!) I've come accross some problems that I'm not entirely sure on how to get around.
Basicaly I'm trying to create a Head unit for car audio purposes. Features such as mp3 support, Hard drive and radio will eventually be included but I want to get a working hardware base doing basic CD tasks first.
The input sources are S/PDIF, MP3 (output of STA013 in PCM) and analog radio. To convert S/PDIF and the radio into a useable digital format would require a DIR1703 and PCM1804 (192khz ADC) respectively. From there a TAS3103 would select the wanted source, apply any EQ, and output to a TAS5012 and then onto a speaker with the appropriate power stage.
My problem is with the clocking sceme of the entire project. Assuming the best option would be to have the TAS3103 as the master, a 92khz Fs would be provided. To connect to the TAS3103, the 48khz output of the DIR1703 would need to be reclocked into 92khz. Would the easiest way to do this be through a SRC4192?
I think the SRC4192 can automaticaly select the ratio when both the input and output are in slave mode. The reference clock can be obtained from either the DAP or DIR1703. This would also provide 'clock isolation', yes? Where the 92khz clock would not need to be synchronised to the 48khz one generated by the DIR1703?
I think the STA013 has an Fs the same as the mp3 stream which also poses a problem when playing 44.1 ad 48khz mp3's. From reading the SRC4192 datasheet, if you put both the input and output into slave modes the reference clock has to be either 128Fs, 256Fs or 512Fs. So is it ok to use the bit clock for this from both the STA013 and DIR1703 outputs to thier SRC4192's?
Also, as a general rule, when do things need to be reset?
As in, loss of S/PDIF input, end of song etc.
or, will the SRC4192 abolish the need for this by returning a stream of zero's at the correct frequency as soon as the input isn't ready(no reference or bit clock on the input)?
Thanks for you help
Matt
After much research and thinking(!) I've come accross some problems that I'm not entirely sure on how to get around.
Basicaly I'm trying to create a Head unit for car audio purposes. Features such as mp3 support, Hard drive and radio will eventually be included but I want to get a working hardware base doing basic CD tasks first.
The input sources are S/PDIF, MP3 (output of STA013 in PCM) and analog radio. To convert S/PDIF and the radio into a useable digital format would require a DIR1703 and PCM1804 (192khz ADC) respectively. From there a TAS3103 would select the wanted source, apply any EQ, and output to a TAS5012 and then onto a speaker with the appropriate power stage.
My problem is with the clocking sceme of the entire project. Assuming the best option would be to have the TAS3103 as the master, a 92khz Fs would be provided. To connect to the TAS3103, the 48khz output of the DIR1703 would need to be reclocked into 92khz. Would the easiest way to do this be through a SRC4192?
I think the SRC4192 can automaticaly select the ratio when both the input and output are in slave mode. The reference clock can be obtained from either the DAP or DIR1703. This would also provide 'clock isolation', yes? Where the 92khz clock would not need to be synchronised to the 48khz one generated by the DIR1703?
I think the STA013 has an Fs the same as the mp3 stream which also poses a problem when playing 44.1 ad 48khz mp3's. From reading the SRC4192 datasheet, if you put both the input and output into slave modes the reference clock has to be either 128Fs, 256Fs or 512Fs. So is it ok to use the bit clock for this from both the STA013 and DIR1703 outputs to thier SRC4192's?
Also, as a general rule, when do things need to be reset?
As in, loss of S/PDIF input, end of song etc.
or, will the SRC4192 abolish the need for this by returning a stream of zero's at the correct frequency as soon as the input isn't ready(no reference or bit clock on the input)?
Thanks for you help
Matt