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#1
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Can you think of a way to combine these two sources without resampling?
I'm going to be copying some music from video tape recorded with a PCM-F1 to CD. Most of these tapes have ambient channels recorded on the HiFi audio tracks of the video tape. I've aquired a PCM-601ES unit with digital output, but I understand that it's 44.05593846 kHz. I'll sample the analog tracks through my Cranesong Spider (best converters I own) at true 44.1 kHz. The front source is 44.056K. I don't have another clock I can set to that rate and the 601 has no word clock output, so there's no way to send both streams synched and clocked. Am I right? My computer will accept both streams simultaneously through separate SPDIF and Lightpipe inputs. I can then resample the back channels to 44.056 and then just tell the file it's 44.1 without resampling. I don't need sample accurate alignment of the front and back channels, and I don't think the slight pitch change (0.1%) will bother anyone. |
#2
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Carey Carlan wrote:
Can you think of a way to combine these two sources without resampling? Not if they have materially different sample rates. If I was going to combine *anything* that was already in the digital domain, I'd be very prone to do it with Audition/CE. I'm going to be copying some music from video tape recorded with a PCM-F1 to CD. Most of these tapes have ambient channels recorded on the HiFi audio tracks of the video tape. I've aquired a PCM-601ES unit with digital output, but I understand that it's 44.05593846 kHz. It misses 44.1 KHz by about 1%. It obviously needs to be resampled. As close as I can get with Audition CE is 44056 Hz resampled to 44.1. That is 0.00014 % off. Seems like close enough. I'll sample the analog tracks through my Cranesong Spider (best converters I own) at true 44.1 kHz. OK The front source is 44.056K. I don't have another clock I can set to that rate and the 601 has no word clock output, so there's no way to send both streams synched and clocked. Am I right? Seems like, but why worry? My computer will accept both streams simultaneously through separate SPDIF and Lightpipe inputs. Why bother - why not read them one at a time and combine in the digital domain? |
#3
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In article ,
Carey Carlan wrote: Can you think of a way to combine these two sources without resampling? I'm going to be copying some music from video tape recorded with a PCM-F1 to CD. Most of these tapes have ambient channels recorded on the HiFi audio tracks of the video tape. I've aquired a PCM-601ES unit with digital output, but I understand that it's 44.05593846 kHz. I'll sample the analog tracks through my Cranesong Spider (best converters I own) at true 44.1 kHz. This is bad. The thing is that SRC is very hard to do with sample rates that are extremely similar. This is probably the worst case scenario for SRC. The front source is 44.056K. I don't have another clock I can set to that rate and the 601 has no word clock output, so there's no way to send both streams synched and clocked. Am I right? Well, that's a question... if you COULD split the output of the 601 with a DA, you would be able to use the S-PDIF signal as a reference clock. The problem is just that it's going to be a lousy reference clock because it'll be referenced to the data clock in the 601 which is not exactly good. If you had an Apogee AD-1000, or some other converter that could sample at 44.056, you could just use it to record the analogue tracks wild and just expect that there might be a little drift here and there. If you had a master reference clock that would lock to an external source, like the Big Ben, I think you could use that to clean up the clock coming out of the 601's S-PDIF connection before putting it into the Cranesong. I think Audio Alchemy made a little reclocking box back in the late eighties, and it sold for a couple hundred bucks when it was new. It took a dirty S-PDIF input and give you a reasonably cleanly-clocked output. The real question is how the Cranesong will deal with a lousy clock input, and you should ask them that. They may have some solution that I haven't thought of. My computer will accept both streams simultaneously through separate SPDIF and Lightpipe inputs. I can then resample the back channels to 44.056 and then just tell the file it's 44.1 without resampling. I don't need sample accurate alignment of the front and back channels, and I don't think the slight pitch change (0.1%) will bother anyone. Alternatively, you could just take the rear channels, which I assume are on the analogue tracks and which probably don't sound all that wonderful anyway, and just run them through the SRC and see what happens. They can't get all THAT much worse. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#4
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why not record each at their own clock rate, save as 44.1 and then pitch
shift the wrong one ? "Carey Carlan" wrote in message . 191... Can you think of a way to combine these two sources without resampling? I'm going to be copying some music from video tape recorded with a PCM-F1 to CD. Most of these tapes have ambient channels recorded on the HiFi audio tracks of the video tape. I've aquired a PCM-601ES unit with digital output, but I understand that it's 44.05593846 kHz. I'll sample the analog tracks through my Cranesong Spider (best converters I own) at true 44.1 kHz. The front source is 44.056K. I don't have another clock I can set to that rate and the 601 has no word clock output, so there's no way to send both streams synched and clocked. Am I right? My computer will accept both streams simultaneously through separate SPDIF and Lightpipe inputs. I can then resample the back channels to 44.056 and then just tell the file it's 44.1 without resampling. I don't need sample accurate alignment of the front and back channels, and I don't think the slight pitch change (0.1%) will bother anyone. |
#5
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![]() Scott Dorsey wrote: This is bad. The thing is that SRC is very hard to do with sample rates that are extremely similar. This is probably the worst case scenario for SRC. True if done with polyphase filters but not if done by sinc interpolation. I believe that Adobe Audition uses the latter at its maximum quality setting. Bob -- "Things should be described as simply as possible, but no simpler." A. Einstein |
#6
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daz[at]roughdiamondmarketing[dot]com wrote:
why not record each at their own clock rate, save as 44.1 and then pitch shift the wrong one ? In essence, that's what I recommended. |
#7
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#8
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Carey Carlan wrote:
(Scott Dorsey) wrote in Well, that's a question... if you COULD split the output of the 601 with a DA, you would be able to use the S-PDIF signal as a reference clock. The problem is just that it's going to be a lousy reference clock because it'll be referenced to the data clock in the 601 which is not exactly good. That's the best idea I've heard so far. The 601 clock is no worse than the signal recorded on the video hifi audio tracks. A bit of jitter really won't mess that sound up much. What kind of DA do I need? Would an active video splitter do it? (got one of those). Try and it see. It will probably work. It will add clock jitter, though. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
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