Linear PCM audio...what the heck is it ?

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Originally Posted by Woochifer

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SACD uses a different format from PCM. Rather than sampling multibit words, SACD uses the DSD carrier, which uses one-bit word lengths sampled at very high rates. Supposedly this approximates analog waveforms better, but any valid comparison using SACD won't really happen until we start seeing more recordings done using DSD. Right now, almost all SACDs are converted from analog or PCM sources.

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These Wooch observations were made 5 years ago! The plot has only thickened with the advent of Blu-ray.

And today there is a pretty good selection of pure DSD SACDs. Over 1300 according to SA-CD.NET; see here; mostly classical genre as you'd expect.

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Originally Posted by Feanor

These Wooch observations were made 5 years ago! The plot has only thickened with the advent of Blu-ray.

And today there is a pretty good selection of pure DSD SACDs. Over 1300 according to SA-CD.NET; see here; mostly classical genre as you'd expect.

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That is because SACD has received a following in Europe, where most of the newer stuff is coming from, and its mostly classical.
And 1300 titles aint that "great", in fact its a sad statement on a decent format that,
barring a stupid "format" war, might have had a chance.
One hopes that its re-incarnation in Europe is the start of a new life, it would be a worthy
replacement for the turntable.:1:

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Originally Posted by Feanor

These Wooch observations were made 5 years ago! The plot has only thickened with the advent of Blu-ray.

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think the statements made 5 years ago hold true?

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Originally Posted by Tarheel_

think the statements made 5 years ago hold true?

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Had the music industry actually supported these formats, yes it would have held true.

I frequently work with outside media sent in for a compilation. The usual ones that I have issues are from the Networks...I usually encounter poor audio, clicks, ticks and taps. When I use AAC output to convert to AppleProRes 422(HQ) Ithe clicks ect are ever present however if I export/convert using Linear PCM I get better quality audio to compose in FCP..why is this?

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Originally Posted by Woochifer

PCM is basically the carrier format used in CD audio and most other uncompressed digital audio formats. There are several varieties of PCM and they vary based on the bit depth and sampling rate. CD audio uses a 16-bit word length and a 44.1 kHz sampling rate.

The DVD format requires the use of EITHER a Dolby Digital or PCM soundtrack. Because DD is a compressed format that can fit a 5.1 soundtrack into one-third of the space required by a two-channel 44.1/16 PCM (CD audio) track, most DVDs use DD as the audio carrier.

However, there are a number of DVDs, mostly with music, that use PCM audio tracks with higher resolution than the 44.1/16 resolution of CDs. Subjectively, this can sound subtlely better than Dolby Digital. The higher resolution can range from bumping up the sampling rate to 48 or 96 kHz or increasing the bit depth from 16 to 20 or 24 bits. Some audiophile labels have issued discs with 96/24 PCM audio tracks, and the sound quality on these discs is superb. The advantage of these discs is that they can be played back on any DVD player and unlike DVD-A or SACD, the signal can be output digitally. Major record labels never got on board with 96/24 discs because it doesn't have the encryption schemes that you see with SACD and DVD-A.

Your Denon DVD player allows for 96/24 resolution audio signals to get passed through the digital outputs. However, some discs with copyright protection do not allow for the audio to get output digitally at a 96 kHz sampling rate and require that the sampling rate step down to 48 kHz before the signal can be sent through the digital outs. Only with a few discs that contain PCM tracks with a 96 kHz sampling rate and copyright restrictions will you ever need to bother with that 96 ----> 48 kHz downconversion option in the DVD setup. On copyright protected discs, the signal output can only use the full 96 kHz sampling rate if it's decoded by the DVD player's internal DACs and output through the analog outputs.

DVD-A is basically a format that plays back PCM tracks encoded at 192/24 (two-channel) or 96/24 (5.1) resolution. The DVD-A tracks are compressed, but during the playback the DVD-A player will restore the data discarded by the compression process prior to the digital-to-analog conversion.

SACD uses a different format from PCM. Rather than sampling multibit words, SACD uses the DSD carrier, which uses one-bit word lengths sampled at very high rates. Supposedly this approximates analog waveforms better, but any valid comparison using SACD won't really happen until we start seeing more recordings done using DSD. Right now, almost all SACDs are converted from analog or PCM sources.

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Originally Posted by pweaver1

I frequently work with outside media sent in for a compilation. The usual ones that I have issues are from the Networks...I usually encounter poor audio, clicks, ticks and taps. When I use AAC output to convert to AppleProRes 422(HQ) Ithe clicks ect are ever present however if I export/convert using Linear PCM I get better quality audio to compose in FCP..why is this?

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Perhaps it is because you are going from a packet driven audio media, to a bitstream based one. There could also be a filter during encoding to PCM that eliminates or just does not recognize the clicks, pops, and various other audio artifacts.

Without know more detail, it is impossible to tell why the audio artifacts disappear when converting to PCM.