The Antelope Audio portable studio interface Zen Studio was used to record the Leman Music Masterclass at the Conservatoire de Musique de Geneve. “Because of portability, digitally controlled preamps and links for the stereo pairs, it was the obvious choice for its ergonomy” said the audio engineer David Trotti. The pristine quality of the resulting audio blowed away recording engineer David and made the day of this yearly event.
Next, ACR’s Zen Studio teamed up with its ORION 32 brother for a redundancy recording at Geneva’s Victoria Hall of “Camerata Armin Jordan” with 112 musicians from the Orchestre de la Suisse Romande.
Direct Stream Digital
Direct-Stream Digital (DSD) is an audio format developed by Sony and Philips for the Super Audio CD system (based on ideas initially described in a 1954 patent). The technology was then later developed by Playback Designs and pioneered the transfer of DSD files over USB connections.
PCM is usually 16-bit to 24-bit (CD standard is 16-bit and 44.1kHz) whereas DSD is commonly 1-bit or in some cases 8-bit and has a sampling rate of 2.8224MHz. The output from a DSD recorder alternates between levels representing ‘on’ and ‘off’ states, and is a binary signal (called a bitstream).
To minimize quantization errors during the ADC process, the DSD format utilizes noise shaping algorithms (filters), which allow shifting of the quantization distortion up to ultrasonic values, frequencies far outside the human range of hearing. DSD bitstreaming allows the SACD players to be made with a simple 1-bit design and using a low-order analog filter during the DAC process. Although the SACD format achieves a dynamic range of 120dB for all frequencies in the range of human hearing (20Hz to 20kHz) and provides an extended frequency response up to 100kHz, most players in the market offer a maximum of 80-90kHz.
Ever since his art school days in the late 1990′s while attending the prestigious Chicago Art Institute, William Close pursued his dream of building and performing with unique, handmade instruments whose sounds have never been heard anywhere in the world. Now, The Earth Harp, his masterpiece instrument creation of unprecedented physical scale and sonic beauty, has been captured in astonishing fidelity in a brand new recording — thanks to the digital clocking and conversion technology of Antelope Audio.
His new album with The Earth Harp Collective, Behind the Veil, captures the authentic sound of this spectacular instrument — from its lavish root notes to its rich harmonics and heavenly overtones. Close attributes the success of the recording in large part to Antelope Audio’s new Rubicon A to D converter, which was used as the primary mastering device, and its Orion³² multi-channel interface, which was used during playback. “I’ve never heard The Earth Harp sounding so good on a recording,” he says. “The instrument has so many beautiful harmonics and overtones, and many times these are lost in the process. The Antelope equipment was awesome and helped us finally achieve a true representation of how The Earth Harp actually sounds.”
What devices need clocking?
In a simple system featuring one audio interface with built-in mic preamps connected to a computer-based DAW, the interface clocks the DAW since the most clock-critical element of the audio chain is the A/D converter, as that’s built in to the interface. If you were to add an external digital device to the equation (reverb, multi-effects processor etc), it should be configured to work as a clock slave to the interface.
Following that logic, even in bigger and more complex studios, it’s generally best to use an A/D as the master clock. If there’s more than one of these you’ll need to decide which one to use as the master, and everything else will have to be slaved to that. It’s quite possible that there will be audible differences between various configurations, because most A/Ds will perform slightly differently when configured as clock master and slave.
Why do we need clocks?
In order for an analog signal to be digitized, it must be sampled precisely and accurately in repeating intervals. The master clock provides that timing information and allows the waveform to be reconstructed as an analog signal correctly (assuming the sample rate is more than twice the highest frequency component of the audio signal being sampled). The clock identifies when each individual sample should be recorded or re-played (word clock).
If the clock timing varies, the audio samples will possibly be replayed or recorded at the wrong time resulting in sound distortion, jitter and aliasing. Jitter is the erroneous capture of a wave form over time. Although the apparent error to the clarity of the audio this creates there are other artifacts that maybe introduced with bad clocking.
Another type of clock is the ‘bit clock’. This is used in serial data interfaces like AES, S/PDIF and ADAT, where there is basically only one ‘conduit’ over which to pass the audio data. The bit clock ensures that the receiving device does not lose track of when each data bit stops and the next begins, eliminating the potential result in corrupted data values being received.