Sony Sound Processor
Ene Vinson
Get lost again in the richness of the melodies and dynamics of your favorite albums by giving back the character of vinyl to your digital tracks. Subtle reproduction of low-frequency tone-arm resonance, surface and scratch noise, and resonance on the vinyl deliver an authentic listening experience.
Audio upscaling is improved even further by our new DSEE HXTM processor which intelligently recognises instruments, voices and musical genres. By identifying these and the relative energy of the audio, it can accurately rebuild audio lost during digital compression, for a full fidelity experience, even with compressed formats.
What the effect does is mostly giving some warmth and body to the sound, making it less transparent. The result is a little more relaxed, laid back sound, with a more analogue flavor, which is more than welcomed by many.
Even hardcore nerds like our staff use after-market DSP boxes to do things like correct performance for bookshelf speakers, headphones, and even calibrate virtual surround sound systems included in products like the Sennheiser Ambeo Soundbar. By using a DSP unit with an appropriate microphone, you can measure the output of your audio gear in any environment, and correct the output to sound the way you want it to automatically.
Essentially, by using a modern DSP, you no longer need to hope that your audio gear will sound good, you can force it to at any time by having the electronics compensate for shortcomings on the fly. This is a big departure from the past, as the use of DSP boxes used to be the domain of only the hobbyist or obsessive. Not anymore!
In a nutshell, a DSP is optimized for the most common tasks used in digital signal processing workloads. The list includes floating-point mathematics, the modulo operation, saturating arithmetic, multiply-accumulate (MAC), and fused multiply-add (FMA) operations. These functions are often required in the filter, Fourier transform, codec encoding, and other DSP algorithms. Digital signal processors are typically built to run a number of these operations in parallel (a superscalar architecture) for much faster processing with lower clock speeds than a typical CPU.
Overall, DSPs are optimized in two key areas compared to general-purpose CPUs. They accelerate common DSP mathematical operations in hardware and boast specific memory architectures designed for real-time data streams. The net result is faster and more efficient processing of audio and some other data types.
Many more high-profile companies are starting to embrace the transformative power of a properly-used DSP. From accurately making 3D audio, to automatically optimizing music, to enabling the next generation of Bluetooth audio codecs, the increased amount of development in the DSP field is going to change how we listen in very stark ways.
The Sony PCM-1630 is a digital audio processor for professional use, designed to be used with a Sony BVU-800DA/800DB videocassette recorder, a DMR-2000/4000 digital master recorder or any other Sony professional VTR to create a professional PCM recording and playback system.
The Sony PCM-1630 was the third generation of the Sony PCM-1600 system released in 1978, (and shortly after this the Sony DRE-2000 digital reverb was released) which was followed by the Sony PCM-1610, and ultimately the Sony PCM-1630. The Sony PCM-1600 used a U-Matic format VCR for its transport, connected to external digital audio processing hardware.
The Sony PCM-1600, PCM-1610 and PCM-1630 were widely used for the production and mastering of many of the first CDs in the early 1980s. Once CDs were commercially introduced in 1982, tapes recorded on the PCM-1600 were sent to the CD pressing plants to be used to make the glass master disc for CD replication.
Digital dubbing with no deterioration
When the unit is connected to two recorders, sound can be dubbed digitally with no deterioration, due to the digital dubbing function of the unit.
Electronic editing
When the unit is used with a DAE-1100/1100A digital audio editor and two recorders, a program can be automatically and electronically edited with precision, and the quality of the editing is more excellent than splice-editing of an analogue tape.
Serial data format and interchangeability
A serial data format is employed as a digital input/output format. Since this format is interchangeable with that of a PCM recording and playback system using a Sony PCM-1610 digital audio processor, it is possible to directly transmit and receive digital data between this unit and the Sony PCM-1610 system.
Tapes recorded with this unit can be played back with a Sony PCM-1610, and vice versa. This unit can be used instead of a PCM-1610 in a PCM recording and playback system using a PCM-1610. (The remarkable difference in the two units is that the PCM-1630 does not incorporate a time code generator, while a PCM-1610 has a built-in time code generator)
Two sampling rates selectable
A sampling rate is selectable for recording at either 44.056 kHz (corresponding to the NTSC TV system) or 44.1 kHz (for a compact disc and digital audio system). In an external sync mode, the unit is automatically synchronised with either frequency by synchronising with an NTSC composite sync signal or a word sync signal.
Linear phase response
To improve the phase response, phase compensation filters are incorporated in the A/D section, and over-sampling FIR (finite impulse response) filters are incorporated in the D/A section.
How it works
A PCM processor is a device that encodes digital audio as video for recording on a videocassette recorder. The adapter also has the ability to decode a video signal back to digital audio for playback. This digital audio system was used for mastering early compact discs.
High-quality PCM audio requires a significantly larger bandwidth than a regular FM analogue audio signal. For example, a 16-bit PCM signal requires an analogue bandwidth of about 1-1.5 MHz (compared to about 15-20 kHz of analogue bandwidth required for an analogue audio signal), and, clearly, a standard analogue audio recorder could not meet that requirement. One solution arrived at in the early 1980s, was to use a video tape recorder, which is capable of recording signals with this high bandwidth, to store the audio information, but a means of converting the digital audio into pseudo-video was necessary.
Such an audio recording system therefore includes two devices, namely the PCM processor, which converts audio into pseudo-video, and the video tape recorder itself. A PCM processor has the analogue audio (stereo) signal as its input, and translates it into a series of binary digits, which, in turn, is coded and modulated into a monochrome (black and white) video signal, appearing as a vibrating checkerboard pattern, modulated with the audio, which can then be recorded as a video signal.
This video signal can be stored on any ordinary analogue video tape recorder, since these were the only widely available devices with sufficient bandwidth. This helps to explain the choice of sampling frequency for the CD, because the number of video lines, frame rate and bits per line end up dictating the sampling frequency one can achieve, that sampling frequency of 44.1 kHz was thus adopted in the Compact Disc, as at that time, there was no other practical way of storing digital sound than by a PCM Converter & video recorder combination.
The sampling frequencies of 44.1 and 44.056 kHz were thus the result of a need for compatibility with the 25-frame (CCIR 625/50 countries) and 30-frame black and white (EIAN 525/60 countries) video formats used for audio storage at the time. (Note that neither PAL nor NTSC was itself used, only the luminance signal, or black and white information, of the composite video output from the PCM processor was used with no colour subcarrier present.)
Most video-based PCM processors record audio at 14 bits quantisation, and a sampling frequency of 44.056 kHz for EIAN countries (or 44.1 kHz for CCIR countries.) However, some of the earlier models, such as the Sony PCM-100, recorded 16-bits quantisation as well, but used only 14 of the bits for the audio, with the remaining 2 bits used for error correction, in case of dropouts or other anomalies being present on the videotape. A PCM adaptor can only store a single stereo signal and is not capable of studio multi-track recording.
Shortly after the arrival of the Sony PCM-1630, an integrated system was developed, known as DAT (Digital Audio Tape) which did not require the external VTR to record, instead it used an inbuilt 4mm wide taped based helical scan system, which all but wiped out these larger PCM-1600 systems.
I recently added a Roku Premier to my Home Theater. The video quality is great, but the audio is always PCM Stereo. My receiver indicates it is not getting any kind of surround information, as would occur in Dolby Digital or Dolby Digital+. I mostly watch Hulu, but I've also tried numerous other apps including the Roku channel. All of them have the same issue -- always PCM Stereo. I know it's not my receiver because if I watch the same show via the same app through my BluRay player instead of the Roku, then I get Dolby. It's only the Roku that sends as PCM Stereo. I've tried all combinations of the Roku audio options from the settings menu -- audio, only Dolby, only Dolby+, only... It's always the same result -- everything is PCM stereo. Rebooting the Roku and/or installing latest updates also doesn't change anything.
Thanks. I had leveling on. After disabling volume modes, I can get Dolby audio. I still get PCM during Hulu commercials and Hulu show intros, which can result in a dramatic change in volume between show and commercial. If I don't hit mute during commercials, the sound is quite unpleasant. However, this volume change sounds like a Hulu problem or something intrinsic to Dolby vs PCM stereo, rather than one with Roku.
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