Saidera's Mastering

 DSD Mastering (Part 1) "Transparent Analog Sound" I'm Morisaki, the chief engineer. Today, I would like to introduce the best DSD mastering method for genres such as club music, R & B, HIP HOP, and ROCK. How to use the KORG MR-2000S to play "fat sounds like an analog recorder" for genres other than jazz and classical music? [What is "fat sound like an analog recorder"] (1) Thickness of mid-low range (2) Size of sound image (3) I think it is a glossy overtone. Even through analog equipment and vacuum tube equipment, an analog feeling cannot be expressed unless this condition is met. The sound of DSD has a wide range, a very smooth high range, and a wide range of low range. Therefore, I challenged that if I could increase the sound density, I should get closer to the ideal sound. [ Method ] Instead of playing the sound flat, it plays the thick low range and thick mid range. First, place the MR-2000S on a 42mm thick plywood board to take thorough measures against vibration. Circuits with analog stages (including power supplies) can reproduce dense, well-defined sounds by thoroughly tuning the housing vibration countermeasures and insulators (foot), whether it is an amplifier or DAC. The power cable used for the MR-2000S is an Allegro cable. The compatibility is outstanding, and the center components such as vocals, snares, and kicks are solid and the sound image becomes large. DSD is often thought of as a "beautiful sound," but with this method, the MR-2000S makes a sound pressure-oriented sound. For the MR-2000S analog out line cable, select the Acrotech 8N cable, which has a slightly grainy texture, and the transparent cable, which has power in the low range, to add gloss to the sound. Finally, lock all digital devices with a word clock for a soft yet tight texture. The sound comes out in front of you. It's a warm but clear sound, so there's almost no need for high-frequency EQ. Since the low range is also firmly stopped, it is okay to do almost no low cut. Conventional methods to create an analog feel include passing through an analog comp or analog EQ, or passing through an analog tape recorder amplifier, but these methods provide an analog feel but tend to lose the delicate texture and speed. did. With this method, you can create a fat sound while maintaining transparency and omission, which you could not make even if you made full use of analog equipment. This is my favorite mastering method these days. Not only DSD sound sources recorded directly on MR-2000S etc., but even if the mix master is a PCM sound source, it is possible to obtain the same nuance by up-converting DSD mastering! P.S. When up-converting a PCM file to a DSD file, a sound source with a level that is not at the limit will not be hard and you can play back a good sound. We recommend that you allow enough headroom for your mix. DSD Mastering (Part 2) "Expression of low mids that expresses the thickness of musical instruments" The characteristic of DSD sound is reality. In particular, it has the expressive power of low mids that expresses the thickness of musical instruments. The sound of the acoustic guitar body, the sound of the distortion guitar amp box, and the sound of the vocal chest. Since the gap between sounds is connected by sound, you can create a warm sound. An artist who was remastered the other day. The request is "I want Gibson's acoustic guitar to have low and mid-low frequencies to make the sound thicker, but I want to keep the original atmosphere for the overall omission." The first setting I came up with was the combination of the KORG MR-2000S and the Allegro cable. The material was a 48kHz, 24-bit WAV file, so I up-converted it to 2.8MHz DSD data. The reason why 2.8MHz is adopted from 5.6MHz is to express the size and thickness of the sound image. I wanted to add an analog feel to the line cable, so I connected the MR-2000S to the Prism Sound ADA8 (AD / DA converter) with the Acrotech 8N. At first, I passed through the analog comp Prism Sound MLA-2, but the TD sound source was thick enough, so I finally removed it. For sound omission, the power of ADA8 was taken from the power supply box of Audio Prism Power Foundation 1A. This power supply box has a built-in noise filter, so it has a very clean sound. Another feature of DSD sound is that it has a large sound image and a quick rise (although it seems to be persistent every time). High can be removed without EQ, and the attack time of the comp can be slowed down, so you can create a fat drum sound. However, low-end processing is very severe. A good monitor environment is absolutely necessary as the difference of 1Hz affects the sound. I really liked the sound. I was happy when I was told "I wanted this sound" the moment I played it back. When I first started using the MR-2000S, I made a lot of trial and error to produce a gutsy sound, but now I use PCM, DSD5.6MHz, and DSD2.8MHz according to the genre and artist's taste. I think the most important thing in mastering is to bring out the best sound quality in playback. DSD Mastering (Part 3) "Transparent voice with little distortion, Noriyuki Makihara / Ringo no Hana" Takayuki Makihara's new single "Ringo no Hana", which is already familiar to everyone in the CM of the opening of Shin-Aomori Station on the JR East Tohoku Shinkansen The TD sound source of "Hana" was 5.6MHz DSD data recorded by KORG MR-2000S. Recently, I've been using a lot of "DSD mastering" methods to up-convert PCM data such as WAV to DSD data with KORG AudioGate, but the sound directly TD to the DSD recorder is different! Vocals that are gentle and warm. When I asked engineer T, "Since I started TD on the MR-2000S, I have almost stopped using EQ. When I dropped it on PCM, I used to apply EQ to improve the voice and outline it, but DSD is high. It's getting simpler and simpler because you can exit without EQing the area. " "Since the transparency of the sound is different, the degree of reverb application has also changed." And that! In the case of PCM, the sound quality changes slightly even with bounce on the DAW or high sampling recording on the master recorder, but when recording on the MR-2000S, the nuance of the sound is almost the same before and after recording. If you do TD work while listening to the input through of the MR-2000S, you can finish the sound as close as possible to the engineer's image. This is one of the factors that makes DSD more musical than other formats. In the DSD mastering of this work, I made a solid foundation of the sound before applying EQ and COMP. First, the word clock ROSENDAHL Nanoclocks used for recording to reproduce the sound that engineer T was listening to during TD was also used during playback, which increased the thickness of the low range and the range of the high range, and the pyramid type was well-balanced. It became a sound. In addition, Allegro's power cable is used to give the contour and presence of the voice. (See DSD Mastering (Part 1) "Transparent Analog Sound") With this cable, the sound image is large and mono instruments such as vocals, kicks, and snares come out in front. EQ is about correction. I added 1.4dB to 1kHz to get the core of the vocal. It's exactly "you don't have to EQ the high frequencies", so you can create a clear and transparent voice with less distortion. COMP was very shallow with a ratio of 1.6: 1, an attack time of 50ms, and a release time of 300ms. The attack time is delayed and the rhythm is tightly finished by reacting to the kick instead of the vocal. DSD Mastering (Part 4) "Selection of the key analog cable" You know the sound quality of DSD recording in "Sound & Recording Magazine presents Premium Studio Live", which is presided over by Sanreco magazine. "DSD recording" allows you to record the atmosphere and reality of the performance as it is. You can record the fine nuances of the artist as it is. The sound is natural, has a quick rise and is glossy. I want you to know that the sound of DSD mastering is powerfully good even if you are recording and mixing with PCM! Today is about making sounds for DSD mastering. "DSD mastering" in mastering for CD refers to mastering in which playback during mastering is performed in DSD. Not only when the TD master is DSD from the beginning, but also when it is PCM such as 48/24, PCM → DSD up-conversion is performed with KORG AudioGate and it is played back with KORG MR-2000S. However, it is not possible to express the goodness of DSD simply by playing back with the MR-2000S. The MR-2000S is firmly equipped with vibration countermeasures, laying the foundation for sound creation when playing back using analog equipment, line cables, and power cables, and making adjustments that match the genre maximizes the goodness of DSD. DSD mastering that makes the most of it is possible. The KORG MR-2000S has a built-in DSD / Analog converter, and the out is a Canon or pin analog. The choice of analog out cable is key to DSD mastering. In PCM mastering so far, I used to work on the transparent MUSICLINK Super to express the low range richly, but DSD is a sound with a sufficiently extended low end / high end, so around 100Hz to 4kHz, kick / Uses an Acrotech 8N cable with a snare / vocal core. The power cable uses an Allegro cable that allows the center component to come out firmly. By firmly producing the center component, not only large audio systems but also DSD mastering makes it possible to create punchy sounds even when listening to small-diameter systems such as headphones and PC speakers. (1) Thickness of mid-low range (2) Size of sound image (3) Glossy overtones (4) Analog sound that firmly produces center component and (5) Natural and fast rise, DSD sound You can listen to a compatible finish. That is my DSD mastering. I suppose he ’s no longer using these though. profile This is the blog of "Sidera Mastering", a mastering studio in Shibuya-ku, Tokyo. From recording to mixing to mastering, we offer the best technology and hospitality for everything related to music and sound.These are the basics of the basics, so especially students who are learning acoustics should master them now! On-site, fast, safe and reliable are required. And don't forget to smile. In rare cases, the TD sound source may be on the rise. Since the low end and high end are overstretched and there is no core of sound, there is no sense of sound pressure even if the volume is raised. It is possible that the TD was performed only with small speakers, or the monitor was "kamaboko". Today is the mastering at that time, Donshari's prescription. First of all, to suppress the low end. Try listening by lowering 1dB to 2dB from around 100Hz by band division such as multi-band compression (in the case of no compression, only level control). If you don't feel the attack of the kick, cut gently from around 30Hz with a low cut filter. Next is the high frequency check. Instruments that are easy to shake are hi-hats, cymbals, shakers, tambourines, etc. First, let these instruments fit into the orchestra so that they do not come out too much. Cut a little from around 4kHz with multi-band. If the sound quality changes, increase the frequency to 8kHz and 10kHz. It is possible to create a contoured sound by cutting from a low frequency. If you adjust the balance so that 100Hz to 10kHz is flat, and frequencies above and below it roll off with a gentle curve, you can finish each instrument with a good balance. Once the range is balanced, it's time to create the sounds for each instrument. The key to getting a well-balanced sound on any system is how to listen to the center component instruments, kicks, snares, vocals, and bass. I decide the texture of the vocal first. The EQ point varies depending on the voice quality, but 1kHz to 1.5kHz is added to bring out the edge of the voice. 1kHz is the clearest outline, but if you hear a hard voice, raise the frequency a little. Listen to the balance with the orchestra while changing the frequency to 1kHz, 1.1kHz, 1.2kHz, 1.3kHz, 1.4kHz, 1.5kHz. There are moments when the voice sounds softer as the frequency is raised. The sweet spot is a little lower than that. If this point is found, you can express a vocal with a strong presence without changing the voice quality even if you EQ other bands. The kick is 120Hz to 180Hz plus, the snare drop is 6kHz plus, and the bass cuts around 60Hz to adjust the fog with the kick and improve the pullout. If you adjust the balance with a multi-band, low-cut / high-cut filter, you can create sounds with less EQ. Mastering work tends to focus on the sound creation of individual instruments, but the first balance adjustment is the most important work. For that purpose, a monitor speaker that can accurately judge the frequency range is required. If the balance is difficult to understand with the small monitor alone, check it in various environments such as headphones, radio cassette recorders, and PC speakers. The piano is the second most difficult instrument to master after vocals. The fundamental tone of the 88-key piano range is 27.5Hz for the lowest note and 4186Hz for the highest note, making it an instrument with a very wide frequency range. In other words, it tends to overlap with the frequency bands of other musical instruments. Not only the types of pianos such as Bösendorfer and Steinway, but also the size of the hall and recording booth, and the arrangement of the microphones will greatly affect the tone. Today is a difficult task, about making sounds by mastering the piano. Since the piano is an instrument that strikes the strings with a hammer, the key point is how to express the attack feeling that the hammer hits the strings. If there is no sense of attack, it will sound like a squishy piano. Let's understand the following points to make a piano attack. The highest note of an 88-key piano is 4186Hz (remember about 4kHz). This means that the band above that is composed of overtones, so in order to get the core of the sound, make the sound in the band below this frequency. 1. Put out the core of the sound 1.2kHz to 1.6kHz 2. Gloss 2kHz-4kHz 3. Gives a sense of transparency 4kHz to 8kHz 4. Warmth in the low range 120Hz-250Hz [Explanation] 1. The core of the sound uses a band slightly above the vocals. As with vocals, use a lower frequency if you want to outline the piano, and a higher frequency if you want to blend in with the orchestra. 2. This frequency is the band most sensitive to the human ear, as you can see from the equal loudness curve. By emphasizing it, the feeling of volume will increase, but since it is a band where the tone color is likely to change, first put out the core of the sound and then use it for correction. 3. If you emphasize this frequency a little, the transparency will increase, but since hardware such as hats and cymbals also exist in this band, adjust it last while balancing with the orchestra. 4. When you feel the tone is hard, raising the low range will enrich the sound and suppress the attack feeling of the high range relatively. If the EQ is not caught in the piano sound source that is driven in because there is no low range, the high range is suppressed. The best way to make piano sounds is to listen to live music and experience it. Let's experience the splendor and tone of real performance at live jazz and piano solo concerts. PS. DSD mastering is really good for piano mastering! The TD sound source contains a lot of musical elements such as the thoughts that the artist put into the performance, the groove, the equipment of the engineer, and the commitment of the studio. However, it can contain unexpected non-musical elements. One of the most difficult things to judge is distortion. There are two types of distortion: "musical distortion" and "electrical distortion". The former is a sound intentionally created by artists and engineers using guitar amps, vacuum tube equipment, and analog equipment. The latter is a sound that has been distorted due to input overload of microphones and digital equipment. Sounds input to vacuum tube equipment and analog equipment at an appropriate level add a pleasant "musical distortion". "Musical distortion" is a very important element that adds volume, power, and momentum in mastering J-POP, ROCK, R & B, etc. However, if it exceeds a certain level, it will cause an unpleasant and unpleasant distortion. In mastering, a monitor environment that can judge the critical point of this level is a very important factor. In mastering work, it is necessary to listen to every corner of the musical elements of artists and engineers. And in the case of distortion, by judging whether it is musical or not and incorporating it into the sound, it is possible to create a sound that is surely transmitted to the listener. The monitor system of Sidera Mastering perfectly clears the above elements, so anyone can judge with confidence. Of course, to reach this sound, we went through hundreds of trials and errors, including how to install all the equipment, cable selection, vibration countermeasures, and room acoustic adjustment. Even now, daily adjustments are indispensable. The quickest way to adjust the monitor speakers is to make small adjustments and make small adjustments. It also helps to train your ears. The other day, we had a "line cable / AD converter thorough viewing" at the "Sound Morning Activity" Sidera Morning Session. We asked the participants to experience that the accumulation of attention to small differences in sound greatly affects the final finish. A "high resolution, uncolored monitor" is essential for mastering to correctly distinguish "small differences in sound". Today, I will explain some of the points of a thorough monitor environment for sidera mastering. [Speaker settings] Place the speaker on a solid table and adjust the left-right angle and distance from the listening point in millimeters. The key low-frequency adjustment is performed by vibration countermeasures with lead plates and diffusion by interlocking. The sound of the entire room is closer to tone adjustment than sound absorption. I pasted SONEX and glass wool on a place with standing waves and made fine adjustments while repeating the audition. Since it does not absorb too much sound, there is no loss and it is possible to obtain a considerable monitor level even with a small wattage. The studio is large enough on both sides of the speakers and has a ceiling height of 4.4 meters, providing a deep and spacious monitor. [Equipment settings] The points of equipment setting are vibration countermeasures and heat dissipation. The equipment is placed on TAOC boards, flower boards, cypress laminated wood, thick plywood, etc. We are particular about the type of screws and how to tighten the rack mount equipment (see Two methods for improving sound quality that can be done immediately (Basics of equipment installation)). To add warmth and thickness to the low end, most equipment is placed on wood and, of course, partially metal insulators are used to adjust the range and balance. Install the equipment with as much space as possible to improve heat dissipation. Fans are installed where necessary to improve air flow. If the equipment gets too hot, the sound will drip, so you need to take proper measures. [Cable selection] The line cable of the monitor line is based on Saidera Ai SD-9003, and the power cable is based on ACROTEC's product. The criteria for selection is "a cable that is uncolored, has a wide range, and rises quickly." Adjust the monitor environment in the order of (1) room acoustic adjustment, (2) equipment setting / adjustment, and (3) cable selection at the end. Adjust the room acoustics and all other equipment based on the sound of the reference CD you are accustomed to listening to. It takes time, but this is the fastest way to create the correct monitor environment. Home mastering tips. As I explained in "Secret Mastering Technique Top Secret Disclosure" (January 27, 2011 Rock oN Seminar), my mastering has three major processes.

1. Determine the character and nuance of the sound. 2. Turn on the volume so that it is not distorted to the required level. 3. Fine balance adjustment. Sound character addition in mastering is done by selecting analog equipment, AD, DA converter, cable selection, etc. The TC Electronic SYSTEM 6000's MD3 has a bit of a Brick Wall Limiter applied so that the peak doesn't eventually hit as the level increases. For fine balance adjustment, I like GML Model 8200 (analog EQ) or MDW HiRes Parametric EQ (digital EQ). The same thing applies to mastering work with pluginsHow? 1. Insert a plug-in that adds a sound character just by passing it through even if the parameters are flat. For example, Pultec Bundle EQ and JOE MEEK. This is just to add the unique sound character of the plug-in by inserting it. Listen and compare different plugins. 2. For the level limiter, choose a plug-in that is easy to use and that you like the sound quality, such as WAVES L1, L2, Digidesign Maxim, and McDSP ML4000. However, excessive limiting is prohibited. Watch your input level closely. 3. For fine balance adjustment, we recommend using an EQ that allows you to visually see which band you are operating. Of course, you can give priority to sound quality and choose one that does not. I use 3 digital EQs in the System6000 series so that the EQ curves are connected neatly. And I store about 20 kinds of frequently used frequencies, Q, and level settings for each genre. If you make sounds while calling and listening to this memory, the work is faster than creating from scratch, and appropriate sound processing that suits the genre is possible. Setting store and recall is one of the plugins' strongest points! The YAMAHA NS-10M is the most famous monitor speaker in a recording studio. (The photo is an improved NS-10M STUDIO) Although the input resistance is only 50W, the output of the standard Amcron PSA-2 of the power amplifier that drives is as much as 150W. A smaller amp will sound good, but recording engineers are looking for instant sound. The speaker must also come when the kick sounds. If it is a car, the difference in acceleration between a light car and a car with a large displacement is the feeling that it reacts instantly when you step on the accelerator. The frequency response of the NS-10M is 60 to 20,000Hz, and the crossover is 2,000Hz, so if you input a sound with a slight peak, or if you EQ too much in one band, especially in the high range, the tweeter will stop. It is very difficult to master the monitor system with such settings. Experience and know-how are required to make the NS-10M sound in a well-balanced and loud volume. If you can make this speaker sound, you are a full-fledged person. At that time, the NS-10M speaker unit was considered a consumable item, and as soon as the sound was lost, it was replaced with a new unit. So there was plenty of stock of replacement units. Replacing the speaker unit changes the character of the sound. I used it in the session after aging and getting used to it, but the NS-10M for backup was still waiting for a few pairs. The engineers were completely familiar with the characteristics of the monitor system for each studio, such as the 2ST sound and the 6St sound. Through this experience, I learned that "understanding the characteristics of monitor systems is essential for engineers." The most attractive thing about NS-10M is its bright and energetic sound. A musical sound that excites when a musician listens to a playback in the control room. You can record a good performance if the artist comes along. Getting the artist to play comfortably is of utmost importance in music production. That's why recording engineers are particular about the sound of playback and cue box returns in order to record good performances. "Turn the teleco when the other person wants to play" "We are recording music instead of recording sound" "Then assistants can participate in music production" "Musicians and teleco Talk to me. " I remember this word when deciding between songs in mastering. Through the sound of NS-10M, I learned the heart of a first-class engineer. * Teleco: Tape recorder. SONY-PCM3348, STUDER-A820, etc. "Put it without play. Tighten the screws firmly." These two are the basics when installing equipment. Today, I will talk about removing backlash from speakers and speaker stands and tightening screws for rack mount equipment as the basics of vibration countermeasures. If the equipment is properly equipped with vibration countermeasures, the focus will be clear and a transparent sound can be reproduced. The basic is to put it firmly so that there is no backlash. Business cards and word cards are useful for removing backlash from speakers and speaker stands. First, shake it by hand to check if there is any play in the speaker and insulator, speaker stand and floor. If there is no play, no adjustment is necessary. If there is any play, I will put a word card in the gap. Adjust the thickness as it is, fold it in half, fold it in three, and adjust it according to the gap. Use a spirit level to check the level. A spirit level of 100 is sufficient, not a professional one. Be sure to bite the washer when mounting an outboard such as an analog EQ or comp in a rack. By doing so, it can be firmly fixed. Even if you intend to tighten the screws firmly, they will loosen over time, so tighten them once every few months. Furthermore, the sound output varies depending on the material of the rack mount screws. The tendency is that iron screws have a gutsy core sound, and stainless steel screws have a wide range and a beautiful sound that extends to high frequencies. This is an advanced technique, but I use it properly according to the equipment. If the equipment is placed directly instead of rack-mounted, some professional equipment does not have rubber feet. Place thin rubber sheets of about 4 cm x 4 cm on the four corners of such equipment and place them firmly. "Put it without backlash. Tighten the screws firmly." Please challenge the two methods of improving sound quality that can be done immediately. Raise your ears to listen to music I'm Morisaki, the chief engineer. Today, I would like to talk about "how to grow ears". It is very important to know the sound of live instruments not only for recording but also for TD and mastering. When I was a student, I often listened to jazz at Yokohama Airegin, orchestra at Suntory Hall, and live music at the local civic hall for studying piano solos and string quartets. In addition, I became a recording assistant at Onkio Haus, and listening to a lot of live drums is especially useful for my current job. Not to mention the performance, the sound changed in various ways due to different instruments, different rooms, different kick beaters, different sticks, different mic arrangements, and different equipment. Mastering applies EQ to the finished 2MIX, so if you EQ easily, unnecessary bands will rise, and conversely, instruments that you do not want to lower will drop. It is OK to increase the ideal sound pressure, spread, and sense of speed, but it is NG to change the texture and balance of the instrument. When I do EQ, I imagine the situation where the instrument is ringing in detail. If it is a kick, consider it separately from the sound of the beater hitting, the body ringing, and the room ringing. By doing so, for example, when you want to make the kick sound tight, it will be easier to judge whether it is better to emphasize the attack of the beater to make the rhythm stand, or whether it sounds tighter to cut the low end and suppress the room noise. If you emphasize the ringing of the body, you will get a sense of volume, so you can see that this band is not touched. Similarly, if it is a snare sound, it is better to emphasize the rim sound or the snappy sound, etc. If you make the sound while imagining the artist playing, it is musical regardless of frequency. You will be able to make sounds. By listening to live music, you will naturally acquire the original tone, sound, and balance of the instrument. Also, by understanding the pronunciation mechanism and playing method of musical instruments, it will be possible to create more realistic sounds. When communicating with a P.S. artist, it's easier to convey "I tried to emphasize the attack of Vita" rather than "I tried to raise 100Hz". The other day, there was mastering of Kenichiro Nishihara's new album "Rugged Mystic Jazz For TALISKER", which presides over the label "Unprivate Acoustic"! We received a comment on Twitter about the finish of DSD mastering! Please listen to it! ”Speaking of the essential mastering, I experienced the first DSD in my life. It will be familiar to those who are viewing the sidera blog etc. OLD NEVE played an active part in this production, but thanks to DSD, the analog feeling has further increased I got closer to the ideal sound quality. "These figures are just images, but the sound is invisible (the waveform can only visualize the dynamic range, but the balance, content and color cannot be seen), so please use it for mastering requests, etc. .. The appeal of music will be greatly enhanced by finishing it according to the genre and purpose. Next time, I would like to talk about points to note when making sounds. This is the point when DSD up-converting the PCM sound source of KORG AudioGate. As I mentioned in "DSD mastering for distribution" the other day, up-converting a PCM sound source to DSD will increase the resolution. At that time, what you have to be careful about is that not only fine nuances but also electrical distortions are up-converted at the same time. Therefore, distortion may be noticeable in a sound source with the level set to the limit with PCM. To take advantage of the DSD format, such as spaciousness and depth, it is necessary to leave headroom for TD. A margin of about 0.5 dB to 1 dB is sufficient for headroom, but first be careful not to turn red. It is an important point to express a natural and elastic kick, a stretchy vocal, etc. However, when directly TD to the KORG MR-2000S, DSD is stronger against clips than PCM, so it is okay to record at a level where red is lit momentarily. As I wrote in "Use of PCM and DSD up-conversion properly" before, when mastering a PCM master (96KHz 24bit / 48KHz, 192Khz, etc.), DSD up-conversion is performed and it is played back on MR-2000S. There are two types of DSD, 5.6MHz and 2.8MHz DSD, depending on the music. 5.6MHz is characterized by a wide range and fine sound, but 2.8MHz is used when you want to give a slightly more extruded feeling. If you still want the volume to go up to the maximum level, you can still play it in Pro Tools. Please also refer to the past articles on DSD mastering & recording. It is used in various works. If you would like to experience up-converted DSD mastering, please feel free to contact us. P.S. Have you read the November issue of Sound & Recording Magazine? Special project "Record with DSD, listen to DSD (P78-86)" This is a must-read content from the basics of DSD to mastery. Mr. Tomohiko Gondo talks about the changes in production since he started recording masters with DSD, and Mr. okuda supa also talks about the difference and proper use of the mix between DSD and PCM. For example, an image taken with a mobile phone. There are not enough pixels for the designer to use as-is for print data. By up-converting to the resolution of large format film, you can adjust the contrast and even the slightest color delicately or boldly without breaking the lines. Similarly, if you up-convert a sound source such as PCM 48kHz 24bit to DSD 5.6MHz, the breathing of vocals that was difficult to hear until then and the buried groove will be reproduced vividly. Curiously, it even feels like the decimated sound is back when it's converted to PCM. I think the real goodness of DSD is reality. The sound rises quickly, and even fine sounds are reproduced firmly. It has the expressive power as if you were playing in front of you. After up-converting, I often get the impression that "the sound image of the song can be heard louder", but that is because DSD has less peaks in the sound and the level can be higher than PCM, and the rise is quick, so the voice is natural and missing. Because it will come. Since it can be removed without contouring the voice, it is possible to create a sound with a solid mid-low range. A sound source mastered in this way does not easily lose its nuance even when compressed to MP3. In other words, it is also very effective for mastering for distribution. There was a big difference in the way of hearing after compression between mastering in the PCM area and up-converting and then DSD mastering. By up-converting, it is possible to create detailed sounds. I think it's because the sound that conveys what the artist wants to convey has finally improved the level of excitement. P.S. DSD 5.6MHz up-conversion DSD mastering can flexibly support high-quality sound distribution. When creating a 96kHz 24bit distribution master, the up-converted sound source (5.6MHz DSD) is output in analog form from the KORG MR-2000S. If necessary, make a sound with analog comp and EQ, and then convert it to 96kHz 24bit with an AD converter. Digital EQ and COMP can also operate at 96kHz 24bit, so more detailed final touch is possible. Of course, you can also create the 2.8MHz DSD (DSF) master used in OTOTOY's DSD distribution at the same time. Each of the PCM and DSD sounds has its own advantages, and we use them properly according to the genre and the direction of the sound required by the artist / producer. Mastering with a PCM sound source has sound pressure and punch, and is suitable for rock and hip hop where you need to turn on the volume. Especially for works where the sound is created by mixing, PCM is more cohesive, and the part that is not too visible becomes a cool sound, so I often select that. On the other hand, up-converting to DSD increases the resolution of the sound and expands the depth, so it is ideal for works that make use of the nuances of vocals and rock one-shot instruments. Up-converting to club music such as house, which is based on the sound of analog records, will make the sound comfortable and make you want to raise the volume. Or if you want to master each song by making the best use of its goodness. The songs that want to suppress the spread a little, the songs that want to go with power rather than the omission are PCM, the songs that want to widen the distance between the artist and the orchestra because it is a ballad, and the songs that want to make the chorus strings magnificent are up-converted to DSD. However, in this case, it is necessary to firmly determine the standard music in order to achieve a sense of unity as a work. Mastering PCM and DSD is not "which is better or worse", but it is important to understand the goodness of each and use them properly. By performing the optimal processing in the optimal way, the song can be finished to reach the listener most. In the actual witness mastering, be sure to listen to the sound of PCM / DSD mastering, check the direction, and get closer to the best finish. About the basics of using EQ. To learn how to operate the EQ, you need to understand what kind of sound components exist in each frequency band. The graphic equalizer is the best way to master the basics. Let's check the bandwidth with the Glyco that comes with iTunes. iTunes Glyco is a 1 / 2oct, 12Band type. The low range starts from 16Hz and the frequency doubles. "Frequency band and sound characteristics" This is just my idea. Ultra low range (32Hz or less) → Low range air feeling, room noise Low range (32Hz-125Hz) → Volume and thickness of low range Mid-low range (125Hz-250Hz) → Thickness of low range Mid range (250Hz ~ 1kHz) → Volume Mid-high range (1kHz to 4kHz) → Thickness of high range and center of sound High range (4kHz to 16kHz) → Shine, luster, transparency Ultra high range (16kHz or more) → High range air feeling For example, if you boost / cut each band with Glyco while listening to the reference CD, you can check which band is used to create the sound. You can also check the sound you have mastered and TD. If you cut the low frequencies a little with iTunes Glyco and it gets better, go back to TD work and try cutting in the same way. If you can understand the band to move, the range of sound creation should be greatly expanded. iTunes Glyco is very simple, so if you have a plugin or analog Glyco, please try it. How to use the plug-in. "Multiband Comp" One of the selection criteria for multi-band comps is to first check the input-through sound. And check for low ratio settings like 1: 1.2. 1: 4 and 1: 8 are sound-making, but with input-through and low ratios, the character itself can be used to make sound, just like changing cables to change nuances. Each has its own character, such as transparent sound, thick sound, and clear sound. In addition, by combining plug-ins, it is possible to use both sound quality and operability. One is input-through to characterize the sound, and the other is fine-tuning. Rather than setting parameters for each song from scratch, create several types of default settings according to the genre in advance, and if you start from there, not only will you be able to work faster, but multi-band parameter adjustment will be more severe. Can be "Comprehensive tool" In mastering, the amount of information of the master is maximized with the simplest possible setting. The same is true for plug-ins, starting with EQ only, then gradually increasing parameters such as comp limiter, multi-band maximizer, and M / S processing. However, once the settings are decided, the plug-ins that are not needed are completely removed from the insert. EQ is the basic method for creating low-frequency sounds such as kicks and basses, rather than using COMP or limiters. For example, 26Hz is 0.2dB, 42Hz is 0.2dB, 121Hz is 1.3dB, and so on. Be especially careful when creating the sound of the center component. The multi-band maximizer is often used to adjust the level of each band rather than fine-tuning it. Use exciters, amp simulators, etc. only if you want to give the final touch. This is useful when you want to add brilliance and overtones that cannot be expressed by EQ. I would like to talk about how to use the power outlet.The biggest difference between a PCM sound source and a DSD sound source is the speed at which the sound rises, especially the rise in the low range. The soft, crisp bass has the nuances of an analog record and is fascinating, but you need a perfect monitor system to take advantage of or kill it. As I wrote in the blog the other day about the introduction of digital amplifiers, cable routing and settings have also been completely improved. Because, the sound of DSD has a more severe balance of low and high frequencies than PCM. We aimed for such a monitor system that reproduces the information contained in the material without adding extra elements.It's easy to think that it's fine-tuned, wide-range, and suitable for high-end audio, but I think the real goodness of DSD is reality. The sound rises quickly and even the finest sounds are reproduced, giving you the expressive power of playing in front of you. By up-converting the 44.1kHz, 16-bit sound source of PCM to 1bit / 5.6MHz, the breathing of vocals and the buried groove that could not be heard until now are vividly reproduced. It's strange, but it feels like the decimated sound is back when it's converted to PCM. In CD mastering, the final format is 44.1kHz, 16Bit, but if you master it as it is with PCM and master it after up-converting, there will be a big difference in how you hear it. I think that is because not only the way of hearing but also the level of impression is improved. This method is also very effective for mastering for distribution because the nuances are not easily lost even if converted to MP3. In Sidera Mastering, Sony multi-channel integrated amplifier TA-DA9100ES is used as the monitor amplifier. This amp is an inspirational amp developed and designed by Sony, which Mr. Takashi Kanai is involved in sound quality.Mr. Kanai is a super engineer representing Sony in the good times, who has been indebted to us through work such as the flute ensemble "LYNX" with our company Ono. Ono has a long interview with Mr. Kanai and other related parties about the development story in a series of articles on "Pro Sound". (← If you cannot find the back number by all means, please let us know your name, company name, and e-mail address, and we will send PDF materials to a limited number of 5 people.) The difference in sound between AV amplifiers and audio amplifiers is that AV amplifiers have less coloring. In the case of audio amplifiers, there is some coloring even if it is a flat balance. The TA-DA9100ES was also used as a mastering reference amplifier during the studio renewal in late 2008. (Until then, I used the TA-DA9100ES for the surround monitor.) I myself used the analog amplifier Boulder 500AE with a BTL connection for sidera mastering, so the TA-DA9100ES can sound the PMC MB1 sufficiently. I was a little worried if I had enough power. However, it disappeared altogether when I first heard the sound of the TA-DA9100ES. Anyway, the range is wide, the speed is high and the rise is fast. And overwhelmingly little distortion. It responds accurately to the detailed sound creation of mastering. It's a digital amp, but I thought the sound nuance was close to that of a vacuum tube amp. The TA-DA9100ES supports surround and has a large power of 200 watts for each channel. The instantaneous power of sound is especially important for expressing the intonation of music. The sound of ff and R & B's fast-starting kick in a classical orchestra rang lively. It is very important to fix the volume in mastering work, but since it is a digital amplifier, it can be fixed to a certain value every time with a volume of 0.5 dB step, and the time from turning on the power until the sound stabilizes is extremely long. There were many benefits unique to digital amplifiers, such as shortness and low power consumption. I will explain how to use it and detailed explanations from now on, so please look forward to it. ★ Development stories that are not in the catalog are also available in "Kaimaru". http://homepage3.nifty.com/kanaimaru/ What I pay attention to when mastering is to listen carefully to the nuances of the TD master. Artist's voice quality, balance between orchestra and song, breadth, depth, etc. Mastering work is not possible unless the recording engineer understands the sound that is precisely created at the time of MIX. Then what direction do you take to finish it? We will have a discussion. I try to understand first whether I want "loudness of sound image" or "loudness of volume". For example, when it is said that A (song) has a lot of volume but the sound is low, and B has a low volume but the sound is louder, first check the size of the sound image. If you need a loud sound image instead of volume, make the outline of the drum or song clear or make the mid-low range thicker. If you can give power and punch to the orchestra, you can lower the volume, so you can create a clear sound with less distortion. Inserting analog equipment is also an effective method. I often insert an SSL passive fader between DA and AD. The warmth, overtones, and transparency peculiar to analog are added. What I am particular about mastering is listening carefully to the TD sound source. Whenever I play back a TD sound source, I always listen to it on a separate line with the direct settings without the EQ and compression. Even if the EQ and comp are turned off, the sound will be colored if you pass them. The points I listen to carefully are the focus of the sound, the size of the sound image, the pitch of the instrument, and the voice quality of the vocals. If it is out of focus, even if the volume is raised, the sound will not be heard sharply and the sound will not have a sense of speed. Also, if the master sound source is large, it is necessary to increase the volume of the sound image while maintaining it. If the sound image is loud, the song can be heard loudly without increasing the volume so much. Regarding the sound quality, the kick pitch does not change, and the vocal texture is particular about the balance between the core of the voice and the overtones so that it can be heard by the artist himself. I'm always aiming for mastering where the vocals are the main characters, but I can hear the orchestra well, and the songs come to my mind without worrying about the sound.