A complete guide of and debunking of audio on Linux, ALSA and Pulse

Hey fellow penguins,
A few days ago, an user asked about audio quality on Linux, and whether it is worse or better than audio on Windows. The thread became a mess quickly, full of misconceptions and urban myths about Linux. I figured it would be worthwhile to create a complete guide to Linux audio, as well as dispelling some myths and misconceptions.
To all be on the same page, this is going to be thorough, slowly introducing more concepts.

What is sound? How and what can I hear?

You might remember from high school that sound is waves traveling through the air. Vibrations of any kind cause molecules in the air to move. When that wave form finds your ears, it causes little hairs in your ear to move. Different hairs are susceptible to different frequencies, and the signals sent by these hairs are turned into sound you hear by your brain.
In reality it is a little more complicated, but for the sake of this post, that's all you need to know.
The pitch of sound comes from its frequency, the 'shorter' the waves are in a waveform, the higher the sound. The volume of sound comes from how 'tall' the waves are. Human hearing sits in a range between 20Hz and 20,000 Hz, though it deviates per person. Being the egocentric species we are, waves below 20 Hz are called 'infrasound' and waves above 20kHz are called 'ultrasound.' Almost no humans can hear beyond ultrasound, you will find that your hearing probably cuts off at 16kHz.
To play around with this, check out this tone generator, you can prove anything above with this yourself. As a fun fact: human hearing is actually really bad, we've among the most limited frequency ranges. A cat can hear up to 40kHz, and dolphins can even hear up to 160kHz!!
FACT: Playing loud music is dangerous! If you listen to music and you are feeling a discomfort, you should turn the volume down. A true alert is when you hear a beep - this is called tinnitus, and that beep you're hearing is pretty much the death cry of the cells that can hear that frequency. That beep is the last time you will hear that very specific frequency ever again. Please, listening to loud music is not worth the permanent hearing damage, please dial it down for your own sake! <3

How does my computer generate sound?

To listen to sound, you will probably be using headphones or speakers, inside of them are cones that are driven by an electromagnet, causing them to vibrate at very precise frequencies. This is essentially how sound works, though modern headphones certainly can be pretty complex.
To drive that magnet, an audio source will send an analog signal (a waveform) over a wire to the driver, causing it to move at the frequency of that waveform. This is in essence how audio playback works; and we're not going to get into it much deeper than this.
Computers are digital - which is to say, they don't do analog; processors understand ON and OFF, they do not understand 38.689138% OFF and 78.21156% ON. When converting an analog signal (like sound) to a digital one, we make use of a format called PCM. For PCM to be turned into an analog signal, you need a DAC - or as you probably know it: a sound card. DAC stands for 'Digital to Analog Converter', or some people mistakenly call it "Digital Audio ConverteChip"
PCM stands for Pulse-code Modulation, which is a way to represent sampled analog signals in a digital format. We're not going to get into it too much here, but imagine taking a sample of a waveform at regular intervals and storing the value, and then rounding that value to a nearest 'step' (remember this). That's PCM.
The fidelity of PCM comes from two elements, which we are going to discuss next: sampling rate and bit depth.

What is sampling rate? Or: HOW SOUND GOOD?

Sampling rate is the most important part of making PCM sound good. Remember how humans hear in a range of 20Hz to 20kHz? The sample rate of audio has a lot to do with this. You cannot capture high frequencies if you do not capture samples often enough. Since our ears can hear up to 20 kHz, you would imagine that 20kHz would be ideal for capturing audio; however, a result of sampling is that you actually need twice the sample rate, this is called the Nyquist-Sannon sampling theorem, which is a complicated thing. Just understand that to reproduce a 20kHz frequency, you need a sample rate of 40kHz.
To have a little bit of room and leeway, we settled on a sample rate of 48kHz (a multiple of 8) for playback, and 96kHz for recording. We record at this frequency only to make sure absolutely no data is lost. You might be more familiar with 44.1kHz for audio, which is a standard we settled on for CD playback and NTSC. A lot of scientific research has been done on sound quality, and there is no evidence to suggest people can tell the difference between 48kHz or anything higher.
MYTH BUST: Humans cannot hear beyond 20 kHz, period. Anyone who claims to be able to is either supernatural or lying to you - I'll let you choose which.

What is bit-depth? Or: HOW IT MAKE SOUND REALLY NICE?

Remember how I told you to remember that PCM rounds values to the nearest step? This has to do with how binary works. The more bits, the bigger the number you can store. In PCM, the bit-depth decides the number of bits of information in each sample. With 16-bit, the range of values that can be stored is 0 to 65535. Going beyond this is pointless for humans, with no scientific research showing any proven benefit, though marketeers would like you to believe there's benefits.
MYTH BUST: 24-bit depth is often touted as 'high-resolution audio', claiming benefits of a better sonic experience. Such is nothing more than marketing speech, there is no meaningful data 24-bit can capture that 16-bit cannot.


We'll briefly touch on the last part of PCM audio, channels. This is very self explanatory, humans have two ears and can hear separate sounds on both of them, which means we have stereo hearing. As a result, most music is recorded with 2 channels. For some surround settings, you need more channels, this is why you may have heard of 5.1 or 7.1; the first digit is the amount of channels the PCM carries.
For most desktop usage, the only sound we care about is 2-channel PCM.


So, we've covered all the elements of PCM sound. Let's go over it quickly: sample rate is expressed in Hz and is how often a sample of a waveform is captured, representing the x-axis of a waveform. Bit-depth is the bits of information stored in each sample, and represents the y-axis of the waveform. Channels decide how many simultaneous outputs the PCM can drive separately, since we have 2 ears, you need at least two channels.
As a result, the standard audio playback for both consumers and professionals is 48kHz, 16-bit, 2 channel PCM. This is more than enough to fully represent the full range of human hearing.

How it works in Linux

So, now that we know how PCM works, how does Linux make sound? How can you make Linux sound great? A few important components come into play here, and we'll need to discuss each of them in some detail.


ALSA is the interface to the kernel's sound driver. ALSA can take a PCM signal and send it to your hardware by talking to the driver. Something important to know about most DACs is that they can only take one signal at a time, actually. That means that only a single application can send sound to ALSA at once. Long ago, in a darker time, you couldn't watch a movie while listening to music!
This problem was solved a long time ago with the use of alsalib, but doing mixing at a library level isn't a very good solution to the problem. This gave rise to sound servers, of which many have existed. Before PulseAudio, esound was a very popular one but had many problems, eventually it was succeeded by PulseAudio.


When you think audio on Linux, PulseAudio is probably among the first things you think of. PulseAudio is NOT a driver, nor does it talk to your drivers. Actually, PulseAudio only does two things that we'll discuss in detail later. PulseAudio talks to ALSA, taking control of its single audio stream, and allows other applications to talk to PulseAudio instead. Pulse is an 'audio multiplexer', turning multiple signals into one through a process that is called mixing. Mixing is an incredibly complicated subject that we won't talk about here.
To be able to mix sounds, one must make sure that all the PCM sources are in the same format (the one that's being sent to ALSA); if the PCM format being sent to Pulse does not match the PCM format being sent to ALSA, pulse does a step before mixing it called resampling. Resampling is another very complicated subject that can turn a 8kHz, 4-bit, 1-channel PCM stream into a 24kHz, 24-bit, 2-channel PCM stream.
These two things allow you to play a game, listen to music and watch YouTube, and notifications to produce a sound all at the same time. PulseAudio is the most critical element of the Linux sound stack.
FACT: PulseAudio is a contentious subject, many people have a dislike for this particular bit of software. In all honesty, PulseAudio was brought to the general public in a bit of a premature state, breaking audio for many people. PulseAudio these days is a very stable, solid piece of software. If you have audio issues these days, it's usually a problem in ALSA or your driver.

What about JACK and PipeWire?

PulseAudio isn't the only sound servedaemon available for Linux, though it is certainly the most popular and most likely the default of whatever distribution you are using. PulseAudio has become a bit of a standard for Linux sound and has by far the best compatibility with most applications, but that doesn't mean there aren't alternatives.
JACK (JACK Audio Connection Kit, a recursive acronym like GNU) is a sound server focused primarily on low latency. If you are doing professional audio work on Linux, you will already be very familiar with JACK. JACK's development is very focused on low latency, real-time audio and is critical for such people. JACK is available on most distros as an alternative, and you can try it for yourself if you so want; but you might find some applications do not work nicely with JACK.
PipeWire is a project that is currently in development, looking to solve key problems that exist in current sound servers. PipeWire isn't just a sound server but also handles the multiplexing of video sources (like a camera). Special attention has been put into working with sandboxed applications (like Flatpaks), which is an area where PulseAudio is lacking. PipeWire is a very promising project that might very well succeed PulseAudio in the future and you should expect to see appearing in distribution repositories very soon. You can try it yourself right now, though it isn't quite as easy to get started with as JACK is.
More audio servers exist, but are beyond the scope of this post.

What is resampling?

Resampling is the process of turning a PCM stream into another PCM stream of a different resolution. Your DAC only accepts a limited range of PCM signals, and it is up to the software to make sure the PCM stream is compatible. There is almost no DAC out there that doesn't support 44.1kHz, 16-bit, 2-channel PCM, so this tends to be the default. When you play an audio source (like an OggVorbis file), the PCM stream might be 96kHz, 24-bit, 2-channel PCM.
To fix that, PulseAudio will use a resampling algorithm. There are two kinds of resampling methods: upsampling and downsampling. Upsamling is lossless, since you can always represent less data with more data. Downsampling is lossy by definition, you cannot represent 24-bit PCM with 16-bit PCM.
MYTH: Downsampling is a loss in quality! This is only true in a technical sense, or if you are downsampling to less than 48kHz, 16-bit PCM. When you downsample a 96kHz, 24-bit PCM stream to a 48kHz, 16-bit stream, no meaningful data is lost in the process; because the discarded data lies outside of the human ear's hearing range.
FACT: Resampling is expensive. Good quality resampling algorithms actually take a non-trivial amount of processing power. PulseAudio defaults to a resampling method with a good balance between CPU time used and quality.

What is mixing?

Mixing is the process of taking two PCM streams and combining them into one. This is extremely complicated and not something we're going to discuss at length. It is not important to understand how this works, only to understand that it exists. Without mixing, you wouldn't be able to hear sounds from multiple sources. This is true not just for PulseAudio and computer sound, this is true for anything. In real life, you might use an A/V receiver to accept sound from your TV and music player at once, the receiver then mixes the signals and plays it through your speakers.

What is encoding?

Finally we can talk a little about encoding. Encoding is the process of taking a PCM stream and writing it to a permanent format, two types exist. You have lossy encoding and lossless encoding. Lossy encoding removes data from the PCM stream to safe space. Usually the discarded data is useless to you, and will not make a difference in sound quality; examples of lossy encoding are MP3, AAC and Ogg Vorbis. Lossless encoding takes a PCM stream and encodes it in such a way that no data is lost, examples of lossless encodings are FLAC, ALAC and WAV.
Note that lossy and lossless do not mean compressed and uncompressed. A lossless format can be compressed and usually is, as uncompressed lossless encoding would be very large; it would just be the raw PCM stream. An example of lossless uncompressed audio is WAV.
A new element encodings bring is their bit rate, not to be confused with samplerate and bit depth. Bit rate has to do with how much data is stored in every second of audio. For a lossless, uncompressed PCM stream this is easy to calculate with the formula bit rate = sample rate * bit depth * channels, for 16-bit, 48kHz, 2 channel PCM this is 1,5 Mbit. To get the value in bytes, divide by 8, thus 192kB per second.
The bit rate of an encoder means how much the audio will be compressed. PCM compression is super complicated, but it generally involves discarding silence, cutting off frequencies you cannot hear, and so forth. Radio encoding has a bit rate of roughly 128 Kbps, while most CDs have a bit rate of 1360kbps.
Lastly, there is the concept of VBR and CBR. VBR stands for Variable Bit Rate, which CBR stands for Constant Bit Rate. In a VBR encoding, the encoder aim for a target bit rate that you set, but it can deviate if it thinks it needs more or less. CBR will encode a constant bit rate, and will never deviate.
MYTH: Lossless sounds better than lossy. This is blatantly untrue, lossless audio formats were created for perservation and archival reasons. When you encode a lossy file from a lossless source, and you make sure that it's a 48kHz, 16-bit PCM encoding, you will not lose any important information. What is enough depends on the quality of the encoder. For OggVorbis, 192kbps is sufficient, for MP3, 256kbps should be preferred. 320kbps is excessive and the highest quality supported by MP3. In general, 256kbps does the trick, but with storage being abundant these days, you can play it safe and use 320kbps if it makes you feel better.
MYTH: CBR is better than VBR. There is no reason not to use VBR at all, there is no point in writing 256Kbps of data if there is only silence or a constant tone. Let your encoder do what it does best!
FACT: Encoding a lossy format to another lossy format will result in a loss of data! You will compress data that is already compressed, which is really bad. When encoding to a lossy format, always use a high quality recording in a lossless format as the source!
I DON'T BELIEVE YOU: This article from the guys Xiph (the people who brought you FLAC and Ogg Vorbis) explain it better than I can:


Here is a quick guide to achieving great sound quality on Linux with the above in mind.
As you can see, there's little you can do in Linux in the first place, so what can you do if you want better sound?
MYTH: Linux sound quality is worse than Windows. They are exactly the same, Pulse doesn't work that different from how Windows does mixing and resampling.
MYTH: Linux sound quality can be better than Windows. They are exactly the same. All improvements in quality come from the driver and your DAC, not the sound server. Pulse and ALSA do not touch the PCM beyond moving it around and resampling it.
I hope this (long) guide was of help to you, and helped to dispell some myths. Did I miss anything? Ask or let me know, and I'll answer the best I can. Did I make any factual errors? Please correct me with a source and I'll amend the post immediately.
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Comprehensive Guide for getting into Home Recording

I'm going to borrow from a few sources and do my best to make this cohesive, but this question comes up a lot. I thought we had a comprehensive guide, but it doesn't appear so. In the absence of this, I feel that a lot of you could use a simple place to go for some basics on recording. There are a couple of great resources online already on some drumming forums, but I don't think they will be around forever.
Some background on myself - I have been drumming a long time. During that time, home recording has gone from using a cassette deck to having a full blown studio at your finger tips. The technology in the last 15 years has gotten so good it really is incredible. When I was trying to decide what I wanted to do with my life, I decided to go to school for audio engineering in a world-class studio. During this time I had access to the studio and was able to assist with engineering on several projects. This was awesome, and I came out with a working knowledge of SIGNAL CHAIN, how audio works in the digital realm, how microphones work, studio design, etc. Can I answer your questions? Yes.

First up: Signal Chain! This is the basic building block of recording. Ever seen a "I have this plugged in but am getting no sound!" thread? Yeah, signal chain.

A "Signal Chain" is the path your audio follows, from sound source, to the recording device, and back out of your monitors (speakers to you normies).
A typical complete signal chain might go something like this:
1] instrument/sound source 2] Microphone/TransducePickup 3] Cable 4] Mic Preamp/DI Box 5] Analog-to-Digital Converter 6] Digital transmission medium[digital data get recoded for usb or FW transfer] 7] Digital recording Device 8] DSP and Digital summing/playback engine 9] Digital-to-Analog Converter 10] Analog output stage[line outputs and output gain/volume control] 11] Monitors/Playback device[headphones/other transducers]
Important Terms, Definitions, and explanations (this will be where the "core" information is):
1] AD Conversion: the process by which the electrical signal is "converted" to a stream of digital code[binary, 1 and 0]. This is accomplished, basically, by taking digital pictures of the audio...and this is known as the "sampling rate/frequency" The number of "pictures" determines the frequency. So the CD standard of 44.1k is 44,100 "pictures" per second of digital code that represents the electrical "wave" of audio. It should be noted that in order to reproduce a frequency accuratly, the sampling rate must be TWICE that of the desired frequency (See: Nyquist-Shannon Theorem). So, a 44.1 digital audio device can, in fact, only record frequencies as high as 22.05khz, and in the real world, the actual upper frequency limit is lower, because the AD device employs a LOW-PASS filter to protect the circuitry from distortion and digital errors called "ALIASING." Confused yet? Don't worry, there's more... We haven't even talked about Bit depth! There are 2 settings for recording digitally: Sample Rate and Bit Depth. Sample rate, as stated above, determines the frequencies captured, however bit depth is used to get a better picture of the sample. Higher bit depth = more accurate sound wave representation. More on this here. Generally speaking, I record at 92KHz/24 bit depth. This makes huge files, but gets really accurate audio. Why does it make huge files? Well, if you are sampling 92,000 times per second, you are taking each sample and applying 24 bits to that, multiply it out and you get 92,000*24 = 2,208,000 bits per second or roughly 0.26MB per second for ONE TRACK. If that track is 5 minutes long, that is a file that is 78.96MB in size. Now lets say you used 8 inputs on an interface, that is, in total, 631.7MB of data. Wow, that escalates quick, right? There is something else to note as well here: Your CPU has to calculate this. So the amount of calculations it needs to perform for this same scenario is ~17.7 million calculations PER SECOND. This is why CPU speed and RAM is super important when recording digitally.
2] DA conversion: the process by which the digital code (the computer representation of a sound wave) is transformed back into electrcal energy in the proper shape. In a oversimplified explanation, the code is measured and the output of the convertor reflects the value of the code by changing voltage. Think of a sound wave on a grid: Frequency would represent the X axis (the horizontal axis)... but there is a vertical axis too. This is called AMPLITUDE or how much energy the wave is generating. People refer to this as how 'loud' a sound is, but that's not entirely correct. You can have a high amplitude wave that is played at a quiet volume. It's important to distinguish the two. How loud a sound is can be controlled by the volume on a speaker or transducer. But that has no impact on how much amplitude the sound wave has in the digital space or "in the wire" on its way to the transducer. So don't get hung up on how "loud" a waveform is, it is how much amplitude it has when talking about it "in the box" or before it gets to the speakeheadphone/whatever.
3] Cables: An often overlooked expense and tool, cables can in fact, make or break your recording. The multitudes of types of cable are determined by the connector, the gauge(thickness), shielding, type of conductor, etc... Just some bullet points on cables:
- Always get the highest quality cabling you can afford. Low quality cables often employ shielding that doesnt efectively protect against AC hums(60 cycle hum), RF interference (causing your cable to act as a gigantic AM/CB radio antenna), or grounding noise introduced by other components in your system. - The way cables are coiled and treated can determine their lifespan and effectiveness. A kinked cable can mean a broken shield, again, causing noise problems. - The standard in the USA for wiring an XLR(standard microphone) cable is: PIN 1= Cold/-, PIN 2= Hot/+, PIN 3=Ground/shield. Pin 3 carries phantom power, so it is important that the shield of your cables be intact and in good condition if you want to use your mic cables without any problems. - Cables for LINE LEVEL and HI-Z(instrument level) gear are not the same! - Line Level Gear, weather professional or consumer, should generally be used with balanced cables (on a 1/4" connector, it will have 3 sections and is commonly known as TRS -or- TipRingSleeve). A balanced 1/4" is essentially the same as a microphone cable, and in fact, most Professional gear with balanced line inputs and outputs will have XLR connectors instead of 1/4" connectors. - Hi-Z cable for instruments (guitars, basses, keyboards, or anything with a pickup) is UNBALANCED, and should be so. The introduction of a balanced cable can cause electricity to be sent backwards into a guitar and shock the guitar player. You may want this to happen, but your gear doesn't. There is some danger here as well, especially on stage, where the voltage CAN BE LETHAL. When running a guitabass/keyboard "Direct" into your interface, soundcard, or recording device, you should ALWAYS use a "DIRECT BOX", which uses a transformer to isolate and balance the the signal or you can use any input on the interface designated as a "Instrument" or "Hi-Z" input. It also changes some electrical properties, resulting in a LINE LEVEL output (it amplifies it from instrument level to line level).
4] Digital Data Transmissions: This includes S/PDIF, AES/EBU, ADAT, MADI. I'm gonna give a brief overview of this stuff, since its unlikely that alot of you will ever really have to think about it: - SDPIF= Sony Phillips Digital Interface Format. using RCA or TOSLINK connectors, this is a digital protocol that carries 3 streams of information. Digital audio Left, Digital Audio Right, and CLOCK. SPDIF generally supports 48khz/20bit information, though some modern devices can support up to 24bits, and up to 88.2khz. SPDIF is the consumer format of AES/EBU - AES/EBU= Audio Engineering Society/European Breadcasters Union Digital protocol uses a special type of cable often terminated with XLR connectors to transmit 2 channels of Digital Audio. AES/EBU is found mostly on expensive professional digital gear. - ADAT= the Alesis Digital Audio Tape was introduced in 1991, and was the first casette based system capable of recording 8 channels of digital audio onto a single cartridge(a SUPER-VHS tape, same one used by high quality VCR's). Enough of the history, its not so important because we are talking about ADAT-LIGHTPIPE Protocol, which is a digital transmission protocol that uses fiberoptic cable and devices to send up to 8 channels of digital audio simultaneously and in sync. ADAT-Lightpipe supports up to 48khz sample rates. This is how people expand the number of inputs by chaining interfaces. - MADI is something you will almost never encounter. It is a protocol that allows up to 64 channels of digital audio to be transmitted over a single cable that is terminated by BNC connectors. Im just telling you it exists so in case you ever encounter a digital snake that doesnt use Gigabit Ethernet, you will know whats going on.
digital transmission specs: SPDIF -> clock->2Ch->RCA cable(consumer) ADAT-Lightpipe->clock->8Ch->Toslink(semi-pro) SPDIF-OPTICAL->clock->2Ch->Toslink(consumer) AES/EBU->clock->2Ch->XLR(Pro) TDIF->clock->8Ch->DSub(Semi-Pro) ______________ MADI->no clock->64Ch->BNC{rare except in large scale pofessional apps} SDIF-II->no clock->24Ch->DSub{rare!} AES/EBU-13->no clock->24Ch->DSub
5] MICROPHONES: There are many types of microphones, and several names for each type. The type of microphone doesn't equate to the polar pattern of the microphone. There are a few common polar patterns in microphones, but there are also several more that are less common. These are the main types- Omni-Directional, Figure 8 (bi-directional), Cardioid, Super Cardioid, Hyper Cardioid, Shotgun. Some light reading.... Now for the types of microphones: - Dynamic Microphones utilize polarized magnets to convert acoustical energy into electrical energy. there are 2 types of dynamic microphones: 1) Moving Coil microphones are the most common type of microphone made. They are also durable, and capable of handling VERY HIGH SPL (sound pressure levels). 2) Ribbon microphones are rare except in professional recording studios. Ribbon microphones are also incredibly fragile. NEVER EVER USE PHANTOM POWER WITH A RIBBON MICROPHONE, IT WILL DIE (unless it specifically requires it, but I've only ever seen this on one Ribbon microphone ever). Sometimes it might even smoke or shoot out a few sparks; applying phantom power to a Ribbon Microphone will literally cause the ribbon, which is normally made from Aluminum, to MELT. Also, windblasts and plosives can rip the ribbon, so these microphones are not suitible for things like horns, woodwinds, vocals, kick drums, or anything that "pushes air." There have been some advances in Ribbon microphones and they are getting to be more common, but they are still super fragile and you have to READ THE MANUAL CAREFULLY to avoid a $1k+ mistake. - CondenseCapacitor Microphones use an electrostatic charge to convert acoustical energy into electrical energy. The movement of the diaphragm(often metal coated mylar) toward a ceramic "backplate" causes a fluctuation in the charge, which is then amplified inside the microphone and output as an electrical signal. Condenser microphones usually use phantom power to charge the capacitors' and backplate in order to maintain the electrostatic charge. There are several types of condenser microphones: 1) Tube Condenser Microphones: historically, this type of microphone has been used in studios since the 1940s, and has been refined and redesigned hundreds, if not thousands of times. Some of the "best sounding" and most desired microphones EVER MADE are Tube Condenser microphones from the 50's and 60's. These vintage microphones, in good condition, with the original TUBES can sell for hundreds of thousands of dollars. Tube mics are known for sounding "full", "warm", and having a particular character, depending on the exact microphone. No 2 tubes mics, even of the same model, will sound the same. Similar, but not the same. Tube mics have their own power supplies, which are not interchangeable to different models. Each tube mic is a different design, and therefore, has different power requirements. 2) FET Condenser microphones: FET stands for "Field Effect Transistor" and the technology allowed condenser microphones to be miniturized. Take for example, the SHURE beta98s/d, which is a minicondenser microphone. FET technology is generally more transparant than tube technology, but can sometimes sound "harsh" or "sterile". 3) Electret Condenser Microphones are a condenser microphone that has a permanent charge, and therefore, does not require phantom power; however, the charge is not truly permanent, and these mics often use AA or 9V batteries, either inside the mic, or on a beltpack. These are less common.
Other important things to know about microphones:
- Pads, Rolloffs, etc: Some mics have switches or rotating collars that notate certain things. Most commonly, high pass filters/lowcut filters, or attenuation pads. 1) A HP/LC Filter does exactly what you might think: Removes low frequency content from the signal at a set frequency and slope. Some microphones allow you to switch the rolloff frequency. Common rolloff frequencies are 75hz, 80hz, 100hz, 120hz, 125hz, and 250hz. 2) A pad in this example is a switch that lowers the output of the microphone directly after the capsule to prevent overloading the input of a microphone preamplifier. You might be asking: How is that possible? Some microphones put out a VERY HIGH SIGNAL LEVEL, sometimes about line level(-10/+4dbu), mic level is generally accepted to start at -75dbu and continues increasing until it becomes line level in voltage. It should be noted that linel level signals are normally of a different impedance than mic level signals, which is determined by the gear. An example for this would be: I mic the top of a snare drum with a large diaphragm condenser mic (solid state mic, not tube) that is capable of handling very high SPLs (sound pressure levels). When the snare drum is played, the input of the mic preamp clips (distorts), even with the gain turned all the way down. To combat this, I would use a pad with enough attenuation to lower the signal into the proper range of input (-60db to -40 db). In general, it is accepted to use a pad with only as much attentuation as you need, plus a small margin of error for extra “headroom”. What this means is that if you use a 20db pad where you only need a 10db pad, you will then have to add an additional 10db of gain to achieve a desireable signal level. This can cause problems, as not all pads sound good, or even transparent, and can color and affect your signal in sometimes unwanted ways that are best left unamplified. - Other mic tips/info: 1) when recording vocals, you should always use a popfilter. A pop filter mounted on a gooseneck is generally more effective than a windscreen made of foam that slips over the microphone. The foam type often kill the highfrequency response, alter the polar pattern, and can introduce non-linear polarity problems(part of the frequency spectrum will be out of phase.) If you don't have a pop filter or don't want to spend on one, buy or obtain a hoop of some kind, buy some cheap panty-hose and stretch it over the hoop to build your own pop filter. 2) Terms Related to mics: - Plosives: “B”, “D”, “F”, “G”, “J”, “P”, “T” hard consonants and other vocal sounds that cause windblasts. These are responsible for a low frequency pop that can severly distort the diaphragm of the microphone, or cause a strange inconsistency of tonality by causing a short term proximity effect.
- Proximity effect: An exponential increase in low frequency response causes by having a microphone excessivly close to a sound. This can be cause by either the force of the air moving actually causes the microphone’s diaphragm to move and sometimes distort, usually on vocalists or buy the buildup of low frequency soundwaves due to off-axis cancellation ports. You cannot get proximity effect on an omnidirectional microphone. With some practice, you can use proximity effect to your advantage, or as an effect. For example, if you are recording someone whispering and it sounds thin or weak and irritating due to the intenese high mid and high frequency content, get the person very close to a cardioid microphone with two popfilters, back to back approx 1/2”-1” away from the mic and set your gain carefully, and you can achieve a very intimite recording of whispering. In a different scenario, you can place a mic inside of a kick drum between 1”-3” away from the inner shell, angled up and at the point of impact, and towards the floor tom. This usually captures a huge low end, and the sympathetic vibration of the floor tom on the kick drum hits, but retains a clarity of attack without being distorted by the SPL of the drum and without capturing unplesant low-mid resonation of the kick drum head and shell that is common directly in the middle of the shell.
6) Wave Envelope: The envelope is the graphical representation of a sound wave commonly found in a DAW. There are 4 parts to this: Attack, Decay, Sustain, Release: 1) Attack is how quickly the sound reaches its peak amplitude; 2) Decay is the time it takes to reach the sustain level; 3) Sustain how long a sound remains at a certain level (think of striking a tom, the initial smack is attack, then it decays to the resonance of the tom, how long it resonates is the sustain); 4) Release is the amount of time before the sustain stops. This is particularly important as these are also the settings on a common piece of gear called a Compressor! Understanding the envelope of a sound is key to learning how to maniuplate it.
7) Phase Cancellation: This is one of the most important concepts in home recording, especially when looking at drums. I'm putting it in this section because it matters so much. Phase Cancellation is what occurs when the same frequencies occur at different times. To put it simply, frequency amplitudes are additive - meaning if you have 2 sound waves of the same frequency, one amplitude is +4 and the other is +2, the way we percieve sound is that the frequency is +6. But a sound wave has a positive and negative amplitude as it travels (like a wave in the ocean with a peak and a swell). If the frequency then has two sources and it is 180 degrees out of phase, that means one wave is at +4 while the other is at -4. This sums to 0, or cancels out the wave. Effectively, you would hear silence. This is why micing techniques are so important, but we'll get into that later. I wanted this term at the top, and will likely mention it again.

Next we can look at the different types of options to actually record your sound!

1) Handheld/All in one/Field Recorders: I don't know if portable cassette tape recorders are still around, but that's an example of one. These are (or used to) be very popular with journalists because they were pretty decent at capturing speech. They do not fare too well with music though. Not too long ago, we saw the emergence of the digital field recorder. These are really nifty little devices. They come in many shapes, sizes and colors, and can be very affordable. They run on batteries, and have built-in microphones, and record digitally onto SD cards or harddiscs. The more simple ones have a pair of built-in condenser microphones, which may or may not be adjustable, and record onto an SD-card. They start around $99 (or less if you don't mind buying refurbished). You turn it on, record, connect the device itself or the SD card to your computer, transfer the file(s) and there is your recording! An entry-level example is the Tascam DR-05. It costs $99. It has two built in omni-directional mics, comes with a 2GB microSD card and runs on two AA batteries. It can record in different formats, the highest being 24-bit 96KHz Broadcast WAV, which is higher than DVD quality! You can also choose to record as an MP3 (32-320kbps) if you need to save space on the SD card or if you're simply going to record a speech/conference or upload it on the web later on. It's got a headphone jack and even small built-in speakers. It can be mounted onto a tripod. And it's about the size of a cell phone. The next step up (although there are of course many options that are price and feature-wise inbetween this one and the last) is a beefier device like the Zoom H4n. It's got all the same features as the Tascam DR-05 and more! It has two adjustable built-in cardioid condenser mics in an XY configuration (you can adjust the angle from a 90-120 degree spread). On the bottom of the device, there are two XLR inputs with preamps. With those, you can expand your recording possibilities with two external microphones. The preamps can send phantom power, so you can even use very nice studio mics. All 4 channels will be recorded independantly, so you can pop them onto your computer later and mix them with software. This device can also act as a USB interface, so instead of just using it as a field recorder, you can connect it directly to your computer or to a DSLR camera for HD filming. My new recommendation for this category is actually the Yamaha EAD10. It really is the best all-in-one solution for anyone that wants to record their kit audio with a great sound. It sports a kick drum trigger (mounts to the rim of the kick) with an x-y pattern set of microphones to pick up the rest of the kit sound. It also has on-board effects, lots of software integration options and smart features through its app. It really is a great solution for anyone who wants to record without reading this guide.
The TL;DR of this guide is - if it seems like too much, buy the Yamaha EAD10 as a simple but effective recording solution for your kit.

2) USB Microphones: There are actually mics that you an plug in directly to your computer via USB. The mics themselves are their own audio interfaces. These mics come in many shapes and sizes, and offer affordable solutions for basic home recording. You can record using a DAW or even something simple like the stock windows sound recorder program that's in the acessories folder of my Windows operating system. The Blue Snowflake is very affordable at $59. It can stand alone or you can attach it to your laptop or your flat screen monitor. It can record up to 44.1kHz, 16-bit WAV audio, which is CD quality. It's a condenser mic with a directional cardioid pickup pattern and has a full frequency response - from 35Hz-20kHz. It probably won't blow you away, but it's a big departure from your average built-in laptop, webcam, headset or desktop microphone. The Audio Technica AT2020 USB is a USB version of their popular AT2020 condenser microphone. At $100 it costs a little more than the regular version. The AT2020 is one of the finest mics in its price range. It's got a very clear sound and it can handle loud volumes. Other companies like Shure and Samson also offer USB versions of some of their studio mics. The AT2020 USB also records up to CD-quality audio and comes with a little desktop tripod. The MXL USB.009 mic is an all-out USB microphone. It features a 1 inch large-diaphragm condenser capsule and can record up to 24-bit 96kHz WAV audio. You can plug your headphones right into the mic (remember, it is its own audio interface) so you can monitor your recordings with no latency, as opposed to doing so with your computer. Switches on the mic control the gain and can blend the mic channel with playback audio. Cost: $399. If you already have a mic, or you don't want to be stuck with just a USB mic, you can purcase a USB converter for your existing microphone. Here is a great review of four of them.
3) Audio Recording Interfaces: You've done some reading up on this stuff... now you are lost. Welcome to the wide, wide world of Audio Interfaces. These come in all different shapes and sizes, features, sampling rates, bit depths, inputs, outputs, you name it. Welcome to the ocean, let's try to help you find land.
- An audio interface, as far as your computer is concerned, is an external sound card. It has audio inputs, such as a microphone preamp and outputs which connect to other audio devices or to headphones or speakers. The modern day recording "rig" is based around a computer, and to get the sound onto your computer, an interface is necessary. All computers have a sound card of some sort, but these have very low quality A/D Converters (analog to digital) and were not designed with any kind of sophisticated audio recording in mind, so for us they are useless and a dedicated audio interface must come into play.
- There are hundreds of interfaces out there. Most commonly they connect to a computer via USB or Firewire. There are also PCI and PCI Express-based interfaces for desktop computers. The most simple interfaces can record one channel via USB, while others can record up to 30 via firewire! All of the connection types into the computer have their advantages and drawbacks. The chances are, you are looking at USB, Firewire, or Thunderbolt. As far as speeds, most interfaces are in the same realm as far as speed is concerned but thunderbolt is a faster data transfer rate. There are some differences in terms of CPU load. Conflict handling (when packages collide) is handled differently. USB sends conflict resolution to the CPU, Firewire handles it internally, Thunderbolt, from what I could find, sends it to the CPU as well. For most applications, none of them are going to be superior from a home-recording standpoint. When you get up to 16/24 channels in/out simultaneously, it's going to matter a lot more.
- There are a number of things to consider when choosing an audio interface. First off your budget, number of channels you'd like to be able to record simultaneously, your monitoring system, your computer and operating system and your applications. Regarding budget, you have to get real. $500 is not going to get you a rig with the ability to multi-track a drum set covered in mics. Not even close! You might get an interface with 8 channels for that much, but you have to factor in the cost of everything, including mics, cables, stands, monitors/headphones, software, etc... Considerations: Stereo Recording or Multi-Track Recording? Stereo Recording is recording two tracks: A left and right channel, which reflects most audio playback systems. This doesn't necessarily mean you are simply recording with two mics, it means that what your rig is recording onto your computer is a single stereo track. You could be recording a 5-piece band with 16 mics/channels, but if you're recording in stereo, all you're getting is a summation of those 16 tracks. This means that in your recording software, you won't be able to manipulate any of those channels independantly after you recorded them. If the rack tom mic wasn't turned up loud enough, or you want to mute the guitars, you can't do that, because all you have is a stereo track of everything. It's up to you to get your levels and balance and tone right before you hit record. If you are only using two mics or lines, then you will have individual control over each mic/line after recording. Commonly, you can find 2 input interfaces and use a sub-mixer taking the left/right outputs and pluging those into each channel of the interface. Some mixers will output a stereo pair into a computer as an interface, such as the Allen&Heath ZED16. If you want full control over every single input, you need to multi-track. Each mic or line that you are recording with will get it's own track in your DAW software, which you can edit and process after the fact. This gives you a lot of control over a recording, and opens up many mixing options, and also many more issues. Interfaces that facilitate multitracking include Presonus FireStudio, Focusrite Scarlett interfaces, etc. There are some mixers that are also interfaces, such as the Presonus StudioLive 16, but these are very expensive. There are core-card interfaces as well, these will plug in directly to your motherboard via PCI or PCI-Express slots. Protools HD is a core-card interface and requires more hardware than just the card to work. I would recommend steering clear of these until you have a firm grasp of signal chain and digital audio, as there are more affordable solutions that will yield similar results in a home-environment.

DAW - Digital Audio Workstation

I've talked a lot about theory, hardware, signal chain, etc... but we need a way to interpret this data. First off what does a DAW do? Some refer to them as DAE's (Digital Audio Editors). You could call it a virtual mixing board , however that isn't entirely correct. DAWs allow you to record, control, mix and manipulate independant audio signals. You can change their volume, add effects, splice and dice tracks, combine recorded audio with MIDI-generated audio, record MIDI tracks and much much more. In the old days, when studios were based around large consoles, the actual audio needed to be recorded onto some kind of medium - analog tape. The audio signals passed through the boards, and were printed onto the tape, and the tape decks were used to play back the audio, and any cutting, overdubbing etc. had to be done physically on the tape. With a DAW, your audio is converted into 1's and 0's through the converters on your interface when you record, and so computers and their harddiscs have largely taken the place of reel-to-reel machines and analog tape.
Here is a list of commonly used DAWs in alphabetical order: ACID Pro Apple Logic Cakewalk SONAR Digital Performer FL (Fruity Loops) Studio (only versions 8 and higher can actually record Audio I believe) GarageBand PreSonus Studio One Pro Tools REAPER Propellerhead Reason (version 6 has combined Reason and Record into one software, so it now is a full audio DAW. Earlier versions of Reason are MIDI based and don't record audio) Propellerhead Record (see above) Steinberg Cubase Steinberg Nuendo
There are of course many more, but these are the main contenders. [Note that not all DAWs actually have audio recording capabilities (All the ones I listed do, because this thread is about audio recording), because many of them are designed for applications like MIDI composing, looping, etc. Some are relatively new, others have been around for a while, and have undergone many updates and transformations. Most have different versions, that cater to different types of recording communities, such as home recording/consumer or professional.
That's a whole lot of choices. You have to do a lot of research to understand what each one offers, what limitations they may have etc... Logic, Garageband and Digital Performer for instance are Mac-only. ACID Pro, FL Studio and SONAR will only run on Windows machines. Garageband is free and is even pre-installed on every Mac computer. Most other DAWs cost something.
Reaper is a standout. A non-commercial license only costs $60. Other DAWs often come bundled with interfaces, such as ProTools MP with M-Audio interfaces, Steinberg Cubase LE with Lexicon Interfaces, Studio One with Presonus Interfaces etc. Reaper is a full function, professional, affordable DAW with a tremendous community behind it. It's my recommendation for everyone, and comes with a free trial. It is universally compatible and not hardware-bound.
You of course don't have to purchase a bundle. Your research might yield that a particular interface will suit your needs well, but the software that the same company offers or even bundles isn't that hot. As a consumer you have a plethora of software and hardware manufacturers competing for your business and there is no shortage of choice. One thing to think about though is compatability and customer support. With some exceptions, technically you can run most DAWs with most interfaces. But again, don't just assume this, do your research! Also, some DAWs will run smoother on certain interfaces, and might experience problems on others. It's not a bad thing to assume that if you purchase the software and hardware from the same company, they're at least somewhat optimized for eachother. In fact, ProTools, until recently would only run on Digidesign (now AVID) and M-Audio interfaces. While many folks didn't like being limited to their hardware choices to run ProTools, a lot of users didn't mind, because I think that at least in part it made ProTools run smoother for everyone, and if you did have a problem, you only had to call up one company. There are many documented cases where consumers with software and hardware from different companies get the runaround:
Software Company X: "It's a hardware issue, call Hardware Company Z". Hardware Company Z: "It's a software issue, call Software Company X".
Another thing to research is the different versions of softwares. Many of them have different versions at different pricepoints, such as entry-level or student versions all the way up to versions catering to the pros. Cheaper versions come with limitations, whether it be a maximum number of audio tracks you can run simultaneously, plug-ins available or supported Plug-In formats and lack of other features that the upper versions have. Some Pro versions might require you to run certain kinds of hardware. I don't have time nor the will to do research on individual DAW's, so if any of you want to make a comparison of different versions of a specific DAW, be my guest! In the end, like I keep stressing - we each have to do our own research.
A big thing about the DAW that it is important to note is this: Your signal chain is your DAW. It is the digital representation of that chain and it is important to understand it in order to properly use that DAW. It is how you route the signal from one spot to another, how you move it through a sidechain compressor or bus the drums into the main fader. It is a digital representation of a large-format recording console, and if you don't understand how the signal gets from the sound source to your monitor (speaker), you're going to have a bad time.

Playback - Monitors are not just for looking at!

I've mentioned monitors several times and wanted to touch on these quickly: Monitors are whatever you are using to listen to the sound. These can be headphones, powered speakers, unpowered speakers, etc. The key thing here is that they are accurate. You want a good depth of field, you want as wide a frequency response as you can get, and you want NEARFIELD monitors. Unless you are working with a space that can put the monitor 8' away from you, 6" is really the biggest speaker size you need. At that point, nearfield monitors will reproduce the audio frequency range faithfully for you. There are many options here, closed back headphones, open back headphones, studio monitors powered, and unpowered (require a separate poweramp to drive the monitor). For headphones, I recommend AKG K271, K872, Sennheiser HD280 Pro, etc. There are many options, but if mixing on headphones I recommend spending some good money on a set. For Powered Monitors, there's really only one choice I recommend: Kali Audio LP-6 monitors. They are, dollar for dollar, the best monitors you can buy for a home studio, period. These things contend with Genelecs and cost a quarter of the price. Yes, they still cost a bit, but if you're going to invest, invest wisely. I don't recommend unpowered monitors, as if you skimp on the poweramp they lose all the advantages you gain with monitors. Just get the powered monitors if you are opting for not headphones.

Drum Mic'ing Guide, I'm not going to re-create the wheel.

That's all for now, this has taken some time to put together (a couple hourse now). I can answer other questions as they pop up. I used a few sources for the information, most notably some well-put together sections on the Pearl Drummers Forum in the recording section. I know a couple of the users are no longer active there, but if you see this and think "Hey, he ripped me off!", you're right, and thanks for allowing me to rip you off!

A couple other tips that I've come across for home recording:
You need to manage your gain/levels when recording. Digital is NOT analog! What does this mean? You should be PEAKING (the loudest the signal gets) around -12dB to -15dB on your meters. Any hotter than that and you are overdriving your digital signal processors.
What sound level should my master bus be at for Youtube?
Bass Traps 101
Sound Proofing 101
submitted by M3lllvar to drums [link] [comments]

MAME 0.210

MAME 0.210

It’s time for the delayed release of MAME 0.210, marking the end of May. This month, we’ve got lots of fixes for issues with supported systems, as well as some interesting additions. Newly added hand-held and tabletop games include Tronica’s Shuttle Voyage and Space Rescue, Mattel’s Computer Chess, and Parker Brothers’ Talking Baseball and Talking Football. On the arcade side, we’ve added high-level emulation of Gradius on Bubble System hardware and a prototype of the Neo Geo game Viewpoint. For this release, Jack Li has contributed an auto-fire plugin, providing additional functionality over the built-in auto-fire feature.
A number of systems have had been promoted to working, or had critical issues fixed, including the Heathkit H8, Lola 8A, COSMAC Microkit, the Soviet PC clone EC-1840, Zorba, and COMX 35. MMU issues affecting Apollo and Mac operating systems have been addressed. Other notable improvements include star field emulation in Tutankham, further progress on SGI emulation, Sega Saturn video improvements, write support for the CoCo OS-9 disk image format, and preliminary emulation for MP3 audio on Konami System 573 games.
There are lots of software list additions this month. Possibly most notable is the first dump of a Hanimex Pencil II cartridge, thanks to the team. Another batch of cleanly cracked and original Apple II software has been added, along with more ZX Spectrum +3 software, and a number of Colour Genie cassette titles.
That’s all we’ve got space for here, but there are lots more bug fixes, alternate versions of supported arcade games, and general code quality improvements. As always, you can get the source and Windows binary packages from the download page.

MAMETesters Bugs Fixed

New working machines

New working clones

Machines promoted to working

Clones promoted to working

New machines marked as NOT_WORKING

New clones marked as NOT_WORKING

New working software list additions

Software list items promoted to working

New NOT_WORKING software list additions

Source Changes

submitted by cuavas to emulation [link] [comments]

How does this electronics engineering technology degree compare to an electronics engineering degree?

Im well aware a technology degree isnt an engineering degree, but Ive read on the ABET website that they are closely related. For electronics engineers on here, how closely does this match an ee degree? Which classes are missing?
Calc I, Calc II, Differential Equations, Physics I, Physics II, Chemistry I
Here are the core classes classes with descriptions:

Circuit Theory I

This 4-credit course arms students with the basic knowledge of circuits needed to compete in the industry. Topics covered include: current, voltage, resistance, Ohm’s Law, work and power, series and parallel resistances, resistance networks, Kirchhoff’s law, network theorems, mesh and nodal analysis, inductance, capacitance, and magnetic circuits.

Circuit Theory II

Including a lab component, this course builds on the foundational knowledge learned in Circuit Theory I, a prerequisite, through the learning of principles and applications.

Electronics I

Focusing on semiconductor devices, this lab course serves as an introduction to electronics. At the end of the course you will be able to perform the analysis of DC transistors biasing, small-signal single and multi-stage amplifiers using BJTs, FETs, and MOSFETS, and frequency response of transistor single and multi-stage amplifiers.

Electronics II

Building on the foundations of Electronics I, a prerequisite for this course, this lab course places an emphasis on troubleshooting of test circuits, and analysis based on computer simulation.

Digital Electronics

This lab course will teach students the principles and applications of digital circuits. Topics include number systems, binary arithmetic, logic gates and Boolean algebra, logic families, combinational and synchronous logic circuit design, logic minimization techniques (Karnaugh maps, Quine-McCluskey), counters, shift registers, encoders and decoders, multiplexors and demultiplexors, and interfacing.


This lab course covers 8, 16, and 32-bit microprocessor technology and features. Learn the principles and applications of microprocessors and microprocessor based systems.

Object-Oriented Programming

Options for this requirement include the courses Object-Oriented Programming, which covers problem solving and algorithm development using Java, or Introduction to Programming, which introduces programming in C++.
This course offers an overview of the current theory and practice of business data communications and networks. It places emphasis on the role of the telecommunications industry in the growth of information societies and their reliance on knowledge and information services to stimulate economic growth. The course examines the seven-layered Open Systems Interconnection (OSI) reference model proposed by the International Standards Organization (ISO) and the notion of network architecture to manage information and communications.
Introduction to use of embedded microcontrollers and microprocessors; processor architecture; assembly language programming; use of development systems and/or emulators for system testing and debugging; software and hardware considerations of processor interfacing for I/O and memory expansion; programmed and interrupt driven I/O techniques.
This course presents systematic design methods for synthesizing sequential digital circuits using hardware description language (HDL), while details of its associated languages too are brought to familiar ground. Specification, modeling, and design principles of sequential systems, as well as design implementation and testing using programmable logic devices and Computer Aided Design (CAD) tools are studied. The course includes laboratory experiments and a group project.
This is an introductory course on continuous linear control systems covering analysis, design, and practical applications. Modeling first- and second-order dynamic physical systems, transient response and steady-state analyses, Routh-Hurwitz stability criteria, roles of feedback in controlling steady-state errors, frequency response design methods (Bode, Nyquist), etc. are covered. The course emphasizes the application of established methodology with the aid of examples, calculators, and computer programs such as MATLAB.
This is a technology focused course covering the principles and applications of analog and digital communication circuits. Analysis of modulation and demodulation (AM, FM, PM), radio frequency (RF) transmitters and receivers, digital communication techniques, coding and multiplexing, network communications and protocols, transmission lines and media, wave propagation, optical fibers, wired and wireless communications, communication test equipment and troubleshooting, and communication standards are covered.

Project Management

Students study the skills required of a project manager and learn the methodologies, tools, and processes for success in planning and managing project scope, schedules, costs, quality, risks, communications, purchases, human resources, and stakeholders.
submitted by sage1_18 to ECE [link] [comments]

MAME 0.210

MAME 0.210

It’s time for the delayed release of MAME 0.210, marking the end of May. This month, we’ve got lots of fixes for issues with supported systems, as well as some interesting additions. Newly added hand-held and tabletop games include Tronica’s Shuttle Voyage and Space Rescue, Mattel’s Computer Chess, and Parker Brothers’ Talking Baseball and Talking Football. On the arcade side, we’ve added high-level emulation of Gradius on Bubble System hardware and a prototype of the Neo Geo game Viewpoint. For this release, Jack Li has contributed an auto-fire plugin, providing additional functionality over the built-in auto-fire feature.
A number of systems have had been promoted to working, or had critical issues fixed, including the Heathkit H8, Lola 8A, COSMAC Microkit, the Soviet PC clone EC-1840, Zorba, and COMX 35. MMU issues affecting Apollo and Mac operating systems have been addressed. Other notable improvements include star field emulation in Tutankham, further progress on SGI emulation, Sega Saturn video improvements, write support for the CoCo OS-9 disk image format, and preliminary emulation for MP3 audio on Konami System 573 games.
There are lots of software list additions this month. Possibly most notable is the first dump of a Hanimex Pencil II cartridge, thanks to the team. Another batch of cleanly cracked and original Apple II software has been added, along with more ZX Spectrum +3 software, and a number of Colour Genie cassette titles.
That’s all we’ve got space for here, but there are lots more bug fixes, alternate versions of supported arcade games, and general code quality improvements. As always, you can get the source and Windows binary packages from the download page.

MAMETesters Bugs Fixed

New working machines

New working clones

Machines promoted to working

Clones promoted to working

New machines marked as NOT_WORKING

New clones marked as NOT_WORKING

New working software list additions

Software list items promoted to working

New NOT_WORKING software list additions

Source Changes

submitted by cuavas to MAME [link] [comments]

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Magnet Binary Options Trading Software - If you need a ...

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