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2016 Lexus Ct200h Fsport
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Discussion Starter · #1 · (Edited)
As we are entering into a future where high res audio will become more standard eventually it’s my job to educate people to refer back to this one day.


First let’s talk about file formats

308092


Compression Ratio: the difference between uncompressed and compressed file, applies to Lossless only, the lower - the better.

Decoding/Encoding Speed shows how many audio samples are processed per time unit, the higher - the better. This is an average value of all tests shown below, read section "Performance Tests" for more details.

Audio data inside a WAVE file is stored uncompressed, it doesn't require any decoding or encoding work to do, therefore its compression ratio is 1:1 and the speed is unlimited.

308095


Most people probably don’t Know this but the time it takes decompress decode and encode back isn’t noticeable to the human but all the things it has to do to serve you but at the expense of losing quality to some degree every little bit helps right?

All lossless codecs have similar compression ratio. However, decoding speed is different: FLAC is almost 3 times faster than WavPack. Believe it or not there is always trade offs for these compressed lossless formats.

I present to you the comparison of audio file formats. Audio data compressed with one of the codec listed above is stored within an audio file, each codec uses its own data container format. They all have their similarities and differences, so the table below gathers everything together.

FormatLosslessFast accurate seekingStreamingCRCNo EOF readingSupported meta tags
.wavyyy/nny/nLIST INFO
.flacyyyyyVorbis comments
.apeyyyynAPE tag
ID3v1
.mp4, .m4ay/nyy/nny/nilst box


PERFORMANCE TESTS

Obviously, performance tests don't show the speed of audio format but the speed of specific encoder/decoder implementation. Don't judge too quickly, the results here depend dramatically on an audio library that is used to do the audio processing, operating system, processor architecture and other computer hardware. The size of encoded files can also vary: although I try to use similar settings for each encoder, they still aren't identical.

Heavy Metal (6:53)
Audio FormatSettingsFile SizeDecoding Time (sec)Encoding Time (sec)Decoding Speed (samples/mcs)Encoding Speed (samples/mcs)
WAVEaudio CD7293988400--
FLACq6501130070.561.6365.1222.37
ALAC519628102.55-14.3-
WavPack"normal"499288021.58-23.08-


Classic (5:17)
Audio FormatSettingsFile SizeDecoding Time (sec)Encoding Time (sec)Decoding Speed (samples/mcs)Encoding Speed (samples/mcs)
WAVEaudio CD5602468400--
FLACq6299162780.411.268.3223.34
ALAC308796251.97-14.22-
WavPack"normal"298080981.2-23.34-

Most people will say this is irrelevant and doesn’t matter but maybe to them it isn’t. Every little thing matters think about it this way you do all this work for you car to get it right and you use a compress lossless file. It does retain all the audio but there are delays in between and just because you don’t notice it doesn’t mean it’s not holding you back some.

Look at it this way you may have cancer but you don’t notice it doesn’t mean it’s not holding you back some. Same thing as not getting enough sleep.


Bit Depth , Bitrate , Sampling Rate & how they all tie in together.

I will give you simple definitions as I do not want to confuse y’all. It is more detailed in my message later on explaining why this makes sense even though not everyone will agree to this in here.

Sampling Rate

Sampling Rates come in different numbers
44.1khz , 96khz , 192khz there are more then Just these but I am just doing a simple definition of them.

The sampling rate refers to the number of samples of audio recorded every second. It is measured in samples per second or Hertz (abbreviated as Hz or kHz, with one kHz being 1000 Hz). An audio sample is just a number representing the measured acoustic wave value at a specific point in time.

44.1kHz (most common for music CDs), and 48kHz (most common for audio tracks in movies). Lower sampling rates mean less samples per second, which in turn mean less audio data, since there is a smaller number of sample points to represent the audio. The sampling rate is chosen for a certain application depending on what acoustic artifacts need to be captured. Some acoustic artifacts like speech utterances require a lower sampling rate than an acoustic artifact such as a music tune in a music CD. It’s important to note that higher sampling rates require more storage space and processing power to handle, though this might not be as big of an issue now as it used to be in the old days when digital storage and processing power were of primary considerations.


Bit Depth

These come in different numbers as well but here are a few.
16bit , 24bit , 32bit (Floating Point)

In addition to the sampling rate, which is how many data points of audio we have, there is also the sample depth. Measured in bits per sample, the sample depth, (also known as the sample precision or sample size), is the second important property of an audio file or stream, and it represents the level of detail, or “quality” each sample has. As mentioned above, each audio sample is just a number, and while having a lot of numbers is helpful to represent audio, you also need the range or “quality” of every individual number to be large enough to represent each sample or data point accurately.

What does “quality” mean? For an audio sample, it simply means that the audio sample can represent a higher range of amplitudes. A sample depth of 8 bits means that we have 2^8 = 256 distinct amplitudes that each audio sample can represent, and a sample depth of 16 bits means that we have 2^16 = 65,536 distinct amplitudes that an audio sample can represent, and so on for higher sample depths. The most common sample depths are 16 bits The more distinct amplitudes one has in a digital recording, the closer the digital recording sounds to the original acoustic event.


Bit Rate

Tying the sampling rate and the sample depth together is the bit rate, which is simply the product of both. Since the sampling rate is measured in samples per second and the sample depth is measured in bits per sample, it is therefore measured in (samples per second) x (bits per sample) = bits per second, abbreviated as bps or kbps. It’s worth noting that because the sample depth and the bit rate are related, they frequently, yet erroneously, get used interchangeably.

The bit rate in audio varies according to application. Applications that require high audio quality, like music, usually have a higher bit rate yielding higher quality, or “crisper” audio.


Let’s use an example

I using the song shallow with lady Gaga from the movie a star is born I have a high res audio flac 96khz I am starting with & went to wave as source conversion that file size is 157mb vs the flac 73mb. The 157mb wave as source conversion. Is from pcm_f32le & what the means is in this screenshot

308103



We are using 32bit fp for the purpose of increased dynamic range which will be explained below again.

But wait this 157mb files wave conversion of course means it was at 96khz. The bitrate for this is 6144kbps which is basically 6mbps

The 2nd test was to use a wave conversion from the 96khz upsampling with a 192khz sampling rate because my 8ch can do a 192khz high sampling rate. Once again some people may not agree with this logic at all and more will be explained as your read about over sampling. But if your remember the sampling rate being 192khz made the file size for this song (192khz) 315mb wave vs the (96khz) 157mb wave.

Can I hear a difference yes this will be based off users ears opinions music sources dsps. I can hear the difference in my car how much of a difference well that's the thing double sampling rate add more space yes but it doesn't exactly improve the sq.

Now remember I am using 32 Bit Floating point and because I am The dynamic range that can be represented by a 32-bit (floating point) file is 1528 dB. Which is significant for dynamic range and noise floor as well as clipping. It’s almost impossible to clip with this much extra space for the music to spread over.

Now remember The sampling rate refers to the number of samples of audio recorded every second. It is measured in samples per second or Hertz.


OVERSAMPLING

Oversampling is the process of increasing the sample rate by a factor of two or more, usually temporarily inside a plugin so it can perform a process in higher resolution.
Oversampling means more headroom for harmonics that are introduced by distortion and overdrive effects – which can go far beyond 22 kHz.
As we learned, if the harmonics go beyond the Nyquist and aren’t properly filtered, you get aliasing.

Oversampling helps to avoid aliasing as it increases the Nyquist frequency and gives us more ‘spectral headroom’. So going from 96khz - 192khz gives spectral head room.

Let’s into more details now about all these things to have a fuller understanding of everything coming together.

I am explaining these things to you in the most simplistic forms as there is a lot more in-depth reading to fully under the calculations and why this and that has to be for that to work but the simpler I make this the easier your understanding will be on this and you can research more if you like.

So, 16-bit WAV files can store audio from 0 dBFS down to -96 dBFS. Each audio sample consumes 16 bits of space on a hard disk or memory, and at a 48 kHz sampling rate this means that 16 x 48,000 = 768,000 bits per second are needed to store a single channel 16-bit, 48 kHz file. So for

16 bit audio has a total of 768 thousand values per sample @48khz

24 bit audio can have over 16 million values per sample

32 bit float Point takes things even further, providing over 4 billion values per sample

The difference between 24 bit and 32 bit float point audio is impossible to hear, even in music with a wide dynamic range.

The main advantage of 32 bit float point audio is waveforms can “clip” without losing any data.

With 24 bit audio, once the maximum value of 0 dB is reached, that’s it! There is no longer any data left to record louder signals. But 32 bit float point audio has a much higher ceiling, and if a signal happens to go beyond 0db, it can be safely reduced later on with no clipping.

This is why for example using mp3 compressions sounds so loud and ****ty. No room on That's a cd not a mp3 which is far less because of lower bitrate and high compression.

16 bit audio file can have a maximum dynamic range of 96 dB

24 bit can have up to 144 dB of dynamic range

32 bit (floating point) will be 1528 dB of dynamic range so as you can see it is totally worth it for that reason to run 32bit fp.

And you can Now see a clearer overall picture of how bit depth and sampling rates come together.

One last thing – do not confuse bit depth with the bit rate of MP3 files. They are separate, and while an MP3 file has a sample rate that affects the bit rate, MP3s do not have bit depth in the same way as WAV files.

The sampling rate is the frequency at which samples are played back. We need a high enough sampling rate to ensure all audible frequencies can be represented.


The bit depth is the quality of each sample in terms of its mathematical accuracy. Lower bit depths mean there is less digital information available to describe each sample, and this results in noise and a lack of dynamics.

I know I repeated some things a few times it's because I have written So much I didn't want you to forget but this explains a simple introduction to these things for high resolution audio.
 

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The write up is largely fine… but how many dB of dynamic range to we need?
We should be listening at 80-85 dB for OSH, and the threshold of hearing low.

Then we use speakers that have distortion in the hand fulls of %, and we worry about 192/24 versus CD quality files, our whether the cables have pixie dust on them?

But I can see the write up being useful for people spending thousands on a rack of equipment, and having a million songs on a flash drive, wondering why it doesn’t sound as a good as their portable CD player.
 

· Banned
2016 Lexus Ct200h Fsport
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2,490 Posts
Discussion Starter · #4 ·
The write up is largely fine… but how many dB of dynamic range to we need?
We should be listening at 80-85 dB for OSH, and the threshold of hearing low.

Then we use speakers that have distortion in the hand fulls of %, and we worry about 192/24 versus CD quality files, our whether the cables have pixie dust on them?

But I can see the write up being useful for people spending thousands on a rack of equipment, and having a million songs on a flash drive, wondering why it doesn’t sound as a good as their portable CD player.
I’m that guy who spends thousands and thousands of sq gear and try to do everything to save money because I refuse to pay for someone to do something I can do. But I can hear the difference on my vinyl rips and the dsd file conversions but that’s a totally different things from this and those vinyl rips have been corrected thru a spectrum analyzer basically eq the master vinyl before making a dsd is all. It works great I have several copies of dsd audio files for hotel California they all sound different some are horrible on details some are amazing but lack a little low end which can be fix also thru eq. Regardless these vinyl rips sound better the the save dsd128 actual releases. It’s honestly amazing But yes to each his own I spend a lot of money on this because it’s my hobby it’s better then a gf and I get to continuously change things for fun and try new things out.
 

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2016 Lexus Ct200h Fsport
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Discussion Starter · #6 ·
You will always hear a difference if you know the difference is there.

Even if the difference is not audible..

:)

Just imo, enjoy the ride by all means.
With the gear I run and keep adding on to it I can always hear a difference because at the end of the day the most awesome you will get after all the hardware and tuning comes from the music source itself
 

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The write up is largely fine… but how many dB of dynamic range to we need?
We should be listening at 80-85 dB for OSH, and the threshold of hearing low.

Then we use speakers that have distortion in the hand fulls of %, and we worry about 192/24 versus CD quality files, our whether the cables have pixie dust on them?

But I can see the write up being useful for people spending thousands on a rack of equipment, and having a million songs on a flash drive, wondering why it doesn’t sound as a good as their portable CD player.
No one has OSH or OSHA riding in their car with them, not even you. You know you listen at more than 85 db.
 

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I’m that guy who spends thousands and thousands of sq gear and try to do everything to save money because I refuse to pay for someone to do something I can do. But I can hear the difference on my vinyl rips and the dsd file conversions but that’s a totally different things from this and those vinyl rips have been corrected thru a spectrum analyzer basically eq the master vinyl before making a dsd is all. It works great I have several copies of dsd audio files for hotel California they all sound different some are horrible on details some are amazing but lack a little low end which can be fix also thru eq. Regardless these vinyl rips sound better the the save dsd128 actual releases. It’s honestly amazing But yes to each his own I spend a lot of money on this because it’s my hobby it’s better then a gf and I get to continuously change things for fun and try new things out.
Well the RIAA is one thing…

Assuming you had a nice 198/24 bit, then one could start with that and save the ];samples as floats.
Then convert to the nearest integer (or truncate) at 23, 22, 21… 9, 8, 7, 6 bit.

One could difference the “whatever bit depth” to the 24 bit, and do FFTs of that, tpo get direct quantity of the distortion.
One can also listen to it,.. I would assume that 44k/16 bit and 198/24 are close.
Clearly in a noisy car the levels of distortion could be well below the noise floor… depending on the car.

And when/if the speaker distortion dominates, then it starts to get interesting whether HD or IMD compares to quantisation noise… and is more or less audible??
 

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With the gear I run and keep adding on to it I can always hear a difference because at the end of the day the most awesome you will get after all the hardware and tuning comes from the music source itself
He's referring to psychoacoustics, more specifically if you expect to hear a difference you will. Not saying you don't just saying it's impossible to eliminate confirmation bias without doing a double blind experiment or something similar. Our aural memories are pretty poor in general, maybe yours is better than average.
 

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I feel no need to upgrade my Helix Ultra running at 96k to something running 192k.. Most audio isnt high enough quality to matter... maybe some day.. but we will need some serious remastering of old music.. or a high enough sample rate to where we get the subtle nuances of analog formats like vinyl onto a digital format..
 

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I feel no need to upgrade my Helix Ultra running at 96k to something running 192k.. Most audio isnt high enough quality to matter... maybe some day.. but we will need some serious remastering of old music.. or a high enough sample rate to where we get the subtle nuances of analog formats like vinyl onto a digital format..
Yeah, you can't get that dusty crackling sound from digital formats, or the rolled off treble and bass. Vinyl has serious physical limits. I'm NOT comparing to lossy formats, only lossless. To top it all off, the music is sent to the lathe in DIGITAL form to cut the master. Is vinyl better than streaming?
 

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As we are entering into a future where high res audio will become more standard eventually it’s my job to educate people to refer back to this one day.


First let’s talk about file formats

View attachment 308092

Compression Ratio: the difference between uncompressed and compressed file, applies to Lossless only, the lower - the better.

Decoding/Encoding Speed shows how many audio samples are processed per time unit, the higher - the better. This is an average value of all tests shown below, read section "Performance Tests" for more details.

Audio data inside a WAVE file is stored uncompressed, it doesn't require any decoding or encoding work to do, therefore its compression ratio is 1:1 and the speed is unlimited.

View attachment 308095

Most people probably don’t Know this but the time it takes decompress decode and encode back isn’t noticeable to the human but all the things it has to do to serve you but at the expense of losing quality to some degree every little bit helps right?

All lossless codecs have similar compression ratio. However, decoding speed is different: FLAC is almost 3 times faster than WavPack. Believe it or not there is always trade offs for these compressed lossless formats.

I present to you the comparison of audio file formats. Audio data compressed with one of the codec listed above is stored within an audio file, each codec uses its own data container format. They all have their similarities and differences, so the table below gathers everything together.

FormatLosslessFast accurate seekingStreamingCRCNo EOF readingSupported meta tags
.wavyyy/nny/nLIST INFO
.flacyyyyyVorbis comments
.apeyyyynAPE tag
ID3v1
.mp4, .m4ay/nyy/nny/nilst box


PERFORMANCE TESTS

Obviously, performance tests don't show the speed of audio format but the speed of specific encoder/decoder implementation. Don't judge too quickly, the results here depend dramatically on an audio library that is used to do the audio processing, operating system, processor architecture and other computer hardware. The size of encoded files can also vary: although I try to use similar settings for each encoder, they still aren't identical.

Heavy Metal (6:53)
Audio FormatSettingsFile SizeDecoding Time (sec)Encoding Time (sec)Decoding Speed (samples/mcs)Encoding Speed (samples/mcs)
WAVEaudio CD7293988400--
FLACq6501130070.561.6365.1222.37
ALAC519628102.55-14.3-
WavPack"normal"499288021.58-23.08-


Classic (5:17)
Audio FormatSettingsFile SizeDecoding Time (sec)Encoding Time (sec)Decoding Speed (samples/mcs)Encoding Speed (samples/mcs)
WAVEaudio CD5602468400--
FLACq6299162780.411.268.3223.34
ALAC308796251.97-14.22-
WavPack"normal"298080981.2-23.34-

Most people will say this is irrelevant and doesn’t matter but maybe to them it isn’t. Every little thing matters think about it this way you do all this work for you car to get it right and you use a compress lossless file. It does retain all the audio but there are delays in between and just because you don’t notice it doesn’t mean it’s not holding you back some.

Look at it this way you may have cancer but you don’t notice it doesn’t mean it’s not holding you back some. Same thing as not getting enough sleep.


Bit Depth , Bitrate , Sampling Rate & how they all tie in together.

I will give you simple definitions as I do not want to confuse y’all. It is more detailed in my message later on explaining why this makes sense even though not everyone will agree to this in here.

Sampling Rate

Sampling Rates come in different numbers
44.1khz , 96khz , 192khz there are more then Just these but I am just doing a simple definition of them.

The sampling rate refers to the number of samples of audio recorded every second. It is measured in samples per second or Hertz (abbreviated as Hz or kHz, with one kHz being 1000 Hz). An audio sample is just a number representing the measured acoustic wave value at a specific point in time.

44.1kHz (most common for music CDs), and 48kHz (most common for audio tracks in movies). Lower sampling rates mean less samples per second, which in turn mean less audio data, since there is a smaller number of sample points to represent the audio. The sampling rate is chosen for a certain application depending on what acoustic artifacts need to be captured. Some acoustic artifacts like speech utterances require a lower sampling rate than an acoustic artifact such as a music tune in a music CD. It’s important to note that higher sampling rates require more storage space and processing power to handle, though this might not be as big of an issue now as it used to be in the old days when digital storage and processing power were of primary considerations.


Bit Depth

These come in different numbers as well but here are a few.
16bit , 24bit , 32bit (Floating Point)

In addition to the sampling rate, which is how many data points of audio we have, there is also the sample depth. Measured in bits per sample, the sample depth, (also known as the sample precision or sample size), is the second important property of an audio file or stream, and it represents the level of detail, or “quality” each sample has. As mentioned above, each audio sample is just a number, and while having a lot of numbers is helpful to represent audio, you also need the range or “quality” of every individual number to be large enough to represent each sample or data point accurately.

What does “quality” mean? For an audio sample, it simply means that the audio sample can represent a higher range of amplitudes. A sample depth of 8 bits means that we have 2^8 = 256 distinct amplitudes that each audio sample can represent, and a sample depth of 16 bits means that we have 2^16 = 65,536 distinct amplitudes that an audio sample can represent, and so on for higher sample depths. The most common sample depths are 16 bits The more distinct amplitudes one has in a digital recording, the closer the digital recording sounds to the original acoustic event.


Bit Rate

Tying the sampling rate and the sample depth together is the bit rate, which is simply the product of both. Since the sampling rate is measured in samples per second and the sample depth is measured in bits per sample, it is therefore measured in (samples per second) x (bits per sample) = bits per second, abbreviated as bps or kbps. It’s worth noting that because the sample depth and the bit rate are related, they frequently, yet erroneously, get used interchangeably.

The bit rate in audio varies according to application. Applications that require high audio quality, like music, usually have a higher bit rate yielding higher quality, or “crisper” audio.


Let’s use an example

I using the song shallow with lady Gaga from the movie a star is born I have a high res audio flac 96khz I am starting with & went to wave as source conversion that file size is 157mb vs the flac 73mb. The 157mb wave as source conversion. Is from pcm_f32le & what the means is in this screenshot

View attachment 308103


We are using 32bit fp for the purpose of increased dynamic range which will be explained below again.

But wait this 157mb files wave conversion of course means it was at 96khz. The bitrate for this is 6144kbps which is basically 6mbps

The 2nd test was to use a wave conversion from the 96khz upsampling with a 192khz sampling rate because my 8ch can do a 192khz high sampling rate. Once again some people may not agree with this logic at all and more will be explained as your read about over sampling. But if your remember the sampling rate being 192khz made the file size for this song (192khz) 315mb wave vs the (96khz) 157mb wave.

Can I hear a difference yes this will be based off users ears opinions music sources dsps. I can hear the difference in my car how much of a difference well that's the thing double sampling rate add more space yes but it doesn't exactly improve the sq.

Now remember I am using 32 Bit Floating point and because I am The dynamic range that can be represented by a 32-bit (floating point) file is 1528 dB. Which is significant for dynamic range and noise floor as well as clipping. It’s almost impossible to clip with this much extra space for the music to spread over.

Now remember The sampling rate refers to the number of samples of audio recorded every second. It is measured in samples per second or Hertz.


OVERSAMPLING

Oversampling is the process of increasing the sample rate by a factor of two or more, usually temporarily inside a plugin so it can perform a process in higher resolution.
Oversampling means more headroom for harmonics that are introduced by distortion and overdrive effects – which can go far beyond 22 kHz.
As we learned, if the harmonics go beyond the Nyquist and aren’t properly filtered, you get aliasing.

Oversampling helps to avoid aliasing as it increases the Nyquist frequency and gives us more ‘spectral headroom’. So going from 96khz - 192khz gives spectral head room.

Let’s into more details now about all these things to have a fuller understanding of everything coming together.

I am explaining these things to you in the most simplistic forms as there is a lot more in-depth reading to fully under the calculations and why this and that has to be for that to work but the simpler I make this the easier your understanding will be on this and you can research more if you like.

So, 16-bit WAV files can store audio from 0 dBFS down to -96 dBFS. Each audio sample consumes 16 bits of space on a hard disk or memory, and at a 48 kHz sampling rate this means that 16 x 48,000 = 768,000 bits per second are needed to store a single channel 16-bit, 48 kHz file. So for

16 bit audio has a total of 768 thousand values per sample @48khz

24 bit audio can have over 16 million values per sample

32 bit float Point takes things even further, providing over 4 billion values per sample

The difference between 24 bit and 32 bit float point audio is impossible to hear, even in music with a wide dynamic range.

The main advantage of 32 bit float point audio is waveforms can “clip” without losing any data.

With 24 bit audio, once the maximum value of 0 dB is reached, that’s it! There is no longer any data left to record louder signals. But 32 bit float point audio has a much higher ceiling, and if a signal happens to go beyond 0db, it can be safely reduced later on with no clipping.

This is why for example using mp3 compressions sounds so loud and ****ty. No room on That's a cd not a mp3 which is far less because of lower bitrate and high compression.

16 bit audio file can have a maximum dynamic range of 96 dB

24 bit can have up to 144 dB of dynamic range

32 bit (floating point) will be 1528 dB of dynamic range so as you can see it is totally worth it for that reason to run 32bit fp.

And you can Now see a clearer overall picture of how bit depth and sampling rates come together.

One last thing – do not confuse bit depth with the bit rate of MP3 files. They are separate, and while an MP3 file has a sample rate that affects the bit rate, MP3s do not have bit depth in the same way as WAV files.

The sampling rate is the frequency at which samples are played back. We need a high enough sampling rate to ensure all audible frequencies can be represented.


The bit depth is the quality of each sample in terms of its mathematical accuracy. Lower bit depths mean there is less digital information available to describe each sample, and this results in noise and a lack of dynamics.

I know I repeated some things a few times it's because I have written So much I didn't want you to forget but this explains a simple introduction to these things for high resolution audio.
Thank you for putting this together, it’s a great read.
 

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Sample rate determines the highest frequency that can be perfectly captured--it takes 2 samples to perfectly capture a sine wave. So, a sample rate of 44.1k can perfectly capture frequencies up to 22,500 Hz. The reason for 44.1 instead of 40k is to leave a little room for the low pass filter that's required. This is ALL that the sample rate does. Determines the high frequency bandwidth.

But depth determines the the number of possible values available to represent the level of the signal at each sample. The lower the bit depth, the greater the quantization error. That error shows up in the signal as noise. So, the higher the bit depth, the lower the noise. Bit depth for a CD is 16 bit, which provides 65,536 possible values. 24 bit provides 16,777,216 possible values. 32 bit provides 4,294,967,296 possible values. 16 bit provides a signal to noise ratio of 96.33 dB. That's like no noise. 32 bit is 192.66 dB.

Bit rate is sample rate times bit depth. This is the amount of data that has to be transferred in real time in order to play a file. So, 44.1k x 16 bit x 2 channels is the bit rate for a CD. The bit rate required to play a CD (or to stream "CD Quality") is 1,411.2 kbps. Bluetooth can theoretically transfer at 1kbps. Realistic throughput is 320k because of required dead space between packets, data transfer for control functions and other overhead.

So, there's no CD quality BT. There's certainly no high res bluetooth, FWIW.
 

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Sample rate determines the highest frequency that can be perfectly captured--it takes 2 samples to perfectly capture a sine wave. So, a sample rate of 44.1k can perfectly capture frequencies up to 22,500 Hz. The reason for 44.1 instead of 40k is to leave a little room for the low pass filter that's required. This is ALL that the sample rate does. Determines the high frequency bandwidth.

But depth determines the the number of possible values available to represent the level of the signal at each sample. The lower the bit depth, the greater the quantization error. That error shows up in the signal as noise. So, the higher the bit depth, the lower the noise. Bit depth for a CD is 16 bit, which provides 65,536 possible values. 24 bit provides 16,777,216 possible values. 32 bit provides 4,294,967,296 possible values. 16 bit provides a signal to noise ratio of 96.33 dB. That's like no noise. 32 bit is 192.66 dB.

Bit rate is sample rate times bit depth. This is the amount of data that has to be transferred in real time in order to play a file. So, 44.1k x 16 bit x 2 channels is the bit rate for a CD. The bit rate required to play a CD (or to stream "CD Quality") is 1,411.2 kbps. Bluetooth can theoretically transfer at 1kbps. Realistic throughput is 320k because of required dead space between packets, data transfer for control functions and other overhead.

So, there's no CD quality BT. There's certainly no high res bluetooth, FWIW.
Just curious of your opinion, but are you satisfied/content with 16 bit, 44.1khz media/playback source or do you feel there is a substantial difference in 24 bit and/or 88.2khz or higher ?
 

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Sample rate determines the highest frequency that can be perfectly captured--it takes 2 samples to perfectly capture a sine wave. So, a sample rate of 44.1k can perfectly capture frequencies up to 22,500 Hz. The reason for 44.1 instead of 40k is to leave a little room for the low pass filter that's required. This is ALL that the sample rate does. Determines the high frequency bandwidth.

But depth determines the the number of possible values available to represent the level of the signal at each sample. The lower the bit depth, the greater the quantization error. That error shows up in the signal as noise. So, the higher the bit depth, the lower the noise. Bit depth for a CD is 16 bit, which provides 65,536 possible values. 24 bit provides 16,777,216 possible values. 32 bit provides 4,294,967,296 possible values. 16 bit provides a signal to noise ratio of 96.33 dB. That's like no noise. 32 bit is 192.66 dB.

Bit rate is sample rate times bit depth. This is the amount of data that has to be transferred in real time in order to play a file. So, 44.1k x 16 bit x 2 channels is the bit rate for a CD. The bit rate required to play a CD (or to stream "CD Quality") is 1,411.2 kbps. Bluetooth can theoretically transfer at 1kbps. Realistic throughput is 320k because of required dead space between packets, data transfer for control functions and other overhead.

So, there's no CD quality BT. There's certainly no high res bluetooth, FWIW.

So just to clarify, you are saying I need to drop $20k on the Alpine F1 status 384K/32bit player, correct? 🤣 (Just kidding!)

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So just to clarify, you are saying I need to drop $20k on the Alpine F1 status 384K/32bit player, correct? (Just kidding!)

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I think they left an important word out of the name, "Symbol" should be tacked on at the end. What's up with the 3" mids and 2" tweets, shouldn't the tweets be tweet-sized?
 

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Great discussion everyone, keep it coming.
 

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Just curious of your opinion, but are you satisfied/content with 16 bit, 44.1khz media/playback source or do you feel there is a substantial difference in 24 bit and/or 88.2khz or higher ?
I used to do demos of my car using an ipod, but that generated too many questions. So now I rip all of those [email protected] to a CD so when people ask, "what are we listening to" I say, "A CD".

I listen to a lot of high res stuff, but often because that's what Amazon says its streaming. Sometimes the mix is different.

But I can't find nor hear any benefit to playback at higher "quality" than CD.
 
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