LAME conversion to MP3

[[email protected]]

This message follows on from a previous message to this forum. I need
confirmation that I have used the right parameters saving to MP3, and
that the audio quality of the MP3 file is as good as that of the WAV
file.

This message pertains to; http://c-compiler.com/myfiles/a-mp3.zip

The original WAV is at; x.wav

I have converted this file to MP3. Please listen to; x.mp3

According to Windows Media Player, the bit rate of the source file is
192 Kbps, see "windows-media-player.jpg"

According to VLC, the source file has sample rate 48,000 Hz and bits
per sample 16. This particular codec (IMA WAV ADPCM Audio) actually
has 4 bits per sample, but this is decompressed to 16 bits per sample.
See "vlc.jpg".

According to MediaInfo, the source file has sample rate 48,000 Hz and
bit rate 192 kb/s, with a bit depth of 4 bits (which is decompressed
to 16 bits, as noted above), see "mediainfo.jpg".

According to Total Recorder, the source file has sample rate 48,000 Hz
and bit depth 4 bits, see "totalrecorder.jpg".

I use the LAME encoder with Total Recorder to convert the WAV to MP3,
see "totalrecorderA.jpg".

The media format in Total Recorder specifies sample rate 48,000 Hz and
bit rate 192. This is in keeping with the parameters for the source
WAV file, see "totalrecorderB.jpg".

Finally, opening the new MP3 file (converted from WAV) gives the
screen shown in "totalrecorderC.jpg". Bit rate for the MP3 is 192
kbit/s and sample rate is 48,000 Hz.


There are essentially two questions I need to ask.

(1) I have used the parameters for the source WAV file when creating
the MP3 file. Is this a sensible approach? Audio quality is top
priority.

(2) Please tell me if the audio in the MP3 file is as clear as with
the WAV. I think it is, but I would like to be re-assured.

The words on the recording are, "people like that should be .... I
know, they should be homeless".

Thank you for responses.

[geoff]

On 20/02/2020 10:48 am, wrote:
> This message follows on from a previous message to this forum. I need
> confirmation that I have used the right parameters saving to MP3, and
> that the audio quality of the MP3 file is as good as that of the WAV
> file.

An MP3 can *never* have as good quality as its source WAV file.

That said 192kHz can be pretty much 'as good' for most people on average
playback systems.

Surely you are still over-thinking this. S

> (1) I have used the parameters for the source WAV file when creating
> the MP3 file. Is this a sensible approach? Audio quality is top
> priority.

Yes.

>
> (2) Please tell me if the audio in the MP3 file is as clear as with
> the WAV. I think it is, but I would like to be re-assured.

Probably not quite the same, but nothing glaringly different. Most
people wouldn't know or care.

> The words on the recording are, "people like that should be .... I
> know, they should be homeless".
>
> Thank you for responses.
>

Given the quality of the source material, does it matter ? Or is the
objective to enable people to have the best shot at picking out the words.

Also if the WAV is 83KB and the resultant MP3 93KB, what is the point of
making it an MP3 in the first place - player compatibility maybe ?

geoff

[[email protected]]

On Thu, 20 Feb 2020 11:04:04 +1300, geoff >
wrote:

>On 20/02/2020 10:48 am, wrote:
>> This message follows on from a previous message to this forum. I need
>> confirmation that I have used the right parameters saving to MP3, and
>> that the audio quality of the MP3 file is as good as that of the WAV
>> file.
>
>An MP3 can *never* have as good quality as its source WAV file.
>
>That said 192kHz can be pretty much 'as good' for most people on average
>playback systems.

Yeah, the bit rate of the source file is 192 Kbps, so creating an MP3
file with the same bit rate is obviously the best thing to do.

>Surely you are still over-thinking this. S

I'm not an expert at computer audio, and value the thoughts of people
more knowledgeable than myself.

>> (1) I have used the parameters for the source WAV file when creating
>> the MP3 file. Is this a sensible approach? Audio quality is top
>> priority.
>
>Yes.
>
>>
>> (2) Please tell me if the audio in the MP3 file is as clear as with
>> the WAV. I think it is, but I would like to be re-assured.
>
>Probably not quite the same, but nothing glaringly different. Most
>people wouldn't know or care.
>
>> The words on the recording are, "people like that should be .... I
>> know, they should be homeless".
>>
>> Thank you for responses.
>>
>
>Given the quality of the source material, does it matter ? Or is the
>objective to enable people to have the best shot at picking out the words.

Yes, I want good audio quality, to allow people to hear on the MP3 the
same words which are on the source WAV.

>Also if the WAV is 83KB and the resultant MP3 93KB, what is the point of
>making it an MP3 in the first place - player compatibility maybe ?

Exactly. The source WAV doesn't work with MS Edge and other browsers,
whereas the created MP3 should work with everything.

[Brian Gaff \(Sofa 2\)]

You will not ever get the same quality from any lossy compression as you do
from a wav or flac compressed file, alac on Apple, but the lower the bitrate
etc the worse it will get of course. Some of the variable bit rate mp3s do a
good job especially at level 3 very 44.1khz and 256kbits/sec max or greater.
Brian

--
----- --
This newsgroup posting comes to you directly from...
The Sofa of Brian Gaff...

Blind user, so no pictures please
Note this Signature is meaningless.!
> wrote in message
...
> This message follows on from a previous message to this forum. I need
> confirmation that I have used the right parameters saving to MP3, and
> that the audio quality of the MP3 file is as good as that of the WAV
> file.
>
> This message pertains to; http://c-compiler.com/myfiles/a-mp3.zip
>
> The original WAV is at; x.wav
>
> I have converted this file to MP3. Please listen to; x.mp3
>
> According to Windows Media Player, the bit rate of the source file is
> 192 Kbps, see "windows-media-player.jpg"
>
> According to VLC, the source file has sample rate 48,000 Hz and bits
> per sample 16. This particular codec (IMA WAV ADPCM Audio) actually
> has 4 bits per sample, but this is decompressed to 16 bits per sample.
> See "vlc.jpg".
>
> According to MediaInfo, the source file has sample rate 48,000 Hz and
> bit rate 192 kb/s, with a bit depth of 4 bits (which is decompressed
> to 16 bits, as noted above), see "mediainfo.jpg".
>
> According to Total Recorder, the source file has sample rate 48,000 Hz
> and bit depth 4 bits, see "totalrecorder.jpg".
>
> I use the LAME encoder with Total Recorder to convert the WAV to MP3,
> see "totalrecorderA.jpg".
>
> The media format in Total Recorder specifies sample rate 48,000 Hz and
> bit rate 192. This is in keeping with the parameters for the source
> WAV file, see "totalrecorderB.jpg".
>
> Finally, opening the new MP3 file (converted from WAV) gives the
> screen shown in "totalrecorderC.jpg". Bit rate for the MP3 is 192
> kbit/s and sample rate is 48,000 Hz.
>
>
> There are essentially two questions I need to ask.
>
> (1) I have used the parameters for the source WAV file when creating
> the MP3 file. Is this a sensible approach? Audio quality is top
> priority.
>
> (2) Please tell me if the audio in the MP3 file is as clear as with
> the WAV. I think it is, but I would like to be re-assured.
>
> The words on the recording are, "people like that should be .... I
> know, they should be homeless".
>
> Thank you for responses.

[Brian Gaff \(Sofa 2\)]

One of the big problems you see are many cycles of conversion, and this
then adds some awful artefacts such as a gritiness or a modulation in out
tones and the swizzle effect in stereo where the phase is mangled in a
similar way to what happens to treble on a stretched cassette tape as it
snakes across the head. Another issue is just dull and uninteresting audio.
Its fine for non critical stuff, but I'd not want it to be used in a very
dynamic situation.
Brian

--
----- --
This newsgroup posting comes to you directly from...
The Sofa of Brian Gaff...

Blind user, so no pictures please
Note this Signature is meaningless.!
"geoff" > wrote in message
...
> On 20/02/2020 10:48 am, wrote:
>> This message follows on from a previous message to this forum. I need
>> confirmation that I have used the right parameters saving to MP3, and
>> that the audio quality of the MP3 file is as good as that of the WAV
>> file.
>
> An MP3 can *never* have as good quality as its source WAV file.
>
> That said 192kHz can be pretty much 'as good' for most people on average
> playback systems.
>
> Surely you are still over-thinking this. S
>
>> (1) I have used the parameters for the source WAV file when creating
>> the MP3 file. Is this a sensible approach? Audio quality is top
>> priority.
>
> Yes.
>
>>
>> (2) Please tell me if the audio in the MP3 file is as clear as with
>> the WAV. I think it is, but I would like to be re-assured.
>
> Probably not quite the same, but nothing glaringly different. Most people
> wouldn't know or care.
>
>> The words on the recording are, "people like that should be .... I
>> know, they should be homeless".
>>
>> Thank you for responses.
>>
>
> Given the quality of the source material, does it matter ? Or is the
> objective to enable people to have the best shot at picking out the words.
>
> Also if the WAV is 83KB and the resultant MP3 93KB, what is the point of
> making it an MP3 in the first place - player compatibility maybe ?
>
> geoff

[John Williamson]

On 19/02/2020 22:19, wrote:

> Yes, I want good audio quality, to allow people to hear on the MP3 the
> same words which are on the source WAV.
>
To be honest, the only way I can tell what is being said on either file
wile listening on the laptop speakers is by reading the text.


--
Tciao for Now!

John.

[[email protected]]

On Thu, 20 Feb 2020 09:56:39 +0000, John Williamson
> wrote:

>On 19/02/2020 22:19, wrote:
>
>> Yes, I want good audio quality, to allow people to hear on the MP3 the
>> same words which are on the source WAV.
>>
>To be honest, the only way I can tell what is being said on either file
>wile listening on the laptop speakers is by reading the text.

Do the two files sound the same? Can you detect any loss of quality
between the source WAV and destination MP3?

You might try using earphones, that will aid comprehension.

[[email protected]]

On Thu, 20 Feb 2020 08:05:25 -0000, "Brian Gaff \(Sofa 2\)"
> wrote:

>You will not ever get the same quality from any lossy compression as you do
>from a wav or flac compressed file, alac on Apple, but the lower the bitrate
>etc the worse it will get of course. Some of the variable bit rate mp3s do a
>good job especially at level 3 very 44.1khz and 256kbits/sec max or greater.
> Brian

I've used the source WAV parameters; sample rate 48,000 Hz and bit
rate 192 kb/s; when converting to the MP3. I don't know if this is
proper for the destination MP3 file.

Advice? Thank you.

What I really want to know is whether the two files sound similar. I
think they do. Again, thanks for advice.

[John Williamson]

On 20/02/2020 12:00, wrote:
> On Thu, 20 Feb 2020 09:56:39 +0000, John Williamson
> > wrote:
>
>> On 19/02/2020 22:19, wrote:
>>
>>> Yes, I want good audio quality, to allow people to hear on the MP3 the
>>> same words which are on the source WAV.
>>>
>> To be honest, the only way I can tell what is being said on either file
>> wile listening on the laptop speakers is by reading the text.
>
> Do the two files sound the same? Can you detect any loss of quality
> between the source WAV and destination MP3?
>
> You might try using earphones, that will aid comprehension.
>
By the way, I just opened your original .wav file in Adobe Audition, and
it was a compressed file anyway before you started processing it.

Filename: x.wav
Folder: F:\Downloads\a-mp3 (1)
File Type: 48000Hz, 16-bit, Mono
Uncompressed Size: 374.71 KB (383,708 bytes)
File Format: ACM Waveform
Microsoft ACM: IMA ADPCM
Size on Disk: 94.05 KB (96,316 bytes)
Last Written (local): 2/20/2020 12:53:29.780
Length: 0:03.996
191,854 samples

This page explains what it has already done to your original recording.

https://docs.microsoft.com/en-us/windows/win32/multimedia/audio-compression-manager

Not all .wav files are uncompressed, or even the same format if not
compressed.

--
Tciao for Now!

John.

[John Williamson]

On 20/02/2020 12:00, wrote:

> Do the two files sound the same? Can you detect any loss of quality
> between the source WAV and destination MP3?
>
They sound as near to identical as makes no difference on these speakers.

For a conversion that is as close to the original as possible, I use
320kbps as the MP3 bit rate, starting with 24 or 16 bit depth .wav
files, and most people can't tell the difference even when the original
contains noise sources such as cymbals as well as musical instruments.

192 kbps is good enough for normal listening on domestic equipment or in
a car, in my experience.

> You might try using earphones, that will aid comprehension.
>
Depending on your reason for using the clip, why should it be necessary
to use high quality reproduction to understand it, unless it is part of
an educational process as to how not to record stuff?

--
Tciao for Now!

John.

[Andy Burns[_2_]]

John Williamson wrote:

> I just opened your original .wav file in Adobe Audition, and it was a
> compressed file anyway before you started processing it.
>
> Filename:*** x.wav
> Folder:*** F:\Downloads\a-mp3 (1)
> File Type:*** 48000Hz, 16-bit, Mono
> Uncompressed Size:*** 374.71 KB (383,708 bytes)
> File Format:*** ACM Waveform
> ****Microsoft ACM: IMA ADPCM

Maybe it was originally recorded with ATRAC compression, if originated
on Mike's minidisc recorder?

[[email protected]]

On Thu, 20 Feb 2020 15:36:51 +0000, John Williamson
> wrote:

>On 20/02/2020 12:00, wrote:
>
>> Do the two files sound the same? Can you detect any loss of quality
>> between the source WAV and destination MP3?
>>
>They sound as near to identical as makes no difference on these speakers.

Good. Thank you for the re-assurance.

When I listen to the audio file in WAV and MP3, on earphones, it
sounds exactly the same. I am unable to detect any loss of quality.

Perhaps we could have one or two more people offering opinions on
this?

>For a conversion that is as close to the original as possible, I use
>320kbps as the MP3 bit rate, starting with 24 or 16 bit depth .wav
>files, and most people can't tell the difference even when the original
>contains noise sources such as cymbals as well as musical instruments.
>
>192 kbps is good enough for normal listening on domestic equipment or in
>a car, in my experience.

The bit rate of the source file is 192 Kbps, and the source file has
sample rate 48,000 Hz. I have kept the same parameters when converting
to MP3, so the MP3 file has bit rate 192Kbps and sample rate 48 KHz.

Is this the right approach to take with the conversion? Intuitively I
would think the bit rate and sample rate should be the same for both,
but my intuition may be wrong.

>> You might try using earphones, that will aid comprehension.

I can hear more clearly through earphones than laptop speakers.

[[email protected]]

On Thu, 20 Feb 2020 15:35:35 +0000, John Williamson
> wrote:

>By the way, I just opened your original .wav file in Adobe Audition, and
>it was a compressed file anyway before you started processing it.
>
>Filename: x.wav
>Folder: F:\Downloads\a-mp3 (1)
>File Type: 48000Hz, 16-bit, Mono
>Uncompressed Size: 374.71 KB (383,708 bytes)
>File Format: ACM Waveform
> Microsoft ACM: IMA ADPCM
>Size on Disk: 94.05 KB (96,316 bytes)
>Last Written (local): 2/20/2020 12:53:29.780
>Length: 0:03.996
> 191,854 samples
>
>This page explains what it has already done to your original recording.
>
>https://docs.microsoft.com/en-us/windows/win32/multimedia/audio-compression-manager
>
>Not all .wav files are uncompressed, or even the same format if not
>compressed.

You are correct. The file uses ADPCM compression, I believe, so it is
not CD quality. It uses a bit depth of 4 bits, which is de-compressed
when playing to 16 bit.

It's really just a practical question, is the MP3 audio quality as
good as the (compressed) source WAV.

I would keep the oriiginal WAV file, but web browsers can't play this
particular type of WAV file.

[Scott Dorsey]

If I were you I'd try to do some processing to improve intelligibility rather
than worrying about fidelity. High pass everything below 200 Hz or so,
then low-pass everything above maybe 6KHz, and consider sticking a presence
boost in there. You might then consider an expander and fiddling with the
threshold on the expander to try and boost the voice out of the noise.

I think you are worried about entirely the wrong thing here.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

> wrote:
>
>You are correct. The file uses ADPCM compression, I believe, so it is
>not CD quality. It uses a bit depth of 4 bits, which is de-compressed
>when playing to 16 bit.

ADPCM is not lossy compression at all, it is a sort of encoding method
intended to get more usable dynamic range with fewer bits but it's still
straight PCM... just not linear PCM. I had no idea you could do it with
as few as 4 bits, but 8-bit u-law encoding is typical telephone quality
today.

>I would keep the oriiginal WAV file, but web browsers can't play this
>particular type of WAV file.

Have you considered retracking this with proper encoding and microphone
placement?
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[[email protected]]

On 21 Feb 2020 12:33:06 -0500, (Scott Dorsey) wrote:

>>I would keep the oriiginal WAV file, but web browsers can't play this
>>particular type of WAV file.
>
>Have you considered retracking this with proper encoding and microphone
>placement?
>--scott

The recording device only produces WAV files with sample rate 48 KHz
and bit rate 192 kb/s. The WAV files it produces cannot be read by MS
Edge and other web browsers.

Can you download and unzip the file

http://c-compiler.com/myfiles/a-mp3.zip

and listen to the files x.wav and x.mp3 ? Are they similar?

The audio files are 4 seconds each, and to my untrained ear they sound
much the same.

What do you think, is the encoding scheme I chose for conversion from
WAV to MP3 fit for purpose?

[[email protected]]

On 21 Feb 2020 12:30:08 -0500, (Scott Dorsey) wrote:

>If I were you I'd try to do some processing to improve intelligibility rather
>than worrying about fidelity. High pass everything below 200 Hz or so,
>then low-pass everything above maybe 6KHz, and consider sticking a presence
>boost in there. You might then consider an expander and fiddling with the
>threshold on the expander to try and boost the voice out of the noise.
>
>I think you are worried about entirely the wrong thing here.
>--scott

The actions on the source WAV file must be such as to maintain
authenticity. If I start changing and deleting parts of the audio
file, that would render the output file as different from the source
file, and make it untrustworthy.

Sorry to bug you, but; I've used the source WAV parameters; sample
rate 48,000 Hz and bit rate 192 kb/s; when converting to the MP3. I
don't know if this is proper for the destination MP3 file. What is
your view?

[Scott Dorsey]

> wrote:
>On 21 Feb 2020 12:33:06 -0500, (Scott Dorsey) wrote:
>
>>>I would keep the oriiginal WAV file, but web browsers can't play this
>>>particular type of WAV file.
>>
>>Have you considered retracking this with proper encoding and microphone
>>placement?
>
>The recording device only produces WAV files with sample rate 48 KHz
>and bit rate 192 kb/s. The WAV files it produces cannot be read by MS
>Edge and other web browsers.

Have you considered retracking this with proper encoding and microphone
placement?

>What do you think, is the encoding scheme I chose for conversion from
>WAV to MP3 fit for purpose?

The problem is what you're starting out from.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

> wrote:
>On 21 Feb 2020 12:30:08 -0500, (Scott Dorsey) wrote:
>
>>If I were you I'd try to do some processing to improve intelligibility rather
>>than worrying about fidelity. High pass everything below 200 Hz or so,
>>then low-pass everything above maybe 6KHz, and consider sticking a presence
>>boost in there. You might then consider an expander and fiddling with the
>>threshold on the expander to try and boost the voice out of the noise.
>>
>>I think you are worried about entirely the wrong thing here.
>
>The actions on the source WAV file must be such as to maintain
>authenticity. If I start changing and deleting parts of the audio
>file, that would render the output file as different from the source
>file, and make it untrustworthy.

I don't know about UK law, but as soon as you have made a transfer to
MP3, or even a transfer to flat PCM, your file is no longer admissible in
court. Rules of evidence in the UK are likely different but you can hire
any one of a number of excellent forensic audio people there who can create
an audition file which is separate from the traceable reference file (which
is what is normally done for courtroom proceedings in the US).

>Sorry to bug you, but; I've used the source WAV parameters; sample
>rate 48,000 Hz and bit rate 192 kb/s; when converting to the MP3. I
>don't know if this is proper for the destination MP3 file. What is
>your view?

My view is that you are looking at totally the wrong thing, but since you
refuse to explain why you want to do any of this, it's hard to know.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[geoff]

On 22/02/2020 6:30 am, Scott Dorsey wrote:
> If I were you I'd try to do some processing to improve intelligibility rather
> than worrying about fidelity. High pass everything below 200 Hz or so,
> then low-pass everything above maybe 6KHz, and consider sticking a presence
> boost in there. You might then consider an expander and fiddling with the
> threshold on the expander to try and boost the voice out of the noise.
>
> I think you are worried about entirely the wrong thing here.
> --scott
>

Yeah, the source media content is the only thing getting in the way of
intelligibility. Unfortunately most of the extraneous clutter is not too
far removed from the vocal frequencies.

Looks like more of a job for Spectral Layers, or a Mac or Linux
equivalent, to edit out the unwanted noises. Arduous fiddly work, but if
the content is really that important ....

geoff

[geoff]

On 22/02/2020 8:38 am, wrote:
> On 21 Feb 2020 12:33:06 -0500, (Scott Dorsey) wrote:
>
>>> I would keep the oriiginal WAV file, but web browsers can't play this
>>> particular type of WAV file.
>>
>> Have you considered retracking this with proper encoding and microphone
>> placement?
>> --scott
>
> The recording device only produces WAV files with sample rate 48 KHz
> and bit rate 192 kb/s. The WAV files it produces cannot be read by MS
> Edge and other web browsers.
>
> Can you download and unzip the file
>
> http://c-compiler.com/myfiles/a-mp3.zip
>
> and listen to the files x.wav and x.mp3 ? Are they similar?

Yes they are similar, almost if not totally indistinguishable, not
helped by of the overall clutter or extraneous sounds.

>
> The audio files are 4 seconds each, and to my untrained ear they sound
> much the same.
>
> What do you think, is the encoding scheme I chose for conversion from
> WAV to MP3 fit for purpose?

Kind of depends what your purpose is. The only unsatisfactory part is
the incredibly poor recording.

geoff

[Andy Burns[_2_]]

Scott Dorsey wrote:

> since you refuse to explain why you want to do any of this, it's hard
> to know.

I suspect the answer to that is somewhere between "prove that
mindcontrol is real" and "show that MI5 are out to get me"

[Trevor]

On 22/02/2020 4:33 am, Scott Dorsey wrote:
> > wrote:
>>
>> You are correct. The file uses ADPCM compression, I believe, so it is
>> not CD quality. It uses a bit depth of 4 bits, which is de-compressed
>> when playing to 16 bit.
>
> ADPCM is not lossy compression at all, it is a sort of encoding method
> intended to get more usable dynamic range with fewer bits but it's still
> straight PCM... just not linear PCM. I had no idea you could do it with
> as few as 4 bits, but 8-bit u-law encoding is typical telephone quality
> today.

Yes, and he never said "lossy compression". However there is plenty of
loss already inherent in those low bit encoding schemes anyway. The
whole point of MP3 etc was to *reduce* the audible loss at low data
rates. I'm sure you know this, but the old compression Vs compression Vs
compression linguistic problem raises it's head once again.

[Trevor]

On 22/02/2020 4:30 am, Scott Dorsey wrote:
> If I were you I'd try to do some processing to improve intelligibility rather
> than worrying about fidelity. High pass everything below 200 Hz or so,
> then low-pass everything above maybe 6KHz, and consider sticking a presence
> boost in there. You might then consider an expander and fiddling with the
> threshold on the expander to try and boost the voice out of the noise.
>
> I think you are worried about entirely the wrong thing here.
> --scott

Yes he seems to keep ignoring the fact the recording is lousy in the
first place and therefore worrying so much about MP3 settings is rather
pointless.

[[email protected]]

On Sat, 22 Feb 2020 16:02:44 +1100, Trevor > wrote:

>On 22/02/2020 4:33 am, Scott Dorsey wrote:
>> > wrote:
>>>
>>> You are correct. The file uses ADPCM compression, I believe, so it is
>>> not CD quality. It uses a bit depth of 4 bits, which is de-compressed
>>> when playing to 16 bit.
>>
>> ADPCM is not lossy compression at all, it is a sort of encoding method
>> intended to get more usable dynamic range with fewer bits but it's still
>> straight PCM... just not linear PCM. I had no idea you could do it with
>> as few as 4 bits, but 8-bit u-law encoding is typical telephone quality
>> today.
>
>Yes, and he never said "lossy compression". However there is plenty of
>loss already inherent in those low bit encoding schemes anyway. The
>whole point of MP3 etc was to *reduce* the audible loss at low data
>rates. I'm sure you know this, but the old compression Vs compression Vs
>compression linguistic problem raises it's head once again.

Can you download and unzip the file

http://c-compiler.com/myfiles/a-mp3.zip

and listen to the files x.wav and x.mp3 ? Are they similar?

They are only 4 seconds in length, there is very little work involved
in listening to these two files.

I am looking for one or two more opinions as to whether the source WAV
file and destination MP3 file are similar to the ear. You might try
listening through earphones, for better clarity.

Sorry to keep harping on about this, but I am trying to obtain
re-assurance that the conversion to MP3 gives a file which is
effectively the same to the ear as the source WAV.

[[email protected]]

On 21 Feb 2020 16:21:13 -0500, (Scott Dorsey) wrote:

>Have you considered retracking this with proper encoding and microphone
>placement?
>
>>What do you think, is the encoding scheme I chose for conversion from
>>WAV to MP3 fit for purpose?
>
>The problem is what you're starting out from.
>--scott

Unfortunately the sound source is beyond my control. It always gives a
WAV file with the same parameters.

[[email protected]]

On 21 Feb 2020 16:24:46 -0500, (Scott Dorsey) wrote:

> > wrote:
>>On 21 Feb 2020 12:30:08 -0500, (Scott Dorsey) wrote:
>>
>>>If I were you I'd try to do some processing to improve intelligibility rather
>>>than worrying about fidelity. High pass everything below 200 Hz or so,
>>>then low-pass everything above maybe 6KHz, and consider sticking a presence
>>>boost in there. You might then consider an expander and fiddling with the
>>>threshold on the expander to try and boost the voice out of the noise.
>>>
>>>I think you are worried about entirely the wrong thing here.
>>
>>The actions on the source WAV file must be such as to maintain
>>authenticity. If I start changing and deleting parts of the audio
>>file, that would render the output file as different from the source
>>file, and make it untrustworthy.
>
>I don't know about UK law, but as soon as you have made a transfer to
>MP3, or even a transfer to flat PCM, your file is no longer admissible in
>court. Rules of evidence in the UK are likely different but you can hire
>any one of a number of excellent forensic audio people there who can create
>an audition file which is separate from the traceable reference file (which
>is what is normally done for courtroom proceedings in the US).

I need this audio file for two purposes.

1) to present to a court or tribunal in support of proceedings, as
evidence. For this I would submit the source WAV file, unedited, apart
from being clipped.

2) to post to a web page in a public-facing role. This should be an
MP3 file, preferably indistinguishable to the average human ear from
the WAV,

Thank you for your advice re US law. I don't know what the situation
is in the UK regarding this sort of evidence.


>>Sorry to bug you, but; I've used the source WAV parameters; sample
>>rate 48,000 Hz and bit rate 192 kb/s; when converting to the MP3. I
>>don't know if this is proper for the destination MP3 file. What is
>>your view?

If you could provide some advice regarding parameters for the target
MP3 file, as in the above quoted paragraph, it would be appreciated.

[geoff]

On 23/02/2020 6:45 am, wrote:

>
> I am looking for one or two more opinions as to whether the source WAV
> file and destination MP3 file are similar to the ear. You might try
> listening through earphones, for better clarity.
>
> Sorry to keep harping on about this, but I am trying to obtain
> re-assurance that the conversion to MP3 gives a file which is
> effectively the same to the ear as the source WAV.
>

For **** sake ! The files are effectively the same.

The only thing standing in the way of intelligibility is the original
recorded sound - not the quality of the recording specs, but the
background extraneous noises and their level in comparison to the
'wanted' content.

geoff

[geoff]

On 23/02/2020 8:02 am, wrote:
> On 21 Feb 2020 16:24:46 -0500, (Scott Dorsey) wrote:
>
>> > wrote:
>>> On 21 Feb 2020 12:30:08 -0500, (Scott Dorsey) wrote:
>>>
>>>> If I were you I'd try to do some processing to improve intelligibility rather
>>>> than worrying about fidelity. High pass everything below 200 Hz or so,
>>>> then low-pass everything above maybe 6KHz, and consider sticking a presence
>>>> boost in there. You might then consider an expander and fiddling with the
>>>> threshold on the expander to try and boost the voice out of the noise.
>>>>
>>>> I think you are worried about entirely the wrong thing here.
>>>
>>> The actions on the source WAV file must be such as to maintain
>>> authenticity. If I start changing and deleting parts of the audio
>>> file, that would render the output file as different from the source
>>> file, and make it untrustworthy.
>>
>> I don't know about UK law, but as soon as you have made a transfer to
>> MP3, or even a transfer to flat PCM, your file is no longer admissible in
>> court. Rules of evidence in the UK are likely different but you can hire
>> any one of a number of excellent forensic audio people there who can create
>> an audition file which is separate from the traceable reference file (which
>> is what is normally done for courtroom proceedings in the US).
>
> I need this audio file for two purposes.
>
> 1) to present to a court or tribunal in support of proceedings, as
> evidence. For this I would submit the source WAV file, unedited, apart
> from being clipped.
>
> 2) to post to a web page in a public-facing role. This should be an
> MP3 file, preferably indistinguishable to the average human ear from
> the WAV,
>

Thanks for finally explaining the context.

But careful - Surely # 2) would compromise # 1) ?

geoff

[[email protected]]

Trevor wrote:


>I'm sure you know this, but the old compression Vs compression Vs
>compression linguistic problem raises it's head once again.


That's why, in both professional and layman circles, I refer to compression
ONLY in the context of dynamics and dynamics processing, and 'data size
reduction' or 'data reduction' in the context of lossy file conversion.

If someone else uses the term 'compression' or 'compressed', I automatically
ask what threshold, ratio, and attack and release speeds they used. When they
say 'No, I mean compressed down to a smaller file size' I respond, 'Oh, you mean
data-reduced, or data reduction. When they ask me why I don't use 'compression
to mean the same thing, I just look at them, expressionless and tell them:

Think about why.

That usually convinces them.

[Scott Dorsey]

> wrote:
>Trevor wrote:
>>I'm sure you know this, but the old compression Vs compression Vs
>>compression linguistic problem raises it's head once again.
>
>That's why, in both professional and layman circles, I refer to compression
>ONLY in the context of dynamics and dynamics processing, and 'data size
>reduction' or 'data reduction' in the context of lossy file conversion.

The thing is, the nonlinear encoding used by ADPCM isn't really either of
these things.

You -might- refer to it as "companding" if you had to, but really that's
still pushing it. "Nonlinear encoding" is as good as it gets.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[[email protected]]

Scott Dorsey wrote:

>The thing is, the nonlinear encoding used by ADPCM isn't
>really either of those things.

Then what is it?? Come on, I'm black & white here, no time
for fuzziness, lol!

[[email protected]]

Cat got your tongue, Scottso? :D

[Scott Dorsey]

> wrote:
>Scott Dorsey wrote:
>
>>The thing is, the nonlinear encoding used by ADPCM isn't
>>really either of those things.
>
>Then what is it?? Come on, I'm black & white here, no time
>for fuzziness, lol!

It's a nonlinear encoding. Amplitude step is log(n) instead of n, so
the size of step 0 is smaller than the size of step 127. If you look
for an explanation of u-Law online, that is the most common method used
in the real world. But it's still straight PCM.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

Andy Burns > wrote:
>Scott Dorsey wrote:
>
>> since you refuse to explain why you want to do any of this, it's hard
>> to know.
>
>I suspect the answer to that is somewhere between "prove that
>mindcontrol is real" and "show that MI5 are out to get me"

I thought that's what MI5's job was?
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[geoff]

On 23/02/2020 1:14 pm, wrote:
> Trevor wrote:
>
>
>> I'm sure you know this, but the old compression Vs compression Vs
>> compression linguistic problem raises it's head once again.
>
>
> That's why, in both professional and layman circles, I refer to compression
> ONLY in the context of dynamics and dynamics processing, and 'data size
> reduction' or 'data reduction' in the context of lossy file conversion.
>
> If someone else uses the term 'compression' or 'compressed', I automatically
> ask what threshold, ratio, and attack and release speeds they used. When they
> say 'No, I mean compressed down to a smaller file size' I respond, 'Oh, you mean
> data-reduced, or data reduction. When they ask me why I don't use 'compression
> to mean the same thing, I just look at them, expressionless and tell them:


Compressed files, as in ZIP, FLAC, etc do not involve any data reduction
whatsoever.

geoff

[[email protected]]

geoff wrote:

>Compressed files, as in ZIP, FLAC, etc do not involve any data reduction
>whatsoever.

>geoff


I will still use 'data reduction' to describe them, because too many idiots out there
assume that doing so equates to reducing the dynamics of the audio.

Ninety-percent of the general public - my audience - think that MP3 and other lossy
formats reduce dynamic range and contributed to the current loudness war. That's
what I'm trying to untrain.

[None]

<theckmah @ shortbus . edu> wrote in message
>...
> >geoff wrote:
>
> >Compressed files, as in ZIP, FLAC, etc do not involve any data reduction
> >whatsoever.
>
> I will still use 'data reduction' to describe them, because too many
> idiots out there
> assume that doing so equates to reducing the dynamics of the audio.

So you'll still use incorrect terminology, because you're a ****ing idiot.
True to form.

> Ninety-percent of the general public - my audience - think that MP3 and
> other lossy
> formats reduce dynamic range and contributed to the current loudness war.
> That's
> what I'm trying to untrain.

It's hilarious that a dumb**** like you thinks you're "untraining" anyone by
gibbering about things you refuse to understand. But that's life as a
****ing retard, isn't it, li'l buddy? You don’t even know what
"ninety-percent" means, do you? Numbers are just gibberish to you. FEDJB.

[[email protected]]

On Sun, 23 Feb 2020 11:09:41 +1300, geoff >
wrote:

>On 23/02/2020 6:45 am, wrote:
>
>>
>> I am looking for one or two more opinions as to whether the source WAV
>> file and destination MP3 file are similar to the ear. You might try
>> listening through earphones, for better clarity.
>>
>> Sorry to keep harping on about this, but I am trying to obtain
>> re-assurance that the conversion to MP3 gives a file which is
>> effectively the same to the ear as the source WAV.
>>
>
>For **** sake ! The files are effectively the same.

Thank you for confirming my opinion, that the source WAV and target
MP3 sound exactly the same to my untrained ear.

Regarding the conversion parameters. When I use Total Recorder to open
the WAV file, and save to MP3, the parameters of the created MP3 file
are obtained from the parameters of the WAV file. So if the source WAV
has sample rate 48 KHz and bit rate 192 Kbps, the default for the
created MP3 file will also be sample rate 48 KHz and bit rate 192
Kbps.

This ends the questions I had regarding the conversion. Thank you for
your help.

[[email protected]]

On Wednesday, February 19, 2020 at 3:48:45 PM UTC-6, wrote:
> (2) Please tell me if the audio in the MP3 file is as clear as with
> the WAV. I think it is, but I would like to be re-assured.

To the human ear, the MP3 sounds every bit as clear as the WAV file. I seriously doubt anyone would be able to reliably distinguish between the two.

[Mike Rivers[_2_]]

On 2/23/2020 10:54 PM, wrote:

> To the human ear, the MP3 sounds every bit as clear as the WAV file. I seriously doubt anyone would be able to reliably distinguish between the two.

I would describe it as the MP3 sounds just as unintelligible as the WAV
file. If this recording was made in a controlled situation, it needs
better mic placement. Since apparently the situation is uncontrolled,
what it needs is a good forensic scrubbing with the proper tools and
experience.

Then, the high bit rate MP3 file and WAV file would be equally
intelligible. And if it was a well crafted and well recorded song, the
two file formats would be indistinguishable to most listeners.


--
For a good time, call http://mikeriversaudio.wordpress.com

[Phil W]

2020-02-23 / 20:25 spammed once again:
> On Sun, 23 Feb 2020 11:09:41 +1300, geoff >
> wrote:
>
>> On 23/02/2020 6:45 am, wrote:
>
>>> I am looking for one or two more opinions as to whether the source WAV
>>> file and destination MP3 file are similar to the ear. You might try
>>> listening through earphones, for better clarity.
>>>
>>> Sorry to keep harping on about this, but I am trying to obtain
>>> re-assurance that the conversion to MP3 gives a file which is
>>> effectively the same to the ear as the source WAV.
>
>> For **** sake ! The files are effectively the same.
>
> Thank you for confirming my opinion, that the source WAV and target
> MP3 sound exactly the same to my untrained ear.
>
> Regarding the conversion parameters. When I use Total Recorder to open
> the WAV file, and save to MP3, the parameters of the created MP3 file
> are obtained from the parameters of the WAV file. So if the source WAV
> has sample rate 48 KHz and bit rate 192 Kbps, the default for the
> created MP3 file will also be sample rate 48 KHz and bit rate 192
> Kbps.

A small amount of logical thinking might lead to the following:
if the SOURCE file was encoded with 192 kbps, then decoded and encoded a
second time, why shouldn´t you try to keep conversion artifacts to the
least possible minimum???
So, just like I suggested already several times and you still keep ignoring:
go for the highest possible MP3 bitrate (= 320 kbps CBR) for the least
amount of audio quality loss. Yes, it might be more than necessary, but
there is nood to worry, that it might have come out better. Yes, the
resulting MP3 file gets bigger with higher bitrate. BUT with such a tiny
source WAV, the resulting MP3 will still be tiny enough to embed it on a
webpage.

Can it be so complicated to understand this???

[[email protected]]

Mike Rivers wrote:

>It needs better mic placement

Any mic placement is better than what aounds like banging around
in someone's pants pocket, or inside a book bag!

[John Williamson]

On 24/02/2020 12:11, wrote:
> Mike Rivers wrote:
>
>> It needs better mic placement
>
> Any mic placement is better than what aounds like banging around
> in someone's pants pocket, or inside a book bag!
>
Which is how he makes most of his recordings.

--
Tciao for Now!

John.

[[email protected]]

John Williamson:

But this sounds like recorded evidence of something. To be used
in a legal matter.

Although, the pocket DAT audio recorded by a patron inside that
RI nightclub seventeen years ago was much clearer - perhaps too
clear, if you know what I mean. The tape - and DAT deck with
scorched exterior - were found in its deceased owners closet several
years later, was transferred by forensics to a functioning cassette,
and was used at the victim settlement trials. Very little of it was made
public.