Best places to buy ACD /DSD in Delhi and surroundings

mbhangui

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Master copy is MP3 ???? 😢
Sounds crazy, but very much possible. This IIRC is the erstwhile HMV. They had absolute no shame in letting precious LP records, master tapes get destroyed in their godown in Calcutta. Most likely they no longer have the original source for the old hindi songs.

One of the companies I worked with was an ISP who also had a business that gave internet, cyber cafe, services. One of the services was to chose your own songs and cut a customized CD. This was an arrangement with a company who had all the rights. They sent us a hard disk with each and every hindi song and the format was mp3. This hard disk was not even 1 Tb.

Those were the iPOD days where apple made you believe mp3 320 kpbs was as good as a wav file and believe me, people were very happy with the CD service that the company did.
 

sachinchavan 15865

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But I see no reason to not have best quality source material (And some times good recording and mastering) after having quality amplification and speakers.
If proven, yes. Otherwise it’d just cost four times the amount with a blind belief that since it’s a wav (and larger) file, it has better resolution/detail.

There are others that I would have to search for as I have forgotten.
Please do that. It would be useful to know the definite assessments of the lossiness of a music file.

A related question for the technically inclined. If you take a mp3 file, and rip it into FLAC format, would the resultant file be same in size or larger than the file you started with? If it is larger, what is all that additional data that gets created? Surely not interpolation/upsampling?
 
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mbhangui

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If proven, yes. Otherwise it’d just cost four times the amount with a blind belief that since it’s a wav (and larger) file, it has better resolution/detail.


Please do that. It would be useful to know the definite assessments of the lossiness of a music file.

A related question for the technically inclined. If you take a mp3 file, and rip it into FLAC format, would the resultant file be same in size or larger than the file you started with? If it is larger, what is all that additional data that gets created? Surely not interpolation/upsampling?
It does increase. I used ffmpeg to convert mp3 to wav. This is what saregama could be doing :mad: :mad: :mad: :mad:
The original size of mp3 was 5952762. The wave file became 10x the size
-rw-r--r-- 1 mbhangui mbhangui 5952762 Nov 23 14:18 'Alisha Chinai - Tera Hone Laga Hoon.mp3'
-rw-r--r-- 1 mbhangui mbhangui 52934602 Nov 23 14:19 'Alisha Chinai - Tera Hone Laga Hoon.wav'

$ file "Alisha Chinai - Tera Hone Laga Hoon.wav"
Alisha Chinai - Tera Hone Laga Hoon.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz

$ file "Alisha Chinai - Tera Hone Laga Hoon.mp3"
Alisha Chinai - Tera Hone Laga Hoon.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 128 kbps, 44.1 kHz, Stereo

argos.indimail.org:(mbhangui) /tmp >ls -l Alisha\ Chinai - Tera Hone Laga Hoon.mp3
-rw-r--r-- 1 mbhangui mbhangui 5952762 Nov 23 14:18 'Alisha Chinai - Tera Hone Laga Hoon.mp3'
argos.indimail.org:(mbhangui) /tmp >ffmpeg -i "Alisha Chinai - Tera Hone Laga Hoon.mp3" "Alisha Chinai - Tera Hone Laga Hoon.wav"
ffmpeg version 5.1.2 Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 12 (GCC)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection' --extra-ldflags='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 ' --extra-cflags=' -I/usr/include/rav1e' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --enable-chromaprint --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libdav1d --enable-libass --enable-libbluray --enable-libbs2b --enable-libcdio --enable-libdrm --enable-libjack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libmysofa --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librav1e --enable-librubberband --enable-libsmbclient --enable-version3 --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-version3 --enable-vapoursynth --enable-libvpx --enable-vulkan --enable-libshaderc --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libxml2 --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-avfilter --enable-libmodplug --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-lto --enable-libmfx --enable-runtime-cpudetect
libavutil 57. 28.100 / 57. 28.100
libavcodec 59. 37.100 / 59. 37.100
libavformat 59. 27.100 / 59. 27.100
libavdevice 59. 7.100 / 59. 7.100
libavfilter 8. 44.100 / 8. 44.100
libswscale 6. 7.100 / 6. 7.100
libswresample 4. 7.100 / 4. 7.100
libpostproc 56. 6.100 / 56. 6.100
Input #0, mp3, from 'Alisha Chinai - Tera Hone Laga Hoon.mp3':
Metadata:
publisher : kajalsusi24
title : Tera Hone Laga Hoon
artist : Alisha Chinai
album_artist : Alisha Chinai
genre : Hindi
comment : kajalsusi24
copyright : kajalsusi24
encoded_by : kajalsusi24
FMPS_PlayCount : 0
FMPS_Rating_Amarok_Score: 0
FMPS_Rating : 0.5
Duration: 00:05:00.12, start: 0.025056, bitrate: 158 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 155 kb/s
Metadata:
encoder : LAME3.99r
Stream #0:1: Video: mjpeg (Baseline), yuvj422p(pc, bt470bg/unknown/unknown), 592x596, 90k tbr, 90k tbn (attached pic)
Metadata:
comment : Cover (front)
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'Alisha Chinai - Tera Hone Laga Hoon.wav':
Metadata:
publisher : kajalsusi24
INAM : Tera Hone Laga Hoon
IART : Alisha Chinai
album_artist : Alisha Chinai
IGNR : Hindi
ICMT : kajalsusi24
ICOP : kajalsusi24
ITCH : kajalsusi24
FMPS_PlayCount : 0
FMPS_Rating_Amarok_Score: 0
FMPS_Rating : 0.5
ISFT : Lavf59.27.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Metadata:
encoder : Lavc59.37.100 pcm_s16le
size= 51694kB time=00:05:00.08 bitrate=1411.2kbits/s speed= 479x
video:0kB audio:51694kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000382%
argos.indimail.org:(mbhangui) /tmp >ls -l Alisha\ Chinai\ -\ Tera\ Hone\ Laga\ Hoon.*
-rw-r--r-- 1 mbhangui mbhangui 5952762 Nov 23 14:18 'Alisha Chinai - Tera Hone Laga Hoon.mp3'
-rw-r--r-- 1 mbhangui mbhangui 52934602 Nov 23 14:19 'Alisha Chinai - Tera Hone Laga Hoon.wav'
argos.indimail.org:(mbhangui) /tmp >file "Alisha Chinai - Tera Hone Laga Hoon.wav"
Alisha Chinai - Tera Hone Laga Hoon.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz
 

sachinchavan 15865

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It does increase. I used ffmpeg to convert mp3 to wav. This is what saregama could be doing :mad: :mad: :mad: :mad:
The original size of mp3 was 5952762. The wave file became 10x the size
-rw-r--r-- 1 mbhangui mbhangui 5952762 Nov 23 14:18 'Alisha Chinai - Tera Hone Laga Hoon.mp3'
-rw-r--r-- 1 mbhangui mbhangui 52934602 Nov 23 14:19 'Alisha Chinai - Tera Hone Laga Hoon.wav'

$ file "Alisha Chinai - Tera Hone Laga Hoon.wav"
Alisha Chinai - Tera Hone Laga Hoon.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz

$ file "Alisha Chinai - Tera Hone Laga Hoon.mp3"
Alisha Chinai - Tera Hone Laga Hoon.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 128 kbps, 44.1 kHz, Stereo
So what exactly happens when the file becomes ten times in size? What is the rest of data that gets added?
 

sachinchavan 15865

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here is some info : Link
Regards
Lot of useful information in there. But doesn’t answer my query. It’s quite a simple one: “How can a mp3 file enlarge to ten times when ripped as a wav? What is the rest of the 90% data on that wav file? And where does it come from?”

Please answer if possible.
 

mbhangui

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Lot of useful information in there. But doesn’t answer my query. It’s quite a simple one: “How can a mp3 file enlarge to ten times when ripped as a wav? What is the rest of the 90% data on that wav file? And where does it come from?”

Please answer if possible.
This is done through mathematical interpolation formulas. There have been many formulas developed over the past many years. Depending on the software different formulas are used. e.g. in audacity one gets two options.
Real-Time Conversion (Sample Rate Converter = Fast Sinc Interpolation, Dither = None)
High-Quality Conversion (Sample Rate Converter = High Quality Sinc Interpolation, Dither = Shaped).
Also as default variables get converted from 16bit to 32 bit float which also increases the size.

Converting from compressed to uncompressed will always result in increased noise floor and if not done correctly can render the music unhearable. An interpolation involves taking part of an existing musical work (as opposed to a sound recording) and incorporating it into a new work. While sometimes confused with sampling a sound recording, interpolating a musical work is different because it does not involve using any of the actual audio sounds contained in a preexisting recording. Instead, new audio is recorded.
 
Last edited:

sachinchavan 15865

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This is done through mathematical interpolation formulas. There have been many formulas developed over the past many years. Depending on the software different formulas are used. e.g. in audacity one gets two options.
Real-Time Conversion (Sample Rate Converter = Fast Sinc Interpolation, Dither = None)
High-Quality Conversion (Sample Rate Converter = High Quality Sinc Interpolation, Dither = Shaped).
Also as default variables get converted from 16bit to 32 bit float which also increases the size.

Converting from compressed to uncompressed will always result in increased noise floor and if not done correctly can render the music unhearable. An interpolation involves taking part of an existing musical work (as opposed to a sound recording) and incorporating it into a new work. While sometimes confused with sampling a sound recording, interpolating a musical work is different because it does not involve using any of the actual audio sounds contained in a preexisting recording. Instead, new audio is recorded.
Thanks @mbhangui for the explanation. So it’s kind of like what image restoration algorithms do with old pictures - they seem to remove the granularity by filing in data, but ending up with a smoothened/glossed out image that to me always looked worse than the original low res pic.

I guess the only way to add back data and restore from compressed to uncompressed without any distortion is to establish the exact reverse of the logic that was used in compressing in the first place (so adding back exactly what was removed) if that was even possible to do.
 

Hiten

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Lot of useful information in there. But doesn’t answer my query. It’s quite a simple one: “How can a mp3 file enlarge to ten times when ripped as a wav? What is the rest of the 90% data on that wav file? And where does it come from?”

Please answer if possible.
mbhangui has very well explained it.
 
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