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9

Here is how you calculate the constant bitrate (CBR) of uncompressed audio: Bits Per Second (bps) = Sample Rate (Hz) * Word Length (bits) * Channel Count Which for 44.1 KHz, 16 bit, 2 Ch. (stereo) audio gives you: 44100 * 16 * 2 = 1411200 bps or 1411.2 kbps (kilobits per second, i.e. bps / 1000) To express that in bytes, kilobytes or megabytes use the ...


9

The ultimate answer for this is: It depends on what material you are encoding. And the strongest scientific evidence is in the coders themselves. When encoding an mp3 using VBR (Variable Bit Rate), some encoders show how many frames were encoded using which bit rate. Here's a screenshot from LAME: You will notice that only 10 frames out of 10735 were ...


8

Download Audacity here. Also download the LAME MP3 library here Install Audacity on your system. Install the LAME MP3 library. Open Audacity Click on File > Open... and select the mp3 file in question Click on Track > Stereo track to Mono Click on File > Export. Choose "MP3 Files" as Format on the dropdown menu and click save. Note 1: If you want to save ...


3

The article you read has either been misinterpreted or was monstrously misleading. It is possible that what they were referring to was the fact that the frequency response required from an analogue medium was only enough to record 100kB/s of digital information. That does not mean that the recording quality of the original recording was equivalent to a ...


3

GIGO [garbage in, garbage out] does apply... ...however - a good mix, master or especially restoration engineer could pull apparent newness out of a low quality file - that's essentially the same task as recovering a track from an old 78 RPM record, or remastering the Beatles albums from the original multitracks. You work to eliminate the 'bad' & ...


3

A sample rate is the rate at which samples are taking from the source sound. It says nothing about how much information is stored in those samples. Whether an audio codec is considered lossy or lossless is dependant on how much information is carried over from the original recorded medium usually (for the sake of argument) based on the 1,412 kbit/s bitrate ...


2

It is impossible to upsample the bitrate. Your files have already lost the info from being compressed to 192. It looks to me as if the AC3 is going to give you a similar if not slightly more consistant result to the MP3. Seeing as this option is also easier to do using the program you mentioned... Go with the AC3


2

On Linux systems like Ubuntu, you can use ffmpeg: ffmpeg -i input.mp3 -c:a libmp3lame -q:a 2 -ac 1 output.mp3 where: -c:a libmp3lame: The audio codec to use -q:a 2: The audio quality (bitrate), see LAME Bitrate Overview -ac 1: One audio channel


2

Your understanding of lossy compression is close, but a little bit off. An MP3 (or any lossy compression) doesn't actually reduce the number of samples per second, but rather, it alters the values of those samples. Lossy and lossless in terms of compression refers to the ability of the compression to reproduce the input to the compression algorithm exactly ...


2

Mp3 is a lossy compression format, the removal of frequencies is permanent. There are lossless files such as flac files which 'pack' the file down to a smaller file size that can then be 'unpacked'. But if they are all 128kbps Mp3 then your out of luck. All those effects you listed are symptoms of destructive compression. You could 'improve' them with ...


2

Your assumption that bitrate = bitdepth * channels * samplerate is true for raw PCM encoded audio. It is not true for bitrate reduction encodings, be it lossless (as flac) or lossy (as mp3). The samplerate and bitdepth given to an mp3 encoder are used to set up appropriately the encoder, and will be used at decoding stage to output a decoded audio with the ...


2

Fi: 44.1khz/16bit (1411kbps) Master: 96khz/24bit (4608kbps) Pitchfork Article Note that only ~30,000 tracks are MASTER quality right now You can tell that Master is available by the M symbol and the playback bar in bottom right will change from HI-FI to MASTER


2

It's an MP3 file. You can't deduce anything from that other than the fact that it's probably got a bit depth of 16 bits. Try the same exercise but using a WAV file. You will have better luck. MP3 is a lossy compression format. Lossy means that by compressing, you 'lose' data. Also with a WAV file the bit-depth is contained in the header. Like it just ...


1

Looking at the specs of the IBM 2401 tape drive from that era, you can see it's at least more complicated than that. The fastest of the 3 models recorded 90 kbyte/s while running the tape at 112.5 inch/s (and you thought a 15 inch/s tape recorder was fast), and the data rate rose with tape speed. That does not mean this was the limit of the medium, though. ...


1

There is no way to compare them because one is analog and one is digital. The way to calculate kbps is: bit depth * sample rate * channels Neither vinyl nor cassettes use bit depth or sample rates. The best you can try to compare is the similarity between bit depth of the CD and dynamic range of the cassette or vinyl


1

Here is a tentative answer based on the 192kHz 24-bit Japanese female speech sample: First I generate the test files from the sample file, sample02.wav: for br in 8 16 24 32 40 48 56 64 80; do for lq in 0 2 5 7 9; do printf -v f0 "out_b%02d_q%d" "$br" "$lq" lame -b "$br" -q "$lq" sample02.wav "${f0}.mp3" sox "${f0}.mp3" -r 96k "$...


1

There's no easy way to measure the perceptual differences (i.e. as perceived by an average audience) between different audio encodings or bitrates. The common way to do these tests is ABX tests. ABX tests are commonly used in evaluations of digital audio data compression methods; Some organizations have setup such tests : http://opus-codec.org/...


1

You could take a look at Zynaptiq Unchirp and Unfilter, but they are expensive options. It's not guaranteed to work but it's an option worth demoing. It really depends on how important these files are too you.


1

Let's assume that what you call 5.1 Dolby is Dolby Digital (aka ac3 codec). That is what your receiver expects. 1 - Any s/pdif interface can output ac3 bitstream (it is part of the standard). 2 - As you want to output the ac3 bitstream, you shouldn't worry about the decoding in the interface. 3 - No 4 - I don't know You must in any case check that the ...


1

Source: http://wiki.hydrogenaud.io/index.php?title=LAME#Recommended_settings_details Deeper source(code): https://github.com/rbrito/lame/blob/origin/libmp3lame/lame.c#L211 Update: --preset insane seems to be using a lowpass filter as well... see LAME: Why is a lowpass filter used with --preset insane


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