Editing MP3 (section)

=== Bit rate ===
{| class="wikitable infobox"
|+MPEG Audio Layer III<br />available bit rates ({{nowrap|kbit/s}})<ref name="neuron2-cd-1991" /><ref name="MPEG-2.5" /><ref name="MPEG-2.5-2" /><ref name="mp3tech-iso13818-3" /><ref>{{cite web |title=Guide to command line options (in CVS) |url=https://lame.cvs.sourceforge.net/viewvc/lame/lame/USAGE |url-status=dead |archive-url=https://web.archive.org/web/20130408110355/http://lame.cvs.sourceforge.net/viewvc/lame/lame/USAGE |archive-date=8 April 2013 |access-date=4 August 2010}}</ref>
|-
! MPEG-1<br />Audio Layer III
! MPEG-2<br />Audio Layer III
! MPEG-2.5<br />Audio Layer III
|-
| –
| 8
| 8
|-
| –
| 16
| 16
|-
| –
| 24
| 24
|-
| 32
| 32
| 32
|-
| 40
| 40
| 40
|-
| 48
| 48
| 48
|-
| 56
| 56
| 56
|-
| 64
| 64
| 64
|-
| 80
| 80
| –
|-
| 96
| 96
| –
|-
| 112
| 112
| –
|-
| 128
| 128
| –
|-
| –

| 144
| –
|-
| 160
| 160
| –
|-
| 192
| –
| –
|-
| 224
| –
| –
|-
| 256
| –
| –
|-
| 320
| –
| –
|}

{| class="wikitable infobox"
|+Supported sampling rates<br />by MPEG Audio Format<ref name="neuron2-cd-1991" /><ref name="MPEG-2.5" /><ref name="MPEG-2.5-2" /><ref name="mp3tech-iso13818-3" />
|-
! MPEG-1<br />Audio Layer III
! MPEG-2<br />Audio Layer III
! MPEG-2.5<br />Audio Layer III
|-
| –
| –
| 8&nbsp;kHz
|-
| –
| –
| 11.025&nbsp;kHz
|-
| –
| –
| 12&nbsp;kHz
|-
| –
| 16&nbsp;kHz
| –
|-
| –
| 22.05&nbsp;kHz
| –
|-
| –
| 24&nbsp;kHz
| –
|-
| 32&nbsp;kHz
| –
| –
|-
| 44.1&nbsp;kHz
| –
| –
|-
| 48&nbsp;kHz
| –
| –
|}
{{more citations needed section|date=July 2020}}
Bit rate is the product of the sample rate and number of bits per sample used to encode the music. CD audio is 44100 samples per second. The number of bits per sample also depends on the number of audio channels. The CD is stereo and 16 bits per channel. So, multiplying 44100 by 32 gives 1411200—the bit rate of uncompressed CD digital audio. MP3 was designed to encode this {{nowrap|1411 kbit/s}} data at {{nowrap|320 kbit/s}} or less. If less complex passages are detected by the MP3 algorithms then lower bit rates may be employed. When using MPEG-2 instead of MPEG-1, MP3 supports only lower sampling rates (16,000, 22,050, or 24,000 samples per second) and offers choices of bit rate as low as {{nowrap|8 kbit/s}} but no higher than {{nowrap|160 kbit/s}}. By lowering the sampling rate, MPEG-2 layer III removes all frequencies above half the new sampling rate that may have been present in the source audio.

As shown in these two tables, 14 selected bit rates are allowed in MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and {{nowrap|320 kbit/s}}, along with the 3 highest available sampling rates of 32, 44.1 and 48&nbsp;[[kHz]].<ref name="MPEG-2.5-2" /> MPEG-2 Audio Layer III also allows 14 somewhat different (and mostly lower) bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, {{nowrap|160 kbit/s}} with sampling rates of 16, 22.05 and 24&nbsp;[[kHz]] which are exactly half that of MPEG-1.<ref name="MPEG-2.5-2" /> MPEG-2.5 Audio Layer III frames are limited to only 8 bit rates of 8, 16, 24, 32, 40, 48, 56 and {{nowrap|64 kbit/s}} with 3 even lower sampling rates of 8, 11.025, and 12&nbsp;kHz.{{Citation needed|reason=Based on results from the LAME encoder, these do seem to be the actual bit rates supported by MPEG-2.5, but official documents claim MPEG-2.5 has the same possible bit rates as MPEG-2. Answer: Bitrate switching implies VBR so, it is not CBR anymore. When MPEG-2 frames are needed instead of the smaller 2.5 frames, the former are generated. Can we find a source that mentions this limitation?|date=December 2013}} On earlier systems that only support the MPEG-1 Audio Layer III standard, MP3 files with a bit rate below {{nowrap|32 kbit/s}} might be played back sped-up and pitched-up.

Earlier systems also lack [[fast forward]]ing and rewinding playback controls on MP3.<ref>{{cite web|quote=Search – locating a desired position on thedisc (audio CD only) |url=http://resources.jvc.com/Resources/00/00/95/lvt1213-001b.pdf |archive-url=https://web.archive.org/web/20200820112149if_/http://resources.jvc.com/Resources/00/00/95/lvt1213-001b.pdf  |archive-date=2020-08-20 |language=mul |page=14 |title=JVC RC-EX30 operation manual |date=2004 }} (2004 [[boombox]])</ref><ref>{{cite web |url=https://www.sharp.co.uk/cps/rde/xbcr/documents/documents/om/13_dvd/DVRW250H_OM_GB.pdf |quote=• Fast forward and review playback does not work with a MP3/WMA/JPEG-CD. |page=33 |language=en-gb |title=DV-RW250H Operation-Manual GB |date=2004 |access-date=20 August 2020 |archive-date=20 August 2020 |archive-url=https://web.archive.org/web/20200820113949/https://www.sharp.co.uk/cps/rde/xbcr/documents/documents/om/13_dvd/DVRW250H_OM_GB.pdf |url-status=live }}</ref>

MPEG-1 frames contain the most detail in {{nowrap|320 kbit/s}} mode, the highest allowable bit rate setting,<ref>{{cite web |title=Sound Quality Comparison of Hi-Res Audio vs. CD vs. MP3 |url=https://www.sony.com/electronics/hi-res-audio-mp3-cd-sound-quality-comparison |website=www.sony.com |publisher=[[Sony]] |access-date=11 August 2020 |language=en |archive-date=14 September 2020 |archive-url=https://web.archive.org/web/20200914005253/https://www.sony.com/electronics/hi-res-audio-mp3-cd-sound-quality-comparison |url-status=live }}</ref> with silence and simple tones still requiring {{nowrap|32 kbit/s}}. MPEG-2 frames can capture up to 12&nbsp;kHz sound reproductions needed up to {{nowrap|160 kbit/s}}. MP3 files made with MPEG-2 do not have 20&nbsp;kHz bandwidth because of the [[Nyquist–Shannon sampling theorem]]. Frequency reproduction is always strictly less than half of the sampling rate, and imperfect filters require a larger margin for error (noise level versus sharpness of filter), so an 8&nbsp;kHz sampling rate limits the maximum frequency to 4&nbsp;kHz, while a 48&nbsp;kHz sampling rate limits an MP3 to a maximum 24&nbsp;kHz sound reproduction. MPEG-2 uses half and MPEG-2.5 only a quarter of MPEG-1 sample rates.

For the general field of human speech reproduction, a bandwidth of 5,512&nbsp;Hz is sufficient to produce excellent results (for voice) using the sampling rate of 11,025 and VBR encoding from 44,100 (standard) WAV file. English speakers average 41–{{nowrap|42 kbit/s}} with -V 9.6 setting but this may vary with the amount of silence recorded or the rate of delivery (wpm). Resampling to 12,000 (6K bandwidth) is selected by the LAME parameter -V 9.4. Likewise -V 9.2 selects a 16,000 sample rate and a resultant 8K lowpass filtering. Older versions of LAME and FFmpeg only support integer arguments for the variable bit rate quality selection parameter. The n.nnn quality parameter (-V) is documented at lame.sourceforge.net but is only supported in LAME with the new style VBR variable bit rate quality selector—not average bit rate (ABR).

A sample rate of 44.1&nbsp;kHz is commonly used for music reproduction because this is also used for [[Red Book (audio CD standard)|CD audio]], the main source used for creating MP3 files. A great variety of bit rates are used on the Internet. A bit rate of {{nowrap|128 kbit/s}} is commonly used,<ref name="Woon-Seng" /> at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet [[bandwidth (computing)|bandwidth]] availability and hard drive sizes have increased, higher bit rates up to {{nowrap|320 kbit/s}} are widespread. Uncompressed audio as stored on an audio-CD has a bit rate of {{nowrap|1,411.2 kbit/s}}, (16 bit/sample&nbsp;× 44,100 samples/second&nbsp;× 2 channels&nbsp;/ 1,000 bits/kilobit), so the bit rates 128, 160, and {{nowrap|192 kbit/s}} represent [[Data compression ratio|compression ratios]] of approximately 11:1, 9:1 and 7:1 respectively.

Non-standard bit rates up to {{nowrap|640 kbit/s}} can be achieved with the [[LAME]] encoder and the free format option, although few MP3 players can play those files. According to the ISO standard, decoders are only required to be able to decode streams up to {{nowrap|320 kbit/s}}.<ref name="Bouvigne" /><ref>{{Cite web|title=lame(1): create mp3 audio files - Linux man page|url=https://linux.die.net/man/1/lame|access-date=2020-08-22|website=linux.die.net|archive-date=22 August 2020|archive-url=https://web.archive.org/web/20200822103430/https://linux.die.net/man/1/lame|url-status=live}}</ref><ref>{{Cite web|title=Linux Manpages Online - man.cx manual pages|url=https://man.cx/lame|access-date=2020-08-22|website=man.cx|archive-date=22 August 2020|archive-url=https://web.archive.org/web/20200822103425/https://man.cx/lame|url-status=live}}</ref> Early MPEG Layer III encoders used what is now called [[constant bit rate]] (CBR). The software was only able to use a uniform bit rate on all frames in an MP3 file. Later more sophisticated MP3 encoders were able to use the bit reservoir to target an [[average bit rate]] selecting the encoding rate for each frame based on the complexity of the sound in that portion of the recording.

A more sophisticated MP3 encoder can produce variable bit rate audio. MPEG audio may use bit rate switching on a per-frame basis, but only layer III decoders must support it.<ref name="MPEG-2.5-2" /><ref name="LAME_GPSYCHO" /><ref name="TwoLAME" /><ref name="MPEG-1 and MPEG-2 BC" /> VBR is used when the goal is to achieve a fixed level of quality. The final file size of a VBR encoding is less predictable than with constant bit rate. Average bit rate is a type of VBR implemented as a compromise between the two: the bit rate is allowed to vary for more consistent quality, but is controlled to remain near an average value chosen by the user, for predictable file sizes. Although an MP3 decoder must support VBR to be standards compliant, historically some decoders have bugs with VBR decoding, particularly before VBR encoders became widespread. The most evolved LAME MP3 encoder supports the generation of VBR, ABR, and even the older CBR MP3 formats.

Layer III audio can also use a "bit reservoir", a partially full frame's ability to hold part of the next frame's audio data, allowing temporary changes in effective bit rate, even in a constant bit rate stream.<ref name="MPEG-2.5-2" /><ref name="LAME_GPSYCHO" /> Internal handling of the bit reservoir increases encoding delay.{{citation needed| date=December 2010}} There is no scale factor band 21 (sfb21) for frequencies above approx 16&nbsp;[[kHz]], forcing the encoder to choose between less accurate representation in band 21 or less efficient storage in all bands below band 21, the latter resulting in wasted bit rate in VBR encoding.<ref name="LAME Y" />