Latest revision as of 05:15, 14 December 2018

Recommended Bitrates

Depending on the kind of audio you want to encode with Opus, you may want to use different bitrate (quality) settings.

The settings in the table below are meant to start you off with a decent tradeoff between good quality and small file size (or bitrate usage, if you're streaming).

You should test the suggested bitrate by actually listening to your encoded audio and then:

tweaking the bitrate down if you think the quality is good, but the file size (or bitrate) is too big,
tweaking the bitrate up if you think the quality is bad, and you can afford having bigger files (or a larger streaming bitrate).

Use Case	Channels	Bitrate (Kb/s)	Notes
Low bandwidth HF/VHF digital radio	1 (mono)	Use Codec 2	Opus only supports bitrates down to 6 Kb/s. Codec 2 handles ultra low bitrate speech at 0.7 - 3.2 Kb/s.
VoIP	1	10 - 24	10 Kb/s will deliver narrowband most of the time, 24 Kb/s should give fullband. More details in the relevant table further down this page.
Audiobooks / Podcasts	1	24	Bitrates from here on up tend to deliver fullband audio.
Audiobooks / Podcasts	2 (stereo)	32
Music Streaming / Radio	2	64 - 96	Opus has better quality than MP3, AAC and Vorbis at these rates. (listening test results: 64 Kb/s, 96 Kb/s)
Music Storage	2	96 - 128	Opus at 128 KB/s (VBR) is pretty much transparent.
	6 (5.1 surround)	128 - 256	For surround sound, Opus uses surround-sound bitrate allocation.
	8 (7.1 surround)	256 - 450
Music Archiving	1 - 8	Use FLAC	If you are archiving audio, use a lossless audio format to prevent generation loss.

Technical Details

For the more technical Opus users, here are some details to help you fine-tune your decision on which bitrate best fits your needs.

Mono or Stereo

Opus tends to start downmixing stereo inputs to mono from roughly 19 Kb/s and lower. You can check the details in the opus_encoder.c source file.

You can force downmixing at any bitrate by using the following command-line parameters:

--downmix-mono - downmixes all input channels to mono

--downmix-stereo - downmixes all input channels to stereo (if there are more than 2 input channels, e.g. surround sound)

Bandwidth Transition Thresholds

The following table shows rough bitrates that you might want to use to encode audio that has limited frequency bandwidths. This could be useful if your audio has already been bandpassed, or should go through a bandpass filter (e.g. VoIP speech).

Bandpass Range (Hz)	Rough Bitrate Required (Kb/s)
	Mono		Stereo
	Voice	Music	Voice	Music
NarrowBand (3 - 4000)	12	15	?	?
MediumBand (3 - 6000)	15	18-22	?	?
WideBand (3 - 8000)	16-20	22-28	?	?
SuperWideBand (3-12000)	24-28	28-32	?	?
FullBand (3-20000)	28-40	32-64	32-64	64-128

The details of Opus' bandpass thresholds can be found in the opus_encoder.c source file.

The HydrogenAudio wiki also has some great information on Opus and its usage.

Framesize Tweaking

Opus can encode frames of 2.5, 5, 10, 20, 40, or 60 ms. It can also combine multiple frames into packets of up to 120 ms.

Opus uses a 20 ms frame size by default, as it gives a decent mix of low latency and good quality.

For real-time applications, sending fewer packets per second reduces the overall bitrate, since it reduces the overhead from IP, UDP, and RTP headers. However, it increases latency and sensitivity to packet losses, as losing one packet constitutes a loss of a bigger chunk of audio. Unless operating at very low bitrates over RTP, there is no reason to use frame sizes above 20 ms, as those will have slightly lower quality for music encoding.

For these reasons, the default 20 ms frames are a good choice for most applications.

Trading Coding Efficiency with CPU Time

The Opus encoder uses its maximum algorithmic complexity setting of 10 by default. This means that it does not hesitate to use CPU to give you the best quality encoding at a given bitrate.

If the CPU usage is too high for the system you are using Opus on, you can try a lower complexity setting. The allowed values span from 10 (highest CPU usage and quality) down to 0 (lowest CPU usage and quality).

@@ Line 1: / Line 1: @@
-Depending on what kinds of sounds you want to encode with Opus, you should use different bitrate (quality) settings.
+= Recommended Bitrates =
+Depending on the kind of audio you want to encode with Opus, you may want to use different bitrate (quality) settings.
-The settings in the table below are meant to '''start you off''' with a decent tradeoff between '''good quality''' and '''small filesize''' (or '''bitrate usage''', if you're streaming).
+The settings in the table below are meant to '''start you off''' with a decent tradeoff between '''good quality''' and '''small file size''' (or '''bitrate usage''', if you're streaming).
-You should test the suggested bitrate by actually '''listening''' to your encoded audio and then tweaking the bitrate:
+You should test the suggested bitrate by actually '''listening''' to your encoded audio and then:
-* '''down''' if you think the quality is good, but the filesize (or bitrate) is too big
+* tweaking the bitrate '''down''' if you think the quality is good, but the file size (or bitrate) is too big,
-* '''up''' if you think the quality is bad, and you can afford having bigger files (or bitrate)
+* tweaking the bitrate '''up''' if you think the quality is bad, and you can afford having bigger files (or a larger streaming bitrate).
 {| class="wikitable" style="text-align:center"
@@ Line 14: / Line 15: @@
 !Notes
 |-
-|Ham radio
+|Low bandwidth HF/VHF digital radio
 |1 (mono)
-|use '''[http://www.rowetel.com/blog/?page_id=452 Codec&nbsp;2]'''
+|Use '''[http://www.rowetel.com/?page_id=452 Codec&nbsp;2]'''
-|Opus only supports bitrates down to 6&nbsp;Kb/s. Codec 2 handles speech from 0.7 to 3.2&nbsp;Kb/s.
+|Opus only supports bitrates '''down to 6&nbsp;Kb/s'''.<br>
+Codec 2 handles ultra low bitrate speech at '''0.7&nbsp;-&nbsp;3.2&nbsp;Kb/s'''.
 |-
 |VoIP
 |1
-|10-24
+|10&nbsp;-&nbsp;24
-|10&nbsp;Kb/s will deliver narrowband most of the time, 24&nbsp;Kb/s should give fullband.<br>More details in '''[[Opus_Recommended_Settings#Bandwidth_Transition_Thresholds|the relevant table]]''' further down this page.
+|10&nbsp;Kb/s will deliver narrowband most of the time, 24&nbsp;Kb/s should give fullband.<br>
+More details in '''[[Opus_Recommended_Settings#Bandwidth_Transition_Thresholds|the relevant table]]''' further down this page.
 |-
 |rowspan="2"|Audiobooks / Podcasts
 |1
 |24
-|bitrates from here on up tend to deliver fullband audio.
+|Bitrates from here on up tend to deliver fullband audio.
 |-
 |2 (stereo)
@@ Line 35: / Line 38: @@
 |Music Streaming / Radio
 |2
-|64-96
+|64&nbsp;-&nbsp;96
-|Opus has better quality than MP3, AAC and [[Vorbis]] at these rates.<br>(test results '''[http://listening-tests.hydrogenaud.io/igorc/results.html here]''' and '''[http://listening-test.coresv.net/results.htm here]''')
+|Opus has better quality than MP3, AAC and [[Vorbis]] at these rates.<br>
+(listening test results: '''[http://listening-tests.hydrogenaud.io/igorc/results.html 64&nbsp;Kb/s]''', '''[http://listening-test.coresv.net/results.htm 96&nbsp;Kb/s]''')
 |-
 |rowspan="3"|Music Storage
 |2
-|96-128
+|96&nbsp;-&nbsp;128
-|Opus at 128&nbsp;KB/s (VBR) is pretty much '''[https://en.wikipedia.org/wiki/Transparency_(data_compression) transparent]'''
+|Opus at 128&nbsp;KB/s (VBR) is pretty much '''[https://en.wikipedia.org/wiki/Transparency_(data_compression) transparent]'''.
 |-
 |6 (5.1 surround)
-|128-256
+|128&nbsp;-&nbsp;256
-|rowspan="2"|for surround sound, Opus uses '''[https://xiph.org/~xiphmont/demo/opus/demo3.shtml surround-sound bitrate allocation]'''
+|rowspan="2"|For surround sound, Opus uses '''[https://xiph.org/~xiphmont/demo/opus/demo3.shtml surround-sound bitrate allocation]'''.
 |-
 |8 (7.1 surround)
-|256-450
+|256&nbsp;-&nbsp;450
 |-
 |Music Archiving
-|any
+|1&nbsp;-&nbsp;8
-|use '''[[FLAC]]'''
+|Use '''[[FLAC]]'''
-|if you are archiving audio, use a '''[https://en.wikipedia.org/wiki/Audio_file_format#Lossless_compressed_audio_format lossless audio format]''' to prevent '''[https://en.wikipedia.org/wiki/Generation_loss generation loss]'''
+|If you are archiving audio, use a '''[https://en.wikipedia.org/wiki/Audio_file_format#Lossless_compressed_audio_format lossless audio format]''' to prevent '''[https://en.wikipedia.org/wiki/Generation_loss generation loss]'''.
 |}
@@ Line 60: / Line 64: @@
 == Mono or Stereo ==
-Opus tends to start '''downmixing stereo inputs to mono''' from roughly '''30&nbsp;Kb/s and lower'''.
+Opus tends to start '''downmixing stereo inputs to mono''' from roughly '''19&nbsp;Kb/s and lower'''.
-You can check the details in the '''[https://github.com/xiph/opus/blob/master/src/opus_encoder.c#L139 opus_encoder.c]''' source file.
+You can check the details in the '''[https://github.com/xiph/opus/blob/master/src/opus_encoder.c#L149 opus_encoder.c]''' source file.
 You can force downmixing at any bitrate by using the following command-line parameters:
-<code>--downmix-mono</code> - downmixes input channels to mono
+<code>--downmix-mono</code> - downmixes all input channels to mono
-<code>--downmix-stereo</code> - downmixes input channels to stereo (if there are more than 2 e.g. if your input is surround sound)
+<code>--downmix-stereo</code> - downmixes all input channels to stereo (if there are more than 2 input channels, e.g. surround sound)
 == Bandwidth Transition Thresholds ==
-The following table shows where Opus decides to switch between its different '''[https://tools.ietf.org/html/rfc6716#section-2 audio frequency bandwidths]'''. This could be useful if you are encoding audio that has already been bandpassed, or should go through a bandpass filter (e.g. VoIP speech).
+The following table shows rough bitrates that you might want to use to encode audio that has '''[https://tools.ietf.org/html/rfc6716#section-2 limited frequency bandwidths]'''.
+This could be useful if your audio has already been bandpassed, or should go through a bandpass filter (e.g. VoIP speech).
-The table was copied from the '''[https://github.com/xiph/opus/blob/master/src/opus_encoder.c#L112 opus_encoder.c]''' source file.
 {| class="wikitable" style="text-align:center"
 |-
-|rowspan="2"|(bitrates in Kb/s)
+!rowspan="3"|Bandpass Range (Hz)
+!colspan="4"|Rough Bitrate Required (Kb/s)
+|-
 !colspan="2"|Mono
 !colspan="2"|Stereo
 |-
-! Voice
+!Voice
-! Music
+!Music
-! Voice
+!Voice
-! Music
+!Music
 |-
-!<abbr title="NarrowBand (3-4000&nbsp;Hz)">NB</abbr> ↔ <abbr title="MediumBand (3-6000&nbsp;Hz)">MB</abbr>
+|style="text-align:right;"|NarrowBand (3&nbsp;-&nbsp;4000)
-|11 ± 1
+|12
-|12 ± 1
+|15
-|11 ± 1
+|?
-|12 ± 2
+|?
 |-
-!MB ↔ <abbr title="WideBand (3-8000&nbsp;Hz)">WB</abbr>
+|style="text-align:right;"|MediumBand (3&nbsp;-&nbsp;6000)
-|14 ± 1
+|15
-|15 ± 1
+|18-22
-|14 ± 2
+|?
-|18 ± 2
+|?
 |-
-!WB ↔ <abbr title="SuperWideBand (3-12000&nbsp;Hz))">SWB</abbr>
+|style="text-align:right;"|WideBand (3&nbsp;-&nbsp;8000)
-|17 ± 1
+|16-20
-|18 ± 1
+|22-28
-|21 ± 2
+|?
-|21 ± 2
+|?
 |-
-!SWB ↔ <abbr title="FullBand (3-20000&nbsp;Hz)">FB</abbr>
+|style="text-align:right;"|SuperWideBand (3-12000)
-|21 ± 1
+|24-28
-|22 ± 2
+|28-32
-|28 ± 2
+|?
-|30 ± 2
+|?
+|-
+|style="text-align:right;"|FullBand (3-20000)
+|28-40
+|32-64
+|32-64
+|64-128
 |}
+The details of Opus' bandpass thresholds can be found in the '''[https://github.com/xiph/opus/blob/master/src/opus_encoder.c#L121 opus_encoder.c]''' source file.
 The '''[http://wiki.hydrogenaud.io/index.php?title=Opus HydrogenAudio]''' wiki also has some great information on Opus and its usage.
 == Framesize Tweaking ==
+Opus can encode frames of '''2.5''', '''5''', '''10''', '''20''', '''40''', or '''60&nbsp;ms'''.  It can also combine multiple frames into packets of '''up to 120&nbsp;ms'''.
 Opus uses a '''20&nbsp;ms''' frame size '''[https://tools.ietf.org/html/rfc6716#section-2.1.4 by default]''', as it gives a decent mix of low latency and good quality.
+For real-time applications, sending fewer packets per second reduces the overall bitrate, since it reduces the overhead from '''[https://en.wikipedia.org/wiki/IPv6_packet#Fixed_header IP]''', '''[https://en.wikipedia.org/wiki/User_Datagram_Protocol#Packet_structure UDP]''', and '''[https://en.wikipedia.org/wiki/Real-time_Transport_Protocol#Packet_header RTP headers]'''.
+However, it increases latency and sensitivity to packet losses, as losing one packet constitutes a loss of a bigger chunk of audio.
+Unless operating at very low bitrates over RTP, there is no reason to use frame sizes above 20 ms, as those will have slightly lower quality for music encoding.
+For these reasons, the default 20&nbsp;ms frames are a good choice for most applications.
+== Trading Coding Efficiency with CPU Time ==
+The Opus encoder uses its maximum algorithmic '''complexity''' setting of '''10''' '''[https://tools.ietf.org/html/rfc6716#section-2.1.5 by default]'''. This means that it does not hesitate to use CPU to give you the best quality encoding at a given bitrate.
+If the CPU usage is too high for the system you are using Opus on, you can try a lower complexity setting. The allowed values span from '''10''' (highest CPU usage and quality) down to '''0''' (lowest CPU usage and quality).
 [[Category:Opus]]

Opus Recommended Settings: Difference between revisions