What is Opus?
Opus is a format for compressing audio to efficiently transmit it across networks. Opus allows for music-grade high quality audio at low data rates while not delaying the signal much. Opus is distinguished from most formats for high quality audio (AAC, Vorbis, MP3) by having low delay and it is distinguished from most low delay formats (G.711, GSM, Speex) by supporting high audio quality. The Opus format itself and the reference implementation are available under liberal royalty-free licenses, making it easy to adopt, compatible with free software, and suitable for for usage as part of the basic infrastructure of the Internet.
Who created Opus?
Opus was created by combining Xiph.Org's CELT development codec and Skype's SILK codec as part of a cooperative effort in the IETF codec working group. Opus has been in development since early 2007 and will be published as an IETF proposed standard RFC 6716.
Why make Opus free?
On the Internet, protocol and codec standards are part of the common infrastructure everyone builds upon. Most of the value of a high quality standard is the innovation and interoperation provided by the systems built on top of it. When a few parties have monopoly rights to monetize a standard, that infrastructure stops being so common—everyone else has more reason to use their own solution instead, increasing cost and reducing efficiency. Imagine a road system where each type of car could only drive on its own manufacturer's pavement. We all benefit from living in a world where all the roads are connected. This is why Opus, unlike many codecs, is free.
Is the SILK part of Opus compatible with the SILK implementation shipped in Skype?
No. The SILK codec, as submitted by Skype to the IETF, was heavily modified as part of its integration within Opus. The modifications are significant enough that it is not possible to just write a "translator" and even sharing code between Opus and the "old SILK" would be highly non-trivial.
How do I use Opus / What programs use Opus?
What are the licensing requirements?
The reference Opus source code is released under a three-clause BSD license, which is a very permissive Open Source license. Commercial use and distribution (including in proprietary software) is permitted, provided that some basic conditions specified in the license are met.
Opus is also covered by some patents, for which royalty-free usage rights are granted, under conditions that the authors believe are compatible with most (all?) open source licenses, including the GPL (v2 and v3).
How does the quality of Opus compare to other codecs?
See the Opus comparison page for more on this.
On what platforms does Opus run?
The Opus code base is written in C89 and should run on the vast majority of recent (and not so recent) CPUs. Opus has been tested and is known to run on the following (non-exhaustive list) platforms: x86, x86-64, ARM, Itanium, Blackfin, SPARC.
Is there a fixed-point implementation?
Yes, the fixed-point and floating-point decoder and encoder implementations are part of the same code base. The code defaults to float, so you need to configure with --enable-fixed-point (or defining FIXED_POINT if not using the configure script) to build the code for fixed-point.
Why not keep the SILK and CELT codecs separate?
Opus is more than just two independent codecs with a switch. It can not only run in "SILK mode" and in "CELT mode", but also in "hybrid mode" where speech frequencies up to 8 kHz are encoded with SILK while those above 8 kHz are encoded with CELT. This is what allows Opus to have such high speech quality around 32 kb/s. Another advantage of the integration is the ability to switch mode seamlessly. It is possible to change between all the modes without any "glitch" and without any out-of-band signalling.
Now that Opus is standardized, does it mean the development will stop?
Unlike most ITU-T codecs, Opus is only defined in terms of its decoder. The encoder can keep evolving as long as the bits it produces can be decoded by the reference decoder. This is what made it possible for MP3 encoders to improve far beyond the original l3enc and the dist10 reference implementation. Although it is unlikely that Opus encoders will see such spectacular evolution, we certainly hope that future encoders will become much better than the reference encoder. In fact, there is already an experimental branch that significantly improve on the reference encoder's quality.
Opus for Software developers
What is difference between supporting Opus and supporting Speex/G.711/MP3?
Opus has variable frame durations which can change on the fly, so an Opus decoder needs to be ready to accept packets with durations that are any multiple of 2.5ms up to a maximum of 120ms.
The opus encoder and decoder do not need to have matched sampling rates, bandwidths, or channel counts. Its recommended to always just decode at the highest rate the hardware supports (e.g. 48kHz stereo) so the user gets the full quality of whatever the far end is sending.
My application doesn't work. Can anyone help me?
It's possible to get help, but before doing so, there are a few basic things to try:
- Implement the application with uncompressed audio instead of Opus. If it still doesn't work, then the problem isn't related to Opus
- Read the documentation
- Read the opus_demo.c source code to see how to use the encoder and decoder
If it still doesn't work, the best option is to ask on the mailing list
How do I report a bug?
If you think you have found a bug in Opus (and not in your application), please file a bug report. Please include a way for us to reproduce the problem. The best way to do this is to provide an input file, along with the opusenc/opusdec/opus_demo command line that causes the bug to occur. If the bug cannot be triggered by the command line tools, please provide a simple patch or C file that can help reproduce it. Please also provide any other relevant information, such as OS, CPU, build options, ... Also, don't hesistate to also contact us on the mailing list or on IRC.
Forward Error correction (FEC) doesn't appear to do anything! HELP!
The inband FEC feature of Opus helps reduce the harm of packet loss by encoding some information about the prior packet.
In order to make use of inband FEC the decoder must delay its output by at least one frame so that it can call the decoder with the decode_fec argument on the next frame in order to reconstruct the missed frame. This works best if it's integrated with a jitter buffer.
FEC is only used by the encoder under certain conditions: the feature must be enabled via the OPUS_SET_INBAND_FEC CTL, the encoder must be told to expect loss via the OPUS_SET_PACKET_LOSS_PERC CTL, and the codec must be operated in any of the linear prediction or Hybrid modes. Frame durations of <10ms and very high bitrates will use the MDCT modes, where FEC is not available.
Even when FEC is not used, telling the encoder about the level of loss will help it make more intelligent decisions. By default the implementation assumes there is no loss.
What is Opus Custom?
Opus Custom is an optional part of the Opus standard that allows for sampling rates other than 8, 12, 16, 24, or 48 kHz and frame sizes other than multiples of 2.5 ms. Opus Custom requires additional out-of-band signalling that Opus does not normally require and disables many of Opus' coding modes. Also, because it is an optional part of the specification, using Opus Custom may lead to compatibility problems. For these reasons, its use is discouraged outside of very specific applications, e.g.:
- ultra low delay applications where synchronization with the soundcard buffer is important.
- low-power embedded applications where compatibility with others is not important.
For almost all other types of applications, Opus Custom should not be used.
How do I use 44.1 kHz or some other sampling rate not directly supported by Opus?
Tools which read or write Opus should inter-work with other sampling rate by transparently performing sample rate conversion behind the scenes. It's generally preferable to run the output at 48kHz even when you know the original input was 44.1kHz because many inexpensive audio interfaces have poor quality output for 44.1k.
In particular, software developers should not use Opus Custom for 44.1 kHz support, except in the very specific circumstances outlined above.
The opus-tools package source code contains a small, high quality, high performance, BSD licensed resampler which can be used where resampling is required.
I can't use malloc or much stack on my embedded platform how do I make Opus work?
A normal build of libopus only uses malloc/free in the _create() and _destroy() calls, so Opus is safe for realtime use so long as the codec state is pre-created.
In order to build Opus without any reference to malloc/free at all use init() calls rather than the create() calls in your application and compile with CFLAGS="-DOVERRIDE_OPUS_ALLOC -DOVERRIDE_OPUS_FREE -D'opus_alloc(x)=NULL' -D'opus_free(x)=NULL' " you will get a build which does not use malloc/free.
If libopus is built with -DNONTHREADSAFE_PSEUDOSTACK (instead of VAR_ARRAYS, or USE_ALLOCA) it will use a user provided block of heap instead of stack for many things resulting in much lower the stack usage. However this makes the resulting library non-threadsafe and is not recommend on anything except limited embedded platforms.
Which implementation should I use?
While the implementation in the latest IETF draft (and eventually RFC) of Opus is what defines the standard, it is likely not the best and most up-to-date implementation. The Opus website was set up for the purpose of continually improving the implementation— in terms of speed, encoding quality, device compatibility, etc— while still conforming to the standard. All Opus implementations are compatible by definition.