From 635b5f2f76ed1950e978c373c8a2469ef0637354 Mon Sep 17 00:00:00 2001 From: Gregory Maxwell Date: Tue, 21 Aug 2012 14:24:38 -0400 Subject: Revise README, update AUTHORS emails. --- README | 68 ++++++++++++++++++++++++++++++++++++++++++------------------------ 1 file changed, 43 insertions(+), 25 deletions(-) (limited to 'README') diff --git a/README b/README index 3d4d94b3..fe0bc429 100644 --- a/README +++ b/README @@ -1,3 +1,5 @@ +== Opus audio codec == + Opus is a codec for interactive speech and audio transmission over the Internet. Opus can handle a wide range of interactive audio applications, including @@ -5,11 +7,16 @@ Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bit-rate narrowband speech to very high quality stereo music. -The IETF draft covering Opus can be found at: - http://tools.ietf.org/id/draft-ietf-codec-opus + Opus, when coupled with an appropriate container format, is also suitable +for non-realtime stored-file applications such as music distribution, game +soundtracks, portable music players, jukeboxes, and other applications that +have historically used high latency formats such as MP3, AAC, or Vorbis. + + Opus is specified by IETF RFC 6716: + http://tools.ietf.org/html/rfc6716 -Opus is subject to the royalty-free patent and copyright licenses specified -in the file COPYING. + The Opus format and this implementation of it are subject to the royalty- +free patent and copyright licenses specified in the file COPYING. This package implements a shared library for encoding and decoding raw Opus bitstreams. Raw Opus bitstreams should be used over RTP according to @@ -26,10 +33,14 @@ described at: An opus-tools package is available which provides encoding and decoding of Ogg encapsulated Opus files and includes a number of useful features. + Opus-tools can be found at: - http://git.xiph.org/?p=users/greg/opus-tools.git + https://git.xiph.org/?p=opus-tools.git +or on the main Opus website: + http://opus-codec.org/ == Compiling libopus == + To build from a distribution tarball, you only need to do the following: % ./configure @@ -42,14 +53,14 @@ To build from the git repository, the following steps are necessary: % git clone git://git.opus-codec.org/opus.git % cd opus -1) Compiling +2) Compiling the source % ./autogen.sh % ./configure % make -Once you have compiled the codec, there will be a opus_demo executable in -the top directory. +Once you have compiled the codec, there will be a opus_demo executable +in the top directory. Usage: opus_demo [-e] [options] @@ -58,27 +69,32 @@ Usage: opus_demo [-e] mode: voip | audio | restricted-lowdelay options: --e : only runs the encoder (output the bit-stream) --d : only runs the decoder (reads the bit-stream as input) --cbr : enable constant bitrate; default: variable bitrate --cvbr : enable constrained variable bitrate; default: unconstrained --bandwidth : audio bandwidth (from narrowband to fullband); - default: sampling rate --framesize <2.5|5|10|20|40|60> : frame size in ms; default: 20 --max_payload : maximum payload size in bytes, default: 1024 --complexity : complexity, 0 (lowest) ... 10 (highest); default: 10 --inbandfec : enable SILK inband FEC --forcemono : force mono encoding, even for stereo input --dtx : enable SILK DTX --loss : simulate packet loss, in percent (0-100); default: 0 - -input and output are little-endian signed 16-bit PCM files or opus bitstreams -with simple opus_demo proprietary framing. + -e : only runs the encoder (output the bit-stream) + -d : only runs the decoder (reads the bit-stream as input) + -cbr : enable constant bitrate; default: variable bitrate + -cvbr : enable constrained variable bitrate; default: + unconstrained + -bandwidth + : audio bandwidth (from narrowband to fullband); + default: sampling rate + -framesize <2.5|5|10|20|40|60> + : frame size in ms; default: 20 + -max_payload + : maximum payload size in bytes, default: 1024 + -complexity + : complexity, 0 (lowest) ... 10 (highest); default: 10 + -inbandfec : enable SILK inband FEC + -forcemono : force mono encoding, even for stereo input + -dtx : enable SILK DTX + -loss : simulate packet loss, in percent (0-100); default: 0 + +input and output are little-endian signed 16-bit PCM files or opus +bitstreams with simple opus_demo proprietary framing. == Testing == This package includes a collection of automated unit and system tests -which should be run after compiling the package especially the first +which SHOULD be run after compiling the package especially the first time it is run on a new platform. To run the integrated tests: @@ -89,6 +105,8 @@ included in this package for size reasons but can be obtained from: http://opus-codec.org/testvectors/opus_testvectors-draft11.tar.gz To run compare the code to these test vectors: + +% curl -O http://opus-codec.org/testvectors/opus_testvectors-draft11.tar.gz % tar -zxf opus_testvectors-draft11.tar.gz % ./tests/run_vectors.sh ./ opus_testvectors 48000 -- cgit v1.2.1