739351d476
Change log:95336cb92b..191d55580eFull diff:95336cb92b..191d55580eRoll chromium third_party 4e16929f46..3a8f2a9e1e Change log:4e16929f46..3a8f2a9e1eChanged dependencies: * src/tools:c44a3f5eca..f524a53b81DEPS diff:95336cb92b..191d55580e/DEPS No update to Clang. TBR=titovartem@google.com, BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: Ic9c4a62b050383646e9fcf5cc07a5653c14ac06e Reviewed-on: https://webrtc-review.googlesource.com/76120 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23205}
55 lines
2.1 KiB
Plaintext
55 lines
2.1 KiB
Plaintext
To build this source code, simply type:
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% make
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If this does not work, or if you want to change the default configuration
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(e.g., to compile for a fixed-point architecture), simply edit the options
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in the Makefile.
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An up-to-date implementation conforming to this standard is available in a
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Git repository at https://git.xiph.org/opus.git or on a website at:
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https://opus-codec.org/
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However, although that implementation is expected to remain conformant
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with the standard, it is the code in this RFC that shall remain normative.
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To build from the git repository instead of using this RFC, follow these
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steps:
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1) Clone the repository (latest implementation of this standard at the time
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of publication)
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% git clone https://git.xiph.org/opus.git
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% cd opus
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2) Compile
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% ./autogen.sh
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% ./configure
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% make
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Once you have compiled the codec, there will be a opus_demo executable in
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the top directory.
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Usage: opus_demo [-e] <application> <sampling rate (Hz)> <channels (1/2)>
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<bits per second> [options] <input> <output>
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opus_demo -d <sampling rate (Hz)> <channels (1/2)> [options]
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<input> <output>
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mode: voip | audio | restricted-lowdelay
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options:
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-e : only runs the encoder (output the bit-stream)
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-d : only runs the decoder (reads the bit-stream as input)
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-cbr : enable constant bitrate; default: variable bitrate
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-cvbr : enable constrained variable bitrate; default: unconstrained
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-bandwidth <NB|MB|WB|SWB|FB> : audio bandwidth (from narrowband to fullband);
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default: sampling rate
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-framesize <2.5|5|10|20|40|60> : frame size in ms; default: 20
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-max_payload <bytes> : maximum payload size in bytes, default: 1024
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-complexity <comp> : complexity, 0 (lowest) ... 10 (highest); default: 10
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-inbandfec : enable SILK inband FEC
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-forcemono : force mono encoding, even for stereo input
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-dtx : enable SILK DTX
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-loss <perc> : simulate packet loss, in percent (0-100); default: 0
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input and output are little endian signed 16-bit PCM files or opus bitstreams
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with simple opus_demo proprietary framing.
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