89 lines
4.2 KiB
Plaintext
89 lines
4.2 KiB
Plaintext
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CELT is a very low delay audio codec designed for high-quality communications.
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Traditional full-bandwidth codecs such as Vorbis and AAC can offer high
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quality but they require codec delays of hundreds of milliseconds, which
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makes them unsuitable for real-time interactive applications like tele-
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conferencing. Speech targeted codecs, such as Speex or G.722, have lower
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20-40ms delays but their speech focus and limited sampling rates
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restricts their quality, especially for music.
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Additionally, the other mandatory components of a full network audio system—
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audio interfaces, routers, jitter buffers— each add their own delay. For lower
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speed networks the time it takes to serialize a packet onto the network cable
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takes considerable time, and over the long distances the speed of light
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imposes a significant delay.
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In teleconferencing— it is important to keep delay low so that the participants
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can communicate fluidly without talking on top of each other and so that their
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own voices don't return after a round trip as an annoying echo.
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For network music performance— research has show that the total one way delay
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must be kept under 25ms to avoid degrading the musicians performance.
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Since many of the sources of delay in a complete system are outside of the
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user's control (such as the speed of light) it is often only possible to
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reduce the total delay by reducing the codec delay.
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Low delay has traditionally been considered a challenging area in audio codec
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design, because as a codec is forced to work on the smaller chunks of audio
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required for low delay it has access to less redundancy and less perceptual
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information which it can use to reduce the size of the transmitted audio.
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CELT is designed to bridge the gap between "music" and "speech" codecs,
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permitting new very high quality teleconferencing applications, and to go
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further, permitting latencies much lower than speech codecs normally provide
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to enable applications such as remote musical collaboration even over long
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distances.
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In keeping with the Xiph.Org mission— CELT is also designed to accomplish
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this without copyright or patent encumbrance. Only by keeping the formats
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that drive our Internet communication free and unencumbered can we maximize
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innovation, collaboration, and interoperability. Fortunately, CELT is ahead
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of the adoption curve in its target application space, so there should be
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no reason for someone who needs what CELT provides to go with a proprietary
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codec.
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CELT has been tested on x86, x86_64, ARM, and the TI C55x DSPs, and should
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be portable to any platform with a working C compiler and on the order of
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100 MIPS of processing power.
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The code is still in early stage, so it may be broken from time to time, and
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the bit-stream is not frozen yet, so it is different from one version to
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another. Oh, and don't complain if it sets your house on fire.
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Complaints and accolades can be directed to the CELT mailing list:
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http://lists.xiph.org/mailman/listinfo/celt-dev/
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To compile:
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% ./configure
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% make
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For platforms without fast floating point support (such as ARM) use the
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--enable-fixed argument to configure to build a fixed-point version of CELT.
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There are Ogg-based encode/decode tools in tools/. These are quite similar to
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the speexenc/speexdec tools. Use the --help option for details.
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There is also a basic tool for testing the encoder and decoder called
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"testcelt" located in libcelt/:
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% testcelt <rate> <channels> <frame size> <bytes per packet> input.sw output.sw
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where input.sw is a 16-bit (machine endian) audio file sampled at 32000 Hz to
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96000 Hz. The output file is already decompressed.
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For example, for a 44.1 kHz mono stream at ~64kbit/sec and with 256 sample
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frames:
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% testcelt 44100 1 256 46 intput.sw output.sw
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Since 44100/256*46*8 = 63393.74 bits/sec.
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All even frame sizes from 64 to 512 are currently supported, although
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power-of-two sizes are recommended and most CELT development is done
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using a size of 256. The delay imposed by CELT is 1.25x - 1.5x the
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frame duration depending on the frame size and some details of CELT's
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internal operation. For 256 sample frames the delay is 1.5x or 384
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samples, so the total codec delay in the above example is 8.70ms
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(1000/(44100/384)).
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