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