Asterisk - The Open Source Telephony Project GIT-master-f36a736
|
Files | |
file | codec_a_mu.c |
codec_a_mu.c - translate between alaw and ulaw directly | |
file | codec_adpcm.c |
codec_adpcm.c - translate between signed linear and Dialogic ADPCM | |
file | codec_alaw.c |
codec_alaw.c - translate between signed linear and alaw | |
file | codec_codec2.c |
Translate between signed linear and Codec 2. | |
file | codec_dahdi.c |
Translate between various formats natively through DAHDI transcoding. | |
file | codec_g722.c |
codec_g722.c - translate between signed linear and ITU G.722-64kbps | |
file | codec_g726.c |
codec_g726.c - translate between signed linear and ITU G.726-32kbps (both RFC3551 and AAL2 codeword packing) | |
file | codec_gsm.c |
Translate between signed linear and Global System for Mobile Communications (GSM) | |
file | codec_ilbc.c |
Translate between signed linear and Internet Low Bitrate Codec. | |
file | codec_lpc10.c |
Translate between signed linear and LPC10 (Linear Predictor Code) | |
file | codec_resample.c |
Resample slinear audio. | |
file | codec_speex.c |
Translate between signed linear and Speex (Open Codec) | |
file | codec_ulaw.c |
codec_ulaw.c - translate between signed linear and ulaw | |
Codecs are referenced in configuration files by name