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Property:Speex
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Format
| Format: | Speex | Wiki | Freetz | IPPF | whmf | AVM | Web |
| Short for: | Speex Audio Codec | ||||||
| Location: | Lexicon >> File-Formats | ||||||
| Weblinks: | Wikipedia.int - Wikipedia.de - RFC-5574 | ||||||
| Description: | Lossy audio Codec for human speech | ||||||
Goto: FRITZ!OS - SMW-Browser
Details
Speex is a free audio Codec designed for lossy compression of human speech.
It can be stored in OGG or FLV Containers or streamed using the RTP protocol, ie. for VoIP.
The typical filename extension is .spx, the MIME types are audio/x-speex or audio/speex.
Speex uses CELP encoding and supports these modes:
Speex-8- narrowband 8 kHzSpeex-16- wideband 16 kHzSpeex-32- ultra wideband 32 kHz
Modes can be mixed in a single stream. VBR (Variable Bit-Rate) is supported with bitrates from 2.15 kbit/s to 44 kbit/s.
Speex supports VAD (Voice Activity Detection) if used with VBR.
FRITZ!OS
Software Codecs in FRITZ!OS are libspeex.so (unused) and libmscodex.so which is used by librtpstream.so, libsiplib.so,
libtam.so, the older lib_hansenet_voice_assi.so, as well as voipd, tam_play, ffmpegconv and the new tts_conv.
In kernel there's Speex support in krtp.ko and pcmlink.ko.
VoIP
VoIP uses the RTP protocol for the audio stream which is performed by voipd using librtpstream.so and optionally krtp.ko.
One if the supported Codecs is Speex, provided by the software Codec in libmscodex.so or the hardware Codec in pcmlink.ko.
In the Speech-Quality log voipd_call_stat which is made human readable in libboxlib.so AVM displays Speex (Speex-8) orSpeex-HD (Speex-16). Speex-32 is not supported in FRITZ!OS.
pcmlink.ko TDM support names Speex8, Speex16, SpxNB3, SpxNB4, SpxNB5.
TAM
The TAM (Telephone Ansering Machine) is implemented in the telefon plugin libtam.so, which uses libmscodex.so.
The greeting message and all recordings use the Speex format.
The fvp.*, rvp.* and syl.* Audio-Files used for offline Speech-Synthesis to report the Timestamp of a TAM recording
also use the Speex format. This is performed by tam_play which also uses libmscodex.so.
tam_play is called by libtam.so for replay by telefon and by libwebusb.so for replay in the Webinterface.
Speechd
Starting with fw 7.39 AVM added an online Speech-Synthesis service to improve the speech quality.
It is maintained by the avmspeechd daemon serving the client api libavmspeech_client.so, which is used by dect_manager
and the shell tool tts_conv, which is used by the telefon plugin libtam.so to improve the speech quality of the TAM replay.
One of the Codecs supported by avmspeechd is Speex.
Hansenet
All Alice IAD models use the voice assistant lib_hansenet_voice_assi.so, which asks for a PIN to be dialled during initial setup.
This PIN is then used as one of the credentials to initiate a TR-069 session doing the setup.
The Audio-Files used for the registration and setup dialog use the Speex format:
- 01_reqpin.speex
- 02_pindeny.speex
- 03_wait.speex
- 04_runauth.speex
- 05_authok.speex
- 06_authdeny.speex
- 07_authnoreply.speex
- 08_checkdslcable.speex
- 09_checkdsl.speex
