You're confusing GSM (Global Systems Mobile) with the gsm codec in Asterisk. They have different meanings. The codecs used by your mobile phone are not the same as the gsm codec in Asterisk.

Shachar Shemesh wrote:
Ian Stirling wrote:
There are 2 D/A, 2 A/D flexibly routed, and one D/A that is dedicated
to the earpiece.

About the only limitation is that you can't do things that would
require more IO sources than are available.
For example, playing stereo MP3, and acting as voicemail/answerphone
may not be possible.
(It'd have to drop to mono).
Lost you there. You seem to suggest the following route for recording
voice calls:
1. Call arrives compressed with a GSM codec
2. Phone decompresses codec
3. Phone moves uncompressed stream through D2A
4. Phone further moves stream through A2D
5. Phone compresses the resulting stream
6. Phone saves compressed stream, presumably to the flash

Why not just do:
1. Call arrives compressed with a GSM codec
2. Phone saves compressed stream to flash

I really don't see why the A/D infrastructure needs to be involved in
voice recording at all. In fact, it seems that it should be easier for
the phone to save the call than to play it to the speaker.

Shachar

P.S.
Asterisk, for example, saves most of its recordings (pick up greeting,
extension selection, voice mail greeting etc.) saved while compressed
with GSM codec. As far as I understand things, if OpenMoko did that,
playing a recording would involve getting it off the flash and dump it
into the GSM line. Extremely light on CPU, and thus unintrusive.

Sh.

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Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
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