Silicon Laboratories SI2493/57/34/15/04, SI2494/39 manual Simplex Speakerphone

AN93

Table 132. TAM PSTN Interrupted Answer – OGM Playback with DTMF Menu Entry (Continued)

Host to Modem Commands / Data	Modem to Host Result	Local Modem Actions
	Codes/Data
AT+VTS=[500,0,100]	OK	Play special administrator tone.
		Using the +VTX command, the voice driver
[Playback Menu Options]		can playback a menu and monitor DTMF
		digit events to perform operations
		remotely.

7.6.4.3. Speakerphone Transition

A SP Button On Event will trigger the transition to Speakerphone system voice mode. Based upon the point at which the event is received, the voice driver will vector to a user answer state to gracefully exit the TAM PSTN mode.

7.6.4.4. Handset Transition

A Handset Raised Event will trigger the transition to Handset system voice mode. Based upon the point at which the event is received, the voice driver will vector to a user answer state to gracefully exit the TAM PSTN mode.

7.7. Speakerphone

7.7.1. Overview

The Si2494/39 speakerphone consists of the following function components as shown in Figure 29 on page 188:

Acoustical echo canceller (AEC)

Acoustical echo suppressor (AES)

AEC double-talk detector (DTD)

Line echo canceller (LEC)

Howling controller (HC)

High-pass filter (HPF)

7.7.2. Simplex Speakerphone

Simplex Speakerphone (SSP) is a special case of speakerphone operation. Like Speakerphone, it allows two-way hands-free voice communication over a telephone line, but SSP allows communication in only one direction at a time. The direction of voice transmission can be done automatically, based on the presence of local and remote speech at the modem, or manually using a push-to-talk function. Simplex and full-duplex speakerphone modes are mutually exclusive. SSP supports the Ademco Contact ID and SIA security protocols. It also supports DTMF generation and detection. DTMF detection is enabled only during +VTR. Commonality in control registers between SSP and full duplex facilitates programming across platforms. SSP supports Type II Caller ID, but Caller ID is disabled during +VRX. A DLE<R> event report is sent to the DTE when a SAS Call Waiting tone is detected. If a CAS tone is detected, the modem mutes the Si3000 speaker while Caller ID Type II data are being captured. It then reports Caller ID info to the DTE as a DLE-shielded complex event. This happens automatically, without the need for an AT+VCIDR? command.

1.The microphone sample is taken after U19D; the speaker sample is taken after U19C. However, the scale factors are set to 1.0 internally, so U19C and U19D will not affect the SSP.

2.To mute the local talker, set U199 [1] to 1 and MUTE_DAA_TO_MIXER=1. To mute the remote talker, set U199 [3] to 1 and MUTE_MIXER_TO_DAA to 1.

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