The Adaptive Digital Technologies
conference chip allows the ultimate flexibility. The number of conferences
and number of members per conference is fully programmable.
The conferencing chip adds the active conference input signals together
to form a composite signal. The conference chip makes use of sophisticated
voice activity detection, noise reduction, and dominant speaker selection
algorithms in order to maximize voice quality, even in very large conferences.
Automatic Gain Control (AGC) is used to compensate for channels with different
attenuation characteristics. Before sending the composite signal back to
each conference member, that member’s transmission
is removed from the composite signal to avoid the perception of echo. The composite
signal is also made available for recording purposes. 
In addition to the conferencing feature, the conferencing chip provides tone detection and tone generation capabilities. Tone detection is typically used in a conference server to allow the user to enter a conference ID and password, or to perform other control functions. Tone generation is used to generate alerting signals.
A voice-playback feature allows the host to send pre-recorded speech messages to individual conference members and to broadcast messages to all conference members. The voice record feature allows the host to capture each individual conference member’s TDM input as well as capturing the composite conference output. Voice playback and voice record data transfers occur via the host’s packet interface.
The conference chip uses TDM serial ports, a packet interface, or both, to input and output PCM data to each channel. The TDM serial ports are fully programmable to allow connection to nearly any type of serial bus.
Conferencing Chip ANSI “C” API
The conference chip is controlled by using an ANSI “C” set of API functions that are provided to run on a host processor. The host processor communicates to the conference chip via a host port interface. The API functions configure and control the conference chip as well as return status information to the host application
ADT Conferencing Chip is available on the TMS320™ DSP Family
C5509: 32 channels of conferencing, EC. DTMF Detect, Tone Gen
The conference chip provides all the DSP functionality necessary for a 32 channel conferencing bridge. Up to 10 independent conferences can be set up, and each conference can bridge together 2 to 32 voice channels.
C6416: 496 channels of conferencing, tone detect, tone gen, voice record/playback, Caller ID.
The figure below is a block diagram of the conference chip. The
conference chip is controlled via the DSP’s Host Port Interface (HPI) (or PCI interface on some 64XX devices). In order to simplify the use of the conferencing chip, the conferencing chip’s ANSI “C” API
software resides on the host processor.
This API provides an abstraction
layer that hides the details of the control mechanisms from the host application.
Assigning A Channel To A Conference
Channels are assigned to a conference by the host dynamically. The host issues an API command with parameters indicating the channel ID and conference ID. Channels may not be assigned to a conference if they are already active members of another conference or, if the conference to which they are assigned already has the maximum permissible number of members.
Removing A Channel From A Conference
Channels are removed from a conference by the host dynamically. The host issues an API command with parameters indicating the channel ID and conference ID. The conference is deemed inactive after all conference members have been removed.
Enabling/Disabling A Channel’s Dtmf Tone Detector
The host can enable or disable a channel’s DTMF tone detector dynamically via an API command with parameters indicating the Channel ID and an enabled/disabled flag.
Serial Port Configuration
In order to interface to a wide variety of serial TDM busses, the serial port configuration is programmable. Table 1 below describes the serial port configuration parameters.
| Serial Port Characteristics | ||
| Parameter | Valid Range | Default Value |
| Serial Port 0 | Enabled/Disabled | Enabled |
| Serial Port 1 | Enabled/Disabled | Enabled |
| Serial Port 2 | Enabled/Disabled | Enabled |
| Number of Time Slots | 0..255 | 128 |
| Use Standard Mapping | True / False | True |
| Data Format | u-Law, A-Law, 8 bit Linear, or 16 bit Linear | u-Law |
| Transmit Sync Polarity | Active High or Active Low | Active High |
| Receive Sync Polarity | Active High or Active Low | Active High |
| Transmit Clock Polarity | Rising Edge or Falling Edge | Rising Edge |
| Receive Clock Polarity | Rising Edge or Falling Edge | Falling Edge |
| Transmit Data Delay | 0 to 2 | 1 |
| Receive Data Delay | 0 to 2 | 1 |
| DX Pin Delay | Enable or Disable | Disable |
Table 1: Serial Port Characteristics
Enabling a Conference’s Voice-Record
Each conference’s composite output is always packetized and copied to the host’s packet interface each frame. There is no specific command to enable this action.
Disabling a Channel’s Voice-Record
The host can command a channel to disable it’s voice-recording output dynamically. The host issues an API command with parameters indicating Channel ID. No data is sent to the packet interface if a channel’s voice-record is disabled.
Sending Voice-Playback data to a Channel or Conference
The host can playback voice messages to individual channels, or can send voice messages to all members of a conference simultaneously. Voice playback data is input to each channel via the host’s packet interface. Each channel polls its packet buffer once per frame for new data from the host. If new data is found in the packet buffer, that data will overwrite the transmit TDM data the channel would have written onto the TDM stream.
The host can also playback voice messages to all members of a conference. In this case, the host inputs the voice message to a conference channel’s input packet buffer via the packet interface. Each conference has an associated “phantom” member that allows playback messages to be input to the conference. A “phantom” conference member is not allocated a TDM timeslot resource..
Commanding a Channel to Generate a ToneA channel can be commanded to generate a tone via a host API command. An API tone generation command includes the following parameters:
Channel ID
Tone generation Mode – selections are: Continuous (always on), Burst (single burst of duration on time), and Cadence (follows on, off time)
Tone Level – level in dBm of each frequency (0…-20)
On Time – number of samples of
ON time
Off Time – number of samples of OFF time
Frequency 1 – frequency 1 in Hz
Frequency 2 – frequency 2 in Hz
When a channel is not generating a tone, a silence pattern is sourced onto the channel’s TDM timeslot.
Enabling a Channel’s Voice-Record
The host can command a channel to enable voice-recording dynamically. The host issues an API command with a parameter indicating Channel ID. When a channel’s voice-record is enabled, it packetizes its TDM input data and copies it to the host packet interface each frame.
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Solution BenefitsSystem designers can leverage a proven solution, allowing them to focus efforts on rapid product development.
Includes ANSI "C" API for host controller to ease integration.No DSP programming required.
Scalable to required product specific features to allow use of most cost effective DSP chip.
