TOBY-L2 Audio tuning - U-blox · TOBY-L2 series Audio tuning Application Note Abstract This document provides information and procedures to perform the audio tuning in TOBY-L2 series

TOBY-L2 series Audio tuning Application Note

Abstract

This document provides information and procedures to perform the audio tuning in TOBY-L2 series modules. It gives some hints for a correct enclosure design. It describes the procedures for tuning the audio parameters for the whole audio system, the hands-free algorithm (echo cancellation, automatic gain

control, noise reduction) and the external audio management.

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UBX-15015834 - R01

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TOBY-L2 series - Application Note

UBX-15015834 - R01

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Document Information

Title TOBY-L2 series

Subtitle Audio tuning

Document type Application Note

Document number UBX-15015834

Revision, date R01 03-May-2016

Document status Early Production Information

Document status explanation

Objective Specification Document contains target values. Revised and supplementary data will be published later.

Advance Information Document contains data based on early testing. Revised and supplementary data will be published later.

Early Production Information Document contains data from product verification. Revised and supplementary data may be published later.

Production Information Document contains the final product specification.

This document applies to the following products:

Product name

TOBY-L2

u-blox reserves all rights to this document and the information contained herein. Products, names, logos and designs described herein may in whole or in part be subject to intellectual property rights. Reproduction, use, modification or disclosure to third parties of this

document or any part thereof without the express permission of u-blox is strictly prohibited. The information contained herein is provided “as is” and u-blox assumes no liability for the use of the information. No warranty, either

express or implied, is given, including but not limited, with respect to the accuracy, correctness, reliability and fitness for a particular purpose of the information. This document may be revised by u-blox at any time. For most recent documents, please visit

www.u-blox.com. Copyright © 2016, u-blox AG


UBX-15015834 - R01 Early Production Information Contents

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Contents

Contents .............................................................................................................................. 3

1 Introduction .................................................................................................................. 5

2 Hints for LEM design .................................................................................................... 6

3 System gains tuning ..................................................................................................... 7

3.1 Step 1: defining maximum loudness of speaker (RLR) ........................................................................... 7

3.1.1 Procedure ...................................................................................................................................... 8

3.2 Step 2: define the maximum gain in uplink audio path (SLR) ................................................................ 8

3.2.1 Procedure ...................................................................................................................................... 8

3.3 Step 3: verify mic sensitivity .................................................................................................................. 9

3.3.1 Procedure ...................................................................................................................................... 9

3.4 Step 4: final check ................................................................................................................................ 9

4 EC algorithm and filters tuning ................................................................................. 10

4.1 Block diagram and tunable parameters ............................................................................................... 10

4.1.1 Microphone levels (ML) ............................................................................................................... 11

4.1.2 Echo Cancellation (EC) ................................................................................................................ 11

4.1.3 Residual Echo Suppression (RES) .................................................................................................. 13

4.1.4 Noise Suppression (NS) and Comfort Noise (CN) .......................................................................... 15

4.1.5 Low Pass Filter (LPF) ..................................................................................................................... 17

4.1.6 High Pass Filter (HPF) ................................................................................................................... 18

4.1.7 Equalizer (EQ) .............................................................................................................................. 19

4.1.8 Additional Gain Control (AGC) .................................................................................................... 21

4.1.9 Dynamic Range Compression (DRC) ............................................................................................ 23

4.1.10 Sidetone ...................................................................................................................................... 25

4.2 No-duplex configuration ..................................................................................................................... 26

5 Audio profiles management ...................................................................................... 27

5.1 Personal profiles ................................................................................................................................. 27

5.1.1 Managing personal profiles using m-center ................................................................................. 28

5.2 Factory-programmed profile ............................................................................................................... 28

6 Mobile Voice Tuning tool .......................................................................................... 29

7 External codec management ..................................................................................... 31

7.1 Setting ................................................................................................................................................ 32

7.2 External device configuration +UEXTDCONF ....................................................................................... 33

8 EVK suggested parameters ........................................................................................ 35

9 Supported speech codecs .......................................................................................... 36

Appendix .......................................................................................................................... 37


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A List of acronyms ......................................................................................................... 37

Related documents .......................................................................................................... 38

Revision history ................................................................................................................ 38

Contact .............................................................................................................................. 39


UBX-15015834 - R01 Early Production Information Introduction

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1 Introduction

The audio interface is supported by TOBY-L200-02S, TOBY-L210-02S, TOBY-L210-62S, TOBY-L220-02S and TOBY-L280-02S.

This document provides information and procedures to achieve the best audio performance in systems using TOBY-L2 series modules.

The audio performance of a device depends on a multiplicity of factors related to module, external circuitry and enclosure mechanical characteristics, too. Often these factors are in opposition and a good knowledge of them is needed to find out the best compromise that fulfills the specific needs.

The audio tuning starts with a good design of the LEM (Loudspeaker – Enclosure – Microphone) system. It is not the aim of the document to offer an exhaustive guide on this complex task, although it provides some information about the basic requirements.

The second step concerns the tuning of the gains in overall audio paths. This means that the gains related to all audio subsystems (both internal to the module and in external circuitry) must be adjusted in order to obtain clear, undistorted audio and comfortable volume levels.

Finally, it is possible to tune the hands-free algorithm to reduce possible echo.

For a detailed description of TOBY-L2 series module audio interface, see the TOBY-L2 series System Integration Manual[2].

The following symbols are used to highlight important information within the document:

An index finger points out key information pertaining to integration and performance.

A warning symbol indicates actions that could negatively impact or damage the module.


UBX-15015834 - R01 Early Production Information Hints for LEM design

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2 Hints for LEM design Acoustic echo is one of the most critical issues that afflict hands-free phones. It is a disturbance that occurs when the sound emitted by the loudspeaker is captured by microphone and sent back to the originating user (far-end device).

Its prevention starts with an accurate design for the LEM system: loudspeaker and microphone characteristics have to be carefully chosen and the enclosure has to be designed in order to reduce any possible coupling:

1. Direct Acoustic Coupling from outside 2. Direct Acoustic Coupling from inside 3. Mechanical Coupling

Figure 1: Possible causes of echo

Orient the microphone and speaker to reduce the amount of sound emitted by speaker and captured by microphone.

Place the microphone in a cavity (i.e. a compartment in the enclosure). The partitions delimiting the cavity should be designed to insulate acoustically microphone from speaker.. A rubber grommet has to be used to insulate the mic capsule from mechanical vibrations and to close any possible air gap between mic and the inside of enclosure.

Figure 2: Example of mic cavity design

The speaker does not have to be fixed directly to the enclosure, but a foam has to be placed between the sides of the enclosure to reduce vibrations than could propagate to the enclosure and then to the microphone.

13

2

Microphone

Loudspeaker

Enclosure

Outside Inside

Mic holeMic

Rubber

grommet

Enclosure

Foam for pressure

to push mic into

plastic housing


UBX-15015834 - R01 Early Production Information System gains tuning

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3 System gains tuning The echo canceller algorithm is based on the assumption that the LEM is linear and stable; if these conditions are not satisfied the echo cancellation will be ineffective. Therefore it is crucial to check the external components and remove any source of non linearity that includes:

1. variable gains 2. non linear processing, e.g. compressors on mic and loudspeaker amplifiers 3. distortions, e.g. due to saturations.

Figure 3: Typical audio setup of device under test (DUT)

Points 1 and 2 require an appropriate choice of components during the schematic design.

Distortions can be avoided by calibrating all the gains in both uplink and downlink audio paths. This task requires advanced test suites. The 3GPP TS 26131-32 specification defines standard procedure audio tests and characterization. Approximate tuning can be performed by following the simplified procedure described in next sections.

The procedure consists of different steps aiming to determine the gains for all the audio blocks which satisfy both project and HF requirements. Usually these factors are in opposition and a compromise should be accepted between:

speaker loudness

microphone sensitivity

no-distorted echo

3.1 Step 1: defining maximum loudness of speaker (RLR)

Receive Loudness Rating (RLR) is the signal loudness in receiving direction.

The first step in the gain calibration procedure is to define the gains in downlink path in order to achieve the requested speaker loudness on DUT side when a reference signal is played at the far-end of the device. The reference signal has to simulate the typical use case at the far-end side. It should be an artificial voice so that the test can be repeated.

Figure 4: Gains calibration - step 1

Device

Cellular

module

External

circuitry

DUT

Cellular module

Reference

signal

FarEnd

Signal to be

evaluated



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3.1.1 Procedure

1. Set up a call and send a reference signal 2. Set the loudspeaker volume at max value (e.g. AT+CLVL=100)

The speaker volume control by the end user has to only act on +CLVL.

3. Tune the gains in the downlink audio path in order to achieve the requirements for speaker loudness, but avoiding any distortion. Properly distribute all audio gains among all audio blocks in downlink path: o AT+UAPT=8,<profile>,0,1,8,<rate>,<parameterOffset>,<numberOfParameters>,<MLnominalLevel>,0,0,0

o External gains in downlink path

Overall Speaker Gain Max is the sum of Microphone level gain (set by +UAPT) and external gains:

SpkGMAX

= GML Nominal Level

+ Gexternal downlink

3.2 Step 2: define the maximum gain in uplink audio path (SLR)

The Send Loudness Rating (SLR) is the signal loudness in sending direction.

This step aims to define the maximum gain in the uplink audio path (SLR) applicable without having echo distortions.

Figure 5: Gains calibration - step 2

3.2.1 Procedure

1. Use downlink calibrations values defined in step above (max RLR) 2. Disable the HF facilities

o EC: AT+UAPT=6,<profile>,0,0,0,0 o RES: AT+UAPT=6,<profile>,0,0,1,0 o NS: AT+UAPT=6,<profile>,0,0,2,0 o LPF AT+UAPT=6,<profile>,0,0,3,0 o HPF: AT+UAPT=6,<profile>,0,0,4,0 o EQ: AT+UAPT=6,<profile>,0,0,5,0 o AGC: AT+UAPT=6,<profile>,0,0,6,0 o DRC: AT+UAPT=6,<profile>,0,0,7,0 o AT+UAPT=0

3. Set up a call, send a reference signal as in step 1 and evaluate the echo at far-end 4. Increase the gains in uplink audio path in order to achieve maximum level without having any distortion in

the echo at far-end. 5. Distribute properly the gains all audio among all audio blocks in uplink path.

Overall Microphone Gain is the sum of Microphone level gain (set by +UAPT) and external gains:

MicGMAX

= GUAPT

+ Gexternal uplink

Cellular module

DUT

Reference

signal

FarEnd

Signal to be

evaluated



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3.3 Step 3: verify mic sensitivity

After downlink and uplink gains have been defined, evaluate the microphone sensitivity to verify the requirements for capturing signals at both short and long distances. The distance boundaries must be specified in the project requirements.

The reference signal must simulate the typical use case at near end side. It should be an artificial voice so that the test can be repeated.

Figure 6: Microphone sensitivity at long distance

Figure 7: Microphone sensitivity at short distance

3.3.1 Procedure

1. Set up a call 2. Verify the signal received at the far end when the near-end user is placed far from the DUT 3. Adjust the gains in the uplink audio path aiming to have both enough level at far-end and no distortions.

The overall mic gain applied when the reference signal is placed at maximum distance MicGNE low

has to be in any case lower than MicG

MAX , defined in previous steps, otherwise distortions would be introduced

MicGNE low

< MicGMAX

The same test should be done by placing the reference signal at a minimum distance from the device. Also in this case, the signal at far-end side must not be distorted.

3.4 Step 4: final check

The fundamental constraint is

MicGMAX

> MicGNE low

If it is not satisfied, repeat all the steps (even more and more times) to find out the best compromise among speaker loudness, microphone sensitivity and no-distorted echo.

If a satisfying compromise cannot be found, it will be needed to modify the hardware and/or the mechanical design.

Cellular module

DUT FarEnd

Reference

signal

far

Cellular module

DUT FarEnd

Reference

signal

near


UBX-15015834 - R01 Early Production Information EC algorithm and filters tuning

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4 EC algorithm and filters tuning The EC functionality implemented in TOBY-L2 modules allows the echo removal.

Next sections describe the effects of the various parameters and settings and present a step-by-step procedure for choosing parameters to remove the echo heard on the far-end side.

4.1 Block diagram and tunable parameters

The signal processing blocks perform digital processing of the audio signals in both the Uplink (Tx) and Downlink (Rx) branches of the voice paths. The uplink modules process the digital samples coming out of the codec before their encoding by the voice encoder. The downlink modules process the voice decoder output samples before their transmission into the codec.

Filtering is performed by the HPF, LPF, and equalizers in both the uplink and downlink paths.

Echo and noise cancellation are handled in several uplink modules: Echo Cancellation (EC) and Noise and Residual Echo Suppression (NRES). The noise cancellation is also performed in the Rx path by the Rx-NS module.

Level adjustment and dynamic range compression are handled by the AGC and DRC (both uplink and downlink).

The voice enhancement modules are configured by +UAPT command; it has different sets of parameters assigned to different features.

The parameters are divided into the following categories:

Customer-tunable parameters: these parameters tune the major functions and are useful for enhance audio performance in customer applications. They are listed and described in this document.

Expertly set parameters: these parameters are intended for enhanced use. Although they are mentioned in this document, they are not described here and they should not be changed.

The management and parameters tuning of audio paths can be configured by means of the following AT commands:

+UAPT (Audio Parameter Tuning): audio parameters management for all the signal processing blocks:

o High Pass Filter (HPF)

o Echo Cancellation (EC)

o Noise Suppression (NS)

o Residual Echo Suppression (RES)

o Equalizer (EQ)

o Additional Gain Control (AGC)

o Dynamic Range Compression (DRC)

o Low Pass Filter (LPF)

+USPM (Set Path Mode)

+UVGC (Volume Gain Control): gains of external Maxim MAX9860 audio codec (connected via I2C interface)



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EC[0]

HFP[4]

To Radio TX

From Radio RX

RES[1]

NS[2]

DRC[7]

LPF[3]

EQ[5]

AGC[6]

I2S_RXD

I2S_TXD

Volume & Mute

CNG[2]

AGC[6]

EQ[5]

LPF[3]

DRC[7]

NS[2]

HPF[4]

Volume & Mute

PCM Player

Tone Generator

Legend:

EC = Echo CancellationRES = Residual Echo Suppression

NS = Noise SuppressionLPF = Low Pass FilterHPF = High Pass Filter

TOBY-L2

TxBe

RxAe

TxAe

RxBe

<direction> = 1

<direction> = 0

EQU = EqualizerAGC = Additional Gain ControlDRC = Dynamic Range CompressionCNG = Comfort Noise Generator

[x] = Block ID

Figure 8: TOBY-L2 speech processing block diagram

4.1.1 Microphone levels (ML)

All modules in uplink path share a set of microphone level (ML) parameters. In this document, a signal “level” is defined as the RMS over the specific signal duration, expressed in dBFS. The levels are measured on the I2S_RxD signal.

4.1.1.1 Syntax

Type Syntax Response Example

Modify parameters

Set AT+UAPT=8,<profile>,0,0,8,<rate>,<parameterOffset>,<numberOfParameters>,<nominalLevel>,0,0,0

4.1.1.2 Defined values

Parameter Type Description

<nominalLevel> Number

[dBFS]

Speech level as measured at the primary-microphone input to the uplink voice enhancement path under nominal conditions. Nominal conditions are defined in the SLR test which requires a

specific test setup.

Range: -90 to 0

4.1.1.3 Default values

profile 0(NB) 0(WB) 1(NB) 1(WB) 2(NB) 2(WB) 3(NB) 3(WB) 4(NB) 4(WB)

0 nominalLevel -30 -30 -52 -44 -36 -36 -30 -30 -30 -30

1 dualMicSpeechRatio 0 0 0 0 0 0 0 0 0 0

2 dualMicNoiseRatio 0 0 0 0 0 0 0 0 0 0

3 dualMicEchoRatio 0 0 0 0 0 0 0 0 0 0

4.1.2 Echo Cancellation (EC)

The Echo Canceller (EC) removes the major part of the acoustic echo generated during conversation in the near-end and heard on the far-end. The echo is a result of the acoustic coupling between the near-end loudspeaker and microphone. The EC algorithm cancels the acoustic echo by subtracting an estimated echo from the input (microphone) signal. The estimated echo is computed by passing the far-end RxRef signal (a delayed version of



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RxAe) through a digital FIR filter that simulates the acoustic coupling between the loudspeaker and microphone. An adaptive normalized least mean square (NLMS) algorithm computes the digital filter.

The EC can operate in full-band or sub-band modes. Sub-band mode is often used with car audio devices. Full-band mode is suggested to be used with other audio devices. The EC operation is augmented by a residual echo suppression (RES) post processor that is embedded in the noise and residual echo suppression (NRES) module.

4.1.2.1 Syntax


Modify parameters

Set AT+UAPT=8,<profile>,0,0,0,<rate>,<parameterOffset>,<numberOfParameters>,<sampleDelay>,<typicalEchoToNear>,<t

ypicalERLE>,<ntaps>,<ntaps2>,<nsubbands>,<mu>,<initialMu>,<initPeriod>,<qFir>,<RxThres>,<ecReserved1>,<ecRese

rved2>



<sampleDelay> Number

[16kHz

samples]

It controls the delay that is added to RxRef before filtering, so that the adaptive FIR only captures the dynamic acoustic channel and not the digital delay from the DSP to the speaker

Range: 0 to 32767

<typicalEchoToNear> Number

[dB]

It affects the EC initial convergence and handling of double-talk.

It is a rough estimate of the ratio between the echo and the near-end speech. It should be

determined for the highest Rx volume setting in an anechoic environment. It can be determined based on the TxAe (TX after enhancement recording point) signal recorded during far-end

speech followed by near-end speech with all modules disabled except the HPF.

Range: -90 to 90

<typicalERLE> Number

[dB]

It affects the EC initial convergence and handling of double-talk.

It is a rough estimate of the amount of echo attenuation achieved by the EC after convergence. It can be determined by comparing the above recorded signal with a similar signal where both

HPF and EC are enabled. Measure the residual echo at least 10 seconds after initialization.

The far-end and near-end signals for these experiments are the RLR and SLR signals generated

by the audio testing equipment. Range: 0 to 90

<ntaps> Number It sets the FIR length. Range: 1 to 512

<ntaps2> Number Reserved to expert users, do not modify.

<nsubbands> Number Reserved to expert users, do not modify.

<mu> Number Reserved to expert users, do not modify.

<initialMu> Number Reserved to expert users, do not modify.

<initPeriod> Number Reserved to expert users, do not modify.

<qFir> Number Reserved to expert users, do not modify.

<RxThres> Number Reserved to expert users, do not modify.

<ecReserved1> Reserved

<ecReserved2> Reserved



0 sampleDelay 36 36 36 36 36 36 36 36 36 36

1 typicalEchoToNear -10 -20 16 15 0 0 -10 -20 -10 -20

2 typicalERLE 8 20 8 10 8 20 8 20 8 20

3 Ntaps 64 128 64 25 64 128 64 128 64 128

4 ntaps2 0 0 0 0 0 0 0 0 0 0

5 Nsubbands 0 0 0 16 0 0 0 0 0 0

6 Mu 3276 8192 1638 3276 3276 3276 3276 8192 3276 8192

7 initialMu 8192 16384 3276 8192 8192 8192 8192 16384 8192 16384



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8 initPeriod 10 10 20 20 10 10 10 10 10 10

9 qFir 20 20 16 16 20 16 20 20 20 20

10 RxThres -50 -50 -50 -50 -50 -50 -50 -50 -50 -50

11 ecReserved1 0 0 0 0 0 0 0 0 0 0

12 ecReserved2 0 0 0 0 0 0 0 0 0 0

4.1.3 Residual Echo Suppression (RES)

The residual echo suppressor (RES) suppresses very late echo reflections and non-linear echo components that were not cancelled by the EC. Similar to NS, RES attenuates the uplink (TX) signal in time/frequency regions where the ratio between near-end signal and residual-echo is low.

4.1.3.1 Syntax


Modify parameters

Set AT+UAPT=8,<profile>,0,0,1,<rate>,<parameterOffset>,<nu

mberOfParameters>,<resEnabled>,0,<resEchoAtt>,20,<gammaDT>,<nonLinearityType>,<nonLinearityFactor>,<refRegres

sion1>,<vmThresDT>,<vmThresDT1>,<vmThresDT2>,<vmThresMus>,<hangDT>,<hangFE>,<numMutedChan>,<muRes>

,<initialMuRes>,<vmThreshAdaptSlowdown>,<initialVmThresAdaptSlowdown>,<initPeriod>,<cnGain>,<outputGain>,<

minChNoise>,<maxChNoise>,<dmVadHighThres>,<dmVadLowThres>,<dmProbabilityThres>,<dmProbabilityMode>,<pat

hChangeThres>,<maskingThresholdFactor>



<resEnabled> Number Enables/disables the RES function

Range: 0,1

<resEchoAtt> Number

[dB]

Maximal residual-echo attenuation

Range: 0 to 60

<gammaDT> Number An overestimation factor of the residual echo. Increasing this parameter increases echo suppression at the cost of more distortion of the near end during double-talk.

Range: 0 to 32767

<nonLinearityType> Number Reserved to expert users, do not modify.

<nonLinearityFactor> Number Reserved to expert users, do not modify.

<refRegression1> Number Reserved to expert users, do not modify.

<vmThresDT> Number Affects the operation of a DT detector. It sets the tradeoff between two undesired phenomena of DT false alarms and misdetection. A DT false alarm means that near-end speech was falsely detected, and misdetection means that near-end speech was

misdetected. Higher values would result in fewer false alarms and more misdetection.

Range: 0 to 1000

<vmThresDT1> Number Reserved to expert users, do not modify.

<vmThresDT2> Number Reserved to expert users, do not modify.

<vmThresMus> Number Reserved to expert users, do not modify.

<hangDT> Number Reserved to expert users, do not modify.

<hangFE> Number Reserved to expert users, do not modify.

<numMutedChan> Number Reserved to expert users, do not modify.

<muRes> Number Reserved to expert users, do not modify.

<initialMuRes> Number Reserved to expert users, do not modify.

<vmThreshAdaptSlowdown> Number Reserved to expert users, do not modify.

<initialVmThresAdaptSlowdown> Number Reserved to expert users, do not modify.



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<initPeriod> Number Reserved to expert users, do not modify.

<cnGain> Number Reserved to expert users, do not modify.

<outputGain> Number Reserved to expert users, do not modify.

<minChNoise> Number Reserved to expert users, do not modify.

<maxChNoise> Number Reserved to expert users, do not modify.

<dmVadHighThres> Number Reserved to expert users, do not modify.

<dmVadLowThres> Number Reserved to expert users, do not modify.

<dmProbabilityThres> Number Reserved to expert users, do not modify.

<dmProbabilityMode> Number Reserved to expert users, do not modify.

<pathChangeThres> Number Reserved to expert users, do not modify.

<maskingThresholdFactor> Number Reserved to expert users, do not modify.



0 resEnabled 1 1 1 1 1 1 1 1 1 1

1 dualMicEnabled 0 0 0 0 0 0 0 0 0 0

2 resEchoAtt 40 40 60 60 40 40 40 40 40 40

3 nonStatNoiseAtt 20 20 20 20 20 20 20 20 20 20

4 gammaDT 1024 1024 16384 32767 1024 1024 1024 1024 1024 1024

5 nonLinearityType 2 2 2 2 2 2 2 2 2 2

6 nonLinearityFactor 0 0 32767 32767 0 0 0 0 0 0

7 refRegression1 8192 8192 8192 8192 8192 8192 8192 8192 8192 8192

8 vmThresDT 100 250 50 50 100 100 100 100 100 100

9 vmThresDT1 482 482 482 964 482 482 482 482 482 482

10 vmThresDT2 0 0 50 0 0 0 0 0 0 0

11 vmThresMus 100 100 100 100 100 100 100 100 100 100

12 hangDT 20 20 20 20 20 20 20 20 20 20

13 hangFE 10 10 10 10 10 10 10 10 10 10

14 numMutedChan 0 0 0 0 0 0 0 0 0 0

15 muRes 672 672 168 672 672 672 672 672 672 672

16 initialMuRes 1680 1680 1680 1680 1680 1680 1680 1680 1680 1680

17 vmThreshAdaptSlowdown 50 50 100 100 50 50 50 50 50 50

18 initialVmThresAdaptSlowdown 200 200 500 200 200 200 200 200 200 200

19 initPeriod 40 40 80 40 40 40 40 40 40 40

20 cnGain 0 0 0 0 0 0 0 0 0 0

21 outputGain 0 0 1 1 0 0 0 0 0 0

22 minChNoise -90 -90 -90 -90 -90 -90 -90 -90 -90 -90

23 maxChNoise 0 0 0 0 0 0 0 0 0 0

24 dmVadHighThres 50 50 50 50 50 50 50 50 50 50

25 dmVadLowThres 30 30 30 30 30 30 30 30 30 30

26 dmProbabilityThres 0 0 0 0 0 0 0 0 0 0

27 dmProbabilityMode 0 0 0 0 0 0 0 0 0 0

28 pathChangeThres 0 0 0 0 0 0 0 0 0 0

29 maskingThresholdFactor 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767



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4.1.4 Noise Suppression (NS) and Comfort Noise (CN)

The Noise Suppressor is based on auditory-perception principles that favor attenuation of the noisy signal in time/frequency segments where the SNR is low. This attenuation comes at the price of distortion of the speech signal, and therefore has to be applied with care. The amount of desired suppression is controlled by the parameter <maxSuppress>. It is set as a compromise between some loss of intelligibility and naturalness versus gain in listening comfort due to the suppressed noise. The NS primarily suppresses quasi-stationary noise generated, for example, by wind, air-conditioning, a car engine, or a crowd.

The Comfort Noise generator generates CN to compensate for over suppression of noise by the RES. Optionally, it can generate a CN floor to prevent total silence in the channel or to mask irritating noise that cannot be removed in any other way. The CNG parameters are split between the Uplink NS (tunable) and NRES (expertly) parameter sets.

4.1.4.1 Syntax


Modify parameters

Set AT+UAPT=8,<profile>,0,<direction>,2,<rate>,<parameterOf

fset>,<numberOfParameters>,<maxSuppress>,<cnFloor>,<cnHighFreqBoost>,<noiseInducedEqualizedFactor>,<lowTxNoi

se>,<HighTxNoise>,<equalizer[0]>,...,<equalizer[21]>,<varThId>,<nsReserved>



<maxSuppress> Number

[dB]

Amount of noise suppression

Range: 0 to 40

<cnFloor> Number

[dBFS]

The level (dBFS) of the inserted Confort Noise. The CN is pink, with a spectral tilt of

approximately -10 dB over the full bandwidth. The floor level corresponds to the level of the I2S_RxD signal after being amplified by NRES <outputGain>

Range: -100 to 0

<cnHighFreqBoost> Number

[dB]

A spectral-tilt boost (dB per the full bandwidth) of the generated CN. For example, setting this parameter to 10 generates an approximately white noise.

Range: -90 to 90

<noiseInducedEqualized

Factor>

Number Range: 0 to 32767

<lowTxNoise> Number

[dB]

Range: -90 to 90

<HighTxNoise> Number

[dB]

Range: -90 to 90

<equalizer[0]> … <equalizer[21]>

Number

[dB]

Range: -20 to 20

<varThId> Number Reserved to expert users, do not modify.

<nsReserved> Reserved


For <direction>=0 (uplink)


0 maxSuppress 15 15 15 15 20 20 15 15 15 15

1 cnFloor -100 -100 -100 -100 -100 -100 -100 -100 -100 -100

2 cnHighFreqBoost 0 0 0 0 0 0 0 0 0 0

3 noiseInducedEqualizerFactor 0 0 0 0 0 0 0 0 0 0

4 lowTxNoise -20 -20 -20 -20 -20 -20 -20 -20 -20 -20

5 highTxNoise 0 0 0 0 0 0 0 0 0 0

6 equalizer[0] 0 0 0 0 0 0 0 0 0 0



Page 16 of 39


7 equalizer[1] 0 0 0 0 0 0 0 0 0 0

8 equalizer[2] 0 0 0 0 0 0 0 0 0 0

9 equalizer[3] 0 0 0 0 0 0 0 0 0 0

10 equalizer[4] 0 0 0 0 0 0 0 0 0 0

11 equalizer[5] 0 0 0 0 0 0 0 0 0 0

12 equalizer[6] 0 0 0 0 0 0 0 0 0 0

13 equalizer[7] 0 0 0 0 0 0 0 0 0 0

14 equalizer[8] 0 0 0 0 0 0 0 0 0 0

15 equalizer[9] 0 0 0 0 0 0 0 0 0 0

16 equalizer[10] 0 0 0 0 0 0 0 0 0 0

17 equalizer[11] 0 0 0 0 0 0 0 0 0 0

18 equalizer[12] 0 0 0 0 0 0 0 0 0 0

19 equalizer[13] 0 0 0 0 0 0 0 0 0 0

20 equalizer[14] 0 0 0 0 0 0 0 0 0 0

21 equalizer[15] 0 0 0 0 0 0 0 0 0 0

22 equalizer[16] 0 0 0 0 0 0 0 0 0 0

23 equalizer[17] 0 0 0 0 0 0 0 0 0 0

24 equalizer[18] 0 0 0 0 0 0 0 0 0 0

25 equalizer[19] 0 0 0 0 0 0 0 0 0 0

26 equalizer[20] 0 0 0 0 0 0 0 0 0 0

27 equalizer[21] 0 0 0 0 0 0 0 0 0 0

28 varThld 200 200 200 200 200 200 200 200 200 200

29 nsReserved2 0 0 0 0 0 0 0 0 0 0

For <direction>=1 (downlink)


0 maxSuppress 10 10 10 10 10 10 10 10 10 10

1 cnFloor -100 -100 -100 -100 -100 -100 -100 -100 -100 -100

2 cnHighFreqBoost 0 0 0 0 0 0 0 0 0 0

3 noiseInducedEqualizerFactor 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767

4 lowTxNoise -30 -30 -30 -30 -30 -30 -30 -30 -30 -30

5 highTxNoise 0 0 0 0 0 0 0 0 0 0

6 equalizer[0] 0 0 0 0 0 0 0 0 0 0

7 equalizer[1] 0 0 0 0 0 0 0 0 0 0

8 equalizer[2] 0 0 0 0 0 0 0 0 0 0

9 equalizer[3] 0 0 0 0 0 0 0 0 0 0

10 equalizer[4] 0 0 0 0 0 0 0 0 0 0

11 equalizer[5] 0 0 0 0 0 0 0 0 0 0

12 equalizer[6] 0 0 0 0 0 0 0 0 0 0

13 equalizer[7] 0 0 0 0 0 0 0 0 0 0

14 equalizer[8] 0 0 0 0 0 0 0 0 0 0

15 equalizer[9] 0 0 0 0 0 0 0 0 0 0

16 equalizer[10] 0 0 0 0 0 0 0 0 0 0

17 equalizer[11] 0 0 0 0 0 0 0 0 0 0

18 equalizer[12] 0 0 0 0 0 0 0 0 0 0

19 equalizer[13] 0 0 0 0 0 0 0 0 0 0

20 equalizer[14] 0 0 0 0 0 0 0 0 0 0

21 equalizer[15] 0 0 0 0 0 0 0 0 0 0

22 equalizer[16] 0 0 0 0 0 0 0 0 0 0

23 equalizer[17] 0 0 0 0 0 0 0 0 0 0



Page 17 of 39


24 equalizer[18] 0 0 0 0 0 0 0 0 0 0

25 equalizer[19] 0 0 0 0 0 0 0 0 0 0

26 equalizer[20] 0 0 0 0 0 0 0 0 0 0

27 equalizer[21] 0 0 0 0 0 0 0 0 0 0

28 varThld 160 160 160 160 160 160 160 160 160 160

29 nsReserved2 0 0 0 0 0 0 0 0 0 0

4.1.5 Low Pass Filter (LPF)

Low Pass Filter in both uplink and downlink paths usually consist of single Bi-Quad filters. The coefficients are given by the parameters a[0], a[1], a[2], b[0], b[1], and b[2]. These parameters conform to the standard definition of IIR-filter coefficients and can be calculated by using the MVT Tool.

4.1.5.1 Syntax


Modify parameters


fset>,<numberOfParameters>,<FilterType>,<TapA0>,<TapA1>,<TapA2>,<TapB0>,<TapB1>,<TapB2>

OK AT+UAPT=8,1,0,0,3,0,0,7,1,8192,10201,3772,

5541,11082,5541

HPF, FrequencyCutoff=3300Hz, Q-Filter=0.707



<FilterType> Number Range: 0,1

<TapA0>, <TapA1>, <TapA2>, <TapB0>, <TapB1>, <TapB2>

Number Bi-Quad filter coefficients.

Use the MVT tool to design the filter and calculate the coefficients.

Define the bi-quad filter’s cutoff frequencies (in Hz). To filter noise outside of speech

frequencies, use the following guidelines, depending on whether you are using a WB or NB filter:

Wideband (WB) LPF: 7500 to 8000 Hz

Narrow band (NB) LPF: 3400 to 4000 Hz

The Q-filter factor defines the sharpness of the slope. Define a Q-Filter floating point between 0

and 1; for a 3 dB attenuation at cutoff frequency, define Q as 2 (0.707).

Range TapA0: 4096, 8192, or 16384

Range others: -32768 to 32767




0 filterType 1 1 1 1 1 1 1 1 1 1

1 a[0] 8192 8192 8192 8192 8192 8192 8192 8192 8192 8192

2 a[1] 11913 10201 10201 10201 10201 10201 10201 10201 10201 10201

3 a[2] 4703 3772 3772 3772 3772 3772 3772 3772 3772 3772

4 b[0] 6202 5541 5541 5541 5541 5541 5541 5541 5541 5541

5 b[1] 12404 11082 11082 11082 11082 11082 11082 11082 11082 11082

6 b[2] 6202 5541 5541 5541 5541 5541 5541 5541 5541 5541



0 filterType 1 1 1 1 1 1 0 1 1 1

1 a[0] 4096 4096 4096 4096 4096 4096 4096 4096 4096 4096

2 a[1] 0 0 0 0 0 0 0 0 0 0

3 a[2] 0 0 0 0 0 0 0 0 0 0



Page 18 of 39


4 b[0] 4096 4096 4096 4096 4096 4096 4096 4096 4096 4096

5 b[1] 0 0 0 0 0 0 0 0 0 0

6 b[2] 0 0 0 0 0 0 0 0 0 0

4.1.6 High Pass Filter (HPF)

High Pass Filter in both uplink and downlink paths usually consist of single Bi-Quad filters. The coefficients are given by the parameters a[0], a[1], a[2], b[0], b[1], and b[2]. These parameters conform to the standard definition of IIR-filter coefficients and can be calculated by using MVT Tool.

4.1.6.1 Syntax


Modify parameters


fset>,<numberOfParameters>,<FilterType>,<TapA0>,<TapA1>,<TapA2>,<TapB0>,<TapB1>,<TapB2>

OK AT+UAPT=8,1,0,0,4,0,0,7,0,8192,-13675,587

2,6935,-13870,6935

HPF, FrequencyCutoff=330 Hz, Q-Filter=0.707



<FilterType> Number Range: 0,1

<TapA0>, <TapA1>, <TapA2>, <TapB0>,

<TapB1>, <TapB2>

Number Bi-Quad filter coefficients.

Use MVT tool to design the filter and calculate the coefficients:

Define the bi-quad filter’s cutoff frequencies (in Hz). To filter noise outside of speech

frequencies, use the following guidelines, depending on whether you are using a WB or NB filter:

Wideband (WB) HFP: 0 to 300 Hz

Narrow band (NB) HFP: 0 to 300 Hz

The Q-filter factor defines the sharpness of the slope. Define a Q-Filter floating point between 0

and 1; for a 3 dB attenuation at cutoff frequency, define Q as 2 (0.707).

Range TapA0: 4096, 8192, or 16384

Range others: -32768 to 32767




0 filterType 0 0 0 0 0 0 0 0 0 0

1 a[0] 8192 8192 8192 4096 8192 8192 8192 8192 8192 8192

2 a[1] -14121 -16020 -12788 -7737 -13446 -16020 -15022 -16020 -15022 -16020

3 a[2] 6206 7836 5254 3665 5518 7836 6935 7836 6935 7836

4 b[0] 7130 8012 6559 3875 6788 8012 7537 8012 7537 8012

5 b[1] -14259 -16024 -13117 -7749 -13578 -16024 -15074 -16024 -15074 -16024

6 b[2] 7130 8012 6559 3875 6788 8012 7537 8012 7537 8012



0 filterType 0 0 0 0 0 0 0 0 0 0

1 a[0] 4096 4096 4096 4096 4096 4096 4096 4096 4096 4096

2 a[1] 0 0 0 0 0 0 0 0 0 0

3 a[2] 0 0 0 0 0 0 0 0 0 0

4 b[0] 4096 4096 4096 4096 4096 4096 4096 4096 4096 4096

5 b[1] 0 0 0 0 0 0 0 0 0 0

6 b[2] 0 0 0 0 0 0 0 0 0 0



Page 19 of 39

4.1.7 Equalizer (EQ)

The equalizers in both Uplink (Tx) and Downlink (Rx) paths consist of 31-tap FIR.

4.1.7.1 Syntax


Modify parameters

Set AT+UAPT=8,<profile>,0,<direction>,5,<rate>,<parameter

Offset>,<numberOfParameters>,<numOfTaps>,<EqTap0>,<EqTap1>,…,<EqTap15>



<numOfTaps>, Number Range: 31

<EqTap0> , …,

<EqTap15>

Number Range: 0 to 32767




0 numOfTaps 31 31 31 31 31 31 31 31 31 31

1 taps[0] 0 0 0 0 10 -17 0 0 0 0

2 taps[1] 0 0 0 0 -1 37 0 0 0 0

3 taps[2] 0 0 0 0 0 5 0 0 0 0

4 taps[3] 0 0 0 0 -5 5 0 0 0 0

5 taps[4] 0 0 0 0 9 33 0 0 0 0

6 taps[5] 0 0 0 0 -9 3 0 0 0 0

7 taps[6] 0 0 0 0 16 -4 0 0 0 0

8 taps[7] 0 0 0 0 -24 47 0 0 0 0

9 taps[8] 0 0 0 0 24 -52 0 0 0 0

10 taps[9] 0 0 0 0 -6 35 0 0 0 0

11 taps[10] 0 0 0 0 34 9 0 0 0 0

12 taps[11] 0 0 0 0 23 -105 0 0 0 0

13 taps[12] 0 0 0 0 -18 246 0 0 0 0

14 taps[13] 0 0 0 0 -10 -306 0 0 0 0

15 taps[14] 0 0 0 0 -113 -194 0 0 0 0

16 taps[15] 1024 1024 1024 1024 1241 1823 1024 1024 1024 1024

17 taps[16] 0 0 0 0 -113 -194 0 0 0 0

18 taps[17] 0 0 0 0 -10 -306 0 0 0 0

19 taps[18] 0 0 0 0 -18 246 0 0 0 0

20 taps[19] 0 0 0 0 23 -105 0 0 0 0

21 taps[20] 0 0 0 0 34 9 0 0 0 0

22 taps[21] 0 0 0 0 -6 35 0 0 0 0

23 taps[22] 0 0 0 0 24 -52 0 0 0 0

24 taps[23] 0 0 0 0 -24 47 0 0 0 0

25 taps[24] 0 0 0 0 16 -4 0 0 0 0

26 taps[25] 0 0 0 0 -9 3 0 0 0 0

27 taps[26] 0 0 0 0 9 33 0 0 0 0

28 taps[27] 0 0 0 0 -5 5 0 0 0 0

29 taps[28] 0 0 0 0 0 5 0 0 0 0

30 taps[29] 0 0 0 0 -1 37 0 0 0 0

31 taps[30] 0 0 0 0 10 -17 0 0 0 0



Page 20 of 39



0 numOfTaps 31 31 31 31 31 31 31 31 31 31

1 taps[0] 0 0 0 0 -4 -1 0 0 0 0

2 taps[1] 0 0 0 0 -16 -5 0 0 0 0

3 taps[2] 0 0 0 0 -4 11 0 0 0 0

4 taps[3] 0 0 0 0 -18 6 0 0 0 0

5 taps[4] 0 0 0 0 0 -8 0 0 0 0

6 taps[5] 0 0 0 0 -9 -11 0 0 0 0

7 taps[6] 0 0 0 0 -1 -8 0 0 0 0

8 taps[7] 0 0 0 0 -11 -11 0 0 0 0

9 taps[8] 0 0 0 0 -3 18 0 0 0 0

10 taps[9] 0 0 0 0 22 31 0 0 0 0

11 taps[10] 0 0 0 0 -63 1 0 0 0 0

12 taps[11] 0 0 0 0 14 72 0 0 0 0

13 taps[12] 0 0 0 0 23 55 0 0 0 0

14 taps[13] 0 0 0 0 80 28 0 0 0 0

15 taps[14] 0 0 0 0 108 131 0 0 0 0

16 taps[15] 1024 1024 1024 1024 709 522 1024 1024 1024 1024

17 taps[16] 0 0 0 0 108 131 0 0 0 0

18 taps[17] 0 0 0 0 80 28 0 0 0 0

19 taps[18] 0 0 0 0 23 55 0 0 0 0

20 taps[19] 0 0 0 0 14 72 0 0 0 0

21 taps[20] 0 0 0 0 -63 1 0 0 0 0

22 taps[21] 0 0 0 0 22 31 0 0 0 0

23 taps[22] 0 0 0 0 -3 18 0 0 0 0

24 taps[23] 0 0 0 0 -11 -11 0 0 0 0

25 taps[24] 0 0 0 0 -1 -8 0 0 0 0

26 taps[25] 0 0 0 0 -9 -11 0 0 0 0

27 taps[26] 0 0 0 0 0 -8 0 0 0 0

28 taps[27] 0 0 0 0 -18 6 0 0 0 0

29 taps[28] 0 0 0 0 -4 11 0 0 0 0

30 taps[29] 0 0 0 0 -16 -5 0 0 0 0

31 taps[30] 0 0 0 0 -4 -1 0 0 0 0



Page 21 of 39

4.1.8 Additional Gain Control (AGC)

The AGC regulates the uplink (TX) or downlink (RX) signal level respectively, to compensate for variations in the voice level and the microphone-to-mouth distance.

The parameter <mode> selects whether AGC operates in loudness mode or peak mode.

In loudness mode, the target of the AGC is to bring the perceived loudness of the near-end signal to a given target, set by <loudnessTarget>.

In peak mode, the target of the AGC is to bring the peak power of the near-end signal to a given target, set by <envelopeTarget>. In this mode the gain is limited to <maxAgcGain>.

The loudness and peak-power of the input signal are continuously updated based on a few seconds of the signal history. Background noise and other interfering signals are rejected and have only a minor effect on the estimated loudness and peak-power.

While striving to reach the loudness or peak targets, the gain is limited so that the short-term output power does not exceed the limit set by <rmsLimit>.

At the beginning of a conversation, the gain is initially set assuming that the input speech is:

at the nominal level, as defined by ML <nominalLevel>, in Uplink AGC;

at a nominal level of -23 dBFS, in Downlink AGC.

Optionally, the initial gain can be decreased by the amount <initialGainReduction> to prevent excessive amplification if and when the incoming speech is unexpectedly high.

The loudness and envelope targets can be dynamically boosted in the presence of TX noise; this feature is slight different between AGC block in uplink and downlink path.

In uplink path, AGC compensates for potential over-estimation of the input speech level due to noise. Such an over-estimate of the input level may lead to a negative-bias in the applied AGC gain. Another motivation is to reflect the fact that near-end speakers typically raise their voices in the presence of noise; without a target-boost, the AGC would cancel the effect of raising one’s voice after a few seconds.

In downlink path, AGC acts as “auto-volume”.

4.1.8.1 Syntax


Modify parameters


Offset>,<numberOfParameters>,<mode>,<loudnessTarget>,<envelopeTarget>,<maxAgcGain>,<rmsLimit>,<initialGainReduction>,<noiseInducedBoost>,<rmsToLoudnessOffset

>,<agcReserved1>



<mode> Number It selects whether Rx-AGC operates in loudness mode or peak mode

Range: 0,1

<loudnessTarget> Number

[dBFS]

In loudness mode, it is the target to which the AGC brings the perceived loudness of the near-end signal

Range: -90 to 0

<envelopeTarget> Number

[dBFS]

In peak mode, it is the target to which the AGC brings the peak power of the near-end signal

Range: -90 to 0

<maxAgcGain> Number

[dB]

In peak mode it limits the gain

Range: 0 to 90

<rmsLimit> Number

[dBFS]

Range: -90 to 0



Page 22 of 39


<initialGainReduction> Number

[dB]

Range: 0 to 20

<noiseInducedBoost> Number

[dB]

Range: 0 to 20

<rmsToLoudnessOffset> Number Reserved to expert users, do not modify.

<agcReserved1> Reserved




0 Mode 0 0 0 0 0 0 0 0 0 0

1 loudnessTarget -25 -25 -26 -26 -26 -26 -25 -25 -25 -25

2 envelopeTarget 0 0 0 0 -5 -5 0 0 0 0

3 maxAgcGain 90 90 90 90 90 90 90 90 90 90

4 rmsLimit 0 0 0 0 0 0 0 0 0 0

5 initialGainReduction 0 0 0 0 0 0 0 0 0 0

6 noiseInducedBoost 0 0 0 0 0 0 0 0 0 0

7 rmsToLoudnessOffset 0 -3 -3 0 -5 -3 0 -3 0 -3

8 agcReserved1 0 0 0 0 0 0 0 0 0 0



0 Mode 0 0 1 1 0 0 0 0 0 0

1 loudnessTarget -22 -24 -24 -24 -24 -24 -24 -24 -24 -24

2 envelopeTarget 0 0 -1 -1 -1 -1 0 0 0 0

3 maxAgcGain 20 20 20 20 20 20 20 20 20 20

4 rmsLimit -15 -15 -15 -15 -15 -15 -15 -15 -15 -15

5 initialGainReduction 0 0 0 0 0 0 0 0 0 0

6 noiseInducedBoost 0 0 0 0 0 0 0 0 0 0

7 rmsToLoudnessOffset -1 -1 -1 -1 -1 -1 -1 -1 -1 -1

8 agcReserved1 0 0 0 0 0 0 0 0 0 0



Page 23 of 39

4.1.9 Dynamic Range Compression (DRC)

Dynamic range compression (DRC) compresses the dynamic range of the received speech signal by selectively amplifying low-level segments. Its purpose is to increase the perceived loudness without increasing the peak power).

The Uplink DRC is identical to the Downlink DRC, except RX-TXnoiseFactor is replaced by TX-TXnoiseFactor.

The DRC instantaneous gain is determined by a “gain function” that takes the following inputs:

a long-term peak-level that is passed from the Rx-AGC to the DRC;

a short-term level estimated by the DRC;

an estimate of the Rx noise level;

a dynamic DRC factor that is optionally dependent on the Tx noise.

The short-term level is estimated by an attack/release leaky integrator mechanism, where the attack time is instantaneous and the release time is configurable by the parameter <release>.

The noise level estimator continuously tracks the level of the stationary noise in the downlink signal. The final noise level that is used by the DRC is the maximum between the estimated stationary noise and the input parameter <minNoiseLevel>.

4.1.9.1 Syntax


Modify parameters


Offset>,<numberOfParameters>,<gainFunctionType>,<drcGainFactor>,<release>,<minNoiseLevel>,<noiseLevelGain>,<noiseInducedGainFactor>,<g2Slope>,<progLevels[0]>,...,

<progLevels[9]>,<progGains[0]>,...,<progGains[9]>,



<gainFunctionType> Number 0: analitic

1: Prog

<drcGainFactor> Number Range: 0 to 32767

<release> Number Range: 0 to 32767

<minNoiseLevel> Number

[dBFS]

Range: -90 to 0

<noiseLevelGain> Number

[dB]

Range: -10 to 10

<noiseInducedGainFactor> Number Range: 0 to 32767

<g2Slope> Number Range: 0 to 32767

<progLevels[0]>, ..., <progLevels[9]>

Number

[dBFS]

Range: -90 to 0

<progGains[0]>, ..., <progGains[9]>

Number

[dB]

Range: -90 to 90



Page 24 of 39




0 gainFunctionType 0 0 0 0 0 0 0 0 0 0

1 drcGainFactor 4096 4096 4096 4096 4096 4096 4096 4096 4096 4096

2 release 327 327 327 327 327 327 327 327 327 327

3 minNoiseLevel -50 -50 -50 -50 -50 -50 -50 -50 -50 -50

4 noiseLevelGain 0 0 0 0 0 0 0 0 0 0

5 noiseInducedGainFactor 0 0 0 0 0 0 0 0 0 0

6 g2Slope 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767

7 progLevels[0] 0 0 0 0 0 0 0 0 0 0

8 progLevels[1] 0 0 0 0 0 0 0 0 0 0

9 progLevels[2] 0 0 0 0 0 0 0 0 0 0

10 progLevels[3] 0 0 0 0 0 0 0 0 0 0

11 progLevels[4] 0 0 0 0 0 0 0 0 0 0

12 progLevels[5] 0 0 0 0 0 0 0 0 0 0

13 progLevels[6] 0 0 0 0 0 0 0 0 0 0

14 progLevels[7] 0 0 0 0 0 0 0 0 0 0

15 progLevels[8] 0 0 0 0 0 0 0 0 0 0

16 progLevels[9] 0 0 0 0 0 0 0 0 0 0

17 progGains[0] 0 0 0 0 0 0 0 0 0 0

18 progGains[1] 0 0 0 0 0 0 0 0 0 0

19 progGains[2] 0 0 0 0 0 0 0 0 0 0

20 progGains[3] 0 0 0 0 0 0 0 0 0 0

21 progGains[4] 0 0 0 0 0 0 0 0 0 0

22 progGains[5] 0 0 0 0 0 0 0 0 0 0

23 progGains[6] 0 0 0 0 0 0 0 0 0 0

24 progGains[7] 0 0 0 0 0 0 0 0 0 0

25 progGains[8] 0 0 0 0 0 0 0 0 0 0

26 progGains[9] 0 0 0 0 0 0 0 0 0 0



Page 25 of 39



0 gainFunctionType 0 0 0 0 0 0 0 0 0 0

1 drcGainFactor 4096 4096 16384 4096 4096 4096 4096 4096 4096 4096

2 release 327 327 327 327 327 327 327 327 327 327

3 minNoiseLevel -50 -50 -50 -50 -50 -50 -50 -50 -50 -50

4 noiseLevelGain 0 0 0 0 0 0 0 0 0 0

5 noiseInducedGainFactor 0 0 0 0 0 0 0 0 0 0

6 g2Slope 0 0 0 0 0 0 0 0 0 0

7 progLevels[0] 0 0 0 0 0 0 0 0 0 0

8 progLevels[1] 0 0 0 0 0 0 0 0 0 0

9 progLevels[2] 0 0 0 0 0 0 0 0 0 0

10 progLevels[3] 0 0 0 0 0 0 0 0 0 0

11 progLevels[4] 0 0 0 0 0 0 0 0 0 0

12 progLevels[5] 0 0 0 0 0 0 0 0 0 0

13 progLevels[6] 0 0 0 0 0 0 0 0 0 0

14 progLevels[7] 0 0 0 0 0 0 0 0 0 0

15 progLevels[8] 0 0 0 0 0 0 0 0 0 0

16 progLevels[9] 0 0 0 0 0 0 0 0 0 0

17 progGains[0] 0 0 0 0 0 0 0 0 0 0

18 progGains[1] 0 0 0 0 0 0 0 0 0 0

19 progGains[2] 0 0 0 0 0 0 0 0 0 0

20 progGains[3] 0 0 0 0 0 0 0 0 0 0

21 progGains[4] 0 0 0 0 0 0 0 0 0 0

22 progGains[5] 0 0 0 0 0 0 0 0 0 0

23 progGains[6] 0 0 0 0 0 0 0 0 0 0

24 progGains[7] 0 0 0 0 0 0 0 0 0 0

25 progGains[8] 0 0 0 0 0 0 0 0 0 0

26 progGains[9] 0 0 0 0 0 0 0 0 0 0

4.1.10 Sidetone

TOBY-L2 has not a sidetone generator, thus if a sidetone is needed, it has to be generated by the external codec (connected via I

2C interface).

For sidetone control on Maxim MAX9860 audio codec see +UVGC (Volume Gain Control).



Page 26 of 39

4.2 No-duplex configuration

In case of hands-free systems with high echo coupling and high non-linearity on the loudspeaker (making the EC cancellation ineffective), a so called “no-duplex” configuration could be chosen. This setup accomplishes excellent results in terms of echo reduction, but has strong side effects that have to be understood and carefully evaluated.

No-duplex configuration consists of a setup having a strong AGC. Doing so, when the AGC measure a signal on downlink path (i.e. the far-end user is talking), a null gain is applied to uplink path and the microphone is muted. In this way, the echo will be suppressed, but also the near-end user.

Figure 9 Far and near end user speaking in the meanwhile

Figure 10: No-duplex vs duplex mode

No-duplex mode (strong AGC) Partial Duplex (weak AGC)

Pros: Pros:

Mic/Loudspeaker levels can be popped up at max

The echo at far-end is completely removed

Near-end speech not completely muted during double talk

Near-end more intelligible also

It allows the understanding of the near-end device even if there is some background noise at the far-end device.

Cons: Cons:

The near-end is muted by AGC when the far-end user is speaking but also if there is high background noise (e.g. call

center) at far-end, not filtered out by far-end device.

Some residual echo could still exist, but it does not disturb the conversation.

Loudspeaker at max level could produce clipping during double talk and an increased echo

DUT

Gain

FAR END

sDL

AGC

echo

uplinked

speechnear end

speech

Far-

end

speech

Gain

mute

unmute

ech

o

Near end

speech

UpLi

nked

speech

mute

unmute

Gain

When the far-end MIC capture a sound,

the near-end speech is muted.

The far-end user will not hear anything from the near-end side; no echo, but neither the near

end speech


UBX-15015834 - R01 Early Production Information Audio profiles management

Page 27 of 39

5 Audio profiles management

5.1 Personal profiles

The audio parameters configuration for the audio paths can be saved in two personal profiles. Each of them

consist of two different files (voice<X>.nvm and audio_gain_calibration<X>.xml, with <X>=0 or 1)

stored in the file system and having the “AUDIO” tag.

At the module boot, the module loads the audio parameters from the selected personal profile, depending on AT&Y setting (the default value is 0). Any change to the setting in the current profile is not stored in NVM and, if needed, it can be stored in one of the personal profiles by using AT&W<n> command.

An audio configuration can be loaded at any time from one of the personal profiles using ATZ<n> command.

Factory-programmed values can be recovered at any time issuing the AT&F command.

These audio files within the personal profiles can be managed (stored, retrieved and deleted) by using file system AT commands. See the u-blox AT Commands Manual [1] for further details about personal profiles management by means of the commands AT&W, AT&Y, ATZ and AT&F.

Personal profile 1

Personal profile 0• AUDIO tag files:

voice0.nvm

audio_gain_calibration0.xml

Factory-programmed values• not accessible by user

Current profile•volatile

AT&W<n>

ATZ <n>

AT&F

Figure 11: Profiles

Each Personal profile contains 5 acoustic profiles. An acoustic profile identifies the set of parameters loaded from volatile memory (current personal profile) in the speech processing blocks when an audio path is enabled. The acoustic profile in use can be selected by +USPM command (see <main_uplink>, <main_downlink> description for +USPM command in u-blox AT Commands Manual [1]). The default configuration is:

+USPM: 1,1,0,0,2

Thus <main_uplink>=1, <main_downlink>=1. Thus acoustic profile 1 is the default.

The <profile> parameter in the +UAPT command identifies the corresponding acoustic profile. Audio parameters stored for each acoustic profile are organized in blocks and distinct for NB and WB scenarios, as described in the section 4.1.

Default parameters have been optimized for generic devices:

Acoustic profile 0: generic earpiece

Acoustic profile 1: generic speaker (default)

Acoustic profile 2: generic headset 4 poles

Acoustic profile 3: generic headset 3 poles

Acoustic profile 4: spare


UBX-15015834 - R01 Early Production Information Audio profiles management

Page 28 of 39

5.1.1 Managing personal profiles using m-center

m-center tool (from release 01.06.00) allows users to easily manage audio files included in the personal profiles.

Switch to File System page, type AUDIO in the tag text box and click the "Dir" button to get the list of the audio files in the FS.

Figure 12: m-center File System AUDIO files

[Store File] Read a file from PC and download it to the module FS

[Retrieve File] Read the selected file from the module FS and store it to PC

[Delete File] Deletes the selected file from the module FS

5.2 Factory-programmed profile

A third profile, not accessible to the users, stores the factory-programmed settings. AT&F loads these settings in the current profile and then restores the audio configuration to the factory-programmed values. Since the current profile is volatile, these changes will be permanent only if they will be stored in a personal profile.


UBX-15015834 - R01 Early Production Information Mobile Voice Tuning tool

Page 29 of 39

6 Mobile Voice Tuning tool The Mobile Voice Tuning (MVT) is an audio tool helpful to compose the AT commands to calibrate any voice module in TOBY-L2. It also enables the user to design equalizer and bi-quad filters for uplink and downlink.

MVT allows editing audio parameters stored in TOBY-L2 module voice0.nvm and voice1.nvm files. These files can be downloaded from the module using m-center tool (ver. 01.07.00 or later):

Run n m-center as administrator. You can save the file in the MVT installation folder

Connect the module. Open the m-center File System Page

Define the tag AUDIO in the “Tag” box at the top right side of the panel.

Press the "Dir" button to get a listing of AUDIO tagged files present in the module's memory

Listed files are:

voice0.nvm, audio_gain_calibration0.xml (This is the audio part the personal profile 0).

voice1.nvm, audio_gain_calibration1.xml (This is the audio part the personal profile 1).

Select a voice*.nvm file in the file system you will edit for custom audio system tuning.

Press the "Retrieve File" button to copy voice0.nvm or voice1.nvm file from module's File System to user's PC.

Store the file as voice.nvm in the …\MobileVoiceTuning\Bin\DB folder (replace existing file)

Open MVT

In the MVT tool, use the file selector to choose voice.nvm file; Confirm selection by “Choose” button

MVT allows viewing and editing every parameter in the voice.nvm file. Selectors can be used to switch between profiles, TX/RX, NB/WB and voice modules:

0: Echo Cancellation ("EC") TX

1: Residual Echo Suppression ("RES") TX

2: Noise Suppression ("NS") TX/RX

3: Low Pass Filter ("LPF") TX/RX

4: High Pass Filter ("HPF") TX/RX

5: Equalizer ("EQU") TX/RX

6: Additional Gain Control ("AGC") TX/RX

7: Dynamic Range Compression ("DRC") TX/RX

8: Microphone Levels ("ML") TX

Once a parameter value is edited, the corresponding AT+UAPT command is prepared by pressing the refresh button (green arrow icon). Use the “copy” button to copy the command in a buffer for pasting in the m-center AT–terminal.

If done during a call, add command AT+UAPT=0 to reload voice parameters from the memory.

This is a quick way to test the effect of a parameter change.

Changed parameters can be saved in the voice.nvm file by button “Write Params to NVM file”. (Warning: switching voice module after editing without saving can cause the edited values to be lost).

Copy voice.nvm as voice0.nvm or voice1.nvm and use the “Store File” button in m-center File System Page to store this file in the module. Reload profile by command atz0 or atz1 respectively.

The effectiveness of parameters change can be checked by AT+UAPT=7 (Get parameters list) command; e.g.:

Command generated by MVT to modify parameters on path 2, Tx direction, NB, AGC module is:

AT+UAPT=8,2,0,0,6,0,0,8,0,-26,-5,90,0,0,0,-5

The corresponding command to get parameters can be obtained substituting <op_code> 8 with 7 and using the first 6 parameters only is:


UBX-15015834 - R01 Early Production Information Mobile Voice Tuning tool

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Command Response Description

AT+UAPT=7,2,0,0,6,0 +UAPT:

ParameterID,

Value:

0,0

1,-26

2,-5

3,90

4,0

5,0

6,0

7,-5

8,-24376

OK

To enable or disable a Voice Module:

Set the Property “Control” of the Voice Module to “Enable” or “Disable” and save to NVM.

Alternatively use the command

AT+UAPT=6,<profile>,<app>,<direction>,<blockID>,<enableState>

e.g.: to disable AGC on Tx path of profile 2, use:

AT+UAPT=6,2,0,0,6,0

AT+UAPT=0

The “Control” Property is common for both NB and WB.

To check status of all Voice Modules, use:

AT+UAPT=5,<profile>,<app>,<direction>


AT+UAPT=5,2,0,0 +UAPT:

ModuleID,

State:

0,1

1,1

2,1

3,1

4,1

5,1

6,0

7,1

8,1

OK

Check status on Rx path of profile 2

A certain Voice Module parameter configuration can be exported as a file by "Export Parameters" button and then, e.g. imported in another profile; to do this just switch the profile by the “Profile” selector and use the "Import" button to select the previously saved file.

The voice.nvm file can be exported to text file by a dedicated button of MVT. Vice-versa a text file can be

imported in voice.nvm. This is a readable format with extended names and values of all parameters, ordered

by profiles and Voice Modules and it is useful e.g. to compare quickly different revisions of parameters in text format.

For a complete description see the MVT User Guide [3].


UBX-15015834 - R01 Early Production Information External codec management

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7 External codec management TOBY-L2 series modules provide an I

2S digital audio interface to connect an external audio device, e.g. a codec.

The Application Processor (AP) should manage the codec.

This section includes an example of architecture for the module / external codec / AP system. In the block diagrams the HW implementation is highly simplified; for more detailed examples of HW implementation, see the TOBY-L2 series System Integration Manual [2]

For more details about the AT commands used in the examples below, see the u-blox AT Commands Manual [1].

TOBY-L2 series modules support additional resources to manage the external codec:

Master Clock Control AT command +UMCLK: this command provides the codec with a 13/26 MHz clock generated from the module.

I2S control command +UI2S extension: new connection modes supported by <I2S_port> parameter.

I2C control commands (+UI2CO, +UI2CW, +UI2CR, +UI2CREGR, +UI2CC). These commands allow sending

commands from the module to the codec through a standard I2C interface. In the examples it is considered

an external codec that can be controlled writing in its registers by I2C interface. Addresses, meanings and

field structures and the registers depend on the specific codec and should be chosen used referring to the specific codec manual.

The AP can monitor the audio activity on the module enabling the +CIEV URCs. The +CIEV URCs on the AT terminal can be generated when the audio activities are running on the module, issuing the command AT+CMER=1,0,0,2,1.

Figure 13 shows a possible architecture for the TOBY-L2 series module / external codec / AP system.

TOBY-L2

series module

AP

VCC REG.

SPK_COD

MIC_COD

I2S

VCC_EN

I2C CODEC_CLK

AT +USPM AT+UI2S

AT+CMERAT+UMCLK

AT+UI2C*

+CIEV

EXT CODEC

Figure 13: External codec management for TOBY-L2 series module

Section 7.1 shows an example of the codec management scenario based on this architecture.

The examples proposed are for a Maxim MAX9860 audio voice codec connected to a TOBY-L2 series module through the I

2S interface as in the evaluation board.



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7.1 Setting

At system start-up, the AP should enable the codec supply by VCC_EN

Configure the module and the codec with a sequence of AT commands


AT+USPM=255,255,0,0,2 OK AT+USPM=<main_uplink>,<main_downlink>,<alert_sound>, <headset_indication>,<vmic_ctrl>

<main_uplink>=255: uplink path is NULL (I2S_RX line not enabled)

<main_downlink>=255: downlink path is NULL (I2S_TX not enabled)

Disable I2S. This allows setting of the I2S properties (I

2S properties cannot be

changed while I2S is in use, i.e. the paths are via I

2S).

AT+UI2S=22,1,0,3,0 OK AT+UI2S=<I2S_mode>,<I2S_port>,<I2S_clk_wa>,<I2S_sample_rate>, <I2S_Master_Slave>

<I2S_mode>=22: PCM mode, Rising CLK edge for RX

<I2S_port>=1: connect I2S to I2Sx connection point

<I2S_clk_wa>=0: dynamic mode (I2S_CLK and I2S_WA outputs are active and running only while audio path is active)

<I2S_sample_rate>=3: 16 kHz sampling rate

<I2S_Master_Slave>=0: master mode (default value if parameter is not specified). In master mode I2S_CLK, I2S_WA, I2S_TX are generated by the

module as output. I2S_RX is an input signal

AT+USPM=1,1,0,0,2 OK AT+USPM=<main_uplink>,<main_downlink>,<alert_sound>,

<headset_indication>,<vmic_ctrl>

<main_uplink>=1: uplink path 1 via I2S (1 for e.g.; any path via I2S is ok)

<main_downlink>=1: uplink path 1 via I2S (1 for e.g.; any path via I2S is ok)

Change audio path to I2S. This is the port connected to the external codec

on EVB

AT+UAPT=0 OK AT+UMCLK=<op_code>

<op_code>=0; reloads voice parameters and configurations from the memory

AT+UMCLK=2,0 OK AT+UMCLK=<mclk_mode>,<enabling_mode>

<mclk_mode>=2: codec master clock at 13 MHz

<enabling_mode>=0: “Audio dependent” mode (the clock is applied to the CODEC_CLK pin only when the audio path is active (audio samples are read on the I2S_RX line and written on the I2S_TX line. For this codec

it is not needed to maintain the clock while I2S is not running, since just

the voltage supply is needed to make I2C work. Be aware that other

codecs could need to maintain the clock running also for register programming. This can be achieved by <enabling_mode>=1:

“Continuous” mode

AT+UI2CO=1,0,0,0x10,0 OK AT+UI2CO=<I2C_controller_number>,<bus_mode>,<bit_rate>, <device_address>,<address_width>

Open logical channel for the Maxim external codec on I2C

<I2C_controller_number>=1: controller 1

<bus_mode>=0: Bus Mode Standard (0 – 100 kb)

<bit_rate>=0: I2C bit rate is 100 kb/s

<device_address>=0x10: Device Address in HEX format; this address can be found in the coded datasheet

<address_width>=0: 7 bit address

AT+UI2CW="00000000101E3F2400

000633004A0000008A",18

OK AT+UI2CW=<hex_data>,<nof_byte_to_write>

<hex_data>= first register address, value for 1st register, value for 2

nd

register, etc. (17 registers values)

<nof_byte_to_write>=18 (register address +17 registers values)

Writing in the register configure the codec (gains, I2S configuration, clock

configuration, etc. Adresses, meanings and field structures and the registers depends on the specific codec; This example is for Max9860;

Refer to Max9860 datasheet for details about this example)

AT+UI2CW="049E",2 OK As above; write byte 0x9E in register 0x04; See Max9860 datasheet for meanings of registers values



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AT+UI2CC OK Close logical channel on I2C

The codec is now initialized and ready to work when the module starts audio

activity; e.g. for audio test

AT+UPAR=0,0,0 OK A tone with infinite repetitions can be started by command. The module enables 13 MHz clock signal on CODEC_CLK pin. I

2S is enabled and starts to

transmit tone data. The tone is played on the loudspeaker by the codec.

AT+USAR=0 OK The command stops the tone. The module stops transmission of audio data on I2S and disable 13 MHz clock signal

The codec supply is maintained up also after the codec is not used anymore.

7.2 External device configuration +UEXTDCONF

This command can be used for an external device configuration, e.g. an audio codec, at boot time. The setting (on / off) for each supported device is saved in NVM and applied each time the module is powered on. The configuration for each supported device is hard-coded in the firmware.


AT+UEXTDCONF=0,1 OK <device_id>=0: Maxim Max9860 audio codec, connected via I2C

<configuration_enable>=1: enabled

AT+CPWROFF OK

Once this procedure has been executed, it is no longer necessary to repeat the procedure in the section 7.1 after each system boot.

When enabled, at every startup the module sets the external codec master clock at audio dependent 13 MHz (corresponding to the command AT+UMCLK=2,0) and configures the external codec via I2C.

Initial setting correspond to command

AT+UI2CO=1,0,0,0x10,0

AT+UI2CW="00000000100F20240000103300500000008A ",18

AT+UI2CW="048F",2

AT+UI2CC

In this way external codec is Maxim MAX9860 programmed and reset after boot. Furthermore the external codec is programmed dynamically via I2C commands every time audio path is enabled, according to gains stored in file audio_gain_calibration0.xml or audio_gain_calibration1.xml (depending on profile set by AT&Y) Gains can be edited by Volume Gain Control command +UVGC; Editable gains for Maxim9860 are:

DVA (DAC Level Adjust)

DVST (Sidetone Level)

PAM (Microphone Preamplifier)

PGAM (Microphone amplifier)

(For details, see the u-blox AT Commands Manual [1], +UVGC command)

E.g.: Status after programming can be verified by these read commands:


AT+UMCLK? +UMCLK: 2,0

OK

read master clk setting

codec master clock is set at 13 MHz, audio dependent mode

AT+UI2CO=1,0,0,0x10,0 OK Open logical channel for the Maxim external codec on I2C



Page 34 of 39


AT+UI2CREGR=0x00,17 +UI2CR: 0: 0x00

+UI2CR: 1: 0x00

+UI2CR: 2: 0x00

+UI2CR: 3: 0x10

+UI2CR: 4: 0x8F

+UI2CR: 5: 0x20

+UI2CR: 6: 0x24

+UI2CR: 7: 0x00

+UI2CR: 8: 0x00

+UI2CR: 9: 0x10

+UI2CR: 10: 0x33

+UI2CR: 11: 0x00

+UI2CR: 12: 0x50

+UI2CR: 13: 0x00

+UI2CR: 14: 0x00

+UI2CR: 15: 0x00

+UI2CR: 16: 0x8A

OK

Read 17 registers of codec, starting from register 0x00;

Refer to Max9860 datasheet for meanings and field structures and the

registers

In register 9 of Max9860, bits 7:0 are reserved for DVA gain (DAC attenuation)

In register 11 of Max9860, bits 4:0 are reserved for DVST gain

(sidetone)

In register 12 of Max9860, bits 6:5 are reserved for PAM gain (Left and right microphone preamp gain) ; bits 4:0 are reserved for PGAM gain

(Left and right microphone gain);

AT+UI2CC OK

The setting of Max9860 registers reserved to gains can be changed by Volume Gain Control command +UVGC, e.g:

AT+UVGC=0,0 +UVGC: -5 reads the DVA setting, in dB; see AT manual for details

AT+UVGC =0,0,-8 sets DVA to -8dB;

The driver will write the corresponding code in the codec register; just for didactic purposes, this can be verified e.g. reading value of register directly, by

AT+UI2CO=1,0,0,0x10,0

AT+UI2CREG=0x09,1 +UI2CR: 0: 0x16 Read 1 registers of codec, starting from register 0x09 (thus, the register itself).

Register 9 value is changed bits 7:0 are reserved for DVA gain

AT+UI2CC OK

The +UVGC settings can be stored in audio_gain_calibration0.xml or audio_gain_calibration1.xml by AT&W0 or AT&W1 command respectively. Once saved, they can be downloaded from the module using m-center tool (ver.1.07.00 or later) as explained for voice0.nvm, voice1.nvm files (“Retrieve file” button on the m-center/File System window). Thus they can be uploaded to others modules by “Store file” button on the m-center/File System window. This allows coping audio calibration parameters from a golden sample to all the produced modules.


UBX-15015834 - R01 Early Production Information EVK suggested parameters

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8 EVK suggested parameters

For use with EVK-L2 (Maxim8960 external codec mounted) and companion EVK headset (EAB-729_Rev1 -HDC5 compatible earphones) the following comand sequence is suggested (to be run just once after each FW flashing):

Command Description

AT+UEXTDCONF=0,1 Enable the external codec Maxim8960

AT+USPM=2,2,0,0 Set the headset 3 poles acoustic profile

AT+UVGC=0,3,8 Change the gain of microphone amplifier on Max9860

AT+UAPT=6,2,0,0,5,0 Disable the rqualizer on uplink headset profile

AT+UAPT=6,2,0,1,5,0 Disable the equalizer on downlik headset profile

AT+UAPT=8,2,0,1,6,0,1,1,-20 Set AGC, loudness Target, NB to -20 dB on downlik headset profile

AT+UAPT=8,2,0,1,6,1,1,1,-20 Set AGC, loudness Target, WB to -20 dB on downlik headset profile

AT&W0 Save parameters on the profile 0

AT&Y0 Set the profile 0 as default

AT+CFUN=16 Reboot the module

See the u-blox AT Commands Manual [1] for further details about these commands.


UBX-15015834 - R01 Early Production Information Supported speech codecs

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9 Supported speech codecs TOBY-L2 supports the following speech codecs on the air interface:

for GSM:

o Full Rate speech codec (8 kHz sampling rate)

o Enhanced Full Rate speech codec (8 kHz sampling rate)

o Half Rate speech codec (8 kHz sampling rate)

o NB-AMR speech codec (8 kHz sampling rate)

o WB-AMR speech codec (16 kHz sampling rate)

for UMTS:

o NB-AMR speech codec (8 kHz sampling rate)

o WB-AMR speech codec (16 kHz sampling rate)

The encoder is placed between the Tx After Enhancement (TxAE) audio signal and the Radio Tx signal. The decoder is placed between the Radio Rx signal and the Rx Before Enhancement (RxBE) audio signal (see block diagram in section 4.1).

The enabling of codecs during the call is managed internally by the audio driver. When changing 8 kHz /16kHz sampling rate, the audio driver automatically reload the NB/WB audio parameters stored in the acoustic profile in use.

Allowed speech codecs to be presented to the network during a voice call setup are configured by means of the AT+UDCONF=30 command (see the u-blox AT Commands Manual [1] for the command description).


UBX-15015834 - R01 Early Production Information Appendix

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Appendix

A List of acronyms Abbreviation / Term Explanation / Definition

AP Application Processor

AT AT Command Interpreter Software Subsystem, or attention

CN Comfort Noise

CNG Comfort Noise Generation / Comfort Noise Generator

dBFS dB Full Scale

DRC Dynamic Range Compression

DT Double Talk

EC Echo Cancellation

EP Earpiece

FIR Finite Impulse Response filter

HF Hands-free Algorithm

HPF High Pass Filter

LEM Loudspeaker-Enclosure-Microphone

LPF Low Pass Filter

ML Microphone Levels

NB Narrow Band

NLMS Normalized Least Mean Square

NR Noise Reduction algorithm

NRES Noise and Residual Echo Suppression

RES Residual Echo Suppression

RLR Receiving Loudness Rating

Rx-NS Rx Noise Suppressor

RxAe Rx After Enhancement

RxBe Rx Before Enhancement

SLR Sending Loudness Rating

SNR Signal to Noise Ratio

SP Loudspeaker

Tx-NS Tx Noise Suppressor

TxAe Tx After Enhancement

TxBe Tx Before Enhancement

WB Wide Band


UBX-15015834 - R01 Early Production Information Related documents

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Related documents [1] u-blox AT Commands Manual, Docu No UBX-13002752

[2] TOBY-L2 series System Integration Manual Docu No UBX-13004618

[3] MVT User Guide Docu No UBX-15032323

Revision history

Revision Date Name Status / Comments

R01 03-May-2016 mrod Initial release release


UBX-15015834 - R01 Early Production Information Contact

Page 39 of 39

Contact For complete contact information visit us at www.u-blox.com

u-blox Offices

North, Central and South America

u-blox America, Inc.

Phone: +1 703 483 3180 E-mail: [email protected]

Regional Office West Coast:


Technical Support:


Headquarters

Europe, Middle East, Africa

u-blox AG

Phone: +41 44 722 74 44 E-mail: [email protected] Support: [email protected]

Asia, Australia, Pacific

u-blox Singapore Pte. Ltd.

Phone: +65 6734 3811 E-mail: [email protected] Support: [email protected]

Regional Office Australia:

Phone: +61 2 8448 2016 E-mail: [email protected] Support: [email protected]

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Phone: +86 10 68 133 545 E-mail: [email protected] Support: [email protected]

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Documents

TOBY-L2 Audio tuning - U-blox · TOBY-L2 series Audio tuning Application Note Abstract This document provides information and procedures to perform the audio tuning in TOBY-L2 series