Claims:We Claim:
Claim1:
A method where voice quality of normal calls between users can be monitored and measured. It captures all voice call related experiences, such as muting, poor quality for both downlink and uplink.
Claim 2:
No additional voice calls required to test voice call quality.
Claim 3:
Can be implemented with and without external hardware. It does not require any additional servers to do analysis.
Claim 4:
Adheres to full privacy of user speech. No recording of user speech.
Claim 5:
Analysis done in real time to provide real time inputs.
Claim 6:
Details of coverage, location, telecom network status is linked to poor voice quality which can help telecom service providers (TSPs) to improve their network.
Claim 7:
No data sent to any servers, hence no possibility of compromise of user privacy / users private data.
Claim 8:
No need to call specific switch numbers with specific voice playback to do analysis
Claim 9:
No interference to user’s normal voice call, no inputs from user is required.
Claim 10:
Not restricted to any application / type of voice calls. Can be used to obtain voice quality of calls on normal PSTN (public switched telephone network), VoLTE (Voice over Long Term Evolution), and also for any application that supports VoIP calls.
Claim 11:
The voice quality can be measured for any application that is integrated with a device’s Bluetooth i.e. over telephony network, over data network, over wifi network.
Claim 12:
No additional dialer required to make / receive the phone calls.
, Description:PATENT COMPLETE SPECIFICATION

1. TITLE OF INVENTION:
Intelligent Passive Audio Call Quality Monitoring.
2. FIELD OF INVENTION:
A. The invention relates to the field of digital mobile media delivery by providing information on user experience on audio phone calls done using a mobile device.
B. It relates to a program for providing notification and/or capturing and transmitting pertinent information from an android mobile device to capture voice quality of audio calls.
3. RELATED APPLICATION
A. In the past specialized equipment was required for testing audio quality of voice calls. In the past all methods used for testing voice quality did not have testing of actual live calls, but simulated calls This resulted in dip stick based measurements, wherein additional equipment / manpower / calls had to be made, which also would only give an estimate of approximate voice quality. In the past there was no method to measure the actual voice quality experienced by users during their normal day-to-day calls. The current invention aims at capturing voice call experience without the need for any external equipment nor the need for any additional test calls. There are three related items to this invention, however the current invention is different from these prior arts in the following ways. There are some prior arts that require user to provide feedback based on his/her interpretation of bad voice quality which is then provided to the telecom operator for analysis.
B. Differences between current invention and prior art having U.S. patent No. 5987320:
Sr. No. Prior art (U.S. Patent 5987320) Our Invention
1 User has to initiate the test. Since the application sits in the background, no user-initiated tests are required
2 Additional user voice calls are used Since the application sits in the background, no additional user voice calls are consumed to test voice quality
3 Test requires additional equipment to generate the voice signal and record the voice signal and analyse the same Test requires no additional equipment, since the sound signal is generated with the mobile itself and is merged with actual user voice and then demerged using Fast fourier transformation techniques in order to give the user a seamless experience.
4 User's experience is not captured Since the application captures voice quality of calls made by the user, actual user voice quality can be captured
C. Differences between current invention and prior art having US patent 2006/0073786
Sr. No. Prior art (U.S. Patent 2006/0073786) Our Invention
1 User has to initiate the feedback. Since the application sits in the background, no user-initiated feedback is required
2 User feedback is prone to user’s interpretation of good / bad voice quality Since the invention calculates the feedback, no variance owing to user interpretation is there.
3 User may choose to not provide any input, in which case no input is logged. Since the application sits in the background, quality of all voice calls are logged.
4 User has no access to average voice quality experienced by him/her, nor does the user have a means to know the areas / locations where he/she experiences good / poor voice quality Since the application captures voice quality of all calls made by the user, the invention can provide the user with a means to know which areas / locations the user experiences good / poor voice quality
5 User feedback is generally prone to by biased towards poor quality, since user would only mostly go out of his/her way to provide input when the quality is poor. Since the application captures voice quality of all calls made by the user, there is no bias that can corrupt the analysis of the system.
6 Pressing DTMF or another tone is required to provide feedback, which may disturb the other party while in conversation. The invention uses fast fourier transformation algorithm to bifurcate any additional signal generated to ensure there is no disturbance to either party during their call
7 Since user inputs are required, the system is prone to human error Since there are no user inputs required, the system is not prone to human error

D. Differences between current invention and prior art having U.S. patent No. US2006/0252376:
Sr. No. Prior art (U.S. Patent 5987320) Our Invention
1 User has to initiate the test. Since the application sits in the background, no user initiated tests are required
2 Additional user voice calls are used Since the application sits in the background, no additional user voice calls are consumed to test voice quality
3 Test requires additional equipment to generate the voice signal and record the voice signal and analyse the same Test requires no additional equipment, since the sound signal is generated with the mobile itself and is merged with actual user voice and then demerged using Fast fourier transformation techniques in order to give the user a seamless experience.
4 User's experience is not captured Since the application captures voice quality of calls made by the user, actual user voice quality can be captured

3. BACKGROUND OF INVENTION & PRIOR ART:
A. In the past there were devices which could be used to test voice quality by playing a pre-defined speech sample which is then recorded at the other end and then compared with the original speech sample to check for the degradation in the voice quality over the call. There were many algorithms used to check for the voice quality wherein the original speech sample is compared with the recorded speech sample to obtain the relevant voice quality.
B. However, these devices / products suffered from the following drawbacks:
i. The devices which supported such functionality had to be rooted devices, which makes it out of the reach of general mobile phone users
ii. The application generated voice calls with pre-defined speech samples which meant additional voice calls are made to test the voice quality
iii. Real time testing of voice quality of live calls made / received by the user couldn’t be done with prior arts.
iv. Additional cost had to be borne by users in terms of telecom cost to make additional calls for testing voice quality.
Sr. No. Current applications Our Invention
1 User has to initiate the test. Since the application sits in the background, no user initiated tests are required
2 Additional user Telecom voice minutes are consumed Since the application sits in the background, no additional user voice minutes are consumed to test the audio call experience
3 Test is limited by only rooted devices Test can be done by any device freely available in the market
4 User's experience is not captured Since the application captures audio call experience of all user calls, actual user experience can be captured

4. OBJECT OF INVENTION:
A. To accurately capture audio call quality experience of users using any Bluetooth based programmable phone. Since the app sits in the background, it scores over current available android application in the following areas:
i. No rooting of devices required. General mobile phones with Bluetooth capability is all that is required
ii. No requirement of generating voice calls. Users personal use voice calls can be analyzed
iii. Not a dipstick measurement. Instead all calls made by the user can be analyzed for voice quality
iv. No additional cost has to be borne by users to test the quality of their voice calls
v. Actual user experience is captured

5. SUMMARY OF THE INVENTION
A. The present invention relates to a mobile application of a phone with Bluetooth capability to capture audio call quality experience of users. It emulates Bluetooth connection, generates a test signal, combines the test signal with user voice, splits the combined signal to obtain only the information of the generated signal and then analysis the information of the generated signal. It captures actual user experience of all audio calls made using devices with Bluetooth capability.
B. The present invention comprises not to make any phone call or play any pre-defined speech sample to capture audio call experience. The application accurately captures audio call quality experience of users. Since the application sits in the background, its serves better than the current available android applications which need to root a handset, initiate tests, make additional voice calls to test audio call experience.
6. BRIEF DESCRIPTION OF THE DRAWINGS

A. For the purpose of illustrating the invention, the drawings show a form of the invention that is presently preferred. However, it should be understood that this invention is not limited to the precise arrangements and instrumentalities shown in the drawings.
B. FIG. 1 schematically depicts a system and its alternates according to the present invention for transmitting information and data by using the Application.
C. Application is made with an object to capture actual user experience of audio call quality by emulates Bluetooth connection, generates a test signal, combines the test signal with user voice, splits the combined signal to obtain only the information of the generated signal and then analysis the information of the generated signal as depicted in flow chart hereinabove.

7. DETAILED DESCRITION OF THE INVENTION WITH THE DRAWING/EXAMPLES
A. Referring to FIG. 1, wherein the flow chart of the present invention is shown. The invention is directed to an application for phones having Bluetooth capability being base on technology of 2G/3G/4G/Wifi . It is also contemplated that the App may be part of a system that includes software located remotely from the mobile device and that interacts with the App for providing some of the functions described herein. In one operative configuration, the App is stored and operates on the mobile device. The App may be stored in fixed or removable memory in the mobile device, such as a flashcard, memory stick or in resident memory, EPROM, RAM, ROM or other form of internal memory.
B. The App may be loaded onto the device either through a wired or wireless connection, such as via a download from a remote storage location, like a server hosting a site with multiple applications, e.g., the google Play Store, that are downloaded by user.
C. The algorithm aims to capture actual user experience during audio call to provide users as well as the relevant Telecom service providers (TSP), the details of user experience during audio call. Along with this, since the application sits on-device, it can also provide the TSP, technical parameters such as technology in use during user experience (2G/3G/4G/WiFi), coverage level (RxLev/RSCP/RSRP/dBM),Quality(RxQual/EcNo/RSRQ/SINR) so that the TSP can take necessary steps to improve any adverse user experiences.
D. Owing to the openness of the Android Operating System, this algorithm currently can only be used to capture audio calling experience of audio calls using an Android based phone (version 4.4 and later). The data points required for this algorithm are as given below :
i. Phone state
ii. Emulation of Bluetooth

E. The operating system on Apple phones (iPhones/ ipads), has restrictions in terms of open APIs available, which do not provide emulation of bluetooth which is required as inputs to execute our algorithm. However Apple provides update to their operating system (iOS) for their iPhones on a regular basis. If and when such APIs are made publicly available in the future, our algorithm will be able to capture audio call quality experience of users making audio calls using any iOS device.
F. The real time audio call quality algorithm uses a combination of
i. Bluetooth emulation
ii. Signal generation
iii. Signal combination
iv. Signal Split
v. Signal analysis
G. For this system to work, one needs to have our application installed on both sides of the call, viz the calling party (A-side) and the called party (B-side). Both the phones need to be running an operating system which supports Bluetooth emulation / additional programmable Bluetooth device to be connected via Bluetooth to phone.
1) Both applications will run a frequency generating code which will be based on the following formula:
a. Setup frequency: 20 Hz. Playing and stopping every alternate second
b. Run frequency: (25 * sin(t))+225 calculated every .25 seconds
2) Application can understand its sitting on A-side phone or B-side phone from the below sequence of events:
a. If phone state changes from idle to off-hook, the application is sitting on A-side
b. If phone state changes from ringing to off-hook, the application is sitting on B-side
3) A-side makes a call
4) A-side application emulates Bluetooth connection / (external Bluetooth device takes over the call) and takes over the speaker and mike. The speech by the A-side phone is now being sent to the A-side application (via Bluetooth emulation) / A-side Bluetooth device.
5) A-side application listens to input from the network (received via Bluetooth) and splits the signal into < 50Hz. Frequency and >50 Hz. frequencies using FFT.
6) A-side application analyses < 50 Hz. Frequency to check for 500 msec.
7) If the A-side application doesn’t get a positive 20 Hz. Frequency for 500 Msec., then A-side application will continue scanning for maximum 65 seconds from call-setup and then close the Bluetooth emulation and handover call directly to the phone / Bluetooth device.
8) If the A-side application gets a positive 20 Hz. Signal for 500 msec., then A-side application will start generating setup frequency as per point 1 above for 1 second and then shift to run frequency for the duration of the call.
9) B-side receives the call
10) B-side application emulates Bluetooth connection / (external Bluetooth device) and takes over the speaker and mike. The speech by the B-side phone is now being sent to the B-side application (via Bluetooth emulation) / B-side bluetooth device
11) B-side application / device inserts the setup frequency generated by the formula given in point 1 above.
12) B-side then sends the combined voice and signal to the phone for transmitting over the cellular network
13) B-side application listens to input from the network (received via Bluetooth) and splits the signal into < 50Hz. Frequency and >50 Hz. frequencies using FFT.
14) B-side application analyses < 50 Hz. Frequency to check for 500 msec.
15) If the B-side application doesn’t get a positive 20 Hz. Frequency for 500 Msec., then B-side application will continue scanning for maximum 5 seconds from call-setup and then close the Bluetooth emulation and handover call directly to the phone / Bluetooth device.
16) If the B-side application gets a positive 20 Hz. Signal for 500 msec., then B-side application will start generating run frequency as per point 1 above until end of call.
17) Both applications at this point will be generating frequency as per point 1 above and combining the signals with the voice signals obtained from the mic of their individual device and sending it on the telecom network.
18) Both applications will now take input from the telecom network (obtained via Bluetooth), split the frequency into < 275Hz. and >275 Hz. The sound stream >275 Hz. will be played to the respective speakers
19) The sound stream < 275 Hz. will be processed as given below:

a. Break the signal into samples of 4.44msec. each
b. For each sample record the max amplitude (peak) as an integer between 0-100. For the purpose of this the sound wave can be pulse code modulated with a sampling frequency of 5Khz.
c. Compute Avg : average of all peaks
d. Take 25% of the average as the “Cut-off”
e. Count number of peaks greater than “Cut-off” : good-peaks
f. Count total number of peaks : Total-peaks
g. Comput e Percentage as “Good-peaks / Total-peaks”
i. Good Voice quality : Percentage > 80%
ii. Average Voice quality : Percentage < 80% and > 60%
iii. Poor Voice quality : Percentage < 60%

H. The present invention may be embodied in other specific forms without departing from the spirit or essential attributes and, accordingly, reference should be made to the appended claims, rather than to the foregoing specification, as indicating the scope of the invention

8. STATEMENT OF INVENTION:
The application captures audio voice quality experience of users using any device for making calls. Since the app sits in the background, it is better than the current available applications where user is not required to initiate test, no additional voice calls are made to test the audio call experience, users can capture, monitor and measure the number of mutes, overall voice quality. The existing applications equally comes with the drawbacks like user has to initiate the test, Additional user voice minutes are consumed, Test is limited to only certain types of handsets.
9. ABSTRACT
An Application for analyzing voice quality of phone calls, which aims to capture actual user experience during voice calls to provide users as well as the relevant telecom service providers (TSP), the details of user experience during voice calls.
The application combines voice signal analysis with dynamic thresholds for each voice call to be able to distinguish between what the user will perceive as mute and what is just low volume.
Along with this, since the application sits on-device, it can also provide the TSP, technical parameters such as technology in use during user experience (2G/3G/4G),coverage level (RxLev/RSCP/RSRP), Quality (RxQual/EcNo/RSRQ/SINR) so that the TSP can take necessary steps to improve any adverse user experiences
10. BEST METHOD
There are two methods to capture the voice signal for analysis.
1) Emulate Bluetooth connection on the device itself
2) Have an external programmable Bluetooth device which can be connected to the device and control the voice received and sent from the device.
Since emulating Bluetooth is only possible on certain Android devices, the best method for running the given algorithm would be to have an external programmable Bluetooth device which can play sounds and receive the played sound and analyses the same.

11. CLAIM
We Claim:
Claim1:
A method where voice quality of normal calls between users can be monitored and measured. It captures all voice call related experiences, such as muting, poor quality for both downlink and uplink.
Claim 2:
No additional voice calls required to test voice call quality.
Claim 3:
Can be implemented with and without external hardware. It does not require any additional servers to do analysis.
Claim 4:
Adheres to full privacy of user speech. No recording of user speech.
Claim 5:
Analysis done in real time to provide real time inputs.
Claim 6:
Details of coverage, location, telecom network status is linked to poor voice quality which can help telecom service providers (TSPs) to improve their network.
Claim 7:
No data sent to any servers, hence no possibility of compromise of user privacy / users private data.
Claim 8:
No need to call specific switch numbers with specific voice playback to do analysis
Claim 9:
No interference to user’s normal voice call, no inputs from user is required.
Claim 10:
Not restricted to any application / type of voice calls. Can be used to obtain voice quality of calls on normal PSTN (public switched telephone network), VoLTE (Voice over Long Term Evolution), and also for any application that supports VoIP calls.
Claim 11:
The voice quality can be measured for any application that is integrated with a device’s Bluetooth i.e. over telephony network, over data network, over wifi network.
Claim 12:
No additional dialer required to make / receive the phone calls.