ENHANCING AUDIO USING A MOBILE DEVICE

• US 20160035337A1
• Filed: 08/01/2014
• Published: 02/04/2016
• Est. Priority Date: 08/01/2013
• Status: Active Grant

• Current Assignee: Caavo Inc
• Original Assignee: Snap Networks Pvt Ltd (SenSen Networks Ltd.)
• Inventors: Satheesh, Sharath Hariharpur, Aggarwal, Ashish Dharmpal
• Granted Patent: US 9,565,497 B2
• Time in Patent Office: Days
• US Class Current: 1/1
• CPC Class Codes: G06F 3/165 Management of the audio stream, e.g. setting of volume, audio stream path
• CPC Class Codes: G10L 21/0232 Processing in the frequency domain
• CPC Class Codes: G10L 25/51 for comparison or discrimination
• CPC Class Codes: H03G 3/20 Automatic control H03G3/005 takes precedence; combined with volume compression or expansion H03G7/00
• CPC Class Codes: H03G 3/3005 in amplifiers suitable for low-frequencies, e.g. audio amplifiers H03G3/32, H03G3/34 take precedence
• CPC Class Codes: H03G 3/32 the control being dependent upon ambient noise level or sound level
• CPC Class Codes: H03G 5/005 of digital signals
• CPC Class Codes: H03G 5/165 Equalizers; Volume or gain control in limited frequency bands
• CPC Class Codes: H03G 7/002 in untuned or low-frequency amplifiers, e.g. audio amplifiers H03G7/007, H03G7/001, H03G7/008, H03G7/02, H03G7/06 take precedence
• CPC Class Codes: H03H 17/04 Recursive filters
• CPC Class Codes: H03H 2017/0472 based on allpass structures
• CPC Class Codes: H04R 1/08 Mouthpieces; Microphones; Attachments therefor
• CPC Class Codes: H04R 2430/01 Aspects of volume control, not necessarily automatic, in sound systems
• CPC Class Codes: H04R 2499/11 Transducers incorporated or for use in hand-held devices, e.g. mobile phones, PDA's, camera's
• CPC Class Codes: H04R 25/50 Customised settings for obtaining desired overall acoustical characteristics
• CPC Class Codes: H04R 25/505 using digital signal processing
• CPC Class Codes: H04R 29/00 Monitoring arrangements; Testing arrangements for hearing aids H04R25/30; detection of loudspeaker connection H04R5/04; sound-field adaptation dependent on speaker detection H04S7/308
• CPC Class Codes: H04R 29/001 for loudspeakers H04R29/007 takes precedence
• CPC Class Codes: H04R 29/004 for microphones H04R29/007 takes precedence
• CPC Class Codes: H04R 3/00 Circuits for transducers , loudspeakers or microphones
• CPC Class Codes: H04R 3/04 : for correcting frequency response

Abstract

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

Claims

1. A method for improving audio quality of an audio system comprising of a source device and at least one rendering device using a mobile device, the method comprising of

improving quality of a source audio signal by the mobile device, if the quality of the source signal is below a quality threshold, wherein the source audio signal is being played on the audio system;

calibrating the at least one rendering device by the mobile device; and

compensating for ambient noise by the mobile device.

2. The method, as claimed in claim 1, wherein the method further comprises of obtaining frequency characteristics of at least one microphone associated with the mobile device.

3. The method, as claimed in claim 2, wherein obtaining frequency characteristics of the at least one microphone associated with the mobile device comprises of

communicating a test signal to the source device by the mobile device;

capturing the test signal by the at least one microphone, on the source device playing the test signal through the rendering device;

determining impulse response of the captured test signal by the mobile device;

inverting frequency characteristics of the impulse response by the mobile device;

determining a microphone equalizing filter by the mobile device using the impulse response and the inverted frequency characteristics of the impulse response; and

using the microphone equalizing filter to compensate for frequency characteristics of the at least one microphone by the mobile device.

4. The method, as claimed in claim 1, wherein determining the quality of the source audio signal by the mobile device comprises of

identifying a detected source audio signal by the mobile device;

fetching a small portion of a high quality audio signal of the identified source audio signal by the mobile device;

time aligning the fetched high quality audio signal and the detected source audio signal by the mobile device;

calculating a metric related to difference between the fetched high quality audio signal and the detected source audio signal using

Sq=(H*H+O*O)/(H−

O)*(H−

O)

where H is the fetched high quality audio signal, O is the detected source audio signal and Sq is quality of the detected source audio signal; and

comparing the metric to the quality threshold by the mobile device to determine the quality of the source audio signal.

5. The method, as claimed in claim 1, wherein improving the quality of the source audio signal comprises of

detecting a high quality source of the source audio signal by the mobile device;

streaming the high quality source of the source audio signal by the mobile device; and

transmitting the high quality source of the audio stream to the source device by the mobile device.

6. The method, as claimed in claim 1, wherein calibrating the at least one rendering device by the mobile device comprises of

obtaining absolute angles of location of the at least one rendering device in terms of Cartesian coordinates with respect to a reference position by the mobile device;

determining relative distances of the rendering devices by the mobile device;

capturing a calibration signal by the mobile device, wherein the calibration signal is sent to the rendering device by the mobile device;

calculating an impulse response for the calibration signal by the mobile device, wherein the impulse response is a ratio of the FFT (Fast Fourier Transform) of the calibration signal to the FFT of the received calibration signal;

calculating a crossover filter using the impulse response by the mobile device, wherein the crossover filter is a fourth order Butterworth filter;

determining loudness of the rendering device using the impulse response by the mobile device;

determining a loudness compensation by computing an average of magnitude of all frequency responses for the rendering device by the mobile device; and

correcting response of the rendering devices by the mobile device.

7. The method, as claimed in claim 6, wherein determining the relative distances of the rendering devices by the mobile device further comprises of

capturing a reference signal by the mobile device, wherein the reference signal is provided by the mobile device and is played by the source device;

and

determining the relative distances of the rendering devices by the mobile device by calculating delays in the captured reference signal.

8. The method, as claimed in claim 6, wherein cut-oil of the crossover filter is based on the frequency response of the impulse response.

9. The method, as claimed in claim 6, wherein determining loudness of the rendering device using the impulse response comprises of calculating energy of the impulse response.

10. The method, as claimed in claim 6, wherein correcting response of the rendering devices by the mobile device comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters by the mobile device;

computing a finite impulse response (FIR) filter by the mobile device, wherein the FIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the FIR filter through a set of all-pass filters by the mobile device.

11. The method, as claimed in claim 6, wherein correcting response of the rendering devices by the mobile device comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters by the mobile device;

computing a infinite impulse response (IIR) filter by the mobile device, wherein the IIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the IIR filter through a set of all-pass filters by the mobile device.

12. The method, as claimed in claim 6, wherein correcting response of the rendering devices by the mobile device comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters by the mobile device;

computing a combination filter by the mobile device, wherein the combination filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale and the combination filter is a combination of FIR and IIR filters; and

inverting non-linear frequency scale to yield the final equalization filter by passing the combination filter through a set of all-pass filters by the mobile device.

13. The method, as claimed in claim 6, wherein the method further comprises of synchronizing a plurality of rendering devices in time by

calculating a delay between broadcast of the calibration signal by the mobile device, wherein the delay is the delay between broadcast of the calibration signal from each of the plurality of rendering devices and receipt of the calibration signal at the microphone;

and

determining a delay compensation filter by subtracting the delay from a pre-determined maximum delay by the mobile device by the mobile device.

14. The method, as claimed in claim 1, wherein compensating for ambient noise by the mobile device comprises of

measuring ambient noise characteristics by the mobile device, on the mobile device detecting a period of silence in the source audio signal;

measuring loudness by the mobile device;

calculating frequency response by the mobile device using FFT (Fast Fourier Transform) of the ambient noise characteristics;

calculating a digital filter by the mobile device using inverse of the calculated frequency response; and

estimating an optimum volume level by the mobile device based on the ambient noise characteristics.

15. The method, as claimed in claim 14, wherein ambient noise characteristics comprises of loudness level of noise and frequency characteristics.

16. The method, as claimed in claim 14, wherein a-weights are used to measure loudness by the mobile device.

17. A computer program product comprising computer executable program code recorded on a computer readable non-transitory storage medium, said computer executable program code when executed, causing a method for improving audio quality of an audio system comprising of a source device and at least one rendering device, the method comprising of

improving quality of a source audio signal, if the quality of the source signal is below a quality threshold, wherein the source audio signal is being played on the audio system;

calibrating the at least one rendering device; and

compensating for ambient noise.

18. The computer program product, as claimed in claim 17, wherein the method further comprises of obtaining frequency characteristics of at least one microphone.

19. The computer program product, as claimed in claim 18, wherein obtaining frequency characteristics of the at least one microphone comprises of

communicating a test signal to the source device;

capturing the test signal by the at least one microphone, on the source device playing the test signal through the rendering device;

determining impulse response of the captured test signal;

inverting frequency characteristics of the impulse response;

determining a microphone equalizing filter using the impulse response and the inverted frequency characteristics of the impulse response; and

using the microphone equalizing filter to compensate for frequency characteristics of the at least one microphone.

20. The computer program product, as claimed in claim 17, wherein determining the quality of the source audio signal comprises of

identifying a detected source audio signal;

fetching a small portion of a high quality audio signal of the identified source audio signal;

time aligning the fetched high quality audio signal and the detected source audio signal;

calculating a metric by the mobile device, wherein the metric is related to difference between the fetched high quality audio signal and the detected source audio signal using

Sq=(H*H+O*O)/(H−

O)*(H−

O)

where H is the fetched high quality audio signal, O is the detected source audio signal and Sq is quality of the detected, source audio signal; and

comparing the metric to the quality threshold to determine the quality of the source audio signal.

21. The computer program product, as claimed in claim 17, wherein improving the quality of the source audio signal comprises of

detecting a high quality source of the source audio signal;

streaming the high quality source of the source audio signal; and

transmitting the high quality source of the audio stream to the source device.

22. The computer program product, us claimed in claim 17, wherein calibrating the at least one rendering device comprises of

obtaining absolute angles of location of the at least one rendering device in terms of Cartesian coordinates with respect to a reference position;

determining relative distances of the rendering devices;

capturing a calibration signal, wherein the calibration signal is sent to the rendering device;

calculating an impulse response for the calibration signal, wherein the impulse response is a ratio of the FFT (Fast Fourier Transform) of the calibration signal to the FFT of the received calibration signal;

calculating a crossover filter using the impulse response, wherein the crossover filter is a fourth order Butterworth filter;

determining loudness of the rendering device using the impulse response;

determining a loudness compensation by computing an average of magnitude (wall frequency responses for the rendering device; and

correcting response of the rendering devices.

23. The computer program product, as claimed in claim 22, wherein determining the relative distances of the rendering devices further comprises of

capturing a reference signal, wherein the reference signal is provided and is played by the source device;

and

determining the relative distances of the rendering devices by calculating delays in the captured reference signal.

24. The computer program product, as claimed in claim 22, wherein cut-off of the crossover filter is based on the frequency response of the impulse response.

25. The computer program product, as claimed in claim 22, wherein determining loudness of the rendering device using the impulse response comprises of calculating energy of the impulse response.

26. The computer program product, as claimed in claim 22, wherein correcting response of the rendering devices comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters;

computing a finite impulse response (FIR) filter, wherein the FIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the FIR filter through a set of all-pass filters.

27. The computer program product, as claimed in claim 22, wherein correcting response of the rendering devices comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters;

computing a infinite impulse response (IIR) filter, wherein the IIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the IIR filter through a set of all-pass filters.

28. The computer program product, as claimed in claim 22, wherein correcting response of the rendering devices comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters;

computing a combination filter, wherein the combination filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale and the combination filter is a combination of FIR and IIR filters; and

inverting non-linear frequency scale to yield the final equalization filter by passing the combination filter through a set of all-pass filters.

29. The computer program product, as claimed in claim 22, wherein the method further comprises of synchronizing a plurality of rendering devices in time by calculating a delay between broadcast of the calibration signal, wherein the delay is the delay between broadcast of the calibration signal from each of the plurality of rendering devices and receipt of the calibration signal at the microphone;

and

determining a delay compensation filter by subtracting the delay from a pre-determined maximum delay.

30. The computer program product, as claimed in claim 17, wherein compensating liar ambient noise comprises of:

measuring ambient noise characteristics, on the mobile device detecting a period of silence in the source audio signal;

measuring loudness;

calculating frequency response using FFT (Fast Fourier Transform) of the ambient noise characteristics;

calculating a digital filter using inverse of the calculated frequency response; and

estimating an optimum volume level based on the ambient noise characteristics.

31. The computer program product, as claimed in claim 30, wherein ambient noise characteristics comprises of loudness level of noise and frequency characteristics.

32. The computer program product, as claimed in claim 30, wherein a-weights are used to measure loudness.

33. A method for obtaining frequency characteristics of at least one microphone associated with a mobile device, wherein the method comprises of

communicating a test signal to a source device by the mobile device;

capturing the test signal by the at least one microphone, on the source device playing the test signal through a rendering device;

determining impulse response of the captured test signal by the mobile device;

inverting frequency characteristics of the impulse response by the mobile device;

determining a microphone equalizing filter by the mobile device using the impulse response and the inverted frequency characteristics of the impulse response; and

using the microphone equalizing filter to compensate for frequency characteristics of the at least one microphone by the mobile device.

34. A computer program product comprising computer executable program code recorded on a computer readable non-transitory storage medium, said computer executable program code when executed, causing a method for obtaining frequency characteristics of at least one microphone associated with the computer program product, wherein the method comprises of

communicating a test signal to a source device;

capturing the test signal by the at least one microphone, on the source device playing the test signal through a rendering device;

determining impulse response of the captured test signal;

inverting frequency characteristics of the impulse response;

determining a microphone equalizing filter using the impulse response and the inverted frequency characteristics of the impulse response; and

using the microphone equalizing filter to compensate for frequency characteristics of the at least one microphone.

35. A method for improving audio quality of an audio system comprising of a source device and at least one rendering device using a mobile device by improving quality of source audio signal by the mobile device, if the quality of the source signal is below a quality threshold, wherein the source audio signal is being played on the audio system;

wherein the method further comprises of

identifying a detected source audio signal by the mobile device;

fetching a small portion of a high quality audio signal of the identified source audio signal by the mobile device;

time aligning the fetched high quality audio signal and the detected source audio signal by the mobile device;

calculating a metric related to difference between the fetched high quality audio signal and the detected source audio signal using

Sq−

(H*H+O*O)/(H−

O)*(H−

O)

where H is the fetched high quality audio signal, O is the detected source audio signal and Sq is quality of the detected source audio signal; and

comparing the metric to the quality threshold by the mobile device to determine the quality of the source audio signal.

36. The method, as claimed in claim 35, wherein the method further comprises of

detecting a high quality source of the source audio signal by the mobile device;

streaming the high quality source of the source audio signal by the mobile device; and

transmitting the high quality source of the audio stream to the source device by the mobile device.

37. A computer program product comprising computer executable program code recorded on a computer readable non-transitory storage medium, said computer executable program code when executed, causing a method for improving audio/quality of an audio system comprising of a source device and at least one rendering device using a mobile device by improving quality of a source audio signal, if the quality of the source signal is below a quality threshold, wherein the source audio signal is being played on the audio system:

wherein the method further comprises of

identifying a detected source audio signal;

fetching a small portion of a high quality audio signal of the identified source audio signal;

time aligning the fetched high quality audio signal and the detected source audio signal;

calculating a metric related to difference between the fetched high quality audio signal and the detected source audio signal using

Sq−

(H*H+O*O)/(H−

O)*(H−

O)

where H is the fetched high quality audio signal, O is the detected source audio signal and Sq is quality of the detected source audio signal; and

comparing the metric to the quality threshold to determine the quality of the source audio signal.

38. The computer program product, as claimed in claim 37, wherein the method further comprises of

detecting a high quality source of the source audio signal;

streaming the high quality source of the source audio signal; and

transmitting the high quality source of the audio stream to the source device.

39. A method for improving audio quality of an audio system comprising of a source device and at least one rendering device using a mobile device by calibrating the at least one rendering device by the mobile device, wherein the method further comprises of

obtaining absolute angles of location of the at least one rendering device in terms of Cartesian coordinates with respect to a reference position by the mobile device;

determining relative distances of the rendering devices by the mobile device;

capturing a calibration signal by the mobile device, wherein the calibration signal is sent to the rendering device by the mobile device;

calculating an impulse response for the calibration signal by the mobile device, wherein the impulse response is a ratio of the FFT (Fast Fourier Transform) of the calibration signal to the FFT of the received calibration signal;

calculating a crossover filter using the impulse response by the mobile device, wherein the crossover filter is a fourth order Butterworth filter;

determining loudness of the rendering device using the impulse response by the mobile device;

determining a loudness compensation by computing an average of magnitude or all frequency responses for the rendering device by the mobile device; and

correcting response of the rendering devices by the mobile device.

40. The method, as claimed in claim 39, wherein determining the relative distances of the rendering devices by the mobile device further comprises of

capturing a reference signal by the mobile device, wherein the reference signal is provided by the mobile device and is played by the source device;

and

determining the relative distances of the rendering devices by the mobile device by calculating delays in the captured reference signal.

41. The method, as claimed in claim 39, wherein cut-off of the crossover filter is bused on the frequency response of the impulse response.

42. The method, as claimed in claim 39, wherein determining loudness of the rendering device using the impulse response comprises of calculating energy of the impulse response.

43. The method, as claimed in claim 39, wherein correcting response of the rendering devices by the mobile device comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters by the mobile device;

computing a finite impulse response (FIR) filter by the mobile device, wherein the FIR filter is a minimum phase fitter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the FIR filter through a set of all-pass filters by the mobile device.

44. The method, as claimed in claim 39, wherein correcting response of the rendering devices by the mobile device comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters by the mobile device;

computing a infinite impulse response (IIR) filter by the mobile device, wherein the IIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the IIR filter through a set of all-pass filters by the mobile device.

45. The method, as claimed in claim 39, wherein correcting response of the rendering devices by the mobile device comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters by the mobile device;

computing a combination filter by the mobile device, wherein the combination filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale and the combination filter is a combination of FIR and IIR filters; and

inverting non-linear frequency scale to yield the final equalization filter by passing the combination filter through a set of all-pass filters by the mobile device.

46. The method, as claimed in claim 39, wherein the method further comprises of synchronizing a plurality of rendering devices in time by

calculating a delay between broadcast of the calibration signal by the mobile device, wherein the delay is the delay between broadcast of the calibration signal from each of the plurality of rendering devices and receipt of the calibration signal at the microphone;

and

determining a delay compensation filter by subtracting the delay from a pre-determined maximum delay by the mobile device by the mobile device.

47. A computer program product comprising computer executable program code recorded on a computer readable non-transitory storage medium, said computer executable program code when executed, causing a method for improving audio quality of an audio system comprising of a source device and at least one rendering device by calibrating the at least one rendering device;

wherein the method further comprises of

obtaining absolute angles of location of the at least one rendering device in terms of Cartesian coordinates with respect to a reference position;

determining relative distances of the rendering devices;

capturing a calibration signal, wherein the calibration signal is sent to the rendering device;

calculating an impulse response for the calibration signal, wherein the impulse response is a ratio of the FFT (Fast Fourier Transform) of the calibration signal to the FFT of the received calibration signal;

calculating a crossover filter using the impulse response, wherein the crossover filter is a fourth order Butterworth filter;

determining loudness of the rendering device using the impulse response;

determining a loudness compensation by computing an average of magnitude of all frequency responses for the rendering device; and

correcting response of the rendering devices.

48. The computer program product, as claimed in claim 47, wherein determining the relative distances of the rendering devices further comprises of

capturing a reference signal, wherein the reference signal is provided and is played by the source device;

and

determining the relative distances of the rendering devices by calculating delays in the captured reference signal.

49. The computer program product, as claimed in claim 47, wherein cut-off or the crossover filter is based on the frequency response of the impulse response.

50. The computer program product, as claimed in claim 47, wherein determining loudness of the rendering device using the impulse response comprises of calculating energy of the impulse response.

51. The computer program product, as claimed in claim 47, wherein correcting response of the rendering devices comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters;

computing a finite impulse response (FIR) filter, wherein the FIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the FIR filter through a set of all-pass filters.

52. The computer program product, as claimed in claim 47, wherein correcting response of the rendering devices comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters;

computing a infinite impulse response (IIR) filter, wherein the IIR filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale; and

inverting non-linear frequency scale to yield the final equalization filter by passing the IIR filter through a set of all-pass filters.

53. The computer program product, as claimed in claim 47, wherein correcting response of the rendering devices comprises of

mimicking non-linear frequency scale of a human auditory system by passing the impulse response through a set of all-pass filters;

computing a combination filter, wherein the combination filter is a minimum phase filter whose magnitude response is inverse of the mimicked non-linear frequency scale and the combination filter is a combination of FIR and IIR filters; and

inverting non-linear frequency scale to yield the final equalization filter by passing the combination filter through a set of all-pass filters.

54. The computer program product, as claimed in claim 47, wherein the method further comprises of synchronizing a plurality of rendering devices in time by

calculating a delay between broadcast of the calibration signal, wherein the delay is the delay between broadcast of the calibration signal from each of the plurality of rendering devices and receipt of the calibration signal at the microphone;

and

determining a delay compensation filter by subtracting the delay from a pre-determined maximum delay.

55. A method for improving audio quality of an audio system comprising of a source device and at least one rendering device using a mobile device by compensating for ambient noise;

wherein the method further comprises of

measuring ambient noise characteristics by the mobile device, on the mobile device detecting a period of silence in the source audio signal;

measuring loudness by the mobile device;

calculating frequency response by the mobile device using FFT (Fast Fourier Transform) of the ambient noise characteristics;

calculating a digital filter by the mobile device using inverse of the calculated frequency response; and

estimating an optimum volume level by the mobile device based on the ambient noise characteristics.

56. The method, as claimed in claim 55, wherein ambient noise characteristics comprises of loudness level of noise and frequency characteristics.

57. The method, as claimed in claim 55, wherein a-weights are used to measure loudness by the mobile device.

58. A computer program product comprising computer executable program code recorded on a computer readable non-transitory storage medium, said computer executable program code when executed, causing a method for improving audio quality of an audio system comprising of a source device and at least one rendering device by compensating for ambient noise:

wherein the method further comprises of

measuring ambient noise characteristics, on detecting a period of silence in the source audio signal;

measuring loudness;

calculating frequency response using FFT (Fast Fourier Transform) of the ambient noise characteristics;

calculating a digital filter using inverse of the calculated frequency response; and

estimating an optimum volume level based on the ambient noise characteristics.

59. The computer program product, as claimed in claim 58, wherein ambient noise characteristics comprises of loudness level of noise and frequency characteristics.

60. The computer program product, as claimed in claim 58, wherein a-weights are used to measure loudness.