Method and apparatus for mode balance for analog FM, digital, radio blend logic in an automotive environment

Abstract

A radio includes a first tuner and a second tuner. A processor compares a first perceivable volume level of a station tuned by the first tuner to at least one second perceivable volume level of at least one background station tuned by the second tuner. The processor enables automatic volume knob changes using a pre-calibrated lookup table that associates a volume step of the volume knob with a difference between the first perceivable volume level and the second perceivable volume level.

Claims

1. A radio, comprising: a first tuner and a second tuner; and a processor configured to: compare a first volume level of a station tuned by the first tuner to at least one second volume level of at least one background station tuned by the second tuner; and enable automatic volume knob changes using a pre-calibrated lookup table that associates a volume step of the volume knob with a difference between the first volume level and the second volume level.

2. The radio of claim 1, wherein the automatic volume knob changes comprise volume increases or volume decreases depending upon an existing volume level of the radio.

3. The radio of claim 1, wherein the station tuned by the first tuner is an analog station.

4. The radio of claim 1, wherein the background station is an analog station.

5. The radio of claim 1, comprising temporally computing an average energy of the station tuned by the first tuner, the first volume level of the station tuned by the first tuner being determined dependent upon the temporally computed average energy.

6. The radio of claim 1, comprising temporally computing an average energy of the background station, the second volume level of the background station being determined dependent upon the temporally computed average energy.

7. The radio of claim 1, wherein the processor is configured to implement the automatic volume knob changes in association with switching from audibly playing the station tuned by the first tuner to audibly playing the background station.

8. A radio, comprising: a first tuner and a second tuner; and a processor configured to: compare a first volume level of a station tuned by the first tuner to at least one second volume level of at least one background station tuned by the second tuner; perform a volume change dependent upon a difference between the first volume level and the second volume level; and temporally compute an average energy of the station tuned by the first tuner, the first volume level of the station tuned by the first tuner being determined dependent upon the temporally computed average energy.

9. The radio of claim 8, wherein the processor is configured to enable automatic volume knob changes using a pre-calibrated lookup table that associates a volume step of the volume knob with a difference between the first volume level and the second volume level.

10. The radio of claim 8, wherein the station tuned by the first tuner is an analog station.

11. The radio of claim 8, wherein the background station is an analog station.

12. The radio of claim 8, wherein the first volume level comprises a first perceivable volume level.

13. The radio of claim 8, wherein the processor is configured to temporally compute an average energy of the background station, the second volume level of the background station being determined dependent upon the temporally computed average energy.

14. The radio of claim 8, wherein the second volume level comprises a second perceivable volume level.

15. A radio, comprising: a first tuner and a second tuner; and a processor configured to: compare a first volume level of a station tuned by the first tuner to at least one second volume level of at least one background station tuned by the second tuner; enable automatic volume knob changes dependent upon a volume step of the volume knob and a difference between the first volume level and the second volume level; and temporally compute an average energy of the station tuned by the first tuner, the first volume level of the station tuned by the first tuner being determined dependent upon the temporally computed average energy.

16. The radio of claim 15, wherein the automatic volume knob changes comprise volume increases or volume decreases depending upon an existing volume level of the radio.

17. The radio of claim 15, wherein the station tuned by the first tuner and/or the background station is an analog station.

18. The radio of claim 15, wherein the processor is configured to temporally compute an average energy of the background station, the second volume level of the background station being determined dependent upon the temporally computed average energy.

19. The radio of claim 15, wherein at least one of the first volume level and the second volume level is perceivable.

20. The radio of claim 15, wherein the processor is configured to implement the automatic volume knob changes in association with switching from audibly playing the station tuned by the first tuner to audibly playing the background station.

Description

BRIEF DESCRIPTION OF THE DRAWINGS

(1) The above-mentioned and other features and objects of this invention, and the manner of attaining them, will become more apparent and the invention itself will be better understood by reference to the following description of embodiments of the invention taken in conjunction with the accompanying drawings, wherein:

(2) FIG. 1 is a flow chart illustrating known technologies.

(3) FIG. 2A is a first portion of a table illustrating processes of the prior art.

(4) FIG. 2B is a second portion of the table of FIG. 2A.

(5) FIG. 2C is a third portion of the table of FIG. 2A.

(6) FIG. 3 is a table illustrating service and related signal quality metrics of the prior art.

(7) FIG. 4 illustrates an Audio Channel Path of the present invention.

(8) FIG. 5A is a first portion of a table illustrating some cases to which the present invention may be applied.

(9) FIG. 5B is a second portion of the table of FIG. 5A.

(10) FIG. 6 shows a 512-sample audio signal frame normalized spectrum plot.

(11) FIG. 7 is a table mapping out the appropriate volume level settings that may be applied to ensure that an end user perceives a specified volume level in a digital use case.

(12) FIG. 8 is a table mapping out the appropriate volume level settings that may be applied to ensure that an end user perceives a specified volume level in an analog use case.

(13) FIG. 9 is a Munson curve.

DETAILED DESCRIPTION

(14) The embodiments hereinafter disclosed are not intended to be exhaustive or limit the invention to the precise forms disclosed in the following description. Rather the embodiments are chosen and described so that others skilled in the art may utilize its teachings.

(15) The main audio entertainment path may apply for the currently tuned audio source. The audio therein may be routed to a core which can run the volume detection algorithm. The second audio path which can apply for audio may be decoded/demodulated by background tuners and or wideband front end devices for other than the currently tuned station audio, and may be routed to volume detection to gauge the volume levels of the currently tuned audio as compared to the background neighboring station audio. This may apply to inter station transition mode balance.

(16) The background audio path can be that of analog and/or digital demodulated audio or bandwidth increase to the main audio source if it is digital to ensure that normalization is achieved in anticipation that the user will traverse or skip to it. This volume level metric along with the a priori information based on digital synchronization cues enables tagging of the station as analog and digital, and may be used to decide whether the volume step increment should be triggered automatically if the current station is analog or if the currently tuned audio is a digital source to employ bandwidth extension algorithms for any existing currently tuned audio source. This a priori information for this decision can be realized by the radio head unit.

(17) In North America, with HD IBOC there is the special situation where an analog station on center frequency can have a digital equivalent (Main Program Station). As such, in test drives in the field there may be transitions from analog to digital and from digital to analog. For example, if there is a transition from analog to digital in the case of HD IBOC radio, the radio head unit may be able to send the HD synchronization cue for it to know it is a transition from analog to digital and vice versa. This information may be needed in case a user skips from the current station to another station.

(18) The above-disclosed process may be possible due to four reasons. First, Silicon On Chip (SOC) with multi-core processors supporting additional processing power are providing enabling technology to signal processing algorithms that previously was not possible to implement.

(19) Second, software radio ASICS and SOC designs have increased processing power to support simultaneous audio demodulation for the currently tuned station and background stations, enabling the present invention.

(20) Third, software radio front ends are becoming multiband and wideband as well. What the latter means is that the front end ADC (Analog to Digital) converter can sample the entire frequency band and, depending on the processing capability of the multicore processors, background stations with potential for simultaneous audio demodulation for select stations may be sampled. This technology can especially change the capabilities of single tuner radios without the need for background tuners for background scanning functionality.

(21) Fourth, connectivity may also enable the option for onboard partitioning instead of offboard partitioning for processing power needs. Thus, the radio head unit can include a single tuner and have the algorithm to gauge the current audio source. The back end can perform processing for what used to be dedicated second or third tuner processing on the radio head unit side through an off board server keeping track of the GPS location and the potential local frequency landscape that the car is located in. The back end may also decode the audio for other stations in the vicinity using an off board server to gauge the volume metric.

(22) The invention provides a practical solution to address the use cases defined in the table of FIGS. 5A-B. The invention may be applied to dual and multi tuner environments as well as to single tuner radios.

(23) Use Case with Dedicated on Board Dual and/or Multi Tuner on the Radio Head Unit

(24) Dual or Multi Tuners enable the radio to be tuned to a particular station while the background tuners scan the other frequencies in the band. In the past, the second tuner enabled the background tuner only to do background scanning of station frequencies and did not provide audio decode capability due to a shortage in processing power of either the SOC (System On Chip) or the ASIC (Application Specific IC). The current state of the art, however, includes the capability to do audio demodulation on both the main tuner and background tuners.

(25) With the second tuner performing audio demodulation, it is possible to compare the perceived volume level of the currently tuned station to that of background stations and enable automatic volume knob changes using a pre-calibrated lookup table that associates the volume step with the difference between the perceived volume levels of the currently tuned station and the background stations. This automatic volume change can be either an increase or decrease in volume levels depending on the end user's original volume level setting. This automatic volume change can be performed for analog stations. If stations are digital, the a priori information can be used to ensure bandwidth normalization of the current tuned station using the bandwidth of neighboring stations.

(26) As an example, sampled audio of the analog demodulated audio from background tuners may be decomposed into N number of subbands using PQMF (Psuedo Quadrature Mirror Filters). The value N depends on the band being used. In the case of analog FM where the bandwidth is capped at 15 kHz, N is 32.

(27) In the case of FM, this can be computed using:
P(F)=21.4 log.sub.10(4.37F+1)(8)

(28) Accordingly, if there is a 15 kHz sampling rate, setting F as 7.5 kHz results in thirty-two auditory filters. For example, using the table of FIGS. 2A-B as gauge, for AM HD where the total bandwidth can be 15 kHz there may be thirty-two subbands.

(29) In all situations, the common metric of the formula below may be used. The total loudness metric as applied to demodulated FM audio for a particular volume step may thus be defined as:

(30) $\begin{array}{cc}L={}_0^{P}L\left(p\right)\mathrm{dp},\mathrm{where}p=32\mathrm{for}\mathrm{FM}.& \left(9\right)\end{array}$

(31) FIG. 6 shows a 512 sample audio signal frame normalized spectrum plot. FIG. 6 is a Welch power spectral density estimate as a plot of power/frequency (dB/Hz) vs. normalized frequency: f/(Fs/2). The darkest line represents the filtered response of the masking threshold level. The dashed line represents the masking threshold. The darkest line can in turn be calibrated to take into account the car cabin environment noise through a microphone input. The lighter solid line represents that signal for this particular 512 sample frame, which is audible to the end user and is above the darkest line.

(32) The present invention may make use of the (ISO/CEI norm 11172-3:1993 F) MPEG1 psychoacoustic model in Matlab. MPEG 1 may use a 512 sample frame from an input signal, and may use the psychoacoustic model to compute a global masking threshold by computing the individual tonal and noise masking frequencies utilizing the pre- and post-masking nature for temporal effects and combining them into a final auditory masking threshold for that frame per subband. The algorithm may be applied to each of the thirty-two sub bands and may return twenty-seven signal-to-mask ratios (SMR) in dB scale, and SMR 28 to 32 are not used. For every 512 sample frame, demodulated audio levels which have signal energy above the masking threshold may be computed for that specific subband, and an average energy may be computed temporally across time for the duration that the background tuner is harboring on the station. The same may be done for the main tuned station.

(33) Since the broadcast audio is continuously changing depending on the song that the artist is playing, the background tuner may sample stations at regular intervals and compute the appropriate energy levels above the psychoacoustic masking level, which is what is audible by the human ear for perceived volume, and may thus define a perceived volume level. For example, a total loudness metric as applied to demodulated FM analog audio for a particular volume step may be thus defined as:

(34) $\begin{array}{cc}L={}_0^{P}L\left(p\right)\mathrm{dp},\mathrm{where}p=32\mathrm{for}\mathrm{FM}\mathrm{for}\mathrm{example}.& \left(10\right)\end{array}$

(35) This value may be stored in RAM memory area in a table (FIG. 7) which may be created to map out the appropriate volume level settings that need to be applied to ensure that the end user perceives a specified volume level. The trust timer may be used to ensure that if a station has been sampled, then the background tuners can scan other frequencies in the neighboring area for the duration of the countdown of the trust timer.

(36) The perceived volume level comparison may take into account the following three factors. The first factor is signal energy levels above audio masking levels. To ensure that the algorithm scales for scenarios in the car cabin environment, microphone input to the car radio head unit can be used to ensure sampling of the background noise (e.g., engine noise and road noise) and factoring in of the masking threshold level shifts.

(37) The second factor is the averaged audio bandwidth of the currently tuned station vs. that of secondary neighboring stations. This can be determined by doing a Fast Fourier Transform of the demodulated audio for digital and analog audio or by utilizing the modulation level of demodulated audio for analog audio.

(38) The third factor is quality metrics such as fieldstrength, multipath, adjacent energy and frequency offset in the case of analog FM; fieldstrength, adjacent energy and frequency offset in the case of AM; and BER in the case of DAB signals to ensure that the algorithm does not engage when dealing with weak stations having noise which should not be amplified.

(39) Use Case with Analog AM/FM Single Tuner without Wideband Scanning Support

(40) As mentioned above, the present invention may be applied to analog AM/FM single tuner radios as well as to dual and multi tuner radios. FIG. 8 is a table similar to the table of FIG. 7, but applicable to analog signals. If the radio is tuned to a low modulation station on a single tuner radio, then on each audio pause the front end DSP may perform a quick alternate frequency check on two neighbor frequencies of the low modulation station. The alternative frequency checks occur within seven millisecond intervals and cannot be perceived by the user as the frequency checks are triggered when the present audio tuned station gets a pause. Each frequency quality check may yield the fieldstrength and modulation levels. The fieldstrength (dBuV) may be needed as the modulation level check is only valid for strong quality station frequencies which are characterized by a strong fieldstrength level.

(41) The radio receiver can calculate the average modulation level of the neighboring frequencies and compare it against the modulation level of the currently tuned station frequency. The radio receiver can then determine whether the user will perceive a volume difference in the event that the user does a tune or seek to the neighboring frequency station. In such a circumstance, the present invention provides a mechanism whereby after the tune or seek settles on the higher modulation level, the radio receiver automatically adjusts the volume level without user intervention.

(42) On Multimedia Sources

(43) DAB and HD IBOC digital audio sources are in essence similar to multimedia audio files. In the case of a multimedia source, the same issue arises when a user traverses from different compression rate audio files. Herein a background instance compressed audio decoder can be used to gauge the volume and ensure mode balance for the current multimedia source.

(44) While this invention has been described as having an exemplary design, the present invention may be further modified within the spirit and scope of this disclosure. This application is therefore intended to cover any variations, uses, or adaptations of the invention using its general principles. Further, this application is intended to cover such departures from the present disclosure as come within known or customary practice in the art to which this invention pertains.