EP2536043A1 - Optimierte Verzögerungsdetektion von Simultanrundfunksignalen - Google Patents

Optimierte Verzögerungsdetektion von Simultanrundfunksignalen Download PDF

Info

Publication number
EP2536043A1
EP2536043A1 EP11169843A EP11169843A EP2536043A1 EP 2536043 A1 EP2536043 A1 EP 2536043A1 EP 11169843 A EP11169843 A EP 11169843A EP 11169843 A EP11169843 A EP 11169843A EP 2536043 A1 EP2536043 A1 EP 2536043A1
Authority
EP
European Patent Office
Prior art keywords
signal
downsampling
audio
reproduced sound
signals
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
EP11169843A
Other languages
English (en)
French (fr)
Inventor
Raik Michael
Stephan Hurz
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Panasonic Industry Europe GmbH
Original Assignee
Panasonic Automotive Systems Europe GmbH
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Panasonic Automotive Systems Europe GmbH filed Critical Panasonic Automotive Systems Europe GmbH
Priority to EP11169843A priority Critical patent/EP2536043A1/de
Publication of EP2536043A1 publication Critical patent/EP2536043A1/de
Ceased legal-status Critical Current

Links

Images

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04H—BROADCAST COMMUNICATION
    • H04H20/00—Arrangements for broadcast or for distribution combined with broadcast
    • H04H20/20—Arrangements for broadcast or distribution of identical information via plural systems
    • H04H20/22—Arrangements for broadcast of identical information via plural broadcast systems

Definitions

  • the present invention relates to an apparatus for receiving broadcast signals. More particularly, the present invention relates to a broadcast signal receiving apparatus which is operative to receive broadcast signals that are transmitted via two different broadcast systems.
  • Audio broadcasting systems have been used for a long time in the field of audio broadcasting.
  • Audio broadcasting systems that have been practically employed include AM audio broadcasting systems in which audio information signals are transmitted in the form of an AM (amplitude modulated) audio information signal and FM audio broadcasting systems in which audio information signals are transmitted in the form of FM (frequency modulated) audio information signal.
  • DAB Digital Audio Broadcasting
  • DRM Digital Radio Mondiale
  • audio receivers which include both analogue and digital receiving sections.
  • a control signal is used for switching the different receivers in order to select the receiver which provides the best reception.
  • signals transmitted by different systems such as digital and analogue broadcasting systems
  • signals distributed via digital broadcasting systems have different spectral properties such as bandwidth and volume as compared with the analogue systems.
  • a further particularly important difference between digitally and analogously transmitted signals is a certain time delay of the digital signals with respect to the analogue signals distributing the same broadcast contents (i.e. the same program of an audio service).
  • the reason for said time delay resides in the particulars of the digital broadcast transmission chain.
  • the digital coding results in a noticeable delay of a digital broadcast audio signal compared to the corresponding analogue audio signal.
  • One important source of coding delays in the digital broadcast system is, for instance, time interleaving of audio information data for the purpose of minimizing deterioration.
  • a typical delay of a digital audio broadcast signal with respect to an analogue broadcast signal of the same service is in the order of several hundreds of milliseconds (ms), for example, 400 ms, up to the order of one second (s), for instance 1.4s.
  • the delay between the analogue and the digital transmission varies from broadcasting station to broadcasting station. It is therefore not possible to perform the delay time compensation based on the assumption of a predetermined delay value. On the contrary, the delay has to be dynamically determined during operation, in the audio receiver.
  • a known apparatus for receiving audio broadcast signals is capable of receiving both digital audio broadcast signals and analogue audio broadcast signals by means of respective reception portions, to obtain a first reproduced sound signal, from the digital broadcast signal and a second reproduced sound signal from the analogue broadcast signal, respectively. Further, the apparatus comprises a variable delay portion for delaying the first reproduced sound signal obtained from the analogue audio broadcast signal by a variable delay time. The variable time delay applied by the variable delay portion is controlled so as to reduce a difference in delay time between the first reproduced time signal after being delayed by the variable delay portion and the second reproduced sound signal.
  • Such a conventional apparatus for receiving both analogue and digital broadcasting systems is known from European patent EP 0 863 632 B1 .
  • a similar conventional apparatus is known from document EP 1 227 608 A2 , wherein two signal matching control portions are employed for controlling the variable delay between the two audio signals and the results of the control portions are combined.
  • the apparatus additionally also takes into account an adaptation of further differences between the sound signals reproduced from digital and analogue sources, such as volume.
  • the reproduced sound signals from the analogously received broadcast signals are generally digitalized so that both reproduced sound signals are available in digitalized form, however, generally having different sampling frequencies.
  • a problem in efficiently calculating a delay between broadcast signals received via different broadcast systems does not only occur in case of parallel reception of certain audio contents via analogue and digital transmissions such as FM and DAB or AM and DRM. Similar problems occur in any broadcast environment, wherein one audio service is transmitted via different broadcast systems (a so-called "simulcast transmission"), and wherein an inaudible switching between the two signals shall be performed. Further examples for such a situation include, but are not limited to: DRM and DAB simulcast transmissions, analogue AM or FM and IBOC (In-Band-On-Channel) simulcast transmissions, and satellite and terrestrial audio broadcast systems in general.
  • the present invention aims to provide an improved audio receiver and a respective receiving method enabling a more efficient control of the delay between two audio signals of different sampling rates originating from different broadcast sources.
  • an audio receiver comprising a first signal receiving portion for receiving a first audio broadcast signal to obtain a first reproduced sound signal, and a second signal receiving portion for receiving a second audio broadcast signal to obtain a second reproduced sound signal.
  • the audio receiver further comprises a variable delay portion for delaying the first reproduced sound signal by a variable delay time.
  • the receiver comprises a delay control portion for controlling the variable delay portion.
  • the audio receiver further comprises a first downsampling portion for downsampling the first reproduced sound signal and a second downsampling portion for downsampling the second reproduced sound signal.
  • the delay control portion controls the variable delay portion based on the first and second reproduced sound signals downsampled by the first and second downsampling portions.
  • a method of receiving audio broadcast signals comprises the steps of receiving a first audio broadcast signal, obtaining a first reproduced sound signal from said received first audio broadcast signal, receiving a second audio broadcast signal and obtaining a second reproduced sound signal from the received second audio broadcast signal. Further, the method comprises the step of delaying the first reproduced sound signal by a variable delay time. The method further includes the steps of downsampling the first reproduced sound signal and downsampling the second reproduced sound signal, and controlling the variable delay time based on the downsampled first and second reproduced sound signals.
  • the invention thereby enables to precisely determine an initial delay between the two signals with low calculation effort. Thereby, the initial delay can be efficiently compensated for, in particular, for sound signals originating from simulcast transmissions and having different sampling frequencies.
  • the first audio broadcast signal is an analogue signal and the second audio broadcast signal is a digital signal.
  • the invention enables a seamless switchover from a digital to an analogue broadcast signal, for instance, for a mobile receiver in case when moving into a service area wherein the digital signal becomes weak or unavailable. Due to coding delays in the digital broadcast transmission chain, the digital broadcast audio signal has a noticeable delay compared to an analogue audio signal of the same service, which shall be compensated for to achieve seamless switchover.
  • the analogue signal can be given an initial delay when sent out from the broadcast station.
  • the first audio broadcast signal can be a digital signal
  • the second signal can be an analogue signal.
  • signals to which the present invention is applicable include, but are not limited to cases, wherein either one of the first and second audio broadcast signals are DRM and DAB or analogue AM and DRM, or analogue AM/FM and IBOC.
  • one of the audio broadcast signals is transmitted via satellite and the other is transmitted via terrestrial transmission.
  • the signal transmitted via satellite can generally be assumed to be received with an initial delay with respect to a corresponding terrestrially transmitted signal, due to the considerably larger travelling distance of the signal.
  • the signal transmitted terrestrially corresponds to the first signal and shall be delayed at the receiver, for compensation.
  • the compensation is, in principle, possible at the broadcasting station, such a compensation will generally not be complete, due to the dependency of the differences in transmission paths on the particular location of the moving station. Therefore, at least part of the compensation is always necessary to be performed at the receiving site.
  • the invention may also be applied to a case, wherein the first signal is terrestrially transmitted, and the second signal is transmitted via satellite.
  • the first and second audio broadcast signals respectively transmit one and the same audio service via different broadcast systems.
  • Any broadcast environment, wherein one audio service is transmitted via different broadcast systems such as digital (for instance DAB or DRM) and analogue (for instance FM or AM) are called “simulcast transmissions" in the art.
  • the downsampled versions of the sound signals that are employed for determining a delay therebetween and controlling the variable delay for compensation are obtained by reducing a sampling frequency (also called sampling rate) thereof.
  • a lowpass filter such as a FIR (Finite Impulse Response) filter is applied to each of the sound signals before downsampling.
  • FIR Finite Impulse Response
  • the downsampling of each of the sound signals is performed in two stages, respectively.
  • the first downsampling portion therefore comprises a first and a second decimation stage.
  • the first reproduced sound signal is downsampled with a first downsampling ratio, and the result thereof is further downsampled by the second decimation stage with a second downsampling ratio.
  • the second downsampling portion comprises a third and a fourth decimation stage wherein the second reproduced sound signal is subsequently downsampled with a third and a fourth downsampling ratio.
  • the downsampling ratio is the ratio of a sampling frequency of a signal before a particular downsampling stage to the sampling frequency after the respective downsampling stage.
  • a two-stage downsampling of the two audio signals is particularly advantageous, since FIR filters of low order can be used for average level detecting of signals to be downsampled, without running in problems with aliasing effects.
  • the delay control portion determines an initial delay time of the second reproduced sound signal with respect to the first reproduced sound signal, in order to control the variable delay time of the variable delay portion so as to compensate for the initial delay time.
  • Delay time compensation may be partially (delay reduction) or completely.
  • the initial delay is completely compensated by the variable delay of the first signal, within the limits of the calculation precision of the initial delay time. In accordance with the present invention, a precision of initial delay determination of 1 ms is possible.
  • the delay control portion determines the initial delay time of the second reproduced sound signal with respect to the first reproduced sound signal by detecting the maximum of a cross-correlation function between the downsampled first and downsampled second reproduced sound signals.
  • cross-correlation is performed by convolution in the frequency domain.
  • Cross-correlation in time domain would need a high calculation power for long vectors.
  • transmission gaps, muted audio at the transmitter side or transmitted sine-frequencies are detected in a subsequent silence detection step.
  • artefacts due to signal portions that are not usable for delay time detection by cross-correlation can be eliminated.
  • variable delay portion comprises a ring buffer with variable length.
  • the variable delay time of the ring buffer can be controlled in correspondence with the initial delay time to be compensated, and indicated by a control signal issued by the control portion based on the cross-correlation function of the downsampled sound signals.
  • downsampling is employed to obtain reproduced sound signals originating from the different transmission paths having only minimal differences in sampling frequencies.
  • the invention can reduce differences in the sampling frequencies of the two signals much below 1%.
  • the present invention provides a scheme enabling calculation time reduction for calculating a delay time between two time-shifted audio signals having different sampling frequencies such as a (digitalized) analogue broadcast signal and a digital broadcast signal.
  • the calculation time reduction is achieved by downsampling the original audio signals to one similar sampling frequency.
  • the downsampling is preferably done in two steps to reduce the needed calculation time further.
  • Similar sampling frequency means that the relative amount of the difference in sampling frequencies between both signals with respect to the absolute amount of one of the sampling frequencies is considerably reduced after downsampling with respect to the original relative difference before downsampling. Experiments showed that a relative difference in the sampling frequency below 0.23% is acceptable.
  • Fig. 1 illustrates a general overview of the structure of an exemplary audio receiver according to the present invention.
  • the audio receiver comprises a first antenna 10, a second antenna 20, a first signal reception portion 12 and a second signal reception portion 22, a first downsampling portion 16 and a second downsampling portion 26, a variable delay portion 18, a delay control portion 30, a selection portion 32 and a selection control portion 34.
  • the first and the second signal reception portions 12 and 22 receive and process audio broadcast signals 11 and 21 that come in via antennae 10 and 20, respectively.
  • the audio signals 11 and 21 transmit one and the same audio service (i.e. one and the same audic program) via two different broadcast systems (simulcast).
  • the first received signal is an analogue broadcast signal such as FM
  • the second received signal is a digital audio broadcast signal such as DAB.
  • the delay has to be compensated in order to enable a seamless and inaudible switching between the two signals.
  • the average delay is about 1.4 seconds.
  • the delay may vary from station to station, and has, therefore, to be dynamically determined at the receiving apparatus during operation.
  • the processing performed by the first and second signal reception portions 12 and 22, respectively results in a reproduced sound signal that is obtained on the basis of the received audio broadcast signals.
  • the first and the second signal reception portions 12 and 22 are an analogue audio broadcast signal receiving portion and a digital audio broadcast signal receiving portion, respectively.
  • Typical processing in an analogue audio broadcast signal receiving portion includes the steps to tune in to an analogue audio broadcast signal, such as a FM audio broadcast signal 11, and further includes a demodulation processing and a de-emphasis processing.
  • the reproduced analogue sound signal 14 is digitized and sampled with a first predetermined sampling frequency.
  • Typical processing in a digital audio broadcast signal receiving portion includes tuning in to a digital audio broadcast signal 20, audio and channel decoding, and time de-interleaving to produce a reproduced digital sound signal 24 constituted with time de-interleaved audio information data.
  • the reproduced digital sound signal 24 is sampled with a second predetermined sampling frequency.
  • the variable delay portion 18 is operable to delay the first reproduced sound signal 14 (in a preferred embodiment: obtained from an analogue broadcast signal 12) with a variable delay time.
  • the output of variable delay portion 18 is a time delayed first reproduced sound signal 14'.
  • Control of the variable delay portion 18, and, in particular, the variable delay time is achieved by delay control portion 30.
  • Control of the variable delay time by the delay control portion 30 is performed based on a calculation of an initial delay time of the second reproduced sound signal 24 with respect to the first reproduced sound signal 14.
  • the delay time is defined as the delay between a particular portion of the broadcast program in the sound signal 24 reproduced by the second signal reception portion 22 with respect to the same portion of the program in the first sound signal 14 as reproduced by the first signal reception portion 12.
  • the delay can for instance be determined on the basis of predetermined pilot signals or time stamps included in both received broadcast signals. Such a determination is also suitable, if there is no simulcast situation.
  • the calculation processing by the delay control portion 30 is simplified by downsampling both first reproduced sound signal 14 and second reproduced sound signal 24 by different downsampling ratios. Thereby, a high degree of coincidence between the sampling rates of the first and the second reproduced sound signal can be achieved, i.e. the relative difference between the sampling frequencies of the first and second downsampled sound signals may be reduced to a value of an order of 0.2% or lower.
  • Downsampling is performed by first downsampling portion 16 and second downsampling portion 26, for the first and the second reproduced sound signal 14 and 24, respectively.
  • Selection unit 32 receives both the delayed first reproduced sound signal 14' and the second reproduced sound signal 24. Selection portion 32 operates to select one of the two input audio signals 14' and 24 and output a selected one of the signals for being perceived by the user. Selection portion 32 is controlled by selection control portion 34. Generally, selection control by the selection control unit 34 is performed in such a manner that the user is capable of always perceiving the signal currently having the best audio quality, even in case of a moving receiver such as in a vehicle. In the case when the quality of the currently selected audio signal decreases (for instance if the currently perceived audio signal is from digital broadcast, and the vehicle drives out of the service area of the digital broadcasting station), automatic switchover is performed. Since the timing of the delayed first reproduced sound signal 14' and the second reproduced sound signal 24 coincide to a high degree, switchover by the selection unit 32 can be performed seamlessly.
  • Fig. 2 is a more detailed view of a portion of the audio receiver in a particular embodiment.
  • the first reproduced sound signal (FM_ln 214) is a digitized analogue FM signal with a sampling frequency of 44,100 Hz
  • the second reproduced sound signal (DAB_In 224) is a digital audio signal (DAB signal) with a sampling frequency of 48,000 Hz.
  • the variable delay portion is implemented by means of ring buffer 218 with variable length. Ring buffer 218 outputs a delayed sound signal with 44,100 Hz sampling rate.
  • the variable time delay of the ring buffer 218 is controlled by a control signal 238 for setting of the ring buffer dimension issued by delay control portion 230.
  • Delay control portion 230 operates by correlating the first and the second reproduced sound signal after having been downsampled in first downsampling portion 216 and second downsampling portion 226.
  • First downsampling portion 216 includes first decimation stage 216a and second decimation stage 216b.
  • Decimation stage 216a includes a FIR lowpass filter 216b1 of 32 nd order and downsampling section 216a2 for downsampling the received signal by a ratio of 1:11.
  • Second decimation stage 216b includes 64 th order FIR lowpass filter 216b1 and downsampling section 216b2 with a downsampling ratio of 1:4.
  • Second downsampling section 226 includes third decimation stage 226a and fourth decimation stage 226b.
  • Third decimation stage 226a comprises 32 nd order FIR lowpass filter 226a1 and downsampling section 226a2 having a downsampling ratio of 1:8.
  • Fourth decimation stage 226b includes 64 th order FIR lowpass filter 226b1 and downsampling section 226b2 having a downsampling ratio of 1:6.
  • Delay control portion 230 comprises two Fast Fourier Transform (FFT) sections 231 and 232. Further, delay control portion 230 comprises convolution section 233 and Inverse Fast Fourier Transform (IFFT) section 234. In the particular embodiment, control portion 230 further includes, besides peak detection section 236, a silence detection section 235.
  • FFT Fast Fourier Transform
  • IFFT Inverse Fast Fourier Transform
  • sampling weights between the DAB audio signal and the digitized FM audio signal is due to the specification of digital FM demodulators available in the art. While DAB audio signals are digitally transmitted with a sampling frequency of 48,000 Hz (48 kHz), the available digital FM demodulators use a sampling frequency of 44,100 Hz (44.1 kHz), which is the sampling frequency used by the CD.
  • the sampling frequencies of the two signals have to be matched. This is achieved by a two-step downsampling procedure wherein two different downsampling ratios are applied by the first and second downsampling portions, respectively.
  • the first decimation stage 216a receives the reproduced FM sound signal 214 with a sampling rate of 44.1 kHz. After downsampling the signal with a downsampling ratio of 1:11, the signal has an intermediate sampling frequency of approximately 4 kHz (exactly: 4009.9 Hz). Said signal is further downsampled in second decimation stage 216b with the downsampling ratio of 1:4. The result is a signal with a sampling frequency of approximately 1 kHz (exactly: 1,002.3 Hz).
  • Third decimation stage 226a receives reproduced DAB sound signal sampling frequency 48 kHz. After downsampling with a ratio of 1:8, the sampling frequency is 6 kHz (6,000 Hz). The downsampled signal is further downsampled in the fourth decimation stage 226b with a downsampling ratio of 1:6. The resulting signal of the fourth decimation stage 226b has a sampling rate of 1 kHz (1,000 kHz). Accordingly, the sampling rates after both signals have been downsampled in two stages with different downsampling ratios respectively, coincide to a high degree. The relative difference in sampling frequencies is as low as 0.23%. Experiments showed that such a small difference in the sampling frequencies is acceptable.
  • both signals are converted to almost the same low sampling frequency of 1 kHz in software. Thereby, necessary calculation power to perform the cross-correlation is reduced considerably and at the same time a possible resolution of the determined delay time of 1 ms is possible. Such a high accuracy of delay time determination and compensation is sufficient to perform an inaudible switch between the two signals if so desired.
  • cross-correlation is used for determining the delay in the digitized FM and DAB audio streams. Because the cross-correlation in the time domain needs a high calculation power for long vectors, the downsampled signals input to delay control portion 230 are first transformed into the frequency domain by FFT sections 231 and 232, respectively. Cross-correlation is performed by convolution in the frequency domain in convolution section 233. After correlation, correlation results undergo inverse FFT in IFFT section 234.
  • the resulting signal (cross-correlation function) is processed by silence detection section 235 to perform a "silence test" to ensure useful interpretation of the cross-correlation function.
  • the silence detection section 235 detects transmission gaps, muted audio at the transmitter side or transmitted sine-frequencies. If such audio signals occur during the determination of the delay, the cross-correlation function has no clear maximum peak. Signals the cross-correlation function of which has no clear maximum peak are not usable for delay determination by peak detection.
  • silence detection section 235 determines if the maximum value of the cross-correlation function is much higher than the average value (i.e. a clear peak is present). If this is not the case, it is considered that the currently received signals are not usable for delay determination, and the whole process is started again with new samples of the audio signals.
  • delay control section 230 takes into account that the audio may be muted or otherwise deteriorated so that no clear peak in the cross-correlation function can be detected. In the case that the audio signal is muted, the delay detection must be repeated until the audio signal is not muted.
  • Peak detection section 236 calculates the delay time by detecting the peak (maximum) of the cross-correlation function.
  • the position of the maximum of the cross-correlation function between the FM and the DAB audio signals determines the delay between the two signals, measured in the number of samples (after downsampling).
  • the thus determined initial delay of the second signal with respect to the first signal determines the delay time of ring buffer 218.
  • Ring buffer 218 shall delay the FM audio signal 214 to produce a delayed FM audio signal 214' that is delayed with respect to initial signal 214 by the delay time of signal 224 with respect to signal 214, as determined by control section 230 (delay time compensation).
  • the calculated value of the initial delay time is taken to define the dimension of ring buffer 218 to appropriately delay FM audio signal 214 to compensate for the initial delay of digital signal 224.
  • delayed FM audio signal 214' and DAB audio 224 are aligned, and a switch between the two signals can be performed seamlessly and inaudible to the user.
  • Fig. 3 shows an exemplary flowchart of a method for determining the control signal setting a variable delay time in accordance with the present invention.
  • step S100 and respectively S200 an analogue and a digital audio signal are received.
  • the specifics of the received signals is not limited to an analogue and a digital signal, but other kinds of signals such as two different kinds of digital signals, or audio broadcast signals distributed by satellite and terrestrially, respectively, are generally also possible.
  • step S110 the received signal (in the example: analogue signal) is processed to obtain the digitized analogue sound signal with a first sampling frequency at step S120.
  • a respective signal processing of the second received signal (in the example: digital signal) is performed in step S210 to obtain a reproduced digital sound signal with a second sampling frequency in step S220.
  • Downsampling of the respective signals is performed by corresponding processing steps S130 for the analogue signal and S230 for the digital signal.
  • step S300 cross-correlation is performed between the two downsampling signals.
  • the cross-correlation is performed by convolution in the frequency domain.
  • step S310 detects the position of the peak of the resulting cross-correlation function.
  • silence detection is performed, in order to disregard any correlation result having no clear peak, and therefore not usable for delay determination.
  • the initial delay of the second (digital) audio signal with respect to the first (analogue) audio signal is determined.
  • the determined initial delay forms the basis for generating a control signal in step S330 that determines the necessary amount of variable time delay to be applied by variable delay portion 18, in order to compensate for the detected initial delay.
  • Step S340 instructs variable delay portion 18 by setting the respective variable delay time.
  • the present invention relates to a particular efficient scheme for determining an initial delay of a second received audio signal with respect to a first received audio signal, specifically suitable for a simulcast environment.
  • the initial delay time must be exactly determined and compensated by delaying the first signal at the receiving apparatus by a respective amount.
  • Calculation power requirements are particularly high if the sound signals reproduced from first and second received audio signals have different sampling frequencies.
  • the present invention adapts the sampling frequencies by downsampling both signals for delay time determination and thus considerably reduces calculation effort.

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Circuits Of Receivers In General (AREA)
EP11169843A 2011-06-14 2011-06-14 Optimierte Verzögerungsdetektion von Simultanrundfunksignalen Ceased EP2536043A1 (de)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP11169843A EP2536043A1 (de) 2011-06-14 2011-06-14 Optimierte Verzögerungsdetektion von Simultanrundfunksignalen

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP11169843A EP2536043A1 (de) 2011-06-14 2011-06-14 Optimierte Verzögerungsdetektion von Simultanrundfunksignalen

Publications (1)

Publication Number Publication Date
EP2536043A1 true EP2536043A1 (de) 2012-12-19

Family

ID=44247797

Family Applications (1)

Application Number Title Priority Date Filing Date
EP11169843A Ceased EP2536043A1 (de) 2011-06-14 2011-06-14 Optimierte Verzögerungsdetektion von Simultanrundfunksignalen

Country Status (1)

Country Link
EP (1) EP2536043A1 (de)

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2014118319A1 (en) * 2013-02-01 2014-08-07 Dan Barry A method and system for matching audio and video
JP2016208319A (ja) * 2015-04-23 2016-12-08 アルパイン株式会社 ラジオ放送受信装置
JP2016213695A (ja) * 2015-05-11 2016-12-15 アルパイン株式会社 ラジオ放送受信装置
JP2016213532A (ja) * 2015-04-29 2016-12-15 アルパイン株式会社 ラジオ放送受信装置及びシームレス切替方法

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1227608A2 (de) 2001-01-26 2002-07-31 Robert Bosch Gmbh Verfahren zur Umschaltung eines Rundfunkempfängers auf einer anderen Funkübertragung mit gleichem Audioinhalt
EP1233556A1 (de) * 2001-02-16 2002-08-21 Sony International (Europe) GmbH Empfänger für den Empfang von Rundfunksignalen mit Verwendung von zwei Empfängern, für den Empfang eines Rundfunksignals das auf zwei unterschiedlichen Rundfunkfrequenzen oder mit zwei unterschiedlichen Übertragungssystemen übertragen wird
EP0863632B1 (de) 1997-03-04 2006-06-07 Sony Corporation Vorrichtung für den Empfang von Rundfunkprogrammen

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0863632B1 (de) 1997-03-04 2006-06-07 Sony Corporation Vorrichtung für den Empfang von Rundfunkprogrammen
EP1227608A2 (de) 2001-01-26 2002-07-31 Robert Bosch Gmbh Verfahren zur Umschaltung eines Rundfunkempfängers auf einer anderen Funkübertragung mit gleichem Audioinhalt
EP1233556A1 (de) * 2001-02-16 2002-08-21 Sony International (Europe) GmbH Empfänger für den Empfang von Rundfunksignalen mit Verwendung von zwei Empfängern, für den Empfang eines Rundfunksignals das auf zwei unterschiedlichen Rundfunkfrequenzen oder mit zwei unterschiedlichen Übertragungssystemen übertragen wird

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
RICHARD C CABOT: "A Note on the Application of the Hilbert Transform to Time Delay Estimation", IEEE TRANSACTIONS ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, IEEE INC. NEW YORK, USA, vol. ASSP-29, no. 3, 1 June 1981 (1981-06-01), pages 607 - 609, XP007910107, ISSN: 0096-3518, DOI: 10.1109/TASSP.1981.1163564 *

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2014118319A1 (en) * 2013-02-01 2014-08-07 Dan Barry A method and system for matching audio and video
US9570109B2 (en) 2013-02-01 2017-02-14 Forty Five Web Sound Limited Method and system for matching audio and video
JP2016208319A (ja) * 2015-04-23 2016-12-08 アルパイン株式会社 ラジオ放送受信装置
JP2016213532A (ja) * 2015-04-29 2016-12-15 アルパイン株式会社 ラジオ放送受信装置及びシームレス切替方法
JP2016213695A (ja) * 2015-05-11 2016-12-15 アルパイン株式会社 ラジオ放送受信装置

Similar Documents

Publication Publication Date Title
USRE49210E1 (en) Method and apparatus for level control in blending an audio signal in an in-band on-channel radio system
US6408038B1 (en) Receiver and receiving circuit
KR102486073B1 (ko) 하이브리드 라디오 시스템에서의 자동 오디오 정렬을 위한 방법 및 장치
US20020115418A1 (en) Alternative system switching
EP2858277B1 (de) Vorrichtung und verfahren zur steuerung von audiosignalen
EP2536043A1 (de) Optimierte Verzögerungsdetektion von Simultanrundfunksignalen
JP2004349805A (ja) Ｉｂｏｃ放送受信機
USRE48655E1 (en) Method and apparatus for time alignment of analog and digital pathways in a digital radio receiver
EP2355381A1 (de) Audioverarbeitungsvorrichtung und Verfahren zur Verarbeitung von Audiosignalen
US9893823B2 (en) Seamless linking of multiple audio signals
JP6756588B2 (ja) 放送受信装置
JPH03126326A (ja) ラジオ受信機
JPH04207420A (ja) ラジオ受信機の自動選局方法

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AL AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO RS SE SI SK SM TR

AX Request for extension of the european patent

Extension state: BA ME

17P Request for examination filed

Effective date: 20130614

RBV Designated contracting states (corrected)

Designated state(s): AL AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO RS SE SI SK SM TR

17Q First examination report despatched

Effective date: 20131108

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: THE APPLICATION HAS BEEN REFUSED

18R Application refused

Effective date: 20141001