64results about "In band on channel" patented technology

A low cost DAB multichannel receiver comprising a simplified buffering method for buffering content segments from multiple streams contained within the DAB channel, where the receiver enables the listener to navigate buffered content segments from multiple streams within the DAB channel while enabling the broadcaster to control the timeshift of commercial content to the receiver output stream. The receiver's buffered content grows over time and is cleared when tuning away from the channel, thus encouraging listeners desiring to tune in to new content to instead navigate to new buffered segments. Broadcaster control of the listener experience may be enabled by setting content control fields which are observed in the broadcast by the multichannel receivers. Additional embodiments are disclosed.

A radio receiver includes a first signal path including a first tuner configured to receive a first signal from a first antenna, and a first demodulator configured to demodulate symbols from an output of the first tuner to produce first branch metrics derived from the demodulated symbols; a second signal path including a second tuner configured to receive a second signal from a second antenna, and a second demodulator configured to demodulate symbols from an output of the second tuner to produce second branch metrics derived from the demodulated symbols; a combiner for maximum ratio combining the first branch metrics and the second branch metrics; and processing circuitry to process the combined first and second branch metrics to produce an output signal.

A method and apparatus is disclosed in which delay is applied to analog-sourced audio in a radio simulcast when the analog signal initially leads the digital signal. A radio receiver is configured to receive a simulcast radio program broadcast with an analog signal and a digital signal. The program content can be extracted from either the analog or digital signals, with the audio source eventually being blended to the digital signal. Audio is initially provided based on the analog signal. If the analog signal is initially leading the digital signal, delay is applied a data stream corresponding to the analog signal relative to a data stream corresponding to the digital signal. Delay applied to the data stream corresponding to the analog signal is increased at a rate that avoids audio artifacts of the output audio. The blend is performed when the data streams are aligned in time.

A method and apparatus are provided for blending analog and digital portions of a composite digital radio broadcast signal by processing compressed audio packets to compute corresponding digital audio quality indicator values, storing the compressed audio packets in an audio blend buffer, processing audio information from each compressed audio packet stored in the audio blend buffer with an audio decoder to generate decompressed digital audio signal samples, and using the digital audio quality indicator values to guide a blending process for combining analog audio signal samples with the digital audio signal samples to produce an audio output by preventing unnecessary blending back and forth between analog and digital if the digital audio quality indicator values indicate that the compressed audio packets are degraded or impaired.

A method and apparatus are disclosed for increasing the time and frequency diversity of a multi-stream signal in a DAB system. A plurality of audio streams are divided into four (4) digital sub-streams, C₀₀, C₀₁, C₁₀, and C₁₁. Each sub-stream C₀₀, C₀₁, C₁₀, and C₁₁ is assigned a unique frequency band, and time slot. A first core sub-stream C₁₀ is mapped to one frequency partition and a second core sub-stream C₀₀ is mapped to another frequency partition and delayed relative to the first core sub-stream. Similarly, two enhancement sub-streams C₁₁ and C₀₁ are mapped to different frequency partitions and are time delayed relative to each other and the core sub-streams. The two core sub-streams C₀₀ and C₁₀ can have a maximum separation across both the time and frequency axes. Each core sub-stream C₀₀ and C₁₀ is separate from one of the enhancement sub-streams in the frequency domain and separate in the time domain from the other enhancement sub-stream. Each core sub-stream C₀₀ and C₁₀ can be combined with any other available core or enhancement sub-stream to form a 64 kbps PAC. In addition, a 96 kbps PAC can be obtained by combining the two core sub-streams C₀₀ and C₁₀ with one of the enhancement sub-streams C₀₀ or C₁₁. Finally, the combination of all four sub-streams produces a full-rate 128 kbps PAC.

A method and apparatus are provided for processing a received digital radio broadcast signal to efficiently remove signal interference artifacts from digital and/or analog signals by using signal quality information extracted from audio samples in one or more buffered audio frames to detect audio frames containing clipped noise artifacts and weaker noise artifacts and to selectively apply anti-interference processing to remove the signal interference artifacts.

The present invention is directed to a radio designed to receive both analog and digital subchannels from radio stations that are broadcasting either an analog only signal, a digital only signal, or a hybrid signal containing both analog and digital subchannels. It allows a user to direct the radio to search for either the next active analog or digital subchannel, or alternately to ignore the analog subchannels and search only for digital subchannels. This is accomplished using a single button for either functionality when searching through incrementing frequencies. Another button may be added for decrementing frequencies with the same basic functionality. In the present invention, when the user presses the “Scan Up” button once for a short period of time, the radio will search for the next active analog or digital subchannel above the current location of the virtual channel map. But if the user presses the button twice in quick succession or holds the button down for a longer period of time, the radio will search only for the next digital subchannel above the current location in the virtual channel map.

In one embodiment, a receiver front end circuit can receive and process multiple radio frequency (RF) signals and output downconverted signals corresponding to these signals. In turn, multiple signal processors can be coupled to this front end. Specifically, a first signal processor can receive and process the downconverted signals to output a first signal obtained from content of a first RF signal, and a second signal processor can receive and process the downconverted signals to output a second signal obtained from content of a second RF signal. In addition, the apparatus may include a detection circuit coupled to the receiver front end circuit to detect presence of at least the second signal and enable the second signal processor responsive to the detected presence.

A customer relationship management (CRM) method using IBOC-radio signals is provided. A message in the radio signal is parsed to obtain a key. The key is compared to a plurality of stored keys. When the received key matches a stored key, a data structure associated with the message is outputted. A device comprising a lookup table with a plurality of stored keys, a tuner unit that receives a CRM in an IBOC signal, and a controller in electrical communication with the lookup table and tuner is provided. The controller comprises (i) instructions for comparing a key in the CRM to one or more stored keys in the plurality of stored keys and (ii) instructions for permitting the display of a display text associated with the received key when there is a match between the received key and a key in the plurality of stored keys.

A method for processing a radio signal includes producing first and second streams of audio samples; decimating the first and second streams of audio samples to produce first and second streams of decimated streams of audio samples; estimating a first offset value between corresponding samples in the first and second streams of decimated streams of audio samples; shifting one of the first and second streams of audio samples by a first shift value; decimating the first and second streams of audio samples to produce third and fourth streams of decimated audio samples; estimating a second offset value; determining a final offset value based on an intersection of ranges of valid results of the first and second offset values; and shifting one of the first and second streams of audio samples by the final offset value to align the first and second streams of audio samples.

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval.

Systems and methods for increasing transmission bandwidth efficiency by the analysis and synthesis of the ultimate components of transmitted content are presented. To implement such a system, a dictionary or database of elemental codewords can be generated from a set of audio clips. Using such a database, a given arbitrary song or other audio file can be expressed as a series of such codewords, where each given codeword in the series is a compressed audio packet that can be used as is, or, for example, can be tagged to be modified to better match the corresponding portion of the original audio file. Each codeword in the database has an index number or unique identifier. For a relatively small number of bits used in a unique ID, e.g. 27-30, several hundreds of millions of codewords can be uniquely identified. By providing the database of codewords to receivers of a broadcast or content delivery system in advance, instead of broadcasting or streaming the actual compressed audio signal, all that need be transmitted is the series of identifiers along with any modification instructions to the identified codewords. After reception, intelligence on the receiver having access to a locally stored copy of the dictionary can reconstruct the original audio clip by accessing the codewords via the received IDs, modify them as instructed by the modification instructions, further modify the codewords either individually or in groups using the audio profile of the original audio file (also sent by the encoder) and play back a generated sequence of phase corrected codewords and modified codewords as instructed. In exemplary embodiments of the present invention, such modification can extend into neighboring codewords, and can utilize either or both (i) cross correlation based time alignment and (ii) phase continuity between harmonics, to achieve higher fidelity to the original audio clip.