BACKGROUND OF THE INVENTION

The present invention relates to a decoder for decoding a media signal and an encoder for encoding secondary media data comprising metadata or control data for primary media data.

In other words, the present invention shows a method and an apparatus for distribution of control data or metadata over a digital audio channel. An embodiment shows the convenient and reliable transmission of control data or metadata to accompany an audio signal, particularly in television plants, systems, or networks using standard AES3 (AES: audio engineering society) PCM (pulse code modulation) audio bitstreams embedded in HD-SDI (high definition serial digital interface) video signals.

In the production and transmission of music, video, and other multimedia content, the reproduction of the content can be enhanced or made more useful or valuable by including metadata describing characteristics of the content. For example, music encoded in the MP3 format has been made more useful by including ID3 tags in the MP3 file to provide information about the title or artist of the content.

In video content, it is common to include not only descriptive metadata, but data for controlling the reproduction of the content depending on the consumer's equipment and environment. For example, television broadcasts and video discs such as DVD and Blu-ray include dynamic range control data that are used to modify the loudness range of the content and downmix gains that are used to control the conversion of a surround sound multichannel audio signal for reproduction on a stereo device. In the case of dynamic range control data, gains are sent for each few milliseconds of content in order to compress the dynamic range of the content for playback in a noisy environment or where a smaller range of loudness in the program is advantageous, by optionally multiplying the final audio signal by the gains.

The means of inclusion of such metadata or control data in a digital bitstream or file for delivery to consumers is well established and specified in audio coding standards such as ATSC A/52 (standardized in Advanced Television Systems Committee, Inc. Audio Compression Standard A/52) or MPEG HE-AAC (standardized in ISO/IEC 14496-3 and ETSI TS 101 154).

However, the transmission of metadata or control data in the professional or creative environment, before the content is encoded into a final bitstream, is much less standardized. Until now this information has been primarily static in nature, remaining constant over the duration of the content. Although, loudness control gains are dynamic, in content production standard “encoding profiles” may be established to control the generation of the gains during the final audio encoding process. In this manner, no dynamic metadata may be recorded or transmitted in the content creation environment.

The development of object-oriented audio systems, where sounds in two or three dimensions are described not by levels in traditional speaker channels or Ambisonic components, but by spatial coordinates or other data describing their position and size, now involves the transmission of dynamic metadata that changes continuously, if such sounds move over time. Also, static objects are used to allow the creation of content with disparate additional audio elements, such as alternate languages, audio description for the visually impaired, or home or away team commentary for sporting events. Content with such static objects no longer fits into a uniform model of channels, such as stereo or 5.1 surround, which professional facilities are currently designed to accommodate. Thus, descriptive metadata may accompany each item of content during production or distribution so that the metadata may be encoded into the audio bitstreams for emission or delivery to the consumer.

Ideally, professional content formats would simply include provisions for such position or descriptive metadata in their structure or schema. Indeed, new formats or extensions to existing formats, such as MDA or BWF-ADM have been developed for this purpose. However, such formats are not understood in most cases by legacy equipment, particularly for distribution in systems designed for live or real-time use.

In such systems, legacy standards such as AES 3, MADI, or embedded audio over SDI are common. The use of these standards is gradually being augmented or replaced by IP-based standards such as Ravenna, Dante, or AES 67. All of these standards or techniques are designed to transmit channels of PCM audio and make no provisions for sending dynamic or descriptive metadata.

One technique considered for solving this problem was to encode the audio in a “mezzanine” format using transparent-bitrate audio coding so an appropriately formatted digital bitstream also containing static metadata could be included. This bitstream was then formatted such that it could be sent as PCM coded audio data over the traditional television plant or professional infrastructure. A common implementation of this technique in the television industry is the Dolby E system, carried in a PCM AES3 audio channel according to SMPTE standard ST 337.

Dolby E allowed legacy equipment designed with four PCM audio channels to be used for the 5.1 channels needed for surround sound, and also include provisions for transmitting the “dialnorm” or integrated loudness value of the program.

Use of the Dolby E system revealed several operational shortcomings: One issue was the inclusion of sample rate conversion in many devices used to embed the PCM audio signals in the SDI infrastructure of production or distribution facilities. Sample rate conversion or resampling of the audio signal is commonly performed to insure correct phase and frequency synchronization of the audio data sampling clock with that of the video sampling clock and video synchronization signals used in the facility. Such resampling has a normally inaudible effect on a PCM audio signal, but changes the PCM sample values. Thus, an audio channel used for transmitting a Dolby E bitstream would have the bitstream corrupted by resampling. In such cases, the resampling may be disabled and other means used to insure synchronism of the sample clocks within the facility.

Another issue was the delay introduced by the block-transform nature of the audio codec employed. The Dolby E codec used one video frame (approximately 1/30 second for interlaced ATSC video) for encoding and one video frame for decoding the signal, resulting in a two-frame delay of the audio relative to the video. This involves delaying the video signal to maintain lip-sync, introducing additional delay in the distribution infrastructure.

A third issue is the need to program SDI routing switchers to treat inputs carrying Dolby E bitstreams as data channels instead of audio signals. Although Dolby E contains a “guard band” around the video signal's vertical interval to allow routing switchers to switch to another input without loss of the Dolby E data, many routing switchers perform a cross-fade of the audio signals during such a switch to prevent audible pops or transients in normal PCM audio signals. These crossfades are of 5-20 ms in duration and corrupt the Dolby E bitstream around the switch point.

These operational limitations resulted in most TV facilities abandoning the use of Dolby E in favor of a strategy of normalizing the dialnorm level of all content upon ingest to their network, so that fixed dialnorm values and dynamic range profiles could be programmed into their emission audio encoders.

An alternative technique sometimes used in TV facilities is to insert metadata information into the SDI video signal itself in the VANC data as standardized in SMPTE standard ST 2020.

Often this is combined with carriage of the metadata using the user bits of AES3. However, ordinary SDI embedding equipment does not support the extraction of this metadata from the AES stream for insertion into VANC bits.

An additional technique sometimes used is to encode dynamic control data within a PCM audio signal by inserting it into the LSBs of the audio signal. Such a technique is described in the paper “A Variable-Bit-Rate Buried-Data Channel for Compact Disc” by Oomen and has been employed in implementations of the MPEG Surround audio coding standard. However, such buried data does not survive sample rate conversion or truncation of the LSB.

A related technique is to use extra bits such as User Bits or Auxiliary Sample Bits specified in the AES3 standard as a side data channel suitable for dynamic control data. Unfortunately, many implementations of the AES3 standard discard this information.

A further limitation of the aforementioned techniques is they are intended for use in only in a technical transmission environment. If they were routed through creative equipment, such as an audio console or digital audio workstation, even if no operations were performed on the containing PCM channel, it could not be guaranteed that the data path through the console was bit-exact, as such equipment is not designed for such purposes. Even if such bit-exactness could be assured, the mere accident of touching a control fader and thus inducing a slight gain change in the PCM channel, would corrupt the signal.

Common to all these techniques are the limitations imposed by creation and transport equipment that is designed solely for the purpose of carrying PCM audio signals, without consideration for the embedding of digital control data.

Therefore, there is a need for an improved approach.

SUMMARY

An embodiment may have an encoder for encoding secondary media data having metadata or control data for primary media data, wherein the encoder is configured to encode the secondary media data to obtain a stream of digital words, the encoding including transforming the secondary media data by a digital modulation or including bandlimiting, and wherein the encoder is configured to output the encoded secondary media data as the stream of digital words.

Another embodiment may have a decoder for decoding a media signal including a received stream of digital words representing encoded secondary media data including metadata or control data for primary media data; wherein the decoder is configured to recover the secondary media data, wherein the recovering includes applying a digital demodulation operation or a resampling operation to obtain recovered secondary media data, and wherein the decoder is configured to derive a bitstream from the recovered secondary media data.

According to another embodiment, a media signal may have: a stream of digital words representing encoded secondary media data including metadata or control data for primary media data.

According to another embodiment, a method for decoding a media signal including a received stream of digital words representing encoded secondary media data including metadata or control data for primary media data may have the steps of: recovering the secondary media data, wherein the recovering comprises applying a digital demodulation operation or a resampling operation to obtain recovered secondary media data, and deriving a bitstream from the recovered secondary media data.

According to another embodiment, a method for encoding secondary media data including metadata or control data for primary media data may have the steps of: encoding the secondary media data to obtain a stream of digital words, the encoding including transforming the secondary media data by a digital modulation or including bandlimiting, and outputting the encoded secondary media data as the stream of digital words.

Another embodiment may have a non-transitory digital storage medium having a computer program stored thereon to perform the inventive methods when said computer program is run by a computer.

According to another embodiment, a data processing system may have: an inventive encoder; and an inventive decoder.

The present invention is based on the finding that secondary media data, for example metadata carrying further information of the content of a first media signal (e. g. payload data) or control data comprising data to control the reproduction of the content of the first media data, may be arranged in a stream of digital words that is robust against a significant variety of signal manipulations. Embodiments show the stream of digital words as an audio-like digital signal being able to withstand or to be robust against signal manipulation which is typical for audio signals. The signal processing might be a transformation of the sampling frequency, an amplification or attenuation of the signal or a DC (direct current) offset. The transformation of the sampling frequency may be performed e.g. if the stream of digital words is arranged in a higher order stream such as e.g. an AES3 PCM digital audio channel, where a sampling frequency of the encoder creating the stream of digital words is different from a sampling frequency of a signal processor, such as an AES3 digital audio interface, creating the higher order stream. Therefore, the secondary media data can be treated as a typical audio signal and may be therefore implemented in one of multiple audio channels in present systems, for example in special hardware in television (TV) studios. A special embodiment might be an SDI video signal containing 16 audio channels, where one audio channel is used for metadata or control data. The SDI video signal may also contain one or more video channels. The audio channels may be PCM digital audio channels. Therefore, the metadata or control data may be encoded as a robust analog-like digital signal instead of a standard digital bitstream, to be robust against signal manipulation typical for PCM digital audio channels. Present systems may be extended to comprise control data or metadata by replacing current encoders and decoders with encoders and decoders described below. This replacement can be achieved by a comparably inexpensive software update. Even if the encoder and decoder are realized in hardware, further (expensive) hardware such as broadcast equipment can remain unchanged.

Embodiments show an encoder for encoding secondary media data comprising metadata or control data for primary media data. The encoder is configured to encode the secondary media data to obtain a stream of digital words, the encoding comprising transforming the secondary media data by a digital modulation or comprising bandlimiting. Moreover, the encoder is configured to output the encoded secondary media data as a stream of digital words. Therefore, the stream of digital words may be formed such that it is able to resist a typical processing of a digital audio stream. Furthermore, means for processing a digital audio stream are able to process the stream of digital words, since the stream of digital words may be designed as an audio-like or analog-like digital stream.

Embodiments relate to the encoding. The encoding may comprise adding redundancy by the digital modulation. The digital modulation, e.g. a pulse amplitude modulation, may be so that two or more bits of the secondary media data are transmitted per digital word of the stream of digital words. Moreover, the encoder may output the stream of digital words so that the stream of digital words is transmittable over a PCM audio channel. Furthermore, the encoder might output a further stream of digital words. The further stream of digital words represents the primary media data and the further stream is separated from the stream of digital words. The primary media data may be audio data and the secondary media data could be metadata for the audio data or control data for the audio data. Therefore, the encoder may be configured to output the stream of digital words and the further stream of digital words so that the further stream of digital words is transmittable over a first audio PCM channel and so that the stream of digital words is transmittable over a second audio PCM channel being different from the first audio PCM channel. Each of the digital words of the further stream representing the primary media data might have a predefined number of bits being greater than 8 bits and smaller than 32 bits, and wherein each of the digital words of the stream of digital words may have the predetermined number of bits as well. The encoder may further generate the stream of digital words so that the stream of digital words comprises a timing reference pattern or an amplitude reference pattern.

Further embodiments show an alignment of the secondary media data. Therefore, the encoder outputs a video stream representing a sequence of video images, so that the control data or meta data of the secondary media data related to a certain video image are related to the certain video image. This is advantageous, since the sequence of video images may be cut at any video image or between any of consecutive video images and the following video image still comprises the control data or meta data related to this video image. Furthermore, the encoder may output the stream of digital words as a first stream of digital words associated to a first video image of the sequence of video images, and to output the stream of digital words as a second stream of digital words associated to a second video image of the sequence of video images, wherein the first and second digital words are identical to each other. This may be advantageous, if consecutive video images comprise identical metadata or control data, to ensure that each video image comprises the metadata or control data referring to the video image.

Moreover, embodiments show the encoder to output the encoded secondary media data as the stream of digital words as a control track and to output up to 15 channels of the primary media data as audio tracks, wherein the control track and the audio tracks are formed in accordance with the AES 3 standard.

Further embodiments show the encoder being configured to generate the digital words, the digital words having 12 to 28 bits, or wherein the digital words are sampled at a sampling rate of between 30 kHz to 55 kHz, or wherein the digital words have a dynamic range of 70 to 160 dB, or have a nominal signal level of −20 dB RMS full scale. The encoder may use an upper frequency for bandlimiting the secondary media data being between 15 kHz to 27.5 kHz for a sampling rate between 30 kHz to 55 kHz.

Embodiments further show the encoder comprising a mapper and a stream builder. The mapper is configured for mapping the grouped secondary media data comprising a first number of bits into a data word comprising a second number of bits being greater than the first number of bits. Furthermore, the grouped secondary media data is aligned with a gap to a most significant bit or a least significant bit of the data word. The stream builder is configured for building a stream representing the encoded secondary media data using a reference pattern and a plurality of data words. This is advantageous, since the gap enables an amplification of the grouped secondary media data by about 6 dB (or with a factor of 2) for each bit the gap comprises to the most significant bit and an attenuation of about 6 dB (or with a factor of 0.5) for each bit the gap comprises to the least significant bit of the data word. Therefore, it does not matter whether the amplification or attenuation is applied on purpose or accidentally, since the structure of the data word, with the mapping of the grouped secondary media data (information) to the data word, where at both ends of the grouped secondary media data padding is applied to obtain the data word, enables bit shifting (amplification by factor 2 for each bit shifted to the most significant bit or attenuation by factor 0.5 for each bit shifted to the least significant bit). Therefore, the grouped secondary media data is not corrupted and remains valid until the amplification or attenuation is greater than the padding.

Embodiments further show the encoder comprising a grouper for grouping a bitstream of secondary media data to form grouped secondary media data. Moreover, the encoder may comprise a reference signal generator for generating a reference pattern indicating a reference amplitude or a predetermined timing instant in the primary media data. The stream builder may build a stream of digital words representing encoded secondary media data using the reference pattern or the data word. The reference pattern may indicate a reference amplitude or a predetermined timing instant in the primary media data. An analysis of the reference pattern in a decoder enables the decoder to calculate an amplification or attenuation or a DC offset applied to the stream of digital words after the stream was encoded in the encoder. Furthermore, a sampling rate of the stream of digital words may be determined from the predetermined timing instant in the primary media data.

The stream builder may further comprise a filter to low-pass filter the data words or the reference pattern to obtain digital words comprising a length of more than one sample of a predetermined sample rate, wherein an amplitude of the digital word is weighted according to the data word or the reference pattern, and wherein the filter is configured to add up consecutive digital words at instants of the predetermined sample rate to obtain the stream of digital words. Applying the filter is advantageous, since the secondary media data is more vulnerable to a resampling than normal audio data. Therefore, the filter enables the secondary media data to withstand applied resampling steps between the encoder and the decoder or in the decoder with respect to the encoder, and to withstand the resampling step that may be used, in the decoder period. Moreover, the stream of digital words may be analog and again digital converted during resampling without considerable loss. However, resampling may not be the same as converting a digital signal to an analog signal. Analog conversion may involve filters with impulse responses that would smear the data, and the analog-to-digital conversion might add quantizing noise to the signal, as well as any analog noise (thermal or semiconductor generated noise, hum or interference, etc). A signal which is generated using the inventive concept is able to withstand a resampling and an digital-to-analog conversion.

According to further embodiments, the filter is configured to obtain zero points at instants of a predetermined sample rate of a data pulse, wherein a data pulse comprises a data word comprising grouped secondary media data or the reference pattern. Furthermore, the stream builder is configured to build the stream representing the encoded secondary media data using the reference pattern and a plurality of data words such that zero points of the data pulse are aligned with a maximum of a further data pulse to obtain an inter-symbol-interference-free stream representing the encoded secondary media data. In other words, it is advantageous to use a Nyquist filter, since a Nyquist-filtered signal may be decoded in the decoder without inter-symbol-interference. In other words, it is advantageous to use a filter satisfying the Nyquist criterion for zero inter-symbol interference. According to embodiments, the cutoff frequency of the filter may be less than 1.5 times of a sampling frequency of the primary media data.

According to an embodiment, the reference signal generator generates a grouped reference pattern comprising a first number of bits. The reference signal generator is further configured to map the grouped reference pattern into a data word comprising a second number of bits being greater than the first number of bits. Alternatively, the mapper maps a grouped reference pattern comprising a first number of bits into a data word comprising a second number of bits being greater than the first number of bits. The embodiments describe options to apply the format of the data words comprising metadata or control data to the reference pattern. Advantageously, the reference pattern obtains the same precautions against amplification or attenuation of the media signal than the secondary media data. Therefore, the reference signal generator may provide the reference pattern in a form of the mapped secondary media data, meaning that the reference pattern comprises a first number of bits and is mapped into a reference pattern comprising a second number of bits being greater than the first number of bits and comprising the same gap to the most significant bit and the least significant bit as already described in the decoder and the encoder. Alternatively, the reference signal generator outputs a reference pattern comprising a first number of bits. In accordance with the secondary media data, the mapper maps the reference pattern with a first number of bits into a data word with a second number of bits.

Embodiments further show a decoder for decoding a media signal comprising a received stream of digital words representing encoded secondary media data comprising metadata or control data for primary media data. The decoder is configured to recover the secondary media data using manipulating the received stream of digital words with respect to amplitudes represented by the received digital words or using resampling. The decoder is configured to derive a bitstream from the recovered secondary media data.

Embodiments further show the decoder comprising a reference signal generator, a signal manipulator, and a signal processor. The reference pattern analyzer analyzes a reference pattern of the encoded secondary media data, wherein the reference pattern analyzer is configured to determine an amplitude of the reference pattern or to determine a predetermined timing instant in the primary media data. The signal manipulator manipulates the encoded secondary media data in accordance with the analyzed reference pattern and a computed reference pattern to obtain secondary media data. The signal processor processes the primary media data according to the encoded secondary media data to obtain a decoded media signal. This is advantageous, since the signal processing applied to the media signal during the encoding enables the signal manipulator to accurately regain the media signal from the encoded media signal, independent from typical signal manipulations like amplification etc.

According to embodiments, the signal manipulator comprises a sample rate converter configured to convert a sample rate associated with the digital words, according to a predetermined timing instant of the primary media data indicated in the reference pattern, to a predetermined sample rate to obtain resampled digital words. This is advantageous, since standards for audio sampling rates may be mixed during processing of the media data. Even a small sample rate conversion from e.g. 48 kHz to 48.1 kHz corrupts the secondary media data since, in contrast to audio data, there is no redundancy or dependency in the secondary media data, which comprises metadata or control data. In other words, consecutive symbols of the secondary media data may vary from the highest possible value to the lowest possible value within one sample. This results in very high frequencies due to the strong changes within the secondary media data.

In contrast to the secondary media data, however, audio samples are typically band-limited, meaning that audio data changes are limited to a maximum frequency determined by the sampling frequency.

Further embodiments describe the reference pattern analyzer comprising a timing instant determiner configured to determine the predefined timing instant of the primary media data in the reference pattern in terms of samples of a sample rate, an upsampler configured to upsample a range around the determined timing instant to determine an exact position of a predetermined timing instant, and a sampling accumulator configured to determine an exact position of the digital words within the stream of digital words to obtain an actual sample rate associated to the digital words being different from a predetermined sample rate.

Embodiments further show the reference pattern analyzer comprising a gain factor calculator configured to calculate an amplification or attenuation factor according to the amplitude or the reference pattern and the amplitude of the computed reference pattern and wherein the signal manipulator comprises a multiplier configured to amplify or attenuate the data words according to the amplification or attenuation factor to obtain gain compensated data words. This is advantageous, since an amplification or attenuation of the encoded media signal is one of the main issues which may be caused during transfer of an encoder to the decoder. It may be applied on purpose, for example in an equalizer, if other audio channels should be amplified or attenuated on purpose or accidentally due to a channel with the above mentioned characteristics.

According to a further embodiment, a media signal comprising a stream of digital words is shown. The stream of digital words represents secondary media data comprising metadata and control data for primary media data.

Further embodiments show the reference pattern analyzer comprising an amplitude detector configured to determine the amplitude of the reference pattern and a further amplitude of the reference pattern. The reference pattern analyzer may further comprise an offset compensation unit configured to calculate an offset of the encoded secondary media data according to a drift of the amplitude of the reference pattern and the further amplitude of the reference pattern and wherein the second manipulator comprises an adder configured to add the calculated offset of the encoded secondary media data from the encoded secondary media data to obtain offset compensated encoded secondary media data. The advantages of the embodiment are similar to those of the previous embodiment of the gain factor calculator, where an offset may be applied to the encoded secondary media data instead of a gain, e.g. during an equalization process between the encoder and the decoder, or accidentally from a drift caused by the transmission channel.

Embodiments further show the signal manipulator comprising a demapper configured to demap grouped secondary media data comprising a first number of bits from the data words comprising a second number of bits being greater than the first number of bits. Additionally or alternatively, the signal manipulator comprises an ungrouper configured to ungroup grouped secondary media data comprising a first number of bits to obtain a decoded media data bitstream. The digital words may further comprise the digital words comprising filtered secondary media data comprising a reference pattern and a plurality of data words, wherein the secondary media data is mapped into data words with a gap to the most significant bit of the data word or the least significant bit of the data word. Moreover, the reference pattern may comprise a reference amplitude of the encoded secondary media data and a predetermined timing instant in primary media data and wherein the plurality of data words comprise secondary media data.

Embodiments show the media signal comprising a further stream of the primary media data, wherein the primary media data comprises audio data or video data, wherein the further stream comprising primary media data is aligned to the stream of encoded secondary media data and the predetermined timing instant in the primary media data. This is advantageous, since the timing instant in the primary media data allows an accurate alignment of the secondary media data to the primary media data. In other words, an audio signal and metadata or control data may be aligned to frames of a video signal at a vertical blanking or a further synchronization signal of the video signal. Furthermore, the timing instant may be a synchronization signal in an audio signal, where the secondary media data is aligned to. Therefore, the secondary media data may be also applied to audio-only streams. The idea is to provide any information of the secondary media data within each frame of the video signal. Since the secondary media data is aligned to the time instant in the primary media data where the video stream is cut, the secondary media data remains unchanged and is intact. Therefore, each video frame may contain any information from the secondary media data even if the video signal comprising the video frame is cut.

Embodiments may be developed according to the following considerations. Therefore, it is an advantage of embodiments of the invention to provide a means for carrying static and dynamic control data or metadata accompanying PCM (pulse code modulation) digital audio signals through traditional creative and distribution equipment which only provides PCM audio channels.

This may be accomplished by considering the PCM digital audio channel's fundamental nature as a transmission means for an audio signal. Such audio signals are normally digitized for television use at a bit-depth of 16 to 24 bits and at a sampling rate of 48 kHz and have a resulting dynamic range of 90 to 140 dB, with a nominal signal level of −20 dB RMS (root mean squared) full scale.

Thus, if one considers the typical AES3 transmission channel as a digitized communication channel having these characteristics, the modulation techniques commonly employed in digital communications may be used to send modulated data over the channel. Such techniques are naturally immune to gain changes, moderate time base distortions, and in many cases, frequency response distortions of the channel.

The AES3 PCM digital audio channel differs from the channels used for digital communication. It is strictly a digital channel, and does not suffer from the multipath and rapid channel fading typical of radio communications channels. Given the 90 to 140 dB dynamic range, it is not practically limited in potential transmit power to provide sufficient carrier to noise ratio. When used in video systems, such as embedded in the SDI (serial digital interface) video signal, it has an inherent block nature due to the need to avoid the video vertical sync interval where switching can occur. Also, unlike many communications systems, there is a need for low latency, to avoid lip-sync issues or to avoid difficulties in monitoring audio when producing live broadcasts.

The throughput requirements of the control data or metadata needed for object audio vary by the number of objects, whether they are static or dynamic, and the particular object audio standard employed. One such standard is the MPEG-H Audio specification, ISO/IEC 23008-3. In this standard, typical use cases involve metadata or control data being encoded in streaming packets using the MHAS (MPEG-H Audio Stream (defined in ISO/IEC 23008-3 in Chapter 14 “MPEG-H 3D audio stream)) specification at bitrates of 10-30 kb/s.

For example, each dynamic object in a MPEG-H audio scene may use 1.5 kb/s for transmission. Thus, a program with 16 dynamic objects (a practical maximum given that the SDI interface only supports 16 channels of embedded audio) may use about 25 kb/s of data. Static metadata regarding the audio scene could take another 40-50 kb/s, if it was sent each audio frame.

The potential bit error rate (BER) needed can be estimated by considering the following factors: If a single bit error were permitted in operation once per year, given a bitrate of 75 kb/s, 2.36E12 bits would be sent in a year, requiring a bit error rate of 4.2E-13. However, the information in the control data is highly redundant. In most cases, bit errors will be detected by the underlying MHAS protocol and the control data would be interpolated from surrounding packets. Additionally or alternatively, CRC (cyclic redundancy check) values, e.g. using 16 bit, or other suitable codes or mechanisms to check for bit errors may be used. In this case, a bit error once per hour might be a reasonable upper limit. This latter case would use a BER of 3.7E-9. Thus, a reasonable BER for this transmission scheme would likely need a BER between 1 E-9 and 1E-12, which is easily possible with the high signal to noise ratios available in the AES3 digital audio channel.

It should be noted that the typical expressions for BER for communications channels do not apply here, as the noise in this channel is strictly that of quantization and resampling, with a rectangular or possibly (in the case dither is applied) triangular probability density function.

The time-base error introduced by sample rate conversion (or more precisely, by sources operating asynchronously) is limited by the accuracy of the clock sources employed in each piece of equipment acting as an asynchronous source. Most professional television facilities operate with clock or synchronization signal sources generated from accurate crystal, GPS, or rubidium standards, typically with a maximum frequency tolerance of 0.1 to 1.0 ppm. Typical consumer equipment may have frequency tolerances of 30 ppm. Allowing some margin for the case of consumer equipment operating at temperature extremes, a tolerance of 100 ppm may be safely assumed, for the case of consumer equipment operated in the field being connected to a professional TV plant.

Thus, a possible set of design assumptions and goals for applying this invention for the purpose of transmitting the control data or metadata needed for a common use of the MPEG-H Audio standard are:

Sampling Frequency48 kHzSymbol Frequency16 kbaud (⅓ sample rate forconvenience)Desired bitrate75 kb/sMaximum latency, end to end240 samples or 5 msMaximum time-base error100 ppmChannel Bit Depth14 bits (allowing for poor rounding,extra quantizing noise in poor digitalaudioequipment design, etc.)Channel Gain+15 to −20 dB (to allow for gainerrors in equipment, or inadvertentadjustment of a channel gain inprocessing equipment or an audioconsole or workstation)Nominal RMS or loudness value−30 to −15 dB FS (to allowof signaloperationa personnel to monitorthe signal level of the audiochannel as they would for normalaudio signals)

A further goal of a embodiment of this invention is to allow ease of implementation and debugging by audio coding engineers, who are very familiar with the building blocks used in perceptual audio coding, but who may not have experience with the implementation techniques common to data communications.

Given the channel bandwidth of 24 kHz, and design symbol rate of 16 kbaud, simple classical modulation techniques such as ASK or PSK will not be adequate. Modulation that provides coding efficiency of at least 5 b/s/Hz will be used.

Those skilled in the art will realize that a number of commonly used modulation techniques for digital communications would satisfy these design assumptions and goals. For example, 64 QAM (Quadrature Amplitude Modulation with an alphabet of 64 symbols) could be used, as it provides a coding efficiency of 6 b/s/Hz. However, implementing a QAM demodulator generally uses moderately complex signal processing to recover the carrier frequency and symbol clock, including the use of digital phase lock loops (PLL) which are unfamiliar to audio coding engineers. Such PLLs involve tuning of loop filters or accumulators to avoid loop instability, and may need some time to stably acquire the signal after a transient or switch.

The embodiment presented here uses 32 PAM (Pulse Amplitude Modulation with 32 levels) as an alternative that does not involve PLLs and produces a design that uses signal processing functions commonly employed in audio coding. PAM involves a 6 dB increase in signal to noise ratio for each increment of coding efficiency compare to the 3 dB needed with QAM, but in this system the signal to noise ratio is inherently high, while the design and debugging costs of a PAM receiver are lower.

All of the previously described embodiments may be seen in total or in combination, for example in a television plant where the encoder encodes a video signal with a corresponding audio signal and metadata or control data (secondary media data), for example at a first sampling frequency and wherein the decoder may be applied to a control instance (e. g. monitoring unit) or an emission instance before transmission of the media signal to a consumer.

