EP2321982B1 - Method for multi-channel processing in a multi-channel sound system - Google Patents

Description

1. a) Aufspaltung eines Kanals oder einer Kanalmischung auf Einzelkanäle;
2. b) Bearbeitung der entstandenen Einzelkanäle durch Einstellung des Parameters Kanalfader, der die Lautstärke eines jeden Einzelkanals regelt;
3. c) Limitierung der Einzelkanäle durch Einstellung von Werten der Parameter Kanalfader, Threshold, Release und Output Level,wobei jedem der Einzelkanäle ein Kompressor und/oder Limiter zuzuordnen ist, wobei durch Einstellung der Parameter Kanalfader, Threshold; Release, Output Level innerhalb eines Kompressors und/oder Limiters das Verhältnis der Einzelkanäle zueinander reguliert wird und
4. d) Encodierung der Einzelkanäle.

The invention relates to a method for multichannel processing in a multi-channel sound system, in particular in surround multichannel sound technology, with a matrix coding with the following method steps:

1. a) splitting a channel or a channel mixture into individual channels;
2. b) processing of the resulting single channels by setting the parameter channel fader, which controls the volume of each individual channel;
3. c) limitation of the individual channels by setting values of the parameters channel fader, threshold, release and output level, whereby each of the individual channels is assigned to a compressor and / or limiter, whereby by setting the parameters channel fader, Threshold; Release, Output Level within a compressor and / or limiter, the ratio of the individual channels to each other is regulated and
4. d) Encoding of the individual channels.

The development of multi-channel sound systems has been particularly driven by the Dolby laboratories. Following on from the "Dolby Surround" invented by the Dolby laboratories in the 1970s, there are now so-called "matrix surround processes", such as the. Dolby ProLogic, ProLogic 2, Circle Surround, Circle Surround 2. This creates the ability to play up to 7.1 channels, ie channels FL (FL), C (center), FR (FR = front right ) to encode (surround) the side surround channels LS (Leftsurround), RS (Rightsurround), the surround back channels BL (Back left), BR (Back right) and the channel LFE (low frequency effect). From these channels, two transmission channels Lt (L = left, t = total), Rt (R = right, t = total) are matrices, which contain all the information to be received from corresponding decoders distributed back to the original channels, ie to be rendered as the source channels after decoding. When encoding (encoding), among other things, the sound components of the information in the surround channels LS, RS, BL, BR phase-rotated by +/- 90 ° to the channels L and R added and with slight lowering of the volume level in the front channels R, L embedded.

This encoding of the individual channels is preceded by further process steps in multichannel audio processing: A first step in the usual multichannel processing process in the area of upmixing of audio material from stereo / mono to surround of any configuration is the splitting of a channel or a channel mix into individual channels. This splitting can be realized by appropriate software. After splitting, the individual channels are available for further processing, in particular to ensure the stability of the multi-channel mix compared to the original example stereo mix by a Mehrkanalcompression / limiting. For this purpose, a compressor / limiter is provided. A Compressor / Limiter is a limiter that prevents exceeding a peak level to prevent oversteer. They are also to be understood as volume / loudness regulators of instrument groups / vocals or speech. In addition, they compensate for part of the energy loss that results from the splitting of a channel or a channel mixture onto the individual channels even before the encoding. Each individual channel or channel groups is / are assigned a compressor / limiter. By adjusting the values of the Channel Fader, Threshold, Attack (if present), Release, and Output Level parameters within a compressor and / or limiter, one adjusts the ratio of the individual channels to each other. The result of this regulation is that the volume / loudness balance of the individual channels is stabilized to each other before encoding. The channel fader volume is used to regulate the volume of each individual channel in a mixing console, which is an important component in a recording studio. The normal setting a channel fader is 0 dB. If you set a channel fader to 0 dB, the signal originally present at the single channel will sound as it was originally flattened, assuming a correctly measured and neutral set gain on the corresponding channel strip. The channel gain controls the preamplification of a signal before it passes the channel fader. The threshold value acts like a threshold of the signal applied to the channel fader. A compressor / limiter works in such a way that the compressor / limiter limits the signal as soon as the applied signal exceeds the threshold value. The release value tells you the amount of time it takes the compressor / limiter to reset to zero after the applied signal drops below the threshold. The attack value determines the reaction time when the threshold value is exceeded. Finally, the output level indicates how strong the signal present on the channel is after being processed by the compressor / limiter. The output leveler is sort of a signal amplifier.

The method of the type mentioned in the context of the matrix surround technology outlined here has little or no acceptance. The relevant specialist literature states that the matrix surround technology can not compete with today's discrete digital methods (cf., for example, Christian Birkner "Surround, Introduction to Multichannel Sound Technology", PPV Presse Projekt Verlags GmbH, Bergkirchen, 2002). Among other things, this results from the realization that the Lt, Rt stereo sum created according to the current technical standards for in comparison with the conventionally produced stereo mixes, as they are processed in the TV area, radio and music usually one hundred percent within the programs for qualitatively can not keep up. The phase shifts caused by the encoding weaken the sound and affect the frequency response so that they sound "smaller and more spongy".

On the other hand, the matrix surround technology meets all requirements for a usable compatible surround system.

In view of this problem and in recognition of the cited prior art, the present invention is therefore the object of developing a method of the type mentioned in such a way that the Lt, Rt encoded surround mixes not only act as a data carrier of the mixes to be decoded, but in parallel for stand up to stereo and mono and improve their quality if necessary by working through the invented method.

This object is achieved with the features of claim 1. Advantageous embodiments of the invention are manifested in the dependent claims.

According to the invention, the channel faders are set to a uniform value and in method step c) limits at least two channels with a uniform output level value, wherein a channel C (center) in the output level value can vary and each further channel is limited in such a way that it has an output level value that is at least one decibel less than the uniform output level value of the two channels, wherein subsequent to step d) further compression and / or limitation of the encoded channels by setting a value of at least one of the parameters Channel Fader, Threshold, Release and Output Level.

Accordingly, for example, first in method step b) individual channels, preferably 5 individual channels, can be set to a uniform channel fader value (volume) in the basic position. The sixth channel within a 5.1 system would then, for example, receive a higher channel fader value depending on the audio material to be processed. This value is preferably between 0.1 to 5 dB. This basic setting can then be maintained within the scope of the invention either over the entire processing or manually changed. This would apply to all six channels (example of a 5.1 configuration). After standardized or manual adjustment of the channel fader values, the signals are then forwarded for limiting in method step c).

Thus, according to the invention, at least two channels can be limited with a uniform output level value with the same setting of the threshold and release, wherein a channel is freely set within certain thresholds both in the threshold and in the output level, depending on the audio material to be processed.

The invention is based on the recognition that the splitting on individual channels is accompanied by an uncontrolled splitting of the energy, which in turn has an effect on the volume / loudness constellation, as well as the phase position within the mix. If one splits a finished stereo mix, which has been harmoniously tuned on the channels FL and FR and has been optimized in volume and loudness, then this has the consequence that the original balance is sustainably destroyed. Neither the balance, the phase position nor the volume / loudness constellation within the mix are retained. The splitting of energy thus results in the weakening of the individual channels. The basic idea of the invention is now to set the balance between the individual channels by limiting them so that the original style mix is restored, or improved in sound and expression. The phase position within the surround mix is stabilized according to the invention even before the encoding by the constellation of the channel faders (step b)) and limitation. According to the invention, the optimal loudness is achieved by a further limitation subsequent to method step d) by a value of at least one of the parameters channel fader, Threshold, Release and Output Level is set.

The first limitation known from the prior art and manifested in method step c) is generally used in the surround region in order to maintain as high a volume as possible in the mix. However, it does not serve to stabilize the split channels according to the invention. A further idea of the invention is that, contrary to the prevailing view, following step d), a further limitation of the individual channels takes place. Since encoded surround mixes are poor compared to stereo and monomixes, they are a fluency program. B. less suitable for broadcasting and TV, since the encoded mix with stereo and monomixes competes here. The invention therefore provides, after encoding, to turn on a limiter in the form of a maximizer in order to control the loudness, i. H. to optimize the energy of the mix, as well as the soundscape. The result is surroundecoded mixes that are absolutely equivalent in stereo and monocontent. This limitation also preserves the energy of a stereo or mono mix, producing a sound that matches or enhances that of a stereo guitar.

In addition, an advantage of the invention is that the surroundecoded mix is more phase stable

Another advantage is that stereo encoded mixes after editing by the invented method sound much better than the previously known stereo. This method also includes the auditory space of the consumer so that the thus encoded stereo mixes sound more transparent, multi-dimensional and intense.

It is also possible by this method, already in a matrix surround process of any kind encoded stereo mixes to edit so that they in the new stability and sound quality in any Audio configuration.

Another area of application is the conversion of existing, created for a digital surround process single channels in the new result. Thus, e.g. Dolby Digital and dt's productions are made audible to consumers who do not have digital decoding equipment-with full stereo and mono compatibility.

Within the scope of the invention, a uniform value of the channels FL and FR can be set for the output level in method step c), wherein preferably an output level value between -8.0 dB (dB = decibel) and -24.0 dB is set. The output level value of channel C is freely adjusted according to the audio material to be processed. Preferably, the output level value of this channel to the channels FR and FL differs by +1 - +6 dB. In order to limit up to 7.1 channels, a further advantageous embodiment of the invention provides that in method step c) additionally a uniform output level value of the channels Ls, Rs, Bl, Br, LFE is set. The setting of an output level value between -9.0 dB and -25.0 dB, a channel fader value between 7.0 dB and 10.0 dB, a threshold value between -1.0 dB and -10 has proven particularly effective dB and a release value between 0.5 and 2.0 proved.

In addition, for the further limitation following process step d), the setting of a threshold value between -1.0 dB and -10.0 dB, an output level value between 0 dB and -1.0 dB and a release value between 0, 5 and 2.0 proved to be particularly advantageous.

The optional use of a multi-band compressor per channel can still influence the frequencies of a Mixes are taken. By emphasizing or lessening a corresponding Frequency range one can specifically repeat the lost frequencies, which have arisen in the change of the sound image in the context of the decoding of, for example, 2 to 6 channels. This effect is therefore equivalent to that of an equalizer, which can also be used as an alternative to the multi-band compressor.

Likewise, it may be advantageous to carry out a decompression per individual channel with corresponding starting material in method step b). The invention has come to the realization that a decompression per single channel after splitting provides a higher-quality result than the previously often used decompression of the total to be split before method step a). A decompressor is a compressor, which is equipped with the parameters Threshold, Attack, Release, Ratio and Output Level. The setting of a Ratio value below the value 1.00, as well as long Attack- and release values can lead to more dynamic results with appropriate starting material, which was inherently too rough or uncomfortably mastered by Brickwall compression. A threshold and output level setting adapted to the source material is required.

Finally, the invention provides the use of the method for a recording medium, a sound carrier, a digital data set and a conversion automation of conventional audio formats to the new sound possibilities. This can be built into audio / video equipment, or run as stand-alone hardware or software, as well as implemented in software and hardware-based host applications.

In the following, a particularly preferred embodiment of the invention is represented by means of a 5.1-channel surround system with the following set parameters:

Limitation before encoding using example 5.1 a) The channel faders FL, FR, Ls, Rs, LFE of the previously split audio tracks have a setting of 0db, the channel fader value of C deviates upwards, depending on the audio material to be processed

In the mixer, the channel faders control the volume of each channel. The normal sound level of a fader is 0 dB. The channel fader is set to 0 dB, so that the signal present at the channel will sound as it was originally leveled out, assuming a correctly adjusted and neutral set gain value.
After this configuration, the signals are sent to the channel faders with limiters

b) Each channel fader of the 6 channels with limiter has a setting of +8.3 dB.

So you drive over each of the 6 channels by 8.3 dB. This is important because in this way the actually too quiet 6 individual channels are strengthened again.

c) Each of the 6 limiters has a "Threshold" value of -3.3 dB.

The "Threshold" value shows the threshold value of the signal present at the channel fader, from which the limiter begins to work. A limiter works by limiting the signal as soon as the signal exceeds the threshold value. Unlike a compressor, however, this is rather simplistic because it reacts immediately, while a compressor has several deficient individual settings, how exactly should be compressed. The limiter has more of the characteristics of a signal compression. If the value, in this case -3.3 dB, is strictly limited from above (and the signal wins) by loudness that the energy within the signal is increased because the level can not increase any further) in such a way that it nevertheless does not change its character. If all channels were simply limited, the result would seem strong, but flat and without depth and with a "pumping" effect. The Threshold value on channel C may vary in the threshold value depending on the audio being edited.

d) Each limiter has a release value of 1.00

The "Release" value tells you how long the limiter needs to return to the zero position after the applied signal drops below the threshold value again. If a signal exceeds the Threshold limit, the limiter limits the level. If the signal drops again, there are several ways to select the "Release" value.

e) The "Output Level" value

The "Output Level" value indicates the strength of the signal applied to the channel after processing by the limiter. It is basically a normal signal amplifier / limiter. The advantage of this value is that a fixed configuration can be found, which makes it possible - in later stereo / mono but also in all surround formats - to find the right volume ratios again.

Advantageous settings are: - Channel "FL": - 17.5 dB - Channel "FR": - 17.5 dB - Channel "C": - 17.5 dB - Channel "LS": - 18.5 dB - Channel "RS": - 18.5 dB - Channel "LFE": - 18.5 dB

The "Output Level" value of channel C may vary, depending on the audio material being edited

This configuration stabilizes the mix again. The basically very low output level values are due to the fact that the mix recharges enormously in terms of energy, since in the encoding process 6 again become 2 channels (Lt, Rt). The potentiation of energy is thus the result. In the case of higher output level values, there is a strong overload in the encoding, with the result that the mix is destroyed.

For the further limitation by means of a compressor / limiter / maximizer provided after process step d), the following values are preferably to be selected:

a) Threshold

You can not standardize the Threshold value Here, your hearing dictates: The further you pull the value down (that is, the earlier the Maximizer starts to work), the more loudness the encoded mix will get as the frequency and sound patterns change , This setting is audio dependent. The more loudness the original signal has (for example, Dance Music), the less threshold is needed. A threshold value of -2.6 dB has proven to be advantageous

b) Output Level

The out-ceiling value of -0.1 dB has proven itself in almost all mixes. After the threshold value is set and the signal (minus) is 0.1 dB, the level peaks within the mix are intercepted due to the single channel addition within the surround mix. An override of the 0 dB mark is thereby avoided.

This produces a result that sounds the same or better than the original stereo, but looks different in amplitude. The result is more dynamic with increasing loudness, which usually excludes each other. Normally, when audio is louder by compression / limitation, but less dynamic at the same time, this does not exclude the method discussed here, but one can gain both dynamics and loudness.

c) release value

The Maximizer uses a "Release" value of 1.00. This has proven itself insofar as the result sounds original.

By optimally setting all the values within the invented method, one produces a higher sound quality in any audio environment compared to the original.

This procedure can also be applied to ready-made single tracks of a surround mix (discrete tracks) to form a Lt, Rt track.

Claims (13)

1. Method for multichannel processing in a multichannel sound system, particularly in multichannel surround sound engineering, using matrix coding, having the following method steps:

   a) a channel or channel mix is split into single channels;

   b) the resultant single channels are processed by setting the channel fader parameter, which regulates the volume of each single channel;

   c) the single channels are compressed and/or limited by setting values for the parameters channel fader, threshold, release and output level, each of the single channels being able to be assigned a compressor and/or limiter, wherein setting the parameters channel fader, threshold, release and output level within a compressor and/or limiter regulates the ratio of the single channels to one another, and

   d) the single channels are encoded,
   characterized in that the channel faders are set to a standard value in method step b) and at least two channels are limited using a standard output level value in method step c), wherein a channel C (C=centre) may have a variable output level value and each further channel is limited such that it has an output level value that is at least one decibel lower than the standard output level value of the two channels, wherein method step d) is followed by further compression and/or limiting of the encoded channels being effected by setting a value for at least one of the parameters channel fader, threshold, release and output level.
2. Method according to Claim 1, characterized in that at least one channel fader is set above the standard (identical) value.
3. Method according to Claim 1, characterized in that the values of the channel faders after a channel or a channel group has been split into single channels is identical for all channels.
4. Method according to Claim 3, characterized in that at least one channel fader is set above the identical value.
5. Method according to one of Claims 1 to 4, characterized in that channel C (C=centre) is above the standard output level value following limiting.
6. Method according to one of Claims 1 to 5, characterized in that in method step c) a standard output level value for the channels FL (FL=front left), FR (front right) is set.
7. Method according to Claim 6, characterized in that in method step c) a standard output level value for the channels LS (LS=left surround), BL (BL=back left), BR (BR=back right), RS (RS=right surround), LFE (LFE=low frequency effect) is additionally set.
8. Method according to one of the preceding claims, characterized in that in method step c) a channel fader value between 7 dB and 10 dB is set for the channels FL, FR, C, Ls, Rs, LFE.
9. Method according to Claim 8, characterized in that the channel fader value 8.3 dB is set.
10. Method according to one of the preceding claims, characterized in that in method step c) a threshold value between -1 dB and -10 dB is set for the channels FL, FR, C, Ls, Rs, LFE.
11. Method according to Claim 10, characterized in that the threshold value -3.3 dB is set.
12. Method according to one of the preceding claims, characterized in that method step d) is followed by a threshold value between -1.0 dB and -10.0 dB being set for the channels FL, FR, C, Ls, Rs, LFE.
13. Method according to Claim 12, characterized in that the threshold value -2.6 dB is set.