EP1415297B1 - Automatic generation of musical scratching effects - Google Patents

Abstract

The invention relates to a method for generating electrical sounds and to an interactive music player. According to the invention, an audio signal in digital format, which lasts for a predeterminable length of time, is used as the starting material. The reproduction position and/or the reproduction direction and/or the reproduction speed of said signal is/are modulated automatically with respect to the rhythm using control information in different predeterminable ways, based on information concerning the musical tempo.

Description

The invention relates to a method for electrical sound generation and an interactive music player, in which the starting material is a predefined time duration lasting, present in digital format audio signal.

The profession of Disk Jockeys (short: DJ) experiences in today's, characterized by modern electronic music dance culture an enormous technical appreciation. The crafting of this profession involves arranging the music titles into a complete work (the set, the mix) with its own suspense.

In the vinyl DJ field, the technique of scratching has established itself far-reaching. It is a technique in which the sound material on the record is used for the rhythmic generation of sound by combined movement of the record with the hand and one of the volume controls on the mixer (so-called fader). Great masters of scratching do this on two or even three turntables simultaneously, which requires the dexterity of a good drummer or piano player.

Increasingly, hardware manufacturers with effect mixers also come into the real-time effects sector. There are also DJ mixing consoles that have sample units with which parts of the audio signal can be reused as a loop or as a one-shot sample. There are also CD players that allow scratching on a CD using a large jog wheel.

However, no device or method is known with which both the playback position of a digital audio signal, as well as the volume curve or other sound parameters of this signal can be automatically controlled so that while a rhythmic, tact-precise "scratch effect" from the currently listening audio Material is generated. However, this would be desirable, because on the one hand, successful scratch effects would be reproducible and additionally transferable to other audio material. On the other hand, a DJ could be so relieved and focus more on other artistic issues such as the composition of the pieces of music.

From the US 5512704 For example, a method for reproducing a digital piece of music is known, which based on scratch control data, the rate of speed and the direction of reproduction of the digital piece of music is determined.

It is therefore an object of the present invention to provide a method and a music player which enable automatic generation of musical scratch effects.

This object is achieved in each case by the independent claims.

Further advantageous embodiments are in the dependent. Claims specified.

FIG 1

ein Zeit-Raum-Diagramm aller sich miteinander im Takt befind- lichen Abspiel-Varianten eines mit Normalgeschwindigkeit wie- dergegebenen Tracks als parallele Geraden der Steigung 1,

FIG 2

ein Detail-Ausschnitt des Zeit-Raum-Diagramms nach FIG 1 zur Beschreibung der geometrischen Zusammenhänge eines Full-Stop- Scratch-Effekts,

FIG 3

einen Ausschnitt eines Zeit-Raum-Diagramms zur Beschreibung der geometrischen Zusammenhänge eines Back-and-For-Scratch- Effekts,

FIG 4

verschiedene mögliche Lautstärke-Hüllkurven zur Realisierung eines Gater-Effektes auf einen Back-and-For Scratch-Effekt,

FIG 5

ein Blockschaltbild eines interaktiven Musik-Abspielers gemäß der Erfindung mit Eingriffsmöglichkeit in eine aktuelle Ab- spielposition,

FIG 6

ein Blockschaltbild einer zusätzlichen Signalverarbeitungsket- te zur Realisierung eines Scratch-Audio-Filters gemäß der Er- findung,

FIG 7

ein Blockschaltbild zur Veranschaulichung der Gewinnung rhyth- musrelevanter Informationen und deren Auswertung zur nähe- rungsweisen Ermittlung von Tempo und Phase eines Musikdaten- stroms,

FIG 8

ein weiteres Blockschaltbild zur sukzessiven Korrektur von er- mitteltem Tempo und Phase und

FIG 9

einen Datenträger, der Audiodaten und Steuerdateien zur Repro- duktion von aus den Audiodaten gemäß der Erfindung erstellten Scratch-Effekten oder Gesamtwerken vereint.

Advantages and details of the invention will become apparent from the following description of advantageous embodiments and in conjunction with the figures. It shows in schematic representation:

FIG. 1

a time-space diagram of all play variants of a normal-speed reproduced track in parallel with each other as a parallel straight line of the slope 1,

FIG. 2

a detail section of the time-space diagram FIG. 1 to describe the geometric relationships of a full-stop-scratch effect,

FIG. 3

a section of a time-space diagram describing the geometric relationships of a back-and-for-scratch effect,

FIG. 4

various possible volume envelopes for realizing a Gater effect on a Back-and-For Scratch effect,

FIG. 5

1 is a block diagram of an interactive music player according to the invention with the possibility of engaging in a current play position;

FIG. 6

2 is a block diagram of an additional signal processing chain for realizing a scratch audio filter according to the invention;

FIG. 7

a block diagram to illustrate the acquisition of rhythm-relevant information and its evaluation for the approximate determination of tempo and phase of a music data stream,

FIG. 8

a further block diagram for the successive correction of the determined tempo and phase and

FIG. 9

a data carrier which combines audio data and control files for the reproduction of scratch effects or complete works created from the audio data according to the invention.

For playing pre-recorded music, various types of devices are conventionally used for various storage media such as vinyl, compact disk or cassette. These formats were not designed to intervene in the playback process to work with the music in a creative way. However, this possibility is desirable, and is practiced today despite the limitations imposed by the aforementioned DJ's. Vinyl records are preferred because they are the easiest way to influence the playback speed and position with your hand.

Today, however, digital formats such as Audio CD and MP3 are mostly used to store music. MP3 is a compression method for digital audio data according to the MPEG standard (MPEG 1 Layer 3). The method is asymmetric, i. the coding is much more complicated than the decoding. It is also a lossy process. The present invention now makes possible the aforementioned creative handling of music on any digital formats by a suitable interactive music player who makes use of the new possibilities created by the inventive measures described above.

There is a fundamental need to have as much helpful information as possible in the graphical representation in order to intervene in a targeted manner. In addition, one would like to be able to intervene ergonomically in the playback process, in a manner similar to DJ's often practiced "scratching" on vinyl record players, with the turntable paused during playback and moved forwards and backwards.

In order to be able to intervene, it is important to have a graphical representation of the music, in which one recognizes the current play position and also recognizes a certain period in the future and in the past. This is usually done by plotting the amplitude envelope of the sound waveform over a period of several seconds before and after the playhead position. The rendering shifts in real time at the rate at which the music plays.

In principle, one would like to have as much helpful information as possible in the graphical representation in order to intervene in a targeted manner. In addition, one would like to be able to intervene as ergonomically as possible in the playback process, in a comparable way to the so-called "scratching" on vinyl record players. The term "scratching" refers to the stopping and moving the turntable forwards or backwards during playback.

In the interactive music player created by the invention, musically relevant times, in particular the beats, can now be used with the later ( 7 and FIG. 8 ) extracted from the audio signal and displayed as markers in the graphical representation, for example on a display or on a screen of a digital computer on which the music player is realized by an appropriate programming.

1. a) Musik läuft frei, mit konstantem Tempo,
2. b) Abspielposition und -geschwindigkeit wird vom Anwender direkt oder automatisch beeinflusst.

Furthermore, a hardware control element R1 is provided, eg a button, in particular the mouse button, with which one switches over between two operating modes:

1. a) music is free, with a constant tempo,
2. b) Playback position and speed are directly or automatically influenced by the user.

The mode a) corresponds to a vinyl record that you do not touch and whose speed is equal to the turntable. The mode b), however, corresponds to a vinyl record, which one holds by hand and pushes back and forth.

In an advantageous embodiment of an interactive music player, the playback speed in mode a) is further influenced by the automatic control for synchronizing the clock of the music being played to another clock (cf. 7 and FIG. 8 ). The other clock may be synthetically generated or given by another concurrently playing music.

In addition, a further hardware control element R2 is provided with which, in operating mode b), one determines the disk position as it were. This can be a continuous controller or the computer mouse.

The representation after FIG. 5 shows a block diagram of such an arrangement with the explained below signal processing means, with which an interactive music player according to the invention is created with possibility of intervention in a current play position.

The position data given with this further control R2 usually has a limited temporal resolution, i. A message will be sent only at regular or irregular intervals to convey the current position. However, the play position of the stored audio signal should change smoothly, with a temporal resolution corresponding to the audio sample rate. Therefore, at this point, the invention uses a smoothing function which generates a high-resolution, uniformly changing signal from the step signal given by the control element R2.

• vorgegebenes Stufensignal -> Rampenglättung -> Tiefpassfilter -> exakte Abspielposition

oder
vorgegebenes Stufensignal -> Tiefpassfilter -> Rampenglättung -> exakte AbspielpositionOne method for doing this is to trigger a ramp with a constant slope with each given position message, which moves the smoothed signal from its old value to the value of the position message in a predetermined time. Another possibility is to send the stepped waveform into a linear digital low-pass filter LP whose output represents the desired smoothed signal. For this purpose, a 2-pole resonance filter is particularly suitable. A combination (series connection) of the two smoothings is also possible and advantageous and allows the following advantageous signal processing chain:

• preset step signal -> ramp smoothing -> lowpass filter -> exact playback position

or
preset step signal -> low pass filter -> ramp smoothing -> exact playback position

The block diagram after FIG. 5 illustrates the in an advantageous embodiment in the form of a schematic diagram. The control element R1 (here a button) serves to change the operating modes a) and b) by triggering a switch SW1. The controller R2 (here a continuous slider) supplies the position information with temporally limited resolution. This serves as a low-pass filter LP for smoothing as an input signal. The smoothed position signal is now differentiated (DIFF) and provides the playback speed. The switch SW1 is driven with this signal at a first input IN1 (mode b). The other input is IN2 with a tempo value A, as in 7 and FIG. 8 described can be determined, applied (mode a). The control element R1 is used to change between the input signals.

In addition, via a third control element (not shown), the control information described above for the automatic manipulation of playback position and / or playback direction and / or playback speed can be specified. Another control is then used to trigger the predetermined with the third control automatic manipulation of the playback position and / or playback direction and / or playback speed.

When changing from one to the other mode (equivalent to holding and releasing the turntable), the position must not jump. For this reason, the proposed interactive music player assumes the position reached in the previous mode as the home position in the new mode. Likewise, the playback speed (1st derivative of the position) should not change rapidly. Therefore, you also take the current speed and guide you through a smoothing function, as described above, to the speed that corresponds to the new mode. To FIG. 5 This is done by a Slew Limiter SL, which triggers a ramp with a constant slope, which moves the signal from its old value to the new value in a given time. This position or speed-dependent signal then controls the actual PLAY playback unit for playback of the audio track by influencing the playback speed.

The complicated movements, in which the record and the crossfader have to work together in a very precise, tempo - adapted way, are now thanks to the FIG. 5 shown arrangement with the corresponding controls and a later described in more detail meta-file format automated. Through a series of presets the length and type of scratches can be selected. The actual sequence of the scratches is controlled in a tempo-accurate manner by the method according to the invention. The movements are either recorded beforehand in a real scratch or they are designed in a graphic editor "on the drawing board".

The automated scratch module now uses the above based on FIG. 5 described so-called scratch algorithm.

The method described above requires only one parameter, namely the position of the hand with which the virtual record moves becomes (see appropriate control), and calculates therefrom by means of two smoothing procedures the current play position in the audio sample. The use of these smoothing methods is not of theoretical necessity but of technical. Without its use, it would be necessary for the unaltered playback to perform the calculation of the current playhead position in the audio rate (44kHz), which would require a significant increase in computational power. Thanks to the algorithm, the playback position can be calculated at a much lower rate (eg 344 Hz).

In the following, it will be explained with reference to the two simplest scratch automations how the inventive method for automatically generating scratch effects works. The same procedure can also be applied to much more complex scratch sequences.

FULL STOP

This scratch is an effect that stops the record (either by hand or by turning on the record player's Stop button). After a certain time, the record will be released, or the engine will be switched on again. After the record has returned to its original rotational speed, it must again be in time with the "well-thought-out" measure before the scratch or again in time with a second, untouched during the full-stop, reference clock.

• sowohl Abbremsen als auch Beschleunigen erfolgen linear, d.h. mit konstanter Beschleunigung.
• Abbremsen und Beschleunigen erfolgen mit derselben Beschleunigung jedoch mit umgekehrtem Vorzeichen.

The following simplifying assumptions were made to calculate the deceleration, standstill and acceleration phases. (However, it is also possible to calculate more complex traces of the scratch without any effort):

• Both deceleration and acceleration take place linearly, ie with constant acceleration.
• Deceleration and acceleration take place with the same acceleration but with the opposite sign.

The representation according to FIG. 1 shows a time-space diagram of all mutually synchronous or in time with each other play variants of a normal speed reproduced track. The duration of a quarter note of a current track is called beat.

If you set all the beat variants of a normal speed track to each other as parallel Straight line of slope 1 in a time-space diagram (X-Axisc: time t in [ms], Y-axis sample position SAMPLE in [ms]), then a FULL STOP Scratch as connection curve (dashed line) between two of the parallel play lines are displayed. The linear velocity transition between the motion phases and the standstill phase of the scratch is represented in the time-space diagram as a parabola segment (linear velocity change = quadratic position change).

Some geometric considerations based on the in FIG. 1 The representation shown now makes it possible to calculate the duration of the various phases (deceleration, standstill, acceleration) in such a way that, after the scratches have been completed, the playback position comes to lie on a straight line parallel to the original straight line and displaced by a whole multiple of a quarter note (beat). which is the graphic equivalent to the above demand for a timely resumption of movement. This shows the FIG. 2 a section of FIG. 1 , in which the following mathematical considerations can be understood.

Let the duration of the deceleration and acceleration process 'ab', v the velocity, x the playback position correlated with the time t and the duration of a quarter note of the current track beat, then calculate the duration of the standstill phase c to be met as follows: $$\mathrm{c}\mathrm{=}\mathrm{beat}\mathrm{-}\mathrm{from}\mathrm{,}$$

und besteht also aus 3 Phasen: Abbremsen von v=1 auf v=0: Dauer: ab Stillstand: Dauer: beat-ab Beschleunigen von v=0 auf v=1: Dauer: ab (für ab<= beat)The total duration T of the scratch is $$\mathrm{T}\mathrm{=}\mathrm{beat}\mathrm{+}\mathrm{from}$$

and thus consists of 3 phases: Decelerating from v = 1 to v = 0: Duration: from standstill: Duration: beat-off Speed up from v = 0 to v = 1: Duration: from (for ab <= beat)

From this it follows that at first normal speed v = 1 is used, before a linear deceleration f (x) = - ½ x ² takes place, which takes the time 'off'. For the duration 'beat-down', standstill v = 0, before a linear acceleration takes place f (x) = ½ x ² , which in turn takes the time 'off'. Thereafter, the procedure is repeated at normal speed V = 1.

The duration 'from' for the deceleration and the acceleration was deliberately kept variable, because by changing this parameter you can intervene decisively in the "sound" (the quality) of the scratches (see default settings).

If the standstill phase c is extended by a multiple of beat, you can generate clock-synchronous full-stop scratches of any length.

BACK AND FOR

The purpose of this scratch is to move the virtual record forward and backward in one tempo-sync mode and to be back in time with the original or reference clock when the scratches are finished. You can look back at the same time-space diagram FIG. 1 use and scratch this in its simplest form
Speed = +/- 1; Frequency = 1 / beat,
as shown in the illustration FIG. 3 represent that on FIG. 2 is ajar. Of course, much more complex motion sequences can be calculated in this way.

The deceleration from v = + 1 to v = -1 and vice versa now requires twice the duration = 2 * ab. With geometrical considerations, the duration of the reverse run phase "rü" and the subsequent forward run phase "vo" can be determined as from FIG. 3 be traceable determined: $$\mathrm{ru}=\mathrm{vo}=1/*\mathrm{beat}-2\mathrm{from}$$

The total duration of the scratches this time is exactly T = beat and consists of 4 phases: Decelerating from v = 1 to v = -1: Duration: 2ab Reverse run: Duration: 1/2 * beat - 2ab Speed up from v = -1 to v = 1: Duration: 2ab Forward: Duration: 1/2 * beat - 2ab

This scratch can be repeated any number of times and returns again and again to the start play position that moves the virtual record on the whole, not further. With every iteration, this means a shift by p = -beat compared to the reference clock.

Also in this scratch, the duration of the deceleration and acceleration process remains "down" variable, since the change of a, the characteristics of the scratch can be greatly changed.

GATER

In addition to the actual manipulation of the original playback speed, a scratch receives its diversity through additional rhythmic highlighting of certain passages of the motion sequence by means of volume or EQ / Filter (sound characteristic) manipulations. For example, in a BACK AND FOR Scratch only the reverse phase can be made audible and the forward phase can be hidden.

This process has also been automated in the present process by the tempo information extracted from the audio material (cf. 7 and FIG. 8 ) is used to rhythmically control these parameters.

• RATE (Frequenz des Gate-Vorgangs),
• SHAPE (Verhältnis von "An"- zu "Aus"-Phase) und
• OFFSET (Phasen-Verschiebung, relativ zum Referenztakt)

eine große Vielfalt an Effekt-Variationen möglich ist. Diese 3 Parameter können anstatt nur auf die Lautstärke des Scratches zu wirken, natürlich auch auf EQs/Filter oder jeden anderen Audio-Effekt, wie Hall, Delay und ähnliches angewendet werden.Here, too, will be illustrated by way of example only, as based on three parameters

• RATE (frequency of gate operation),
• SHAPE (ratio of "on" to "off" phase) and
• OFFSET (phase shift, relative to the reference clock)

a wide variety of effect variations is possible. These 3 parameters can be applied to EQs / filters or any other audio effect such as reverb, delay and the like, instead of just affecting the volume of the scratches.

The Gater himself already exits in many effect devices. However, the combination with a tempo-synchronous Scratch algorithm for the generation of fully automatic Scratch processes, to which necessarily also volume gradients belong, is used for the first time in this procedure.

In FIG. 4 is a simple 3-fold BACK AND FOR Scratch
shown. Including different volume envelopes, which result from the adjacent gate parameters. Also shown is the resulting rendering curve to illustrate how different the end result can be through the application of different gate parameters. If the BACK AND FOR Scratch is now varied in its frequency and the acceleration parameter "down" (not shown in the drawing), there are an extremely large number of possible combinations.

• RATE = 1/4
• SHAPE = 0
• OFFSET = 0

The first trace below the original shape (3-fold BACK AND FOR Scratch) emphasizes only the second half of the playback movement, while eliminating the first half of each. The Gater values for this history are:

• RATE = 1/4
• SHAPE = 0
• OFFSET = 0

The course of the volume envelope is in each case continuously drawn, while the thus selected areas of the playback movement are each shown in dashed lines.

• RATE = 1/4
• SHAPE = - 1/2
• OFFSET = 0,4

In the course below, only the backward movements of the playback movement are selected with the Gater parameters:

• RATE = 1/4
• SHAPE = - 1/2
• OFFSET = 0.4

• RATE = 1/8
• SHAPE = - 1/2
• OFFSET = 0,2

The underlying course is another variant in which the upper and lower reversal points of the playback movement are respectively selected by:

• RATE = 1/8
• SHAPE = - 1/2
• OFFSET = 0.2

In another operating mode of the scratch automatism, it is conceivable to also optimize the selection of the audio sample with which the scratch is performed and thus make it user-independent. In this mode, pressing the button would start the procedure, but this only. completed, if in the audio material a suitable Bcat event is found, which is particularly suitable for the execution of the selected Scratches.

"SCRATCH SYNTHESIZER"

Everything described so far treats the procedure with which any section of an audio material is modified

Can be played back (in the case of rhythmic material synonymous tempo-sync) - Since now, however, the result (the sound) of a Scratches directly related to the selected audio material, the resulting sound diversity is in principle as large as the audio material used itself. Since the process is parameterized, it can even be described as a new sound-synthesis process.

When "scratching" vinyl records, which means playing at a fast and rapidly changing speed, the tone waveform changes in a characteristic way, due to the peculiarities of the recording method used by default for records. When creating the press master for the record in the recording studio, the sound signal goes through a RIAA standard pre-emphasis (pre-emphasis) filter that raises the treble (so-called "cutting line"). In each system used to play records, there is a corresponding de-emphasis filter (back-equalizing filter) that reverses the effect to approximate the original signal.

But if the playback speed is no longer the same as when recording, what u.a. occurs during "scratching", all frequency components of the signal are shifted accordingly on the record and therefore damped differently by the de-emphasis filter. This results in a characteristic sound.

In order to achieve the most authentic reproduction possible when playing with fast and rapidly changing speed, similar to "scratching" with a vinyl turntable, another advantageous embodiment of the interactive music player according to the invention uses a scratch audio filter for an audio signal. wherein the audio signal is subjected to pre-emphasis filtering (pre-emphasis) and stored in a buffer from which it can be read at variable tempo depending on the respective playing speed, then subjected to de-emphasis filtering and reproduced become.

In this advantageous embodiment of the inventive interactive music player according to the invention with a structure accordingly FIG. 5 Therefore, a scratch audio filter is provided for simulating the described characteristic effect. For this purpose, in particular for a digital simulation of this process, the audio signal within the playback unit PLAY FIG. 5 subjected to another signal processing, like this in FIG. 6 is shown. For this purpose, after the digital audio data of the music piece to be played has been read from a medium D or sound carrier (eg CD or MP3) and DEC decoded (especially in the case of the MP3 format), the audio signal is subjected to a corresponding pre-emphasis filtering PEF , The signal prefiltered in this way is then stored in a buffer memory B, from which it is stored in a further processing unit R depending on the operating mode a) or b), as in FIG. 5 is read out according to the output signal of SL with varying speed. The read out signal is then treated with a de-emphasis filter DEF and then reproduced (AUDIO_OUT).

For the pre- and de-emphasis filters PEF and DEF, which should have the same frequency response as defined in the RIAA standard, it is convenient to use a second-order digital IIR filter, i. with two favorably chosen poles and two favorably chosen zeros. If the poles of one filter are equal to the zeroes of the other filter, the two filters cancel each other out exactly as desired when the audio signal is played back at the original speed. In all other cases, the filters mentioned generate the characteristic sound effect during "scratching". Of course, the described scratch audio filter can also be used in conjunction with any other type of music players with "scratching" function.

The information from the audio material requires the tempo of the track to determine the size of the "beat" variable as well as the "timing" of the gate. For example, the following is used for the audio track tempo determination method described below.

In this context, the technical problem of tempo and phase alignment of two pieces of music or audio tracks in real time. It would be desirable if a possibility for automatic tempo and phase equalization of two pieces of music or audio tracks in Echtzoit would be available to free the DJ of this technical aspect of mixing, or a mix automatically or semi-automatically, without the help of a senior Create DJ's.

So far, this problem has been solved only in part aspects. There are MP3 format software players (a standard format for compressed digital audio data) that realize pure real-time tempo recognition and adaptation. The detection of the phase must, however, continue to be done manually by hearing and adjusting the DJ. This requires a considerable amount of DJ attention, which would otherwise be available for artistic aspects such as music composition, etc.

An object of the present invention is thus to provide a possibility for automatic tempo and phase equalization of two pieces of music or audio tracks in real time with the highest possible accuracy.

An essential technical hurdle to overcome is the accuracy of a tempo and phase measurement, which decreases with the time available for this measurement. The problem is thus primarily for a determination of the tempo and the phase in real time, as it u.a. when live mixing is the case.

In the following, a possible realization of the approximate tempo and phase detection as well as tempo and phase adaptation according to the invention will be illustrated.

The first step of the procedure is a first, approximate determination of the tempo of the piece of music. This is done by a statistical evaluation of the time intervals of the so-called beat events. One possibility for obtaining rhythm-relevant events from the audio material is by narrow bandpass filtering of the audio signal in different frequency ranges. In order to determine the tempo in real time, only the beat events of the last seconds are used for the following calculations. There are 8 to 16 events in about 4 to 8 seconds.

Due to the quantized structure of music (16th note scale) not only quarter note beat intervals can be used to calculate the tempo. Also other intervals (16th, 8th, ½ and whole notes) can be transformed into a predefined frequency octave (eg 80 - 160 bpm, English for beats per minute) by octavation (eg by multiplying their frequency by powers of 2) and thus temporally relevant Provide information. Faulty octaves (eg of triplet intervals) are of late importance because of their relative rarity in the statistical evaluation. In order to also capture triplets or ruffled rhythms (single notes offset slightly from the 16th scale), the time intervals obtained in the first point are additionally grouped in pairs and triplets by adding their time values before being octaved. By this method, the rhythmic structure between the bars is calculated out of the time intervals.

The amount of data thus obtained is examined for accumulation points. As a rule, three accumulation maxima arise as a result of the octavation and grouping methods whose value is in a rational relationship (2/3, 5/4, 4/5 or 3/2) to one another. Should it not be clear enough from the strength of one of the maxima that this indicates the actual tempo of the piece of music, the correct maximum can be determined from the rational ratios of the maxima among themselves.

For approximate determination of the phase, a reference oscillator is used. This vibrates at the previously determined pace. Its phase is advantageously chosen to give the best match between beat events of the audio and zero crossings of the oscillator.

This is followed by a successive improvement of the tempo and phase determination. Due to the natural inadequacy of the first approximate tempo determination, after a few seconds the phase of the reference oscillator will shift relative to the audio track. This systematic phase shift provides information about the amount by which the tempo of the reference oscillator has to be changed. A correction of the tempo and the phase is advantageously carried out at regular intervals to remain below the audibility limit of the shifts and the correction movements.

All phase corrections made from the approximate phase correlation are accumulated over time so that the calculation of the tempo and phase is based on an ever-increasing time interval. As a result, the tempo and phase values become increasingly more precise and lose the initially mentioned flaw of the approximate real-time measurement. After a short time (about 1 min), the error of the tempo value determined with this method drops below 0.1%, a measure of accuracy which is a prerequisite for the calculation of loop lengths.

The representation according to FIG. 7 shows a possible technical realization of the described approximate tempo and Phsenerkennung a music data stream in real time using a block diagram. The structure shown can also be referred to as a 'Beat Detector'.

As input are two streams of audio events and audio events E _i with value 1 before meeting the peaks in the frequency bands F1 and F2 at 150 Hz or 9000 Hz corresponding to at 4000Hz. These two event streams are initially treated separately by filtering them through respective bandpass filters having respective cutoff frequencies F1 and F2.

If an event follows the previous one within 50 ms, the second event will not be considered. A time of 50 ms corresponds to the duration of a 16-bit at 300 bpm, which is well below the duration of the shortest interval in which the music pieces are usually located.

From the stream of filtered events E _i , a stream of the simple time intervals T _i between the events is formed in respective processing units BD1 and BD2.

From the stream of simple time intervals T _1i , two further streams of the band-limited time intervals are additionally formed in identical processing units BPM_C1 and BPM_C2, namely with time intervals T _2i , the sums of two successive time intervals, and with time intervals T _3i , the sums of three in each case successive time intervals. The events used may also overlap.

• T_2i: (t₁+t₂), (t₂+t₃), (t₃+t₄), (t₄+t₅), (t₅+t₆),... und
• T_3i: (t₁+t₂+t₃), (t₂+t₃+t₄), (t₃+t₄+t₅), (t₄+t₅,+t₆),...

Thereby, from the stream: t _1, t _2, t _2, t _4, t _5, t _6, ... in addition generates the following two streams:

• T _2i : (t ₁ + t ₂ ), (t ₂ + t ₃ ), (t ₃ + t ₄ ), (t ₄ + t ₅ ), (t ₅ + t ₆ ), ... and
• T _3i : (t ₁ + t ₂ + t ₃ ), (t ₂ + t ₃ + t ₄ ), (t ₃ + t ₄ + t ₅ ), (t ₄ + t ₅ , + t ₆ ), .. ,

The three streams T _1i , T _2i , T _3i are now time-octaved in corresponding processing units OKT. The time octave OKT is performed such that the individual time intervals of each current are doubled until they lie in a predetermined interval BPM_REF. In this way one obtains three data streams T _1io , T _2io , T _3io , ... The upper limit of the interval is calculated from the lower bpm limit according to the formula: $${\mathrm{t}}_{\mathrm{Hi}\left[\mathrm{m}\right]}\mathrm{=}\mathrm{60000}/{\mathrm{bpm}}_{\mathrm{low}}\mathrm{,}$$

The lower limit of the interval is 0.5 * t _hi . Each of the three streams thus obtained is now checked for consistency for both frequency bands F1, F2 in respective further processing units CHK. This is used to determine whether a certain number of consecutive, time-octaved interval values are within a predetermined error limit. For example, the following values are checked in detail:

1. a) (t_11o - t_12o)² + ( t_11o - t_13o)² + ( t_11o - t_14o)² < 20
   Ist dies der Fall, wird der Wert t_11o als gültiges Zeitintervall ausgegeben.
   Für T_2i überprüft man dessen letzte 4 Events t_21o, t_22o, t_23o, t_24o daraufhin, ob gilt:
2. b) (t_21o - t_22o)² + (t_21o - t_23o)² + (t_21o - t_24o)² < 20
   Ist dies der Fall, wird der Wert t_11o als gültiges Zeitintervall ausgegeben.
   Für T3i überprüft man dessen letzte 3 Events t_31o, t_32o, t_33o, daraufhin, ob gilt:
3. c) (t_31o - t_32o)² + ( t_31o - t_33o)² < 20
   Ist dies der Fall, wird der Wert t₃₁₀ als gültiges Zeitintervall ausgegeben.

For T _1i check its last 4 events t _11o , t _12o , t _13o , t _14o then whether

1. a) (t _11o - t _12o ) ² + (t _11o - t _13o ) ² + (t _11o - t _14o ) ² <20
   If this is the case, the value t _{11o is output} as a valid time interval.
   For T _2i, check its last 4 events t _21o , t _22o , t _23o , t _24o to _see if:
2. b) (t _21o - t _22o ) ² + (t _21o - t _23o ) ² + (t _21o - t _24o ) ² <20
   If this is the case, the value t _{11o is output} as a valid time interval.
   For T3i check its last 3 events t _31o , t _32o , t _33o , then, if:
3. c) (t _31o - t _32o ) ² + (t _31o - t _33o ) ² <20
   If this is the case, the value t _{310 is} output as a valid time interval.

In this case, the consistency check a) takes precedence over b) and b) takes precedence over c). If a value is output at a), b) and c) are no longer examined. If no value is output at a), b) is examined, etc. If, however, neither a) nor b) nor c) finds a consistent value, the sum of the last 4 non-octave individual intervals (t ₁ + t ₂ + t ₃ + t ₄ ).

The value stream of consistent time intervals determined in this way from the three streams is in turn octaved in a downstream processing unit OKT into the predetermined time interval BPM_REF. Subsequently, the octaved time interval is converted into a BPM value.

As a result, there are now two streams BPM1 and BPM2 of bpm values - one for each of the two frequency ranges F1 and F2. In a prototype these currents are interrogated with a fixed frequency of 5 Hz and the last eight events from both streams are used for the statistical evaluation. However, you can also use a variable (event-driven) sampling rate at this point, and you can also use more than just the last 8 events, such as 16 or 32 events.

These last 8, 16 or 32 events from each frequency band F1, F2 are merged and considered in a downstream processing unit STAT on accumulation maxima N. In the prototype version, an error interval of 1.5 bpm is used, i. as events differ less than 1. 5 bpm from each other, they are considered to belong together and add up in weighting. The processing unit STAT determines here at which BPM values accumulations occur and how many events are to be assigned to the respective accumulation points. The most heavily weighted cluster point can be considered the local BPM measurement and provides the desired tempo value A.

In a first development of this method, in addition to the local BPM measurement, a global measurement is performed by expanding the number of events used to 64, 128, etc. With alternating rhythm patterns, in which the tempo comes through only every fourth bar, an event number of at least 128 can often be necessary. Such a measurement is more reliable, but also takes more time.

Another crucial improvement can be achieved by:

Not only the first accumulation maximum is taken into consideration, but also the second accumulation maximum. This second maximum is almost always due to triplets present and may even be stronger than the first maximum. However, the tempo of the triplets has a clearly defined relationship to the tempo of the quarter notes, so that it can be determined from the ratio of the tempi of the first two maxima, which accumulation maximum is assigned to the quarters and which to the triplets.

• wenn T2 = 2/3 * T1, dann ist T2 das Tempo.
• Wenn T2 = 4/3 * T1, dann ist T2 das Tempo.
• Wenn T2 = 2/5 * T1, dann ist T2 das Tempo.
• Wenn T2 = 4/5 * T1, dann ist T2 das Tempo.
• Wenn T2 = 3/2 * T1, dann ist T1 das Tempo.
• Wenn T2 = 3/4 * T1, dann ist T1 das Tempo.
• Wenn T2 = 5/2 * T1, dann ist T1 das Tempo.
• Wenn T2 = 5/4 * T1, dann ist T1 das Tempo.

Assuming T1 as the tempo of the first maximum in bpm and T2 as that of the second maximum, the following rules apply:

• if T2 = 2/3 * T1 then T2 is the tempo.
• If T2 = 4/3 * T1 then T2 is the tempo.
• If T2 = 2/5 * T1, then T2 is the tempo.
• If T2 = 4/5 * T1 then T2 is the tempo.
• If T2 = 3/2 * T1 then T1 is the tempo.
• If T2 = 3/4 * T1 then T1 is the tempo.
• If T2 = 5/2 * T1 then T1 is the tempo.
• If T2 = 5/4 * T1 then T1 is the tempo.

An approximate phase value P is determined from one of the two filtered simple time intervals T _i between the events, preferably from those values filtered at the lower frequency F1. These are used to roughly determine the frequency of the reference oscillator.

The representation after FIG. 8 shows a possible block diagram for the successive correction of the detected tempo A and phase P, hereinafter referred to as 'CLOCK CONTROL'.

First, the reference oscillator or the reference clock MCLK is allowed in a first step 1 with the coarse phase values P and tempo values A from the beat detection, which is virtually a reset of the in FIG. 2 equaled control loop. Subsequently, in a further step 2, the time intervals between beat events of the incoming audio signal and the reference clock MCLK are determined. For this purpose, the approximate phase values P are compared with a reference signal CLICK, which has the frequency of the reference oscillator MCLK, in a comparator V.

oder $$\mathrm{A}\left(\mathrm{i}\mathrm{+}\mathrm{1}\right)\mathrm{=}\mathrm{A}\left(\mathrm{i}\right)-\mathrm{q}$$

entgegen der Abweichung (wieder) an das Audio-Signal angepasst, wobei q die verwendete Absenkung oder Anhebung des Tempos darstellt. Andernfalls (-) wird das Tempo konstant gehalten.In the case of systematic exceeding (+) of a "critical" deviation in the case of several consecutive events having a value of, for example, more than 30 ms, in a further processing step 3 the reference clock MCLK is changed by a brief change in tempo $$\mathrm{A}\left(\mathrm{i}\mathrm{+}\mathrm{1}\right)\mathrm{=}\mathrm{A}\left(\mathrm{i}\right)\mathrm{+}\mathrm{q}$$

or $$\mathrm{A}\left(\mathrm{i}\mathrm{+}\mathrm{1}\right)\mathrm{=}\mathrm{A}\left(\mathrm{i}\right)-\mathrm{q}$$

against the deviation (again) adapted to the audio signal, where q represents the used lowering or raising the tempo. Otherwise (-) the tempo is kept constant.

In the further course, in a further step 4, a summation of all correction events from step 3 and the time elapsed since the last "reset" takes place in separate memories (not shown). At approximately every 5th to 10th event of approximately accurate sync (difference between the audio data and the reference clock MCLK below approximately 5 ms), the tempo value will be based on the previous tempo value of the accumulated correction events and which has been recalculated since the elapsed time in a further step 5 as follows.

• q als der in Schritt 3 verwendeten Absenkung oder Anhebung des Tempos (beispielsweise um den Wert 0.1),
• dt als der Summe der Zeit, für welche das Tempo insgesamt abgesenkt oder angehoben wurde (Anhebung positiv, Absenkung negativ),
• T als dem seit dem letzten Reset (Schritt 1) verstrichenen Zeitintervall, und
• bpm als dem in Schritt 1 verwendeten Tempowert A

errechnet sich das neue, verbesserte Tempo nach folgender einfachen Formel: $$\mathrm{bpm\_neu}\mathrm{=}\mathrm{bpm}\mathrm{*}\left(\mathrm{1}\mathrm{+}\left(\mathrm{q}\mathrm{*}\mathrm{dt}\right)\mathrm{/}\mathrm{T}\right)$$

With

• q as the decrease or increase in the tempo used in step 3 (for example by the value 0.1),
• dt as the sum of the time for which the tempo has been lowered or raised altogether (increase positive, decrease negative),
• T as the time interval since the last reset (step 1), and
• bpm as the tempo value A used in step 1

The new, improved pace is calculated according to the following simple formula: $$\mathrm{bpm\_neu}\mathrm{=}\mathrm{bpm}\mathrm{*}\left(\mathrm{1}\mathrm{+}\left(\mathrm{q}\mathrm{*}\mathrm{dt}\right)\mathrm{/}\mathrm{T}\right)$$

It is also checked whether the corrections in step 3 are always negative or positive over a certain period of time. In such a case, there is probably a tempo change in the audio that can not be corrected using the above procedure. This status is detected, and upon reaching the next near perfect synchronization event (step 5), the time and correction memories are cleared in step 6 to reset the starting point in phase and tempo. After this "reset", the procedure starts again with a touchdown on step 2 to optimize the tempo.

A synchronization of a second piece of music is now done by adjusting its tempo and phase. The adaptation of the second piece of music is done indirectly via the reference oscillator. After the above Approximated tempo and phase detection of the piece of music, these values are successively adapted to the reference oscillator according to the above method, but this time the playback phase and the playback speed of the track itself is changed. The original tempo of the track can be easily calculated backwards from the necessary change in its playback speed compared to the original playback speed.

Furthermore, the information obtained about the tempo and the phase of an audio track allows the control of so-called tempo-synchronous effects. The audio signal is manipulated according to your own rhythm, which allows rhythmically effective real-time sound change. In particular, the tempo information can be used to cut out loops with exact timed lengths in real time from the audio material.

As already mentioned, conventionally, when mixing several pieces of music, the audio sources of sound carriers are played on several playback devices and mixed via a mixing console. In this procedure, an audio recording is limited to a recording of the final result. A reproduction of the mixing process or scratch operations and a touchdown at a later time exactly at a predeterminable position within a piece of music is thus not possible.

This is precisely what the present invention achieves by proposing a file format for digital control information that provides the ability to record and accurately reproduce the process of interactive mixing and eventual effects processing of audio sources. This is possible in particular with a music player as described above.

The recording of mixing operations or a scratch process is divided into a description of the audio sources used and a timing of control information of the mixing process or scratch process and additional effect processing.

Only the information about the actual mixing process or scratch process and about the source audio sources is needed to render the result. The actual digital audio data is provided externally. This avoids copyrights of copyrighted music pieces that are problematic in terms of copyright. Thus, by storing digital control information, mixing operations of plural pieces of audio with regard to playback positions, synchronization information, real-time interventions with audio signal processing means, etc., can be used as a mix the audio sources and their effects processing eg with scratch effects as a new complete work with a relatively long playing time can be realized.

This has the advantage that the description of the processing of the audio sources compared to the generated audio data of the mixing process are low, the mixing process can be edited anywhere and re-set. In addition, existing audio tracks can be rendered in different summaries or as longer coherent interpretations.

With previous sound carriers and music players, however, it was not possible to record and reproduce the interaction of a user, since the known players lack the technical prerequisites to control them accurately enough. This is made possible only by the present invention by reproducing a plurality of digital audio sources and determining and controlling their playback positions. This makes it possible to process the entire process digitally and store corresponding control data in a file. This digital control information is preferably stored in a resolution that corresponds to the sampling rate of the processed digital audio data.

• eine Liste der verwendeten Audioquellen z.B. digitale Aufgezeichnete Audiodaten in komprimierter und unkomprimierter Form wie z.B. WAV, MPEG, AIFF und digitale Tonträger wie etwa eine Compact Disk und
• den zeitlichen Ablauf der Steuerinformation.

The recording is essentially divided into 2 parts:

• a list of audio sources used, eg digital recorded audio data in compressed and uncompressed form such as WAV, MPEG, AIFF and digital audio recordings such as a compact disk and
• the timing of the control information.

• Informationen zur Identifizierung der Audioquelle
• zusätzlich berechnete Information, die Charakteristiken der Audioquelle beschreibt (z.B. Abspiellänge und Tempoinformationen)
• beschreibende Information zur Herkunft und Urheberinformation der Audioquelle (z.B. Künstler, Album, Verlag etc.)
• Metainformation, z.B. Zusatzinformation die über den Hintergrund der Audioquelle informiert (z.B. Musikgenre, Information zum Künstler und Verlag)

The list of audio sources used includes:

• Information for identifying the audio source
• additionally calculated information describing the characteristics of the audio source (eg playing length and tempo information)
• descriptive information on the source and copyright information of the audio source (eg artist, album, publisher etc.)
• Meta information, eg additional information that informs about the background of the audio source (eg music genre, information about the artist and publisher)

• die zeitliche Abfolge von Steuerdaten
• die zeitliche Abfolge von exakten Abspielpositionen in der Audioquelle
• Intervalle mit kompletter Zustandsinformation aller Stellglieder, um als Wiederaufsetzpunkte der Wiedergabe zu dienen

The control information stores, inter alia:

• the timing of control data
• the timing of exact playback positions in the audio source
• Intervals with complete state information of all actuators to serve as re-points of the replay

 <?xml version="1.0" encoding="ISO-8859-1"?>

 <MJL VERSION="Versions Beschreibung">

 <HEAD PROGRAM="Programmname" COMPANY=" Firmenname"/>

 <MIX TITLE="Titel des Mixes">

 <LOCATION FILE="Kennung der Steuerinformationsdatei" PATH="Speicherort der
 Steuerinformationsdatei"/>

 <COMMENT>Kommentare und Bemerkungen zum Mix</COMMENT>

 </MIX>

 <PLAYLIST>

 <ENTRY TITLE="Titel Eintrag 1" ARTIST="Name des Autors" ID="Kennung des
 Titels">

 <LOCATION FILE=" Kennung der Audioquelle" PATH-"Speicherort der Audioquel-
 le" VOLUME="Speichermedium der Datei"/>

 <ALBUM TITLE="Name des zugehörigen Albums" TRACK="Kennung des Tracks auf
 Album"/>

 <INFO PLAYTIME="Abspieldauer in Sekunden" GENRE_ID="Musik Genre-Kennung"/>
 <TEMPO BPM="Abspieltempo in BPM" BPM_QUALITY="GUte des Tempowerts aus der
 Analyse"/>

 <CUE POINT1="Lage des 1. Markierungspunkts" ... POINTn="Lage des n. Markie-
 rungspunkts"/>

 <FADE TIME-"Überblendzeit" MODE="Überblendmodus">

 <COMMENT>Kommentare und Bemerkungen zum Audiostück>

 <IMAGE FILE="Kennung einer Bilddatei als zusätzliche Kommentarmöglich-
 keit"/>

 <REFERENCE URL="Kennung für weiterführende Informationen zur Audioquel-
 le"/>

 </COMMENT>

 </ENTRY>

 ...

 <ENTRY ...>

 ...

 </ENTRY>

 </PLAYLIST>

 </MJL>

The following is a possible example of managing the list of audio pieces in one form of the XML format. XML stands for Extensible Markup Language. This is a name for a metalanguage for describing pages in the WWW (World Wide Web). In contrast to HTML (Hypertext Markup Language), it is possible for the author of an XML document in the document to define certain extensions of XML in the Document Type Definition part of the document and also to use them in the same document.

 <? xml version = "1.0" encoding = "ISO-8859-1"?><MJL VERSION = "Version Description"><HEAD PROGRAM = "program name" COMPANY = "company name"/><MIX TITLE = "Mix title"><LOCATION FILE = "Control Information File ID" PATH = "Location of the
 Control information file "/><COMMENT> comments and comments about the mix </ COMMENT></MIX><PLAYLIST><ENTRY TITLE = "Title Entry 1" ARTIST = "Author Name" ID = "ID of the
 Title "><LOCATION FILE = "ID of the audio source" PATH- "Location of the audio source
 le "VOLUME =" file storage medium "/><ALBUM TITLE = "Name of the associated album" TRACK = "Identification of the track
 Album "/><INFO PLAYTIME = "Playing time in seconds" GENRE_ID = "Music genre ID"/> 
 <TEMPO BPM = "Play Tempo in BPM" BPM_QUALITY = "Good of the tempo value from the
 Analysis "/><CUE POINT1 = "Location of the 1st marker point" ... POINTn = "Location of the nth marker point
 approximately punkts "/><FADE TIME fade time "MODE =" Fade mode "><COMMENT> comments and comments about the audio piece><IMAGE FILE = "ID of an image file as an additional commentary option
 ness "/><REFERENCE URL = "ID for further information about the audio source
 le "/></COMMENT></ENTRY>

 ...

 <ENTRY ...>

 ...

 </ ENTRY></PLAYLIST></MJL> 

Possible presets or control data for automatically generating scratch effects as described above are described below.

• Scratch Art (Full-Stop, Back & For, Back-Spin, u.v.m.)
• Scratch Dauer (1,2,... beats - auch Druckdauer-Abhängig s.u.)
• Scratch Geschwindigkeit (Spitzengeschwindigkeit)
• Beschleunigungsdauer a (Dauer einer Geschwindigkeitsänderung von +/-1)
• Scratch Frequenz (Wiederholungen pro beat bei rhythmischen Scratches)
• Gate Frequenz (Wiederholungen pro beat)
• Gate Shape (Verhältnis von "An"- zu "Aus"-Phase)
• Gate Offset (Versatz des Gate relativ zum Takt)
• Gate Routing (Zuweisung des Gates auf andere Effekt-Parameter)

These are a series of operating elements with which all parameters of the scratches can be set in advance. Which also includes:

• Scratch Art (Full-Stop, Back & For, Back-Spin, etc.)
• Scratch duration (1,2, ... beats - also printing time-dependent see below)
• Scratch speed (top speed)
• Acceleration duration a (duration of a speed change of +/- 1)
• Scratch frequency (repetitions per beat for rhythmic scratches)
• Gate frequency (repetitions per beat)
• Gate Shape (ratio of "on" to "off" phase)
• Gate Offset (offset of the gate relative to the clock)
• Gate routing (assigning the gate to other effect parameters)

These are just a few of many conceivable parameters, depending on the nature of a realized scratch effect.

The actual scratch will be triggered by a central button / control after presetting and will develop automatically from this point. The user only needs the scratch by the moment in which he presses the key (selection of the scratched audio sample) and by the duration of the keypress (selection of the scratch length).

 [Anzahl der Steuerblöcke N]

 Für [Anzahl der Steuerblöcke N] wird wiederholt (

 [Zeitdifferenz seit letztem Steuerblock in Millisekunden]

 [Anzahl der Steuerpunkte M]

 Für [Anzahl der Steuerpunkte M] wird wiederholt {

 [Kennung des Controllers]

 [Controller Kanal]

 [Neuer Wert des Controllers]

      }

 }

The control information data referenced by the list of audio pieces is preferably stored in binary format. The basic structure of the stored control information in a file can be described as an example as follows:

 [Number of control blocks N]

 For [number of control blocks N], repeat (

 [Time difference since last control block in milliseconds]

 [Number of control points M]

 For [number of control points M] is repeated {

 [Identifier of the controller]

 [Controller channel]

 [New value of the controller]

      }

 } 

[Controller Identifier] denotes a value identifying a controller (e.g., volume, speed, position, play direction, etc.) of the interactive music player. Such controllers may have multiple subchannels [controller channel], e.g. Number of the playback module to be assigned. A unique control point M is addressed by [controller identifier], [controller channel].

As a result, a digital record of the blending process or scratch process is created which can be stored, reproduced, duplicated and transmitted non-destructively with respect to the audio material, e.g. over the internet.

An advantageous embodiment with such control files is a disk D, as this is based on FIG. 9 is illustrated. This has a combination of a normal audio CD with digital audio data AUDIO_DATA a first data area D1 with a stored on another data part D2 of the CD program PRG_DATA for playing such also existing mix files or scratch effect files MIX_DATA directly on the on the CD stored audio data AUDIO_DATA access. In this case, the play or mix application PRG_DATA does not necessarily have to be part of such a data carrier. Also, a combination of a first data area D1 with digital audio information AUDIO DATA and a second data area with one or more files with said digital control data MIX_DATA is advantageous, since such a data carrier contains in connection with a music player of the invention, all necessary information for reproduction a new work created earlier from existing digital audio sources.

However, the invention can be implemented particularly advantageously on a suitably programmed digital computer with corresponding audio interfaces, by a software program performing the method steps described above on the computer system (for example the application PRG_DATA).

All features mentioned in the preceding description or illustrated in the figures should, if the known state of the art permits, be regarded as falling within the scope of the invention alone or in combination.

The foregoing description of preferred embodiments of the invention has been presented for purposes of illustration. These embodiments are not exhaustive. Also, the invention is not limited to the precise form disclosed, but numerous modifications and changes are possible within the scope of the technical teaching given above. A preferred embodiment has been chosen and described in order to clarify the principal details of the invention and practical applications to enable those skilled in the art to practice the invention. A variety of preferred embodiments as well as other modifications come in special application areas into consideration.

LIST OF REFERENCE NUMBERS

beat

Duration of a quarter note of a current track

from

Duration of the deceleration and acceleration process

c

Stationary phase

SAMPLE

Playback position of the audio signal

t

Time

v

speed

x

path

T

Total duration of a scratch

rü

Reverse run Phase

vo

Forward-up phase

RATE

Frequency of a gate operation

SHAPE

Ratio of "on" to "off" phase

OFFSET

Phase shift, relative to the reference clock

E _i

Events of an audio data stream

T _i

time intervals

F1, F2

frequency bands

BD1, BD2

Detectors for rhythm-relevant information

BPM REF

Reference time interval

BPM_C1, BPM_C2

Processing units for tempo detection

T _1i

ungrouped time intervals

T _2i

Pairs of time intervals

T _3i

Triplets of time intervals

October

Time Oktavierungseinheiten

T _1io ... T _3io

Time-octaved time intervals

CHK

consistency check

BPM1, BPM2

independent streams of tempo bpm

STAT

Statistical evaluation of tempo values

N

accumulation points

A, bpm

Approximate determined tempo of a piece of music

P

approximately determined phase of a piece of music

1 ... 6

steps

MCLK

Reference oscillator / master clock

V

comparator

+

Phase match

-

phase shift

q

correction value

bpm_neu

resulting new tempo value A

RESET

Restart with tempo change

CD-ROM

Audio data source / CD-ROM drive

S

central instance / scheduler

TR1 ... TR n

Audio tracks

P1 ... Pn

buffer memory

A1 ... An

current play positions

S1 ... Sn

Beginnings of the data

R1, R2

Controls / Controls

LP

Low-pass filter

DIFF

differentiator

SW1

switch

IN1, IN2

first and second entrance

a

first operating mode

b

second operating mode

SL

Ramp-smoothing / slew limiter

PLAY

playback unit

DEC

decoder

B

buffer memory

R

Reading unit with variable tempo

PEF

Pre-Emphasis Filter / Predistortion Filter

DEF

De-emphasis filter / de-emphasis filter

AUDIO_OUT

Audio output

D

Sound carrier / data medium

D1, D2

data areas

AUDIO_DATA

digital audio data

MIX_DATA

digital control data

PRG_DATA

Computer program data

Claims (24)

1. Method for electrical sound generation by automatically modulating an audio signal so as to produce musical scratch effects, wherein the source material is an audio signal (sample) in digital format with a predefinable time duration, wherein the playback position and/or the playback direction and/or the playback speed and/or the playback volume and/or the sound characteristic of the audio signal are modulated automatically and rhythm-related (beat) based on control information in different predefinable ways, depending on musical tempo information,
   wherein the automatically determined tempo of the used audio material (sample) is used as musical tempo information, or an external reference tempo is used as musical tempo information, and wherein the control information represents motions of a record on a turntable of a record player.
2. Method for electrical sound generation according to claim 1,
   characterized in that
   for generating the control information, motions of a record are recorded during a manual scratch as discrete time values, or
   that for generating the control information, virtual motions of a record for a scratch effect are assembled in form of discrete time values with a predefinable resolution, in particular by way of graphic editing.
3. Method for electrical sound generation according to one of the preceding claims 1 to 2,
   characterized in that
   the control information relating to form, duration and speed of the modulation of the audio signal represents the form, duration and speed of a motion of a record for a scratch effect.
4. Method for electrical sound generation according to one of the preceding claims 1 to 3,
   characterized in that
   an acceleration of a motion of a record is also determined as a time-discrete control value for a scratch effect and defined for modulating the audio signal.
5. Method for electrical sound generation according to claim 4,
   characterized in that
   for generating a control value for the acceleration for a motion of a scratch effect, a deceleration and acceleration of the record with the same acceleration is assumed.
6. Method for electrical sound generation according to one of the preceding claims,
   characterized in that
   based on additional control information, a segment-wise emphasis of certain passages or of the audio signal (sample) or of the motion is performed automatically and rhythm-related (beat) in a different predefinable way depending on the musical tempo information, in particular by emphasizing a corresponding rhythm through manipulation of the volume or the sound characteristic.
7. Method for electrical sound generation according to one of the preceding claims,
   characterized in that
   for determining musical tempo information, tempo and phase of musical information presented in digital form, in particular of the audio signal (sample), is determined according to the following method steps:

   - approximately determining the tempo (A) of the music information by a statistical evaluation (STAT) of the time intervals (Ti) between rhythm-related beat information in the digital audio data (Ei),

   - approximately determining the phase (P) of the piece of music based on the position of the beats in the digital audio data in the time pattern of a reference oscillator (MCLK) oscillating with a frequency that is proportional to the determined tempo,

   - successively correcting the determined tempo (A) and phase (P) of the music information based on a possible phase shift of the reference oscillator (MCLK) relative to the digital audio data by evaluating the resulting systematic phase shift and regulating the frequency of the reference oscillator proportional to the determined phase shift.
8. Method for electrical sound generation according to claim 7,
   characterized in that
   rhythm-related beat information (Ti) is obtained by bandpass filtering (F1, F2) of the underlying digital audio data in different frequency ranges.
9. Method for electrical sound generation according to claim 7 or 8,
   characterized in that
   rhythm intervals of the audio data are transformed, if necessary, by multiplying their frequency with powers of 2 to a predefined frequency octave (OKT), where these time intervals (T1io ... T3io) are used for determining the tempo.
10. Method for electrical sound generation according to claim 9,
    characterized in that
    the frequency transformation (OKT) is preceded by grouping of rhythm intervals (Ti), in particular into pairs (T2i) or into groups of three (T3i), by adding their time values.
11. Method for electrical sound generation according to one of the claims 7 to 10,
    characterized in that
    the obtained quantity of data of time intervals (BPM1, BPM2) of the rhythm-related beat information is analyzed for accumulation points (N), and the tempo is determined approximately from information of a accumulation maximum.
12. Method for electrical sound generation according to one of the claims 7 to 11,
    characterized in that
    for approximately determining the phase (P) of a piece of music, the phase of the reference oscillator (MCLK) is selected so as to produce the greatest possible agreement between the rhythm-related beat information in the digital audio data and the zero crossings of the reference oscillator (MCLK).
13. Method for electrical sound generation according to one of the claims 7 to 11,
    characterized in that
    the determined tempo and phase of the piece of music are successively corrected (2, 3, 4, 5) in regular time intervals, wherein the time intervals are so short that resulting correction movements and/or collection displacements remain below the audibility limit,
    and/or
    that all successive corrections of determined tempo and phase of the piece of music are accumulated over time (4) and used for additional corrections with gradually increasing precision, wherein successive corrections are preferably performed until they are smaller than a predefined acceptable error limit value, in particular until the corrections for the determined tempo are smaller than an error limit value of less than 0.1 %,
    and/or
    that if the respective corrections are always negative or always positive (6) during a predefinable time interval, the tempo (A) and phase (P) are again approximately determined (RESET) by a subsequent successive correction (2, 3, 4, 5).
14. Interactive music player for electrical sound generation by automatically modulating an audio signal so as to produce musical scratch effects, comprising:

    - a processing unit which is constructed and configured so that a source material provided as an audio signal (sample) in digital format with a predefinable time duration, is modulated automatically and rhythm-related (beat) based on musical tempo information with respect to its playback position and/or the playback direction and/or the playback speed and/or the playback volume and/or sound characteristic,
    wherein the automatically determined tempo of the used audio material (sample) is used as musical tempo information, or an external reference tempo provided via an interface is used as musical tempo information, and

    - an interface is provided for receiving control information which is subsequently interpreted, wherein the control information represents motions of a record on a turntable of a record player,

    - a means for graphically representing of the control information.
15. Interactive music player according to claim 14, with

    - a means for graphically representing the actual playback position, with the means capable of displaying an amplitude envelope curve of the sound waveform of the reproduced piece of music over a predefinable time interval before and after the actual playback position, wherein the real-time representation moves with the speed with which the piece of music is reproduced, and with

    - a means for smoothing (LP, SL) a stepped curve of time-limited playback position data defined by the second control element (R2) to form a signal which changes smoothly with a time resolution corresponding to the audio sampling rate.
16. Interactive music player according to claim 15, wherein a means for smoothing a slope (SL) is provided for smoothing a stepped curve of time-limited playback position data, by which a ramp with a constant slope can be triggered for each defined playback position message, with the ramp changing the smoothed signal during a predefinable time interval from its previous value to the value of the playback position message,
    or
    wherein a linear digital low pass filter (LP), in particular a second order resonance filter, is used for smoothing a stepped curve of predefined time-limited playback position data.
17. Interactive music player according to one of the preceding claims 15 to 16, wherein in the event of a change between the operating modes (a, b), the actual playback speed (DIFF) attained in the preceding mode can be changed by a smoothing function, in particular by ramp smoothing (SL) or by a linear digital low-pass filter (LP), to the playback speed that corresponds to the new operating mode,
    and/or
    wherein an audio signal passes through a scratch-audio-filter, whereby the audio signal undergoes a Pre-Emphase Filtering (PEF) and is stored in a buffer memory (B), from which it can be read (R) with variable tempo depending on the respective playback speed, and subsequently undergoes a De-Emphase Filtering (DEF) and is reproduced.
18. Interactive music player according to one of the preceding claims 15 to 17, wherein each reproduced audio data stream can be manipulated in real time by signal processing means, in particular by filter devices and/or audio effects.
19. Interactive music player according to one of the preceding claims 15 to 18, wherein real-time interventions with respect to the temporal curve can be stored as digital control information (MIX_DATA), in particular those of a manual scratch intervention with a separate control element (R2) and/or additional signal processing.
20. Interactive music player according to one of the preceding claims 15 to 19, wherein stored digital control information has a format which includes information for identification of the processed pieces of music and a corresponding associated time-sequence of playback positions and state information of the actutating members of the music player.
21. Interactive music player according to one of the preceding claims 15 to 20, which is realized by a suitably programmed computer system having audio interfaces.
22. Computer program product which can be loaded directly into the internal memory of a digital computer and comprises software segments that can be used to execute the method steps according to one of the claims 1-13, 15, 16, when the program product is executed on a computer, wherein the control data can already be stored on the computer program product, thereby enabling automatic playback.
23. Computer program product according to claim 22, in particular a Compact Disc, comprising:

    - a first data range (D1) with digital audio data (AUDIO_DATA) of one or several pieces of music (TR1 ... TRn), and

    - a second data range (D2) with a control file (MIX_DATA) with digital control information for controlling a music player, wherein

    - the control data (MIX_DATA) of the second data range (D2) make reference to audio data (AUDIO_DATA) of the first data range (D1).
24. Computer program product according to claim 23, wherein the digital control data (MIX_DATA) of the second data range (D2) represent interactive recordings of manual scratch interventions and/or starting points and type of automatic scratch interventions in pieces of music as a new complete work of the digital audio information (AUDIO_DATA) of pieces of music of the first data range (D1),
    and/or
    wherein the stored digital control information (MIX_DATA) of the second data range (D2) has a format that comprises information for identifying the processed pieces of music (TR1 ... TRn) of the first data range (D1) and a corresponding associated time sequence of playback positions and state information of the actutating members of the music player.

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