360 research outputs found
Log complex color for visual pattern recognition of total sound
The present disclosure is generally directed to audio visualization methods for visual pattern recognition of sound. In particular, the present disclosure is directed to plotting amplitude intensity as brightness/saturation and phase-cycles as hue-variations to create visual representations of sound.
For the most up-to-date information about these patents, including the availability of Certificates of Correction, be sure to check the United States Patent and Trademark Office\u27s free, publicly accessible database: Patent Public Search https://ppubs.uspto.gov/pubwebapp/static/pages/landing.htmlhttps://irl.umsl.edu/patents/1022/thumbnail.jp
Exploiting prior knowledge during automatic key and chord estimation from musical audio
Chords and keys are two ways of describing music. They are exemplary of a general class of symbolic notations that musicians use to exchange information about a music piece. This information can range from simple tempo indications such as “allegro” to precise instructions for a performer of the music. Concretely, both keys and chords are timed labels that describe the harmony during certain time intervals, where harmony refers to the way music notes sound together. Chords describe the local harmony, whereas keys offer a more global overview and consequently cover a sequence of multiple chords.
Common to all music notations is that certain characteristics of the music are described while others are ignored. The adopted level of detail depends on the purpose of the intended information exchange. A simple description such as “menuet”, for example, only serves to roughly describe the character of a music piece. Sheet music on the other hand contains precise information about the pitch, discretised information pertaining to timing and limited information about the timbre. Its goal is to permit a performer to recreate the music piece. Even so, the information about timing and timbre still leaves some space for interpretation by the performer.
The opposite of a symbolic notation is a music recording. It stores the music in a way that allows for a perfect reproduction. The disadvantage of a music recording is that it does not allow to manipulate a single aspect of a music piece in isolation, or at least not without degrading the quality of the reproduction. For instance, it is not possible to change the instrumentation in a music recording, even though this would only require the simple change of a few symbols in a symbolic notation.
Despite the fundamental differences between a music recording and a symbolic notation, the two are of course intertwined. Trained musicians can listen to a music recording (or live music) and write down a symbolic notation of the played piece. This skill allows one, in theory, to create a symbolic notation for each recording in a music collection. In practice however, this would be too labour intensive for the large collections that are available these days through online stores or streaming services. Automating the notation process is therefore a necessity, and this is exactly the subject of this thesis. More specifically, this thesis deals with the extraction of keys and chords from a music recording. A database with keys and chords opens up applications that are not possible with a database of music recordings alone. On one hand, chords can be used on their own as a compact representation of a music piece, for example to learn how to play an accompaniment for singing. On the other hand, keys and chords can also be used indirectly to accomplish another goal, such as finding similar pieces.
Because music theory has been studied for centuries, a great body of knowledge about keys and chords is available. It is known that consecutive keys and chords form sequences that are all but random. People happen to have certain expectations that must be fulfilled in order to experience music as pleasant. Keys and chords are also strongly intertwined, as a given key implies that certain chords will likely occur and a set of given chords implies an encompassing key in return. Consequently, a substantial part of this thesis is concerned with the question whether musicological knowledge can be embedded in a technical framework in such a way that it helps to improve the automatic recognition of keys and chords.
The technical framework adopted in this thesis is built around a hidden Markov model (HMM). This facilitates an easy separation of the different aspects involved in the automatic recognition of keys and chords. Most experiments reviewed in the thesis focus on taking into account musicological knowledge about the musical context and about the expected chord duration. Technically speaking, this involves a manipulation of the transition probabilities in the HMMs. To account for the interaction between keys and chords, every HMM state is actually representing the combination of a key and a chord label.
In the first part of the thesis, a number of alternatives for modelling the context are proposed. In particular, separate key change and chord change models are defined such that they closely mirror the way musicians conceive harmony. Multiple variants are considered that differ in the size of the context that is accounted for and in the knowledge source from which they were compiled. Some models are derived from a music corpus with key and chord notations whereas others follow directly from music theory.
In the second part of the thesis, the contextual models are embedded in a system for automatic key and chord estimation. The features used in that system are so-called chroma profiles, which represent the saliences of the pitch classes in the audio signal. These chroma profiles are acoustically modelled by means of templates (idealised profiles) and a distance measure. In addition to these acoustic models and the contextual models developed in the first part, durational models are also required. The latter ensure that the chord and key estimations attain specified mean durations.
The resulting system is then used to conduct experiments that provide more insight into how each system component contributes to the ultimate key and chord output quality. During the experimental study, the system complexity gets gradually increased, starting from a system containing only an acoustic model of the features that gets subsequently extended, first with duration models and afterwards with contextual models. The experiments show that taking into account the mean key and mean chord duration is essential to arrive at acceptable results for both key and chord estimation. The effect of using contextual information, however, is highly variable. On one hand, the chord change model has only a limited positive impact on the chord estimation accuracy (two to three percentage points), but this impact is fairly stable across different model variants. On the other hand, the chord change model has a much larger potential to improve the key output quality (up to seventeen percentage points), but only on the condition that the variant of the model is well adapted to the tested music material. Lastly, the key change model has only a negligible influence on the system performance.
In the final part of this thesis, a couple of extensions to the formerly presented system are proposed and assessed. First, the global mean chord duration is replaced by key-chord specific values, which has a positive effect on the key estimation performance. Next, the HMM system is modified such that the prior chord duration distribution is no longer a geometric distribution but one that better approximates the observed durations in an appropriate data set. This modification leads to a small improvement of the chord estimation performance, but of course, it requires the availability of a suitable data set with chord notations from which to retrieve a target durational distribution. A final experiment demonstrates that increasing the scope of the contextual model only leads to statistically insignificant improvements. On top of that, the required computational load increases greatly
Automatic Transcription of Bass Guitar Tracks applied for Music Genre Classification and Sound Synthesis
Musiksignale bestehen in der Regel aus einer Überlagerung mehrerer
Einzelinstrumente. Die meisten existierenden Algorithmen zur automatischen
Transkription und Analyse von Musikaufnahmen im Forschungsfeld des Music
Information Retrieval (MIR) versuchen, semantische Information direkt aus
diesen gemischten Signalen zu extrahieren. In den letzten Jahren wurde
häufig beobachtet, dass die Leistungsfähigkeit dieser Algorithmen durch
die Signalüberlagerungen und den daraus resultierenden Informationsverlust
generell limitiert ist. Ein möglicher Lösungsansatz besteht darin,
mittels Verfahren der Quellentrennung die beteiligten Instrumente vor der
Analyse klanglich zu isolieren. Die Leistungsfähigkeit dieser Algorithmen
ist zum aktuellen Stand der Technik jedoch nicht immer ausreichend, um eine
sehr gute Trennung der Einzelquellen zu ermöglichen. In dieser Arbeit
werden daher ausschließlich isolierte Instrumentalaufnahmen untersucht,
die klanglich nicht von anderen Instrumenten überlagert sind. Exemplarisch
werden anhand der elektrischen Bassgitarre auf die Klangerzeugung dieses
Instrumentes hin spezialisierte Analyse- und Klangsynthesealgorithmen
entwickelt und evaluiert.Im ersten Teil der vorliegenden Arbeit wird ein
Algorithmus vorgestellt, der eine automatische Transkription von
Bassgitarrenaufnahmen durchführt. Dabei wird das Audiosignal durch
verschiedene Klangereignisse beschrieben, welche den gespielten Noten auf
dem Instrument entsprechen. Neben den üblichen Notenparametern Anfang,
Dauer, Lautstärke und Tonhöhe werden dabei auch instrumentenspezifische
Parameter wie die verwendeten Spieltechniken sowie die Saiten- und Bundlage
auf dem Instrument automatisch extrahiert. Evaluationsexperimente anhand
zweier neu erstellter Audiodatensätze belegen, dass der vorgestellte
Transkriptionsalgorithmus auf einem Datensatz von realistischen
Bassgitarrenaufnahmen eine höhere Erkennungsgenauigkeit erreichen kann als
drei existierende Algorithmen aus dem Stand der Technik. Die Schätzung der
instrumentenspezifischen Parameter kann insbesondere für isolierte
Einzelnoten mit einer hohen Güte durchgeführt werden.Im zweiten Teil der
Arbeit wird untersucht, wie aus einer Notendarstellung typischer sich
wieder- holender Basslinien auf das Musikgenre geschlossen werden kann.
Dabei werden Audiomerkmale extrahiert, welche verschiedene tonale,
rhythmische, und strukturelle Eigenschaften von Basslinien quantitativ
beschreiben. Mit Hilfe eines neu erstellten Datensatzes von 520 typischen
Basslinien aus 13 verschiedenen Musikgenres wurden drei verschiedene
Ansätze für die automatische Genreklassifikation verglichen. Dabei zeigte
sich, dass mit Hilfe eines regelbasierten Klassifikationsverfahrens nur
Anhand der Analyse der Basslinie eines Musikstückes bereits eine mittlere
Erkennungsrate von 64,8 % erreicht werden konnte.Die Re-synthese der
originalen Bassspuren basierend auf den extrahierten Notenparametern wird
im dritten Teil der Arbeit untersucht. Dabei wird ein neuer
Audiosynthesealgorithmus vorgestellt, der basierend auf dem Prinzip des
Physical Modeling verschiedene Aspekte der für die Bassgitarre
charakteristische Klangerzeugung wie Saitenanregung, Dämpfung, Kollision
zwischen Saite und Bund sowie dem Tonabnehmerverhalten nachbildet.
Weiterhin wird ein parametrischerAudiokodierungsansatz diskutiert, der es
erlaubt, Bassgitarrenspuren nur anhand der ermittel- ten notenweisen
Parameter zu übertragen um sie auf Dekoderseite wieder zu
resynthetisieren. Die Ergebnisse mehrerer Hötest belegen, dass der
vorgeschlagene Synthesealgorithmus eine Re- Synthese von
Bassgitarrenaufnahmen mit einer besseren Klangqualität ermöglicht als die
Übertragung der Audiodaten mit existierenden Audiokodierungsverfahren, die
auf sehr geringe Bitraten ein gestellt sind.Music recordings most often consist of multiple instrument signals, which
overlap in time and frequency. In the field of Music Information Retrieval
(MIR), existing algorithms for the automatic transcription and analysis of
music recordings aim to extract semantic information from mixed audio
signals. In the last years, it was frequently observed that the algorithm
performance is limited due to the signal interference and the resulting
loss of information. One common approach to solve this problem is to first
apply source separation algorithms to isolate the present musical
instrument signals before analyzing them individually. The performance of
source separation algorithms strongly depends on the number of instruments
as well as on the amount of spectral overlap.In this thesis, isolated
instrumental tracks are analyzed in order to circumvent the challenges of
source separation. Instead, the focus is on the development of
instrument-centered signal processing algorithms for music transcription,
musical analysis, as well as sound synthesis. The electric bass guitar is
chosen as an example instrument. Its sound production principles are
closely investigated and considered in the algorithmic design.In the first
part of this thesis, an automatic music transcription algorithm for
electric bass guitar recordings will be presented. The audio signal is
interpreted as a sequence of sound events, which are described by various
parameters. In addition to the conventionally used score-level parameters
note onset, duration, loudness, and pitch, instrument-specific parameters
such as the applied instrument playing techniques and the geometric
position on the instrument fretboard will be extracted. Different
evaluation experiments confirmed that the proposed transcription algorithm
outperformed three state-of-the-art bass transcription algorithms for the
transcription of realistic bass guitar recordings. The estimation of the
instrument-level parameters works with high accuracy, in particular for
isolated note samples.In the second part of the thesis, it will be
investigated, whether the sole analysis of the bassline of a music piece
allows to automatically classify its music genre. Different score-based
audio features will be proposed that allow to quantify tonal, rhythmic, and
structural properties of basslines. Based on a novel data set of 520
bassline transcriptions from 13 different music genres, three approaches
for music genre classification were compared. A rule-based classification
system could achieve a mean class accuracy of 64.8 % by only taking
features into account that were extracted from the bassline of a music
piece.The re-synthesis of a bass guitar recordings using the previously
extracted note parameters will be studied in the third part of this thesis.
Based on the physical modeling of string instruments, a novel sound
synthesis algorithm tailored to the electric bass guitar will be presented.
The algorithm mimics different aspects of the instrument’s sound
production mechanism such as string excitement, string damping, string-fret
collision, and the influence of the electro-magnetic pickup. Furthermore, a
parametric audio coding approach will be discussed that allows to encode
and transmit bass guitar tracks with a significantly smaller bit rate than
conventional audio coding algorithms do. The results of different listening
tests confirmed that a higher perceptual quality can be achieved if the
original bass guitar recordings are encoded and re-synthesized using the
proposed parametric audio codec instead of being encoded using conventional
audio codecs at very low bit rate settings
Final Research Report on Auto-Tagging of Music
The deliverable D4.7 concerns the work achieved by IRCAM until M36 for the “auto-tagging of music”. The deliverable is a research report. The software libraries resulting from the research have been integrated into Fincons/HearDis! Music Library Manager or are used by TU Berlin. The final software libraries are described in D4.5.
The research work on auto-tagging has concentrated on four aspects:
1) Further improving IRCAM’s machine-learning system ircamclass. This has been done by developing the new MASSS audio features, including audio augmentation and audio segmentation into ircamclass. The system has then been applied to train HearDis! “soft” features (Vocals-1, Vocals-2, Pop-Appeal, Intensity, Instrumentation, Timbre, Genre, Style). This is described in Part 3.
2) Developing two sets of “hard” features (i.e. related to musical or musicological concepts) as specified by HearDis! (for integration into Fincons/HearDis! Music Library Manager) and TU Berlin (as input for the prediction model of the GMBI attributes). Such features are either derived from previously estimated higher-level concepts (such as structure, key or succession of chords) or by developing new signal processing algorithm (such as HPSS) or main melody estimation. This is described in Part 4.
3) Developing audio features to characterize the audio quality of a music track. The goal is to describe the quality of the audio independently of its apparent encoding. This is then used to estimate audio degradation or music decade. This is to be used to ensure that playlists contain tracks with similar audio quality. This is described in Part 5.
4) Developing innovative algorithms to extract specific audio features to improve music mixes. So far, innovative techniques (based on various Blind Audio Source Separation algorithms and Convolutional Neural Network) have been developed for singing voice separation, singing voice segmentation, music structure boundaries estimation, and DJ cue-region estimation. This is described in Part 6.EC/H2020/688122/EU/Artist-to-Business-to-Business-to-Consumer Audio Branding System/ABC D
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