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Music frequently exhibits regular repeating structure because of musical beats, vamps, and rhythms. In this thesis, an audio compression algorithm that takes advantage of this structure is described. After providing some background, a simplified implementation of a perceptual audio compression algorithm is described and then modified to explore this idea. The newly proposed algorithm is evaluated using a publicly available dataset. The results show that the proposed algorithm can improve the compression ratio while retaining audio quality on rhythmic mixtures and background music. Better audio compression has potential applications in virtual music performance. / Graduate

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Dans le cadre de notre mémoire, nous nous intéressons aux moyens d’obtenir des niveaux de sonie élevés, particulièrement lorsqu’une telle pratique est pertinente (dans notre cas, pour la musique électronique et pour le hip-hop), soit des sons qui sont forts du le point de vue de la perception (loudness), et ce, sans obtenir des distorsions relatives à l’écrêtage, soit la coupure horizontale d’une amplitude, et sans perdre trop de gamme dynamique, soit la diminution de l’écart entre la valeur en dB la plus faible et la plus élevée d’une pièce (par l’utilisation agressive de limiteurs/compresseurs). Concrètement, nous tentons trois approches autonomes parallèles qui s’inscrivent dans cette méthode organique : 1) Concevoir la conception sonore en fonction de la perception du niveau de sonie ; 2) Contrôler dynamique des percussions en prévision de la compression ; 3) Mixer et matricer en même temps. / This study is interested in the means of obtaining loud mixes, particularly in musical genres where such practices are pertinent (electronic music and hip-hop, in our case), while avoiding the usual disadvantages and defects of loud mixes. In other words, we are seeking for ways to achieve loud tunes from the listener’s subjective perception, without encountering distortions caused by the squaring of the waves’ amplitudes, usually associated with the heavy usage of compressors and limiters, and without loosing to much dynamic variety and/or dynamic range (the range between the lowest and the highest value [dB] in a mix). For these endeavours we have tried three different—yet parallel—autonomous methods. These are: 1) Sound design in function of our subjective perception of loudness (psychoacoustics); 2) Dynamic control of percussions for an efficient compression at the mastering stage; 3) Mixing and mastering at the same time.

The increasing interest for immersive experiences in areas such as augmented and virtual reality makes high quality 3D sound more important than ever before. A technique for capturing and rendering 3D audio which has received more attention during the last twenty years are Higher Order Ambisonics (HOA). Higher Order Ambisonics is a scene based audio format which has a lot of advantages compared to other standard formats. Hovever, one problem with HOA is that it requires a lot of bandwidth. For example, sending an uncoded high quality HOA signal requires 49 channels to be transmitted at the same time which requires a bandwidth of about 40 Mbps. A lot of effort has been made in the last ten years on coding HOA signals. In this thesis, two different approaches are taken on coding HOA signals. In one approach, called Sound Field Rotation (SFR) in this thesis, the microphone that records the sound field is virtually rotated to see if it is possible to make some of the channels zero. The second approach, called Sound Field Decomposition (SFD) in this thesis, use Principal component analysis to decompose a sound field into a foreground and background component. The Sound Field Decomposition approach is inspired by the emerging MPEG-H 3D Audio standard for coding HOA signals. The result shows that the Sound Field Rotation method only works for very simple sound scenes. It has also been shown that a 49 channels HOA signal can be reduced to as little as 7 channels if the sound scene consists of a point source. The Sound Field Deomposition method worked for more complex sound scenes. It was shown that a MPEG similar system could be improved. Result from MUSHRA (Multiple stimuli with hidden reference and anchor) listening tests showed that an improved MPEG similar system reached a MUSHRA score about 78 while the MPEG similar system reached 55 at a bitrate of 256 kbps. Without coding each monochannels with the 3GPP EVS (Enhanced voice services) codec, the improved MPEG similar system reached the MUSHRA score 85. At 256 kbps, the improved MPEG similar system coded the HOA signal into six channels instead of 49 for the uncoded signal. From objective results, it was shown that the improved MPEG similar system had largest effect at low bitrates.

A game could have thousands of sound assets, to fit all of those files to a manageable storage space it is often necessary to reduce the size of the files to a more manageable size so they have to be compressed. One type of sound that often takes up a lot of disc space (because there is so much of it) is dialogue. In the popular game engine Unreal Engine 4 (UE4) the the audio is compressed to Ogg Vorbis and has as default the bit rate is set to 104 kbit/s. The goal of this paper is to see if untrained listeners find dialogue compressed in Ogg Vorbis 104 kbit/s good enough for dialogue or if they prefer higher bit rates. A game was made in UE4 that would act as a listening test. Dialogue audio was recorded with a male and a female voice-actor and was compressed in UE4 in six different bit rates. 24 untrained subjects was asked to play the game and identify the two out of six robots with the dialogue audio they thought sound the best. The results show that the subjects prefer the higher bit rates that was tested. The results was analyzed with a chi-squared test which showed that the null-hypothesis can be rejected. Only 21% of the answers were towards UE4s default bit rate of 104 kbit/s or lower. The result suggest that the subjects prefer dialogue in higher bit rates and UE4 should raise the default bit rate.

MP3, or MPEG-1 Layer 3, is the most widely-used format for storing compressed audio. MP3 is more advantageous than uncompressed audio (PCM), offering a much smaller size but without a noticeable loss in audio quality. This report will demonstrate decoding and playback of MP3 audio using a TLL5000 FPGA board. / text

Research paper (M.A.) -- Cardinal Stritch College -- Milwaukee, 1972. / A research paper submitted in partial fulfillment of the requirements for the degree of Master of Arts in Education (Education of Mentally Handicapped). Includes bibliographical references (p. 62-64).