Portable smart speakers with wireless connections have in recent years become more popular. These speakers are often moved to new locations and placed in different positions in different rooms, which affects the sound a listener is hearing from the speaker. These speakers usually have microphones on them, typically used for voice recording. This thesis aims to provide a way to compensate for the speaker position’s effect on the sound (so called room correction) using the microphones on the speaker and the speaker itself. Firstly, the room frequency response is estimated for several different speaker positions in a room. The room frequency response is the frequency response between the speaker and the listener. From these estimates, the relationship between the speaker’s position and the room frequency response is modeled. Secondly,an algorithm that estimates the speaker’s position is developed. The algorithm estimates the position by detecting reflections from nearby walls using the microphones on the speaker. The acquired position estimates are used as input for the room frequency response model, which makes it possible to automatically apply room correction when placing the speaker in new positions. The room correction is shown to correct the room frequency response so that the bass has the same power as the mid- and high frequency sounds from the speaker, which is according to the research aim. Also, the room correction is shown to make the room frequency response vary less with respect to the speaker’s position.

In this bachelor thesis, the possiblity to correct for room acousticsbased on frequency analysis is studied. A software to calculate transferfunctions online was constructed and tested. This was done using a ver-sion of the Maximum Length Sequence method, which is a method thatrequires long sequences for rooms with long reverberation. During theproject, it was noted that zero padding the sequences improved the ac-curacy greatly, it was also noted that the length of the zero pad aectedthe results. The software was tested both in computer simulations andin practice. While testing in practice, it was noted that the system haslimitations on which rooms it would work in. All testsignals were recordedand afterwards, compared to the original recording. The constructed soft-ware showed, that it is possible to correct for unknown transfer functionsusing only frequency analysis, to some extent. Further, it does correct forthe room's transfer function, but it is dicult to say if it this is valid forall rooms and transfer functions.

In this Thesis a study on the subject on how to improve the overall sound quality within a room using signal processing, played back using a loudspeaker, was conducted. This is a subject that has gained attention during the recent years, with more and more consumer and professional products including it. The objective was to find techniques that offered perceptually good audio quality covering most of the room, while being robust and stable. The solution was to design a correction system which fulfilled these requirements and took advantage of today’s computing technology. This problem and its solution, as included in this Thesis, expose the reader to an introduction to loudspeaker system design and reproduction, room acoustics, psychoacoustics (how humans perceive sound), signal extraction (pre-processing) and filter design as well as design considerations for all of these components. Different ways that this system can be developed further were also discussed. This thesis was mainly based on the theory explained in Immersive Audio Signal Processing av S. Bharitkar and C. Kyriakakis [1]. The results of experiments show that a well-performing room correction system can be realized using a microphone with a known response and a computer. In most cases the improvement in both audible and measurable audio quality is considerable, with only a few cases where an improvement was not made. Using multiple measurement positions, positions of the microphone, led to a further improvement. On the other hand, it was also shown that having two well-positioned microphones was shown to be close to as performant as covering the whole room, even if a combination measurements over the whole listening area was the best performing approach. / I den här examensuppsatsen utfördes en studie på hur man kan förbättra ljudupplevelsen i ett rum, när ljud spelas upp på en högtalare, genom att använda signalbehanlindning. Detta är ett ämne som blivit mer relevant, med mer och mer avancerade och prisvärda ljudsystem på marknaden. Målet för projektet var att hitta tekniker som gav en förbättring av ljudupplevelsen som både var robust och täckte en större yta av rummet. Lösningen var att designa ett korrektionssystem som uppfyllde kraven och tog vara på de stora beräkningsresurserna som dagens datorer erbjuder. Problemet och dess lösning förklaras tillsammans med en introduktion av varje ämne som påverkar ljuduppspelningen samt vad man kan göra för att motverka de oönskade sidoeffekterna. Det inkluderar områden såsom högtalarsystemkonstruktion, rumsaksustik, signalbearbetning och filterdesign, samt exempel och en diskussion på vidare utvecklingar av projektet. Projektet baserades till stor del på boken Immersive Audio Signal Processing av S. Bharitkar and C. Kyriakakis [1] som beskriver hur man skapar en inneslutande ljudupplevelse via rumskorrigering. Slutresultaten visade att det går att med några få steg bygga ett högtalar- och rumskorrigeringssystem som uppfyller de satta villkoren med mycket god ljudkvalitet. Även de enklare systemen, som bara använder en enstaka mätpunkt, kan korrigera för uppspelningen i ett helt rum med goda resultat. Genom att gå vidare med att undersöka att kombinera flera mätpunkter visades det att bara två välplacerade punkter kan prestera likvärdigt med att mäta över hela lyssningsytan. Däremot visas det att en kombination av mätningar över lyssningytan alltid presterar bäst.