Demo Data from VLDB 2026 / reviewers in the wild / expert
Janne Vainio
dblp:33/9407
· DBLP profile ↗
9ranked-venue papers
1as first author
0since 2021 · last 2010
—ORCID · none
Domains — the database's venue-derived domains; a paper can count in several
Graphics, computer vision, multimedia, augmented reality and games · 5 · 1 first-authorArtificial intelligence and machine learning · 3Computer networks · 1Human-computer interaction and ubiquitous computing · 1Applied, interdisciplinary, general and emerging computing · 1
Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.
| Computer graphics and multimedia
1 paper |
Audio and music processing · 100% | |
| Computer networks
1 paper |
Cellular and mobile networks · 100% |
Topics — the 2 heaviest of 3, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing
speech coding |
0.0 | 1 | 2002 | The adaptive multirate wideband speech codec (AMR-WB) · IEEE Trans. Speech Audio Process. 2002 |
Audio and music processing › speech coding
wideband speech coding |
0.0 | 1 | 2002 | The adaptive multirate wideband speech codec (AMR-WB) · IEEE Trans. Speech Audio Process. 2002 |
Methods — techniques the papers use, named apart from their topics
adaptive multirate coding · 0.1ACELP · 0.1
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2010 | Interacting with the flowabstractMobile devices offer challenges for UI design. Limited screen space leads to deep menus, complex navigation and loss of position. We introduce a new user interface concept that reverses the traditional navigation paradigm. By utilizing context awareness and allowing the user to control the UI via filters, objects of interest navigate past the user instead of the user navigating to the object. The user operates on a single view without the need for deep menu navigation. The new UI is also easy to configure. We implemented the concept on the Nokia S60 5th edition touch platform and conducted user testing with 16 users. Initially, users felt confused because of new ways of accessing things. However, after a short period of usage, majority of the users found it easy to use. Most of the users felt the system to be fun and playful. Janne Bergman, Janne Vainio |
Mobile HCI | 2 |
| 2008 | Mobile Diary for Wellness Management - Results on Usage and Usability in Two User StudiesabstractThe prevalence of lifestyle-related health problems is increasing rapidly. Many of the diseases and health risks could be prevented or alleviated by making changes toward healthier lifestyles. We have developed the Wellness Diary (WD), a concept for personal and mobile wellness management based on Cognitive-Behavioral Therapy (CBT). Two implementations of the concept were made for the Symbian Series 60 (S60) mobile phone platform, and their usability, usage, and acceptance were studied in two 3-month user studies. Study I was related to weight management and study II to general wellness management. In both the studies, the concept and its implementations were well accepted and considered as easy to use and useful in wellness management. The usage rate of the WD was high and sustained at a high level throughout the study. The average number of entries made per day was 5.32 (SD = 2.59, range = 0-14) in study I, and 5.48 (SD = 2.60, range = 0-17) in study II. The results indicate that the WD is well suited for supporting CBT-based wellness management. Elina M. Mattila, Juha Pärkkä, Marion Hermersdorf, Jussi Kaasinen, Janne Vainio, K. Samposalo, Juho Merilahti, Juha Kolari, Minna Kulju, Raimo Lappalainen, Ilkka Korhonen |
IEEE Trans. Inf. Technol. Biomed. | 5 |
| 2002 | The adaptive multirate wideband speech codec (AMR-WB)abstractThis paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec. Bruno Bessette, Redwan Salami, Roch Lefebvre, Milan Jelinek, J. Rotola-Pukkila, Janne Vainio, Hannu Mikkola, Kari Järvinen |
IEEE Trans. Speech Audio Process. | 6 |
| 2001 | Techniques for high-quality ACELP coding of wideband speechabstractWe present in this paper new methods for achieving high- quality wideband speech at low rates using the ACELP algorithm. Several innovations are introduced to optimize the quality and minimize the complexity of the coder. A multi-rate wideband speech encoding algorithm based on these techniques was recently selected by 3GPP as the standard for AMR-WB, and is currently one of the candidates for the ITU- T wideband speech coder standard at around 16 kbit/sec. This standard was jointly developed by VoiceAge and Nokia. Bruno Bessette, Roch Lefebvre, Redwan Salami, Milan Jelinek, Janne Vainio, J. Rotola-Pukkila, Hannu Mikkola, Kari Järvinen |
INTERSPEECH | 5 |
| 2001 | AMR wideband codec - leap in mobile communication voice qualityabstractThe Third Generation Partnership Project (3GPP) and European Telecommunications Standards Institute (ETSI) have carried out development and standardisation of a wideband speech codec for GSM and the third generation mobile communication WCDMA system since 1999. The Adaptive Multi-Rate Wideband (AMR-WB) codec algorithm was selected in December 2000, and the corresponding specifications were approved in March 2001. The AMR-WB codec was jointly developed by Nokia and VoiceAge. AMRWB extends the audio bandwidth from 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to existing 2nd and 3rd generation mobile communication systems. The wideband speech service provided by the AMRWB codec will give mobile communication speech quality that even exceeds (narrowband) wireline quality. J. Rotola-Pukkila, Janne Vainio, Hannu Mikkola, Kari Järvinen, Bruno Bessette, Roch Lefebvre, Redwan Salami, Milan Jelinek |
INTERSPEECH | 2 |
| 1998 | GSM EFR based multi-rate codec familyabstractThis paper describes a multi-rate codec family developed as a potential candidate for the GSM adaptive multi-rate (AMR) codec standard. The codec family consists of the GSM enhanced full rate (EFR) codec and lower bit-rate extensions thereof. The codec family consists of several codecs, i.e., modes that have different bit-rate partitionings between source coding and error protection. All the source codecs use the same ACELP-method (algebraic code excited linear predictive coding) used also in the GSM EFR codec. The codec operates at gross bit-rates of 22.8 kbit/s in the GSM full rate (FR) channel and 11.4 kbit/s in the GSM half rate (HR) channel. In the full rate channel, the codec provides improved error robustness over the GSM enhanced full rate (EFR) codec. It extends wireline quality (equal to or better than G.726-32 ADPCM) to poor channel error conditions with low C/I-ratios of 7 dB or even below. When operated in the half rate channel, the codec provides improved channel capacity while still providing wireline quality at high C/I-ratios above 16-19 dB. Janne Vainio, Hannu Mikkola, Kari Järvinen, Petri Haavisto |
ICASSP | 1 |
| 1997 | Enhanced full rate speech codec for IS-136 digital cellular systemabstractIn this paper, we describe the enhanced full rate (EFR) speech codec that has recently been standardised for the North American TDMA digital cellular system (IS-136). The EFR codec, specified in the IS-641 standard, has been jointly developed by Nokia and University of Sherbrooke. The codec consists of 7.4 kbit/s speech (source) coding and 5.6 kbit/s channel coding (error protection) resulting in a 13.0 kbit/s gross bit-rate in the channel. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec offers speech quality close to that of wireline telephony (G.726 32 kbit/s ADPCM used as a wireline reference) and provides a substantial improvement over the quality of the current speech channel. The improved speech quality is not only achieved in error-free conditions, but also in typical cellular operating conditions including transmission errors, environmental noise, and tandeming of speech codecs. Tero Honkanen, Janne Vainio, Kari Järvinen, Petri Haavisto, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul |
ICASSP | 2 |
| 1997 | GSM enhanced full rate speech codecabstractThis paper describes the GSM enhanced full rate (EFR) speech codec that has been standardised for the GSM mobile communication system. The GSM EFR codec has been jointly developed by Nokia and University of Sherbrooke. It provides speech quality at least equivalent to that of a wireline telephony reference (32 kbit/s ADPCM). The EFR codec uses 12.2 kbit/s for speech coding and 10.6 kbit/s for error protection. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec provides substantial quality improvement compared to the existing GSM full rate and half rate codecs. The old GSM codecs lack wireline quality even in error-free channel conditions, while the EFR codec provides wireline quality not only for error-free conditions but also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile to mobile calls). Kari Järvinen, Janne Vainio, Pekka Kapanen, Tero Honkanen, Petri Haavisto, Redwan Salami, Claude Laflamme, Jean-Pierre Adoul |
ICASSP | 2 |
| 1997 | Description of GSM Enhanced Full Rate Speech CodecabstractThis paper describes the enhanced full rate (EFR) speech codec that has been standardized by ETSI for the GSM mobile communications system in 1996. The codec was developed jointly by Nokia and the University of Sherbrooke. It operates at 12.2 kbit/s speech coding (source coding) bit-rate and provides speech quality equivalent to that of wireline telephony (G.726 32 kbit/s ADPCM). The algorithm is based on the algebraic code-excited linear prediction (ACELP) technology, using 20 ms speech frames. The GSM EFR speech codec provides substantial quality improvement compared to the current GSM full rate (FR) and half rate (HR) codecs. The old GSM codecs lag far behind wireline quality even in error-free conditions, while the EFR codec provides wireline quality also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile-to-mobile calls). The codec was defined using fixed-point basic operators with complexity estimated at 18 WMOPS (below that of the GSM half-rate codec). Redwan Salami, Claude Laflamme, Jean-Pierre Adoul, Kari Järvinen, Janne Vainio, Pekka Kapanen, Tero Honkanen, Petri Haavisto |
ICC (2) | 5 |