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Hannu Mikkola

dblp:46/7162 · DBLP profile ↗
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4ranked-venue papers
0as first author
0since 2021 · last 2002
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 3Graphics, computer vision, multimedia, augmented reality and games · 3

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

TopicWeightPapersLastEvidence papers
Audio and music processing
speech coding
0.012002
The adaptive multirate wideband speech codec (AMR-WB) · IEEE Trans. Speech Audio Process. 2002
Audio and music processing › speech coding
wideband speech coding
0.012002
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
YearPublicationVenuePosition
2002 The adaptive multirate wideband speech codec (AMR-WB)
abstract
This 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.7
2001 Techniques for high-quality ACELP coding of wideband speech
abstract
We 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
INTERSPEECH7
2001 AMR wideband codec - leap in mobile communication voice quality
abstract
The 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
INTERSPEECH3
1998 GSM EFR based multi-rate codec family
abstract
This 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
ICASSP2