WO2009099373A1 - Procédé de transmission de parole - Google Patents

Procédé de transmission de parole Download PDF

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Publication number
WO2009099373A1
WO2009099373A1 PCT/SE2008/051555 SE2008051555W WO2009099373A1 WO 2009099373 A1 WO2009099373 A1 WO 2009099373A1 SE 2008051555 W SE2008051555 W SE 2008051555W WO 2009099373 A1 WO2009099373 A1 WO 2009099373A1
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WO
WIPO (PCT)
Prior art keywords
speech
transmitter
data
rate
transmitting
Prior art date
Application number
PCT/SE2008/051555
Other languages
English (en)
Inventor
Per Synnergren
Daniel ENSTRÖM
Hans Hannu
Original Assignee
Telefonaktiebolaget L M Ericsson (Publ)
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Telefonaktiebolaget L M Ericsson (Publ) filed Critical Telefonaktiebolaget L M Ericsson (Publ)
Publication of WO2009099373A1 publication Critical patent/WO2009099373A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/08—Arrangements for detecting or preventing errors in the information received by repeating transmission, e.g. Verdan system
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm

Definitions

  • the present invention relates to a method and a device for transmitting speech over a circuit switched connection.
  • BACKGROUND Cellular Circuit Switched (CS) telephony was the first service introduced in the first generation of mobile networks. Since then CS telephony has become the largest service in the world.
  • HSPA High Speed Packet Access
  • CPC Continuous Packet Connectivity
  • a originating mobile station connects via HSPA to the base station NodeB.
  • the base station is connected to a Radio Network Controller (RNC) comprising a jitter buffer.
  • RNC Radio Network Controller
  • the RNC is via a Mobile Switching Center (MSC)/Media Gateway (MGW) connected to an RNC of the terminating mobile station.
  • MSC Mobile Switching Center
  • MGW Media Gateway
  • the terminating mobile station is connected to its RNC via a local base station (NodeB).
  • NodeB local base station
  • the mobile station on the terminating side also comprises a jitter buffer.
  • the air interface is using Wideband Code Division Multiple Access (WCDMA) HSPA, which result in that:
  • WCDMA Wideband Code Division Multiple Access
  • the uplink is High Speed Uplink Packet Access (HSUPA) running 2 ms Transmission Time Interval TTI and with Dedicated Physical Control Channel (DPCCH) gating.
  • HSUPA High Speed Uplink Packet Access
  • DPCCH Dedicated Physical Control Channel
  • the downlink is High Speed Downlink Packet Access (HSDPA) and can utilize Fractional Dedicated Physical Channel (F-DPCH) gating and Shared Control Channel for HS-DSCH (HS-SCCH) less operation, where the abbreviation HS-DSCH stands for High Speed Downlink Shared Channel.
  • F-DPCH Fractional Dedicated Physical Channel
  • HS-SCCH Shared Control Channel for HS-DSCH
  • Both uplink and downlink uses Hybrid Automatic Repeat Request H-ARQ to enable fast retransmissions of damaged voice packets.
  • the use of fast retransmissions for robustness, and HSDPA scheduling requires a jitter buffer to cancel the delay variations that can occur due to the H-ARQ retransmissions, and scheduling delay variations.
  • Two jitter buffers are needed, one at the originating RNC and one in the terminating terminal.
  • the jitter buffers use a time stamp that is created by the originating terminal or the terminating RNC to de-jitter the packets.
  • the timestamp will be included in the Packet Data Convergence Protocol (PDCP) header of a special PDCP packet type.
  • PDCP Packet Data Convergence Protocol
  • a PDCP header is depicted in Fig. 2.
  • a method of transmitting speech using a mobile transmitter with a connection to a circuit switched core network via a wireless access is provided.
  • When transmitting speech first speech data is generated.
  • the generated data is then at least partly redundantly transmitted by transmitting the same information more than one time.
  • the robust transmission mode can include application layer redundancy to handle conditions with severe speech frame loss rates.
  • redundancy is obtained by transmitting the same information several times in order to mitigate the problems caused by loss of data units in the transport link. Redundancy can provide better speech quality in links that can introduce loss of speech frames.
  • the invention also extends to a transmitter adapted to transmit speech in accordance with the above.
  • Fig. 1 is a general view of a Circuit switched mobile connection
  • PDCP Packet Data Convergence Protocol
  • - Fig. 3 is a view illustrating redundancy transmission
  • Fig. 4 is a view illustrating speech quality at different transmission rates
  • - Fig. 5 is a general view of a codec
  • - Fig. 6 is a flow chart illustrating procedural steps performed when transmitting speech.
  • redundancy can provide better speech quality in links that can introduce loss of speech frames without increasing the gross transmission rate.
  • Fig.3 an exemplary embodiment for providing redundancy is depicted where the data units Pn, ...,Pn+ 3 are sent twice.
  • Fig. 4 shows the speech quality improvement that can be expected for a few different speech codecs as a function of the packet (data unit) loss percentage.
  • the legend in Fig. 4 marks the speech quality curves without redundancy "RO” while the curves with redundancy are marked "Rl”.
  • redundancy means that the speech frames are sent twice (or more times) the gross bit rate is increased when using redundancy.
  • a speech codec using a lower rate (i.e. the 6.7 kbps AMR mode) with redundancy performs better than a speech codec with higher rate (i.e. the 12.2 kbps AMR mode) without redundancy already at packet loss rates of a few percent.
  • a speech quality perspective it can be better to start use AMR at 6.7 kbps with redundancy giving a AMR gross rate of 13.4 kbps than keep on using AMR 12.2 kbps when starting to encounter losses of speech frames.
  • redundancy can be advantageous to utilize during some transmission conditions.
  • redundancy can be used during conditions suffering from high packet loss rates.
  • redundancy can for example be included by providing a suitable way of signaling that the speech frames are sent as redundant information.
  • CS over HSPA data units in contrast to CS over DCH, includes a PDCP PDU packet type field and an AMR counter field, see Error! Reference source not found.. These fields can be used to signal the use of redundancy.
  • a speech codec 501 of a transmitter is depicted.
  • the speech codec 501 comprises a selector 503 for selecting codec rate and to include redundant transmission based on some predetermined conditions. For example redundant transmission can be activated when the loss rate become higher than some predetermined value. Below some transmission schemes that the speech codec can be adapted to use are described in more detail.
  • Fig. 6 a flow chart illustrating procedural steps performed when transmitting speech using a mobile transmitter with a connection to a circuit switched core network in accordance with the above is shown.
  • speech data is generated.
  • the speech data is transmitted at least partly redundantly by transmitting the same information more than one time.
  • the speech frame and the redundant data of the earlier speech frame are sent in the same transmission.
  • a PDCP PDU type indicating the use of redundant data can be used.
  • a PDCP PDU type is reserved to signal the redundancy.
  • the example shows PDU type 011.
  • a PDCP PDU carrying redundant information can look like the example below.
  • the figure presents transmission #X.
  • the AMR counter represents timing information that is synchronized with the AMR speech frame production rate. This means that in normal transmission, the AMR counter is incremented with 1 every 20 ms. For a redundancy PDCP PDU type of packet the AMR counter value should correspond to the timing when the newest packet of the redundancy packet was produced (i.e. when the Pn packet was produced). Thus transmission X+l can look like:
  • the voice stream may go into a Discontinuous transmission (DTX) state
  • DTX Discontinuous transmission
  • no speech codec frames will be produced until the encoder goes out of the DTX state
  • SID Silence Description
  • later on ordinary SID frames are produced with an interval of 160 ms
  • SID FIRST frame is sent by using a redundant transport mechanism and then only send the ordinary SID frames once
  • the SID FIRST must be sent in the fashion shown below First a Redundancy DATA PDU is sent with the SID first and the previous AMR speech frame Directly after an ordinary PDCP AMR Data PDU is sent with the same AMR counter value as the previous Redundancy DATA PDU
  • a speech frame and the redundant data of an earlier speech frame are sent in two transmissions.
  • the AMR counter value will be the same in two transmissions as shown below.
  • the redundant data can be the same information as the original data, but it can also differ. For instance:
  • the original transmission uses a high codec rate, e.g. AMR12.2, while the redundant transmission uses a low codec rate, e.g. AMR4.75.
  • the original transmission is a complete AMR codec encoded frame, while the redundant data is only parts of the information of the AMR encoded frame. For example selected parameters such as only the A-class bits can be sent redundant. In this case a protocol to inform this to the decoder is needed. An exemplary protocol is depicted below.
  • not all transmissions are transmitted with a redundant transmission scheme. For example only important parts of the speech like speech onsets can be sent redundantly.
  • different codec modes may be used as redundant data for speech onsets compared to stationary voiced speech.
  • the speech quality can be increased in high frame loss rate scenarios.

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

L'invention porte sur un procédé de transmission de parole à l'aide d'un émetteur mobile ayant une connexion à un cœur de réseau à commutation de circuit par l'intermédiaire d'un accès sans fil. Lors d'une transmission de parole, des données de parole sont d'abord générées. Les données générées sont ensuite transmises de façon au moins partiellement redondante par transmission des mêmes informations plus d'une seule fois. Par introduction d'un mode de transmission plus robuste pour des appels à commutation de circuit, une meilleure qualité pour des connexions téléphoniques peut être obtenue. Le mode de transmission robuste peut comprendre une redondance de couche application pour gérer des conditions avec des taux de perte de trame de parole sévères.
PCT/SE2008/051555 2008-02-05 2008-12-23 Procédé de transmission de parole WO2009099373A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US2623408P 2008-02-05 2008-02-05
US61/026,234 2008-02-05

Publications (1)

Publication Number Publication Date
WO2009099373A1 true WO2009099373A1 (fr) 2009-08-13

Family

ID=40504797

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/SE2008/051555 WO2009099373A1 (fr) 2008-02-05 2008-12-23 Procédé de transmission de parole

Country Status (1)

Country Link
WO (1) WO2009099373A1 (fr)

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6167060A (en) * 1997-08-08 2000-12-26 Clarent Corporation Dynamic forward error correction algorithm for internet telephone
WO2003045000A1 (fr) * 2001-11-19 2003-05-30 Motorola, Inc. Procede et appareil de transmission d'informations vocales
WO2006056832A2 (fr) * 2004-10-26 2006-06-01 Nokia Siemens Networks Oy Compensation de perte de paquets
WO2007089183A1 (fr) * 2006-02-03 2007-08-09 Telefonaktiebolaget Lm Ericsson (Publ) Activation de redondance pour la transmission de voix sur internet

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6167060A (en) * 1997-08-08 2000-12-26 Clarent Corporation Dynamic forward error correction algorithm for internet telephone
WO2003045000A1 (fr) * 2001-11-19 2003-05-30 Motorola, Inc. Procede et appareil de transmission d'informations vocales
WO2006056832A2 (fr) * 2004-10-26 2006-06-01 Nokia Siemens Networks Oy Compensation de perte de paquets
WO2007089183A1 (fr) * 2006-02-03 2007-08-09 Telefonaktiebolaget Lm Ericsson (Publ) Activation de redondance pour la transmission de voix sur internet

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
JOHANSSON I ET AL: "Bandwidth efficient amr operation for voip", SPEECH CODING, 2002, IEEE WORKSHOP PROCEEDINGS. OCT. 6-9, 2002, PISCATAWAY, NJ, USA,IEEE, 6 October 2002 (2002-10-06), pages 150 - 152, XP010647243, ISBN: 978-0-7803-7549-9 *

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