draft-ietf-avtext-mixer-to-client-audio-level-01.txt   draft-ietf-avtext-mixer-to-client-audio-level-02.txt 
Network Working Group E. Ivov, Ed. Network Working Group E. Ivov, Ed.
Internet-Draft Jitsi Internet-Draft Jitsi
Intended status: Informational E. Marocco, Ed. Intended status: Informational E. Marocco, Ed.
Expires: September 15, 2011 Telecom Italia Expires: November 10, 2011 Telecom Italia
J. Lennox J. Lennox
Vidyo, Inc. Vidyo, Inc.
March 14, 2011 May 9, 2011
A Real-Time Transport Protocol (RTP) Header Extension for Mixer-to- A Real-Time Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication Client Audio Level Indication
draft-ietf-avtext-mixer-to-client-audio-level-01 draft-ietf-avtext-mixer-to-client-audio-level-02
Abstract Abstract
This document describes a mechanism for RTP-level mixers in audio This document describes a mechanism for RTP-level mixers in audio
conferences to deliver information about the audio level of conferences to deliver information about the audio level of
individual participants. Such audio level indicators are transported individual participants. Such audio level indicators are transported
in the same RTP packets as the audio data they pertain to. in the same RTP packets as the audio data they pertain to.
Status of this Memo Status of this Memo
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
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Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 15, 2011. This Internet-Draft will expire on November 10, 2011.
Copyright Notice Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
skipping to change at page 2, line 16 skipping to change at page 2, line 16
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Protocol Operation . . . . . . . . . . . . . . . . . . . . . . 4 3. Protocol Operation . . . . . . . . . . . . . . . . . . . . . . 4
4. Header Format . . . . . . . . . . . . . . . . . . . . . . . . 6 4. Header Format . . . . . . . . . . . . . . . . . . . . . . . . 6
5. Audio level encoding . . . . . . . . . . . . . . . . . . . . . 6 5. Audio level encoding . . . . . . . . . . . . . . . . . . . . . 6
6. Signaling Information . . . . . . . . . . . . . . . . . . . . 7 6. Signaling Information . . . . . . . . . . . . . . . . . . . . 7
7. Security Considerations . . . . . . . . . . . . . . . . . . . 9 7. Security Considerations . . . . . . . . . . . . . . . . . . . 10
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
9. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 10 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10 10. Changes From Earlier Versions . . . . . . . . . . . . . . . . 11
11. Changes From Earlier Versions . . . . . . . . . . . . . . . . 10 10.1. Changes From Draft -01 . . . . . . . . . . . . . . . . . 11
11.1. Changes From Draft -00 . . . . . . . . . . . . . . . . . 10 10.2. Changes From Draft -00 . . . . . . . . . . . . . . . . . 11
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11 11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
12.1. Normative References . . . . . . . . . . . . . . . . . . 11 11.1. Normative References . . . . . . . . . . . . . . . . . . 11
12.2. Informative References . . . . . . . . . . . . . . . . . 11 11.2. Informative References . . . . . . . . . . . . . . . . . 11
Appendix A. Reference Implementation . . . . . . . . . . . . . . 12 Appendix A. Reference Implementation . . . . . . . . . . . . . . 12
A.1. AudioLevelCalculator.java . . . . . . . . . . . . . . . . 12 A.1. AudioLevelCalculator.java . . . . . . . . . . . . . . . . 13
A.2. AudioLevelRenderer.java . . . . . . . . . . . . . . . . . 14 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction 1. Introduction
The Framework for Conferencing with the Session Initiation Protocol The Framework for Conferencing with the Session Initiation Protocol
(SIP) defined in RFC 4353 [RFC4353] presents an overall architecture (SIP) defined in RFC 4353 [RFC4353] presents an overall architecture
for multi-party conferencing. Among others, the framework borrows for multi-party conferencing. Among others, the framework borrows
from RTP [RFC3550] and extends the concept of a mixer entity from RTP [RFC3550] and extends the concept of a mixer entity
"responsible for combining the media streams that make up a "responsible for combining the media streams that make up a
conference, and generating one or more output streams that are conference, and generating one or more output streams that are
delivered to recipients". Every participant would hence receive, in delivered to recipients". Every participant would hence receive, in
skipping to change at page 5, line 13 skipping to change at page 5, line 13
mechanism to map the CSRC IDs to user identities through the CNAME mechanism to map the CSRC IDs to user identities through the CNAME
field. More advanced mechanisms, may exist depending on the field. More advanced mechanisms, may exist depending on the
signaling protocol used to establish and control a conference. In signaling protocol used to establish and control a conference. In
the case of the Session Initiation Protocol [RFC3261] for example, the case of the Session Initiation Protocol [RFC3261] for example,
the Event Package for Conference State [RFC4575] defines a <src-id> the Event Package for Conference State [RFC4575] defines a <src-id>
tag which binds CSRC IDs to media streams and SIP URIs. tag which binds CSRC IDs to media streams and SIP URIs.
This document describes an RTP header extension that allows mixers to This document describes an RTP header extension that allows mixers to
indicate the audio-level of every conference participant (CSRC) in indicate the audio-level of every conference participant (CSRC) in
addition to simply indicating their on/off status. This new header addition to simply indicating their on/off status. This new header
extension is based on the "General Mechanism for RTP Header extension uses "General Mechanism for RTP Header Extensions"
Extensions" [RFC5285]. described in [RFC5285].
Each instance of this header contains a list of one-octet audio Each instance of this header contains a list of one-octet audio
levels expressed in -dBov, with values from 0 to 127 representing 0 levels expressed in -dBov, with values from 0 to 127 representing 0
to -127 dBov(see Section 4 and Section 5). Appendix A provides a to -127 dBov(see Section 4 and Section 5). Appendix A provides a
reference implementation indicating one way of obtaining such values reference implementation indicating one way of obtaining such values
from raw audio samples. from raw audio samples.
Every audio level value pertains to the CSRC identifier located at Every audio level value pertains to the CSRC identifier located at
the corresponding position in the CSRC list. In other words, the the corresponding position in the CSRC list. In other words, the
first value would indicate the audio level of the conference first value would indicate the audio level of the conference
skipping to change at page 6, line 23 skipping to change at page 6, line 23
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ID | len |0| level 1 |0| level 2 |0| level 3 ... | ID | len |0| level 1 |0| level 2 |0| level 3 ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Audio level indicators extension format Figure 2: Audio level indicators extension format
The 4-bit len field is the number minus one of data bytes (i.e. audio The 4-bit len field is the number minus one of data bytes (i.e. audio
level values) transported in this header extension element following level values) transported in this header extension element following
the one-byte header. Therefore, the value zero in this field the one-byte header. Therefore, the value zero in this field
indicates that one byte of data follows. A value of 15 is not indicates that one byte of data follows. RFC 3550 [RFC3550] only
allowed by this specification and it MUST NOT be used as the RTP allows RTP packets to carry a maximum of 15 CSRC IDs. Given that
header can carry a maximum of 15 CSRC IDs. The maximum value allowed audio levels directly refer to CSRC IDs, implementations MUST NOT
is therefore 14 indicating a following sequence of 15 audio level include more than 15 audio level values. The maximum value allowed
values. in the len field is therefore 14.
Note that use of the two-byte header defined in RFC 5285 [RFC5285] Note that use of the two-byte header defined in RFC 5285 [RFC5285]
follows the same rules the only change being the length of the ID and follows the same rules the only change being the length of the ID and
len fields. len fields.
5. Audio level encoding 5. Audio level encoding
Audio level indicators are encoded in the same manner as audio noise The audio level header extension only carries the level of the audio
level in the RTP Payload Comfort Noise specification [RFC3389] and in the RTP payload of the packet it is associated with. This
audio level in the RTP Extension Header for Client-to-mixer Audio information is carried in an RTP header extension element as defined
Level Notification [I-D.ietf-avtext-client-to-mixer-audio-level] by [RFC5285].
specification. The magnitude of the audio level is packed into the
least significant bits of one audio-level byte with the most The audio level is defined in the same manner as is audio noise level
significant bit unused and always set to 0 as shown below in in the RTP Payload Comfort Noise specification [RFC3389]. The
Figure 3. overall magnitude of the noise level is encoded into the first byte
of the payload, with spectral information about the noise in
subsequent bytes. This specification's audio level parameter is
defined so as to be identical to the comfort noise payload's noise-
level byte.
The magnitude of the audio level is packed into the seven least
significant bits of the single byte of the header extension, shown in
Figure 3. The least significant bit of the audio level magnitude is
packed into the least significant bit of the byte. The most
significant bit of the byte is unused and always set to 0 as shown
below in Figure 3.
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+
|0| level | |0| level |
+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+
Figure 3: Audio Level Encoding Figure 3: Audio Level Encoding
The two-byte header defined in RFC 5285 [RFC5285] may also be used.
The audio level is expressed in -dBov, with values from 0 to 127 The audio level is expressed in -dBov, with values from 0 to 127
representing 0 to -127 dBov. dBov is the level, in decibels, relative representing 0 to -127 dBov. dBov is the level, in decibels, relative
to the overload point of the system, i.e. the maximum-amplitude to the overload point of the system, i.e. the maximum-amplitude
signal that can be handled by the system without clipping. (Note: signal that can be handled by the system without clipping.(Note:
Representation relative to the overload point of a system is Representation relative to the overload point of a system is
particularly useful for digital implementations, since one does not particularly useful for digital implementations, since one does not
need to know the relative calibration of the analog circuitry.) For need to know the relative calibration of the analog circuitry.) For
example, in the case of u-law (audio/pcmu) audio [ITU.G.711], the 0 example, in the case of u-law (audio/pcmu) audio [ITU.G.711], the 0
dBov reference would be a square wave with values +/- 8031. (This dBov reference would be a square wave with values +/- 8031. (This
translates to 6.18 dBm0, relative to u-law's dBm0 definition in Table translates to 6.18 dBm0, relative to u-law's dBm0 definition in Table
6 of G.711.) 6 of G.711.)
The audio level for digital silence, for example for a muted audio
source, MAY be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
Implementations MAY choose to measure audio levels prior to encoding
them in the payload carried in the RTP payload, e.g. on raw linear
PCM input.
The audio level header extension only carries the level of the audio
in the RTP payload of the packet it is associated with, with no long-
term averaging or smoothing applied.
To simplify implementation of the encoding procedures described here, To simplify implementation of the encoding procedures described here,
this specification provides a sample Java implementation (Appendix A) this specification provides a sample Java implementation (Appendix A)
demonstating one way it can be achieved. of an audio level calculator that helps obtain such values from raw
linear PCM audio samples.
6. Signaling Information 6. Signaling Information
The URI for declaring the audio level header extension in an SDP The URI for declaring the audio level header extension in an SDP
extmap attribute and mapping it to a local extension header extmap attribute and mapping it to a local extension header
identifier is "urn:ietf:params:rtp-hdrext:csrc-audio-level". There identifier is "urn:ietf:params:rtp-hdrext:csrc-audio-level". There
is no additional setup information needed for this extension (i.e. no is no additional setup information needed for this extension (i.e. no
extensionattributes). extensionattributes).
An example attribute line in the SDP, for a conference might be: An example attribute line in the SDP, for a conference might be:
skipping to change at page 7, line 50 skipping to change at page 8, line 23
The above mapping will most often be provided per media stream (in The above mapping will most often be provided per media stream (in
the media-level section(s) of SDP, i.e., after an "m=" line) or the media-level section(s) of SDP, i.e., after an "m=" line) or
globally if there is more than one stream containing audio level globally if there is more than one stream containing audio level
indicators in a session. indicators in a session.
Presence of the above attribute in the SDP description of a media Presence of the above attribute in the SDP description of a media
stream indicates that some or all RTP packets in that stream would stream indicates that some or all RTP packets in that stream would
contain the audio level information RTP extension header. contain the audio level information RTP extension header.
Conferencing clients that support audio level indicators and have no Conferencing clients that support audio level indicators and have no
mixing capabilities SHOULD always include the direction parameter in mixing capabilities would not be able to content for this audio level
the "extmap" attribute setting it to "recvonly". Conference focus extension and would hence have to always include the direction
entities with mixing capabilities MAY omit the direction or set it to parameter in the "extmap" attribute with a value of "recvonly".
"sendrecv" in SDP offers. Such entities SHOULD set it to "sendonly" Conference focus entities with mixing capabilities can omit the
in SDP answers to offers with a "recvonly" parameter and to direction or set it to "sendrecv" in SDP offers. Such entities would
"sendrecv" when answering other "sendrecv" offers. need to set it to "sendonly" in SDP answers to offers with a
"recvonly" parameter and to "sendrecv" when answering other
"sendrecv" offers.
This speicification does not define use of the audio level extensions
in video streams. Therefore, the extension defined in this document
SHOULD NOT be advertised in anything but audio streams.
The following Figure 4 and Figure 5 show two example offer/answer The following Figure 4 and Figure 5 show two example offer/answer
exchanges between a conferencing client and a focus, and between two exchanges between a conferencing client and a focus, and between two
conference focus entities. conference focus entities.
v=0 v=0
o=alice 2890844526 2890844526 IN IP6 host.example.com o=alice 2890844526 2890844526 IN IP6 host.example.com
c=IN IP6 host.example.com c=IN IP6 host.example.com
t=0 0 t=0 0
m=audio 49170 RTP/AVP 0 4 m=audio 49170 RTP/AVP 0 4
skipping to change at page 9, line 25 skipping to change at page 9, line 48
i=A Seminar on the session description protocol i=A Seminar on the session description protocol
o=us-focus 2890844526 2890844526 IN IP6 focus.us.example.net o=us-focus 2890844526 2890844526 IN IP6 focus.us.example.net
c=IN IP6 focus.us.example.net c=IN IP6 focus.us.example.net
t=0 0 t=0 0
m=audio 52543 RTP/AVP 0 m=audio 52543 RTP/AVP 0
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level
An example SDP offer/answer exchange between two conference focus An example SDP offer/answer exchange between two conference focus
entities with mixing capabilities negotiating an audio stream with entities with mixing capabilities negotiating an audio stream with
bidirectional flwo of audio level information. bidirectional flow of audio level information.
Figure 5 Figure 5
7. Security Considerations 7. Security Considerations
1. This document defines a means of attributing audio level to a 1. This document defines a means of attributing audio level to a
particular participant in a conference. An attacker may try to particular participant in a conference. An attacker may try to
modify the content of RTP packets in a way that would make audio modify the content of RTP packets in a way that would make audio
activity from one participant appear as coming from another. activity from one participant appear as coming from another.
2. Furthermore, the fact that audio level values would not be 2. Furthermore, the fact that audio level values would not be
protected even in an SRTP session may be of concern in some cases protected even in an SRTP session might be of concern in some
where the activity of a particular participant in a conference is cases where the activity of a particular participant in a
confidential. conference is confidential.
3. Both of the above are concerns that stem from the design of the 3. Both of the above are concerns that stem from the design of the
RTP protocol itself and they would probably also apply when using RTP protocol itself and they would probably also apply when using
CSRC identifiers the way they were specified in RFC 3550 CSRC identifiers the way they were specified in RFC 3550
[RFC3550]. It is therefore important that according to the needs [RFC3550]. It is therefore important that according to the needs
of a particular scenario, implementors and deployers consider use of a particular scenario, implementors and deployers consider use
of a lower level security and authentication mechanism. of header extension encryption
[I-D.lennox-avtcore-srtp-encrypted-header-ext] or a lower level
security and authentication mechanism.
8. IANA Considerations 8. IANA Considerations
This document defines a new extension URI that, if approved, would This document defines a new extension URI that, if approved, would
need to be added to the RTP Compact Header Extensions sub-registry of need to be added to the RTP Compact Header Extensions sub-registry of
the Real-Time Transport Protocol (RTP) Parameters registry, according the Real-Time Transport Protocol (RTP) Parameters registry, according
to the following data: to the following data:
Extension URI: urn:ietf:params:rtp-hdrext:csrc-audio-level Extension URI: urn:ietf:params:rtp-hdrext:csrc-audio-level
Description: Mixer-to-client audio level indicators Description: Mixer-to-client audio level indicators
Contact: emcho@jitsi.org Contact: emcho@jitsi.org
Reference: RFC XXXX Reference: RFC XXXX
9. Open Issues Note to the RFC-Editor: please replace "RFC XXXX" by the number of
this RFC.
At the time of writing of this document the authors have no clear
view on how and if the following list of issues should be address
here:
1. Audio levels in video streams. This specification allows use of
audio level values in "silent" audio streams that don't otherwise
carry any payload thus allowing their delivery within systems
where the various focus/mixer components communicate with each
other as conference participants. The same train of thought may
very well justify audio level transport in video streams.
2. It has been suggested to reference ITU P.56 [ITU.P56.1993] for
level measurement. This needs to be investigated.
10. Acknowledgments 9. Acknowledgments
Lyubomir Marinov contributed level measurement and rendering code. Lyubomir Marinov contributed level measurement and rendering code.
Roni Even, Ingemar Johansson, Michael Ramalho and several others Roni Even, Keith Drage, Ingemar Johansson, Michael Ramalho and
provided helpful feedback over the dispatch mailing list. several others provided helpful feedback over the dispatch mailing
list.
Jitsi's participation in this specification is funded by the NLnet Jitsi's participation in this specification is funded by the NLnet
Foundation. Foundation.
11. Changes From Earlier Versions 10. Changes From Earlier Versions
Note to the RFC-Editor: please remove this section prior to Note to the RFC-Editor: please remove this section prior to
publication as an RFC. publication as an RFC.
11.1. Changes From Draft -00 10.1. Changes From Draft -01
o Removed code related the AudioLevelRenderer from "APPENDIX A.
Reference Implementation" as it was considered an implementation
matter by the working group.
o Modified the AudioLevelCalculator in "APPENDIX A. Reference
Implementation" to take overload as a parameter.
o Clarified non-use of audio levels in video streams
o Closed the P.56 open issue. It was agreed on IETF 80 that P.56 is
mostly about speech levels and the levels transported by the
extension defined here should also be able to serve as an
indication for noise.
o The Open Issues section has been removed as all issues that were
in there are now resolved or clarified.
o Editorial changes for consistency with
[I-D.ietf-avtext-client-to-mixer-audio-level].
10.2. Changes From Draft -00
o Added code for sound pressure calculation and measurement in o Added code for sound pressure calculation and measurement in
"APPENDIX A. Reference Implementation". "APPENDIX A. Reference Implementation".
o Changed affiliation for Emil Ivov. o Changed affiliation for Emil Ivov.
o Removed "Appendix: Design choices". o Removed "Appendix: Design choices".
12. References 11. References
12.1. Normative References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008. Header Extensions", RFC 5285, July 2008.
12.2. Informative References 11.2. Informative References
[I-D.ietf-avtext-client-to-mixer-audio-level] [I-D.ietf-avtext-client-to-mixer-audio-level]
Lennox, J., Ivov, E., and E. Marocco, "A Real-Time Lennox, J., Ivov, E., and E. Marocco, "A Real-Time
Transport Protocol (RTP) Header Extension for Client-to- Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", Mixer Audio Level Indication",
draft-ietf-avtext-client-to-mixer-audio-level-00 (work in draft-ietf-avtext-client-to-mixer-audio-level-01 (work in
progress), February 2011. progress), March 2011.
[I-D.lennox-avtcore-srtp-encrypted-header-ext]
Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)",
draft-lennox-avtcore-srtp-encrypted-header-ext-00 (work in
progress), March 2011.
[ITU.G.711] [ITU.G.711]
International Telecommunications Union, "Pulse Code International Telecommunications Union, "Pulse Code
Modulation (PCM) of Voice Frequencies", ITU- Modulation (PCM) of Voice Frequencies", ITU-
T Recommendation G.711, November 1988. T Recommendation G.711, November 1988.
[ITU.P56.1993] [ITU.P56.1993]
International Telecommunications Union, "Objective International Telecommunications Union, "Objective
Measurement of Active Speech Level", ITU-T Recommendation Measurement of Active Speech Level", ITU-T Recommendation
P.56, March 1988. P.56, March 1988.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E. A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261, Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002. June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002. Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3920] Saint-Andre, P., Ed., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 3920, October 2004.
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the [RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353, Session Initiation Protocol (SIP)", RFC 4353,
February 2006. February 2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006. State", RFC 4575, August 2006.
Appendix A. Reference Implementation Appendix A. Reference Implementation
This appendix contains Java code for a reference implementation of This appendix contains Java code for a reference implementation of
the level calculation and rendering methods.The code is not normative the level calculation and rendering methods.The code is not normative
and by no means the only possible implementation. Its purpose is to and by no means the only possible implementation. Its purpose is to
help implementors add audio level support to mixers and clients. help implementors add audio level support to mixers and clients.
The Java code consists of the following files and methods: The Java code contains an AudioLevelCalculator class that calculates
the sound pressure level of a signal with specific samples. It can
AudioLevelCalculator.java: Calculates the sound pressure level of a be used in mixers to generate values suitable for the level extension
signal with specific samples. Can be used in mixers to generate headers.
values suitable for the level extension headers.
AudioLevelRenderer.java: Helps adjust a sequence of pressure levels
so that they would appear "natural" to users. Can be used by
clients and applied over the values received in a level extension
header so that displayed levels would change smoothly and
correspond to user experience.
The implementation is provided in Java but does not rely on any of The implementation is provided in Java but does not rely on any of
the language specific and can be easily ported to another. the language specific and can be easily ported to another.
A.1. AudioLevelCalculator.java A.1. AudioLevelCalculator.java
/** /**
* Calculates the audio level of specific samples of a singal based on * Calculates the audio level of specific samples of a singal based on
* sound pressure level. * sound pressure level.
*/ */
skipping to change at page 12, line 48 skipping to change at page 13, line 26
{ {
/** /**
* Calculates the sound pressure level of a signal with specific * Calculates the sound pressure level of a signal with specific
* <tt>samples</tt>. * <tt>samples</tt>.
* *
* @param samples the samples of the signal to calculate the sound * @param samples the samples of the signal to calculate the sound
* pressure level of. The samples are specified as an <tt>int</tt> * pressure level of. The samples are specified as an <tt>int</tt>
* array starting at <tt>offset</tt>, extending <tt>length</tt> * array starting at <tt>offset</tt>, extending <tt>length</tt>
* number of elements and each <tt>int</tt> element in the specified * number of elements and each <tt>int</tt> element in the specified
* range representing a 16-bit sample. * range representing a sample of the signal to calculate the sound
* pressure level of. Though a sample is provided in the form of an
* <tt>int</tt> value, the sample size in bits is determined by the
* caller via <tt>overload</tt>.
* *
* @param offset the offset in <tt>samples</tt> at which the samples * @param offset the offset in <tt>samples</tt> at which the samples
* start * start
*
* @param length the length of the signal specified in * @param length the length of the signal specified in
* <tt>samples<tt> starting at <tt>offset</tt> * <tt>samples<tt> starting at <tt>offset</tt>
*
* @param overload the overload (point) of <tt>signal</tt>.
* For example, <tt>overload</tt> may be {@link Byte#MAX_VALUE}
* for 8-bit signed samples or {@link Short#MAX_VALUE} for
* 16-bit signed samples.
*
* @return the sound pressure level of the specified signal * @return the sound pressure level of the specified signal
*/ */
public static int calculateSoundPressureLevel( public static int calculateSoundPressureLevel(
int[] samples, int offset, int length) int[] samples, int offset, int length,
int overload)
{ {
/* /*
* Calcuate the root mean square of the signal i.e. the * Calcuate the root mean square of the signal i.e. the
* effective sound pressure. * effective sound pressure.
*/ */
double rms = 0; double rms = 0;
for (; offset < length; offset++) for (; offset < length; offset++)
{ {
double sample = samples[offset]; double sample = samples[offset];
sample /= Short.MAX_VALUE; sample /= overload;
rms += sample * sample; rms += sample * sample;
} }
rms = (length == 0) ? 0 : Math.sqrt(rms / length); rms = (length == 0) ? 0 : Math.sqrt(rms / length);
/* /*
* The sound pressure level is a logarithmic measure of the * The sound pressure level is a logarithmic measure of the
* effectivesound pressure of a sound relative to a reference * effectivesound pressure of a sound relative to a reference
* value and is measured in decibels. * value and is measured in decibels.
*/ */
double db; double db;
skipping to change at page 14, line 30 skipping to change at page 15, line 19
{ {
db = MIN_SOUND_PRESSURE_LEVEL; db = MIN_SOUND_PRESSURE_LEVEL;
} }
return (int) db; return (int) db;
} }
} }
AudioLevelCalculator.java AudioLevelCalculator.java
A.2. AudioLevelRenderer.java
/**
* Helps adjust a sequence of pressure levels so that they would appear
* "natural" to users. Can be used by clients and applied over the
* values received in a level extension header so that displayed levels
* would change smoothly and correspond to user experience..
*/
public class AudioLevelRenderer
{
/**
* The last audio level displayed by
* {@link AudioLevelCalculator#displayAudioLevel(int, int, int)}.
*/
private int lastAudioLevel = 0;
/**
* Returns a specific sound pressure level as an animated (i.e.
* does not jump up and down too much in a single update) audio
* level.
*
* @param spl the sound pressure level to be displayed
* @param minAudioLevel the minimum of the UI range which is used
* to depict audio levels
* @param maxAudioLevel the maximum of the UI range which is used
* to depict audio levels
* @return a sound pressure level that can be displayed to the user.
*/
public int renderAudioLevel(
int spl, int minAudioLevel, int maxAudioLevel)
{
/*
* The minimum sound pressure level that the UI is interested in
* displaying.
*/
final double MIN_SPL_TO_DISPLAY = 40 /* A WHISPER */;
/*
* The maximum sound pressure level that the UI is interested in
* displaying.
*/
final double MAX_SPL_TO_DISPLAY = 85 /* HEARING DAMAGE */;
int audioLevel;
if (spl < MIN_SPL_TO_DISPLAY)
audioLevel = minAudioLevel;
else if (spl > MAX_SPL_TO_DISPLAY)
audioLevel = maxAudioLevel;
else
{
/*
* Depict the range between "A WHISPER" and the beginning of
* "HEARING DAMAGE".
*/
audioLevel
= (int)
(((spl - MIN_SPL_TO_DISPLAY)
/ (MAX_SPL_TO_DISPLAY - MIN_SPL_TO_DISPLAY))
* (maxAudioLevel - minAudioLevel));
if (audioLevel < minAudioLevel)
audioLevel = minAudioLevel;
else if (audioLevel > maxAudioLevel)
audioLevel = maxAudioLevel;
}
/*
* Animate the audio level so that it does not jump up and down
* too fast.
*/
lastAudioLevel
= (int) (lastAudioLevel * 0.8 + audioLevel * 0.2);
/* Return the displayable audio level. */
return lastAudioLevel;
}
}
AudioLevelRenderer.java
Authors' Addresses Authors' Addresses
Emil Ivov (editor) Emil Ivov (editor)
Jitsi Jitsi
Strasbourg 67000 Strasbourg 67000
France France
Email: emcho@jitsi.org Email: emcho@jitsi.org
Enrico Marocco (editor) Enrico Marocco (editor)
 End of changes. 34 change blocks. 
168 lines changed or deleted 133 lines changed or added

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