Network Working Group E. Ivov, Ed.
Internet-Draft Jitsi
Intended status: Standards Track E. Marocco, Ed.
Expires: February 28, 2012 Telecom Italia
J. Lennox
Vidyo, Inc.
August 27, 2011
A Real-Time Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication
draft-ietf-avtext-mixer-to-client-audio-level-04
Abstract
This document describes a mechanism for RTP-level mixers in audio
conferences to deliver information about the audio level of
individual participants. Such audio level indicators are transported
in the same RTP packets as the audio data they pertain to.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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and may be updated, replaced, or obsoleted by other documents at any
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on February 28, 2012.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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publication of this document. Please review these documents
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Protocol Operation . . . . . . . . . . . . . . . . . . . . . . 4
4. Audio Levels . . . . . . . . . . . . . . . . . . . . . . . . . 5
5. Signaling Information . . . . . . . . . . . . . . . . . . . . 7
6. Security Considerations . . . . . . . . . . . . . . . . . . . 10
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 10
9. Changes From Earlier Versions . . . . . . . . . . . . . . . . 11
9.1. Changes From Draft -03 . . . . . . . . . . . . . . . . . . 11
9.2. Changes From Draft -02 . . . . . . . . . . . . . . . . . . 11
9.3. Changes From Draft -01 . . . . . . . . . . . . . . . . . . 11
9.4. Changes From Draft -00 . . . . . . . . . . . . . . . . . . 11
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
10.1. Normative References . . . . . . . . . . . . . . . . . . . 12
10.2. Informative References . . . . . . . . . . . . . . . . . . 12
Appendix A. Reference Implementation . . . . . . . . . . . . . . 13
A.1. AudioLevelCalculator.java . . . . . . . . . . . . . . . . 13
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15
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1. Introduction
The Framework for Conferencing with the Session Initiation Protocol
(SIP) defined in RFC 4353 [RFC4353] presents an overall architecture
for multi-party conferencing. Among others, the framework borrows
from RTP [RFC3550] and extends the concept of a mixer entity
"responsible for combining the media streams that make up a
conference, and generating one or more output streams that are
delivered to recipients". Every participant would hence receive, in
a flat single stream, media originating from all the others.
Using such centralized mixer-based architectures simplifies support
for conference calls on the client side since they would hardly
differ from one-to-one conversations. However, the method also
introduces a few limitations. The flat nature of the streams that a
mixer would output and send to participants makes it difficult for
users to identify the original source of what they are hearing.
Mechanisms that allow the mixer to send to participants cues on
current speakers (e.g. the CSRC fields in RTP [RFC3550]) only work
for speaking/silent binary indications. There are, however, a number
of use cases where one would require more detailed information.
Possible examples include the presence of background chat/noise/
music/typing, someone breathing noisily in their microphone, or other
cases where identifying the source of the disturbance would make it
easy to remove it (e.g. by sending a private IM to the concerned
party asking them to mute their microphone). A more advanced
scenario could involve an intense discussion between multiple
participants that the user does not personally know. Audio level
information would help better recognize the speakers by associating
with them complex (but still human readable) characteristics like
loudness and speed for example.
One way of presenting such information in a user friendly manner
would be for a conferencing client to attach audio level indicators
to the corresponding participant related components in the user
interface as displayed in Figure 1.
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________________________
| |
| 00:42 | Weekly Call |
|________________________|
| |
| |
| Alice |====== | (S) |
| |
| Bob |= | |
| |
| Carol | | (M) |
| |
| Dave |=== | |
| |
|________________________|
Figure 1: Displaying detailed speaker information to the user by
including audio level for every participant.
Implementing a user interface like the above requires analysis of the
media sent from other participants. In a conventional audio
conference this is only possible for the mixer since all other
conference participants are generally receiving a single, flat audio
stream and have therefore no immediate way of determining individual
audio levels.
This document specifies an RTP extension header that allows such
mixers to deliver audio level information to conference participants
by including it directly in the RTP packets transporting the
corresponding audio data.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. Protocol Operation
According to RFC 3550 [RFC3550] a mixer is expected to include in
outgoing RTP packets a list of identifiers (CSRC IDs) indicating the
sources that contributed to the resulting stream. The presence of
such CSRC IDs allows RTP clients to determine, in a binary way, the
active speaker(s) in any given moment. RTCP also provides a basic
mechanism to map the CSRC IDs to user identities through the CNAME
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field. More advanced mechanisms can exist depending on the signaling
protocol used to establish and control a conference. In the case of
the Session Initiation Protocol [RFC3261] for example, the Event
Package for Conference State [RFC4575] defines a <src-id> tag which
binds CSRC IDs to media streams and SIP URIs.
This document describes an RTP header extension that allows mixers to
indicate the audio-level of every contributing conference participant
(CSRC) in addition to simply indicating their on/off status. This
new header extension uses "General Mechanism for RTP Header
Extensions" described in [RFC5285].
Each instance of this header contains a list of one-octet audio
levels expressed in -dBov, with values from 0 to 127 representing 0
to -127 dBov(see Figure 2 and Figure 3). Appendix A provides a
reference implementation indicating one way of obtaining such values
from raw audio samples.
Every audio level value pertains to the CSRC identifier located at
the corresponding position in the CSRC list. In other words, the
first value would indicate the audio level of the conference
participant represented by the first CSRC identifier in that packet
and so forth. The number and order of these values MUST therefore
match the number and order of the CSRC IDs present in the same
packet.
When encoding audio level information, a mixer SHOULD include in a
packet information that corresponds to the audio data being
transported in that same packet. It is important that these values
follow the actual stream as closely as possible. Therefore a mixer
SHOULD also calculate the values after the original contributing
stream has undergone possible processing such as level normalization,
and noise reduction for example.
It can sometimes happen that a conference involves more than a single
mixer. In such cases each of the mixers MAY choose to relay the CSRC
list and audio-level information they receive from peer mixers (as
long as the total CSRC count remains below 16). Given that the
maximum audio level is not precisely defined by this specification,
it is likely that in such situations average audio levels would be
perceptibly different for the participants located behind the
different mixers.
4. Audio Levels
The audio level header extension carries the level of the audio in
the RTP payload of the packet it is associated with. This
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information is carried in an RTP header extension element as defined
by the "General Mechanism for RTP Header Extensions" [RFC5285].
The payload of the audio level header extension element can be
encoded using the one-byte or the two-byte header defined in
[RFC5285]. Figure 2 and Figure 3 show sample audio level encodings
with each of them.
0 1 2 3
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=2 |0| level 1 |0| level 2 |0| level 3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Sample audio level encoding using the one-byte header format
Figure 2
0 1 2 3
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=3 |0| level 1 |0| level 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| level 3 | 0 (pad) | ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Sample audio level encoding using the two-byte header format
Figure 3
In the case of the one-byte header format, 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 the one-byte header.
Therefore, the value zero in this field indicates that one byte of
data follows. In the case of the two-byte header format the 8-bit
len field contains the exact number of audio levels carried in the
extension. RFC 3550 [RFC3550] only allows RTP packets to carry a
maximum of 15 CSRC IDs. Given that audio levels directly refer to
CSRC IDs, implementations MUST NOT include more than 15 audio level
values. The maximum value allowed in the len field is therefore 14
for one-byte header format and 15 for two-byte header format.
Audio levels in this document are defined in the same manner as is
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audio noise level in the RTP Payload Comfort Noise specification
[RFC3389]. In the comfort noise specification, the overall magnitude
of the noise level in comfort noise 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 itself is packed into the seven
least significant bits of the single byte of the header extension,
shown in Figure 2 and 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.
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
to the overload point of the system, i.e. the maximum-amplitude
signal that can be handled by the system without clipping. (Note:
Representation relative to the overload point of a system is
particularly useful for digital implementations, since one does not
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
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
6 of G.711.)
The audio level for digital silence, for example for a muted audio
source, MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
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. That level is measured as a
root mean square of all the samples in the measured range.
To simplify implementation of the encoding procedures described here,
this specification provides a sample Java implementation (Appendix A)
of an audio level calculator that helps obtain such values from raw
linear PCM audio samples.
5. Signaling Information
The URI for declaring the audio level header extension in an SDP
extmap attribute and mapping it to a local extension header
identifier is "urn:ietf:params:rtp-hdrext:csrc-audio-level". There
is no additional setup information needed for this extension (i.e. no
extensionattributes).
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An example attribute line in the SDP, for a conference might be:
a=extmap:7 urn:ietf:params:rtp-hdrext:csrc-audio-level
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
globally if there is more than one stream containing audio level
indicators in a session.
Presence of the above attribute in the SDP description of a media
stream indicates that RTP packets in that stream, which contain the
level extension defined in this document, will be carrying them with
an ID of 7.
Conferencing clients that support audio level indicators and have no
mixing capabilities would not be able to provide content for this
audio level extension and would hence have to always include the
direction parameter in the "extmap" attribute with a value of
"recvonly". Conference focus entities with mixing capabilities can
omit the direction or set it to "sendrecv" in SDP offers. Such
entities would need to set it to "sendonly" in SDP answers to offers
with a "recvonly" parameter and to "sendrecv" when answering other
"sendrecv" offers.
This specification only defines use of the audio level extensions in
audio streams. They MUST NOT be advertised with other media types
such as video or text for example.
The following Figure 4 and Figure 5 show two example offer/answer
exchanges between a conferencing client and a focus, and between two
conference focus entities.
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v=0
o=alice 2890844526 2890844526 IN IP6 host.example.com
c=IN IP6 host.example.com
t=0 0
m=audio 49170 RTP/AVP 0 4
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=extmap:1/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
v=0
i=A Seminar on the session description protocol
o=conf-focus 2890844730 2890844730 IN IP6 focus.example.net
c=IN IP6 focus.example.net
t=0 0
m=audio 52543 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:csrc-audio-level
A client-initiated example SDP offer/answer exchange negotiating an
audio stream with one-way flow of of audio level information.
Figure 4
v=0
i=Un seminaire sur le protocole de description des sessions
o=fr-focus 2890844730 2890844730 IN IP6 focus.fr.example.net
c=IN IP6 focus.fr.example.net
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level
v=0
i=A Seminar on the session description protocol
o=us-focus 2890844526 2890844526 IN IP6 focus.us.example.net
c=IN IP6 focus.us.example.net
t=0 0
m=audio 52543 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=extmap:1/sendrecv urn:ietf:params:rtp-hdrext:csrc-audio-level
An example SDP offer/answer exchange between two conference focus
entities with mixing capabilities negotiating an audio stream with
bidirectional flow of audio level information.
Figure 5
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6. Security Considerations
1. This document defines a means of attributing audio level to a
particular participant in a conference. An attacker may try to
modify the content of RTP packets in a way that would make audio
activity from one participant appear as coming from another.
2. Furthermore, the fact that audio level values would not be
protected even in an SRTP session might be of concern in some
cases where the activity of a particular participant in a
conference is confidential. Also, as discussed in
[I-D.perkins-avt-srtp-vbr-audio], an attacker might be able to
infer information about the conversation, possibly with phoneme-
level resolution.
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
CSRC identifiers the way they were specified in RFC 3550
[RFC3550]. It is therefore important that according to the needs
of a particular scenario, implementors and deployers consider use
of header extension encryption
[I-D.ietf-avtcore-srtp-encrypted-header-ext] or a lower level
security and authentication mechanism.
7. IANA Considerations
This document defines a new extension URI that, if approved, would
need to be added to the RTP Compact Header Extensions sub-registry of
the Real-Time Transport Protocol (RTP) Parameters registry, according
to the following data:
Extension URI: urn:ietf:params:rtp-hdrext:csrc-audio-level
Description: Mixer-to-client audio level indicators
Contact: emcho@jitsi.org
Reference: RFC XXXX
Note to the RFC-Editor: please replace "RFC XXXX" by the number of
this RFC.
8. Acknowledgments
Lyubomir Marinov contributed level measurement and rendering code.
Keith Drage, Roni Even, John Elwell, Kevin P. Fleming, Ingemar
Johansson, Michael Ramalho, Magnus Westerlund and several others
provided helpful feedback over the avt and avtext mailing lists.
Jitsi's participation in this specification is funded by the NLnet
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Foundation.
9. Changes From Earlier Versions
Note to the RFC-Editor: please remove this section prior to
publication as an RFC.
9.1. Changes From Draft -03
o Addressed editorial comments made on the mailing list.
9.2. Changes From Draft -02
o Removed the no-data use case that allowed sending levels in RTP
packets. Choosing the right RTP payload type for this use case
would have incurred complexity without bringing any real value.
o Merged the "Header Format" and the "Audio level encoding" sections
into a single "Audio Levels" section.
o Changed encoding related text so that it would cover both the one-
byte and the two-byte header formats.
o Clarified use of root mean square for dBov calculation
o Added a reference to [I-D.perkins-avt-srtp-vbr-audio] to better
explain some "Security Considerations" .
o Other minor editorial changes.
9.3. 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].
9.4. Changes From Draft -00
o Added code for sound pressure calculation and measurement in
"APPENDIX A. Reference Implementation".
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o Changed affiliation for Emil Ivov.
o Removed "Appendix: Design choices".
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
10.2. Informative References
[]
Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)",
draft-ietf-avtcore-srtp-encrypted-header-ext-00 (work in
progress), June 2011.
[I-D.ietf-avtext-client-to-mixer-audio-level]
Lennox, J., Ivov, E., and E. Marocco, "A Real-Time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication",
draft-ietf-avtext-client-to-mixer-audio-level-03 (work in
progress), July 2011.
[I-D.perkins-avt-srtp-vbr-audio]
Perkins, C. and J. Valin, "Guidelines for the use of
Variable Bit Rate Audio with Secure RTP",
draft-perkins-avt-srtp-vbr-audio-05 (work in progress),
December 2010.
[ITU.G.711]
International Telecommunications Union, "Pulse Code
Modulation (PCM) of Voice Frequencies", ITU-
T Recommendation G.711, November 1988.
[ITU.P56.1993]
International Telecommunications Union, "Objective
Measurement of Active Speech Level", ITU-T Recommendation
P.56, March 1988.
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[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353,
February 2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
Initiation Protocol (SIP) Event Package for Conference
State", RFC 4575, August 2006.
Appendix A. Reference Implementation
This appendix contains Java code for a reference implementation of
the level calculation and rendering methods.The code is not normative
and by no means the only possible implementation. Its purpose is to
help implementors add audio level support to mixers and clients.
The Java code contains an AudioLevelCalculator class that calculates
the sound pressure level of a signal with specific samples. It can
be used in mixers to generate values suitable for the level extension
headers.
The implementation is provided in Java but does not rely on any of
the language specific and can be easily ported to another.
A.1. AudioLevelCalculator.java
/**
* Calculates the audio level of specific samples of a signal based on
* sound pressure level.
*/
public class AudioLevelCalculator
{
/**
* Calculates the sound pressure level of a signal with specific
* <tt>samples</tt>.
*
* @param samples the samples of the signal to calculate the sound
* pressure level of. The samples are specified as an <tt>int</tt>
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* array starting at <tt>offset</tt>, extending <tt>length</tt>
* number of elements and each <tt>int</tt> element in the specified
* 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
* start
*
* @param length the length of the signal specified in
* <tt>samples<tt> starting at <tt>offset</tt>
*
* @param overload the overload (point) of <tt>signal</tt>.
* For example, <tt>overload</tt> can 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
*/
public static int calculateSoundPressureLevel(
int[] samples, int offset, int length,
int overload)
{
/*
* Calcuate the root mean square of the signal i.e. the
* effective sound pressure.
*/
double rms = 0;
for (; offset < length; offset++)
{
double sample = samples[offset];
sample /= overload;
rms += sample * sample;
}
rms = (length == 0) ? 0 : Math.sqrt(rms / length);
/*
* The sound pressure level is a logarithmic measure of the
* effectivesound pressure of a sound relative to a reference
* value and is measured in decibels.
*/
double db;
/*
* The minimum sound pressure level which matches the maximum
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* of the sound meter.
*/
final double MIN_SOUND_PRESSURE_LEVEL = 0;
/*
* The maximum sound pressure level which matches the maximum
* of the sound meter.
*/
final double MAX_SOUND_PRESSURE_LEVEL
= 127 /* HUMAN TINNITUS (RINGING IN THE EARS) BEGINS */;
if (rms > 0)
{
/*
* The commonly used "zero" reference sound pressure in air
* is 20 uPa RMS, which is usually considered the threshold
* of human hearing.
*/
final double REF_SOUND_PRESSURE = 0.00002;
db = 20 * Math.log10(rms / REF_SOUND_PRESSURE);
/*
* Ensure that the calculated level is within the minimum
* and maximum sound pressure level.
*/
if (db < MIN_SOUND_PRESSURE_LEVEL)
db = MIN_SOUND_PRESSURE_LEVEL;
else if (db > MAX_SOUND_PRESSURE_LEVEL)
db = MAX_SOUND_PRESSURE_LEVEL;
}
else
{
db = MIN_SOUND_PRESSURE_LEVEL;
}
return (int) db;
}
}
AudioLevelCalculator.java
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Authors' Addresses
Emil Ivov (editor)
Jitsi
Strasbourg 67000
France
Email: emcho@jitsi.org
Enrico Marocco (editor)
Telecom Italia
Via G. Reiss Romoli, 274
Turin 10148
Italy
Email: enrico.marocco@telecomitalia.it
Jonathan Lennox
Vidyo, Inc.
433 Hackensack Avenue
Seventh Floor
Hackensack, NJ 07601
US
Email: jonathan@vidyo.com
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