Audio/Video Transport Working Group                            S. Ikonin                            Y. Morzeev
Internet Draft                                                SPIRIT DSP
Intended status: Informational                           August 18,                           September 2, 2009

RTP Payload Format for IP-MR Speech Codec draft-ietf-avt-rtp-ipmr-05.txt draft-ietf-avt-rtp-ipmr-06.txt

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This document specifies the payload format for packetization of SPIRIT
IP-MR encoded speech signals into the Real-time Transport Protocol
(RTP). The payload format supports transmission of multiple frames per
payload and introduced redundancy for robustness against packet loss.

Table of Contents

 1. Introduction......................................................3
 2. IP-MR Codec Description...........................................3
 3. Payload Format....................................................4
    3.1. RTP Header Usage.............................................4
    3.2. Payload Format Structure.....................................5
    3.3. Payload Header...............................................5
    3.4. Speech Table of Contents.....................................6
    3.5. Speech Data..................................................7
    3.6. Redundancy Header............................................7
    3.7. Redundancy Table of Contents.................................8
    3.8. Redundancy Data..............................................9
 4. Payload Examples..................................................9
    4.1. Payload Carrying a Single Frame..............................9
    4.2. Payload Carrying Multiple Frames with Redundancy............10
 5. Media Type Registration..........................................11
    5.1. Registration of media subtype audio/ip-mr_v2.5..............11
    5.2. Mapping Media Type Parameters into SDP......................12
 6. Security Considerations..........................................13
 7. Congestion Control...............................................13
 8. IANA Considerations..............................................14
 9. Normative References.............................................14
 10. Author(s) Information...........................................15
 11. Disclaimer......................................................15
 12. Legal Terms.....................................................15
 APPENDIX A. RETRIEVING FRAME INFORMATION............................17
 A.1. get_frame_info.c...............................................17
 Authors' Addresses..................................................19

1. Introduction

This document specifies the payload format for packetization of SPIRIT
IP-MR encoded speech signals into the Real-time Transport Protocol
(RTP). The payload format supports transmission of multiple frames per
payload and introduced redundancy for robustness against packet loss.

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
document are to be interpreted as described in RFC 2119 [RFC 2119].

2. IP-MR Codec Description

The IP-MR codec is scalable adaptive multi-rate wideband speech codec
designed by SPIRIT for use in IP based networks. These codec is suitable
for real time communications such as telephony and videoconferencing.

The codec operates on 20 ms frames at 16 kHz sampling rate and has an
algorithmic delay of 25ms.

The IP-MR supports six wide band speech coding modes with respective bit
rates ranging from about 7.7 to about 34.2 kbps. The coding mode can be
changed at any 20 ms frame boundary making possible to dynamically
adjust the speech encoding rate during a session to adapt to the varying
transmission conditions.

The coded frame consists of multiple coding layers - base (or core)
layer and several enhancement layers which are coded independently.
Only the core layer is mandatory to decode understandable speech and
upper layers provide quality enhancement. These enhancement layers
may be omitted and remaining base layer can be meaningfully decoded
without artifacts. This makes the bit stream scalable and allows
to reduce bit rate during transmission without re-encoding.

This memo specifies an optional form of redundancy coding within RTP
for protection against packet loss. It is based on commonly known
scheme when previously transmitted frames are aggregated together
with new ones. Each frame is retransmitted once in the following
RTP payload packet. f(n-2)...f(n+4) denotes a sequence of speech
frames, and p(n-1)...p(n+4) is a sequence of payload packets:

     | f(n-2) | f(n-1) |  f(n)  | f(n+1) | f(n+2) | f(n+3) | f(n+4) |

      <---- p(n-1) ---->
               <----- p(n) ----->
                        <---- p(n+1) ---->
                                 <---- p(n+2) ---->
                                          <---- p(n+3) ---->
                                                   <---- p(n+4) ---->

But because of the scalable nature of IP-MR codec there is no need to
duplicate the whole previous frame - only the core layer may be
retransmitted. This reduces redundancy overhead while keeping
efficiency. Moreover, the speech bits encoded in core layer are divided
on six classes (from A to F) of perceptual sensitivity to errors. Using
these classes as introduced redundancy make possible to adjust trade-off
between overhead and robustness against packet loss.

The mechanism described does not really require signaling at the session
setup. The sender is responsible for selecting an appropriate amount of
redundancy based on feedback about the channel conditions.

The main codec characteristics can be summarized as follows:

    o Wideband, 16 kHz, speech codec

    o Adaptive multi rate with six modes from about 7.7 to 34.2 kbps

    o Bit rate scalable

    o Variable bit rate changing in accordance with actual speech

    o Discontinuous Transmission (DTX), silence suppression and
      comfort noise generation

    o In-band redundancy scheme for protection against packet loss

3. Payload Format

The main purpose of the payload design for IP-MR is to maximize the
potential of the codec with as minimal overhead as possible. The payload
format allows changing parameters of the codec  (such as bit rate,
level of scalability, DTX and redundancy mode) without re-negotiation
at any packet boundary. This make possible dynamically adjust streaming
parameters in accordance to changing network conditions. The payload
format also supports aggregation of multiple consecutive frames
(up to 4) in a payload. That allows controlling trade-off between
delay and header overhead.

3.1. RTP Header Usage

The RTP timestamp corresponds to the sampling instant of the first
sample encoded for the first frame-block in the packet. The timestamp
clock frequency SHALL be 16 kHz. The duration of one frame is 20 ms,
corresponding to 320 samples at 16 kHz. Thus the timestamp is increased
by 320 for each consecutive frame. The timestamp is also used to recover
the correct decoding order of the frame-blocks.

The RTP header marker bit (M) SHALL be set to 1 whenever the first
frame-block carried in the packet is the first frame-block in a
talkspurt (see definition of the talkspurt in Section 4.1 [RFC 3551]).
For all other packets, the marker bit SHALL be set to zero (M=0).

The assignment of an RTP payload type for the format defined in this
memo is outside the scope of this document. The RTP profiles in use
currently mandate binding the payload type dynamically for this payload
format. This is basically necessary because the payload type expresses
the configuration of the payload itself, i.e. basic or interleaved mode,
and the number of channels carried.

The remaining RTP header fields are used as specified in [RFC 3550].

3.2. Payload Format Structure

The IP-MR payload format consists of a payload header with general
information about packet, a speech table of contents (TOC), and speech
data. An optional redundancy section follows after speech data. The
redundancy section consists of redundancy header, redundancy TOC and
redundancy data payload.

The following diagram shows the standard payload format layout:

  +---------+--------+--------+- - - - - - +- - - - - - +- - - - - - +
  | payload | speech | speech | redundancy | redundancy | redundancy |
  | header  | TOC    | data   | header     | TOC        | data       |
  +---------+--------+--------+- - - - - - +- - - - - - +- - - - - - +

3.3. Payload Header

The payload header has the following format:

                           0                   1
                           0 1 2 3 4 5 6 7 8 9 0 1
                          |T| CR  | BR  |D|A|GR |R|

    o T (1 bit): Reserved compatibility with future extensions. SHOULD
      be set to 0.

    o CR (3 bits): coding rate of frame(s) in this packet, as per the
       following table:

                          |  CR   | avg. bitrate |
                          |   0   |   7.7 kbps   |
                          |   1   |   9.8 kbps   |
                          |   2   |  14.3 kbps   |
                          |   3   |  20.8 kbps   |
                          |   4   |  27.9 kbps   |
                          |   5   |  34.2 kbps   |
                          |   6   |  (reserved)  |
                          |   7   |   NO_DATA    |

The CR value 7 (NO_DATA) indicates that there is no speech data (and
speech TOC accordingly) in the payload. This MAY be used to transmit
redundancy data only. The value 6 is reserved. If receiving this value
the packet SHOULD be discarded.

    o BR (3 bits): base rate for core layer of frame(s) in this packet
      using the table for CR. Values in the range 0-5 indicate bitrates
      for core layer, same as for packet SHOULD be discarded. The base
      rate is the lowest rate for scalability, so speech payload can
      be scaled down not lower than BR value. If a received packet has
      BR > CR then during decoding it will be assumed that BR = CR.

    o D (1 bit): indicates if the DTX mode is allowed or not. This
      parameter is only required for a correct payload parsing. The
      particular decoder implementation must always include DTX mode
      support and update internal states properly. It is prohibited
      to decoder to rely this parameter is constantly off during session.

    o A (1 bit): byte-aligned payload. If A=1 then all speech frames
      MUST be byte-aligned. This mode speeds up speech data access.
      The A=0 value specifies bandwidth-efficient mode with no byte
      alignment(including end of header).

    o GR (2 bits): number of frames in packet (grouping size). Actual
      grouping size is GR + 1, thus maximum grouping supported is 4.

    o R (1 bit): redundancy presence bit. If R=1 then the packet
      contains redundancy information for lost packets recovery.
      In this case after speech data the redundancy section is present.

3.4. Speech Table of Contents

The speech TOC contains entries for each frame in packet (grouping size
in total). Each entry contains a single field:

    o E (1 bit): frame existence indicator. If set to 0, this indicates
      the corresponding frame is absent and the receiver should set
      special LOST_FRAME flag for decoder. This can be followed by the
      lost frame itself or by empty frames generated by the encoder
      during silence intervals in DTX mode.

Note that if CR flag from payload header is 7 (NO_DATA) then speech TOC
is empty.

3.5. Speech Data

Speech data of a payload contains one or more speech frames or comfort
noise frames, as specified in the speech TOC of the payload.

Each speech frame represents 20 ms of speech encoded with the rate
indicated in the CR and base rate indicated in BR field of the payload

The size of coded speech frame is variable due to the nature of codec.
The Encoder's algorithm decides what size of each frame is and returns
it after encoding. In order to save bandwidth the size is not placed
into payload obviously. The frame size can be determined by frame's
content using a special service function specified in Appendix A.
This function provides complete information about coded frame including
size, number of layers, size of each layer and size of perceptual
sensitive classes.

3.6. Redundancy Header

If a packet contains redundancy (R field of payload header is 1) the
speech data is followed by redundancy header:

                             0 1 2 3 4 5
                            | CL1 | CL2 |

Redundancy header consists of two fields. Each field contains class
specifier for amount of redundancy partly taken from the preceding
packet (CL1) and pre-preceding packet (CL2), e.g. distant from the
current packet by 1 and 2 packets accordingly. The values are listed
in the table below:

                     |  CL   | amount redundancy |
                     |   0   |       NONE        |
                     |   1   |      CLASS A      |
                     |   2   |      CLASS B      |
                     |   3   |      CLASS C      |
                     |   4   |      CLASS D      |
                     |   5   |      CLASS E      |
                     |   6   |      CLASS F      |
                     |   7   |     (reserved)    |

Each specifier takes 3 bits, thus the total redundancy header size is 6

These classes indicate subjective importance of bits from core layer.
Class A contains the bits most sensitive to errors and lost of these
bits results in a corrupted speech frame which should not be decoded
without applying packet loss concealment (PLC) procedure. Class B is
less sensitive than class A and so on to F. Sum of all bit classes
from A to F composes core layer.

Putting some part (classes of bits) from previous frame into current
packet makes possible to partially decode previous frame in case of
it's lost. Than more information is delivered than less speech quality
degradation will be. Flags CL1 and CL2 specify how many classes from
previous frames current packet contain. E.g. CL1=3 (class C), it means
that packet contains bits from classes A, B and C of previous frame.
If CL1=6 (class F) then whole core layer is included.

3.7. Redundancy Table of Contents

                    | Pkt1 Entries| Pkt2 Entries|

The redundancy TOC contains entries for redundancy frames from preceding
and pre-preceding packets. Each entry takes 1 bit like speech TOC entry


    o E (1 bit): frame existence indicator. If set to 0, this indicates
      the corresponding frame is absent.

    o For each preceding and pre-preceding packet the number of entries
      is equal to the grouping size of the current packet. E.g. maximum
      number of entries is 4*2 = 8.

    o If class specifier in the redundancy header is CL=0 (NO_DATA)
      then there is no entries for corresponding packet redundancy.

3.8. Redundancy Data

Redundancy data of a payload contains redundancy information for one or
more speech frames or comfort noise frames that may be lost during
transition, as specified in the redundancy TOC of the payload. Actually
redundancy is the most important part of preceding frames representing
20 ms of speech. This data MAY be used for partial reconstruction of
lost frames. The amount of available redundancy is specified by CL flag
in redundancy header section (3.5). This flag SHOULD be passed to
decoder. The size of redundancy frame is variable and can be obtained
using service function specified in Appendix A.

4. Payload Examples

A few examples to highlight the payload format follow.

4.1. Payload Carrying a Single Frame

The following diagram shows a standard IP-MR payload carrying a single
speech frame without redundancy:

   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
  |0|CR=1 |BR=0 |0|0|0 0|0|1|sp(0)                                |
  |                                                               |
  |                                                               |
  |                                                               |
  |                                                               |
  |                                                               |
  |                      sp(193)|P|

In the payload the speech frame is not damaged at the IP origin (E=1),
the coding rate is 9.7 kbps(CR=1), the base rate is 7.8 kbps (BR=0), and
the DTX mode is off. There is no byte alignment (A=0) and no redundancy
(R=0). The encoded speech bits - s(0) to s(193) - are placed immediately
after TOC. Finally, one zero bit is added at the end as padding to make
the payload byte aligned.

4.2. Payload Carrying Multiple Frames with Redundancy

The following diagram shows a payload that contains three frames, one of
them with no speech data. The coding rate is 7.7 kbps (CR=0), the base
rate is 7.7 kbps (BR=0), and the DTX mode is on. The speech frames are
byte aligned (A=1), so 1 zero bit is added at the end of the header.
Besides the speech frames the payload contains six redundancy frames
(three per each delayed packet).

The first speech frame consists of bits sp1(0) to sp1(92). After that 3
bits are added for byte alignment. The second frame does not contain any
speech information that is represented in the payload by its TOC entry.
The third frame consists of bits sp3(0) to sp3(171).

The redundancy header follows after speech data. The one-packet-delayed
redundancy contains class A+B bits (CL1=2), and two-packet-delayed
redundancy contains class A bits (Cl2=1). The one-packet-delayed
redundancy contains three frames with 20, 39 and 35 bits respectively.

The first frame of two-packet-delayed redundancy is absent, it is
represented in its TOC entry, and two other frames have sizes 15 and 19

Note that all speech frames are padded with zero bits for byte

   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
  |0|CR=0 |BR=0 |1|1|1 0|1|1 0 1|P|sp1(0)                         |
  |                                                               |
  |                                                               |
  |                  sp1(92)|P|P|P|sp3(0)                         |
  |                                                               |
  |                                                               |
  |                                                               |
  |                                                               |
  |                                               sp3(171)|P|P|P|P|
  |CL1=2|CL2=1|1 1 1|0 1 1|red1_1(0)                    red1_1(19)|
  |red1_2(0)                                                      |
  |   red1_2(38)|red1_3(0)                                        |
  |         red1_3(34)|red2_2(0)          red2_2(14)|red2_3(0)    |
  |             red2_3(18)|P|P|P|P|

5. Media Type Registration

This section describes the media types and names associated with this
payload format.

5.1. Registration of media subtype audio/ip-mr_v2.5

Type name: audio

Subtype name: ip-mr_v2.5

Required parameters: none

Optional parameters:
* ptime: Gives the length of time in milliseconds represented by the
media in a packet. Allowed values are: 20, 40, 60 and 80.

Encoding considerations: This media type is framed binary data (see RFC
4288, Section 4.8).

Security considerations: See RFC 3550 [RFC 3550]

Interoperability considerations: none

Published specification: RFC XXXX

Applications that use this media type: Real-time audio applications like
voice over IP and teleconference, and multi-media streaming.

Additional information: none

Person & email address to contact for further information:
Yuri Morzeev

Intended usage: COMMON

Restrictions on usage: This media type depends on RTP framing, and hence
is only defined for transfer via RTP [RFC 3550].

Sergey Ikonin <>
Yury Morzeev <>

Change controller: IETF Audio/Video Transport working group delegated
from the IESG.

5.2. Mapping Media Type Parameters into SDP

The information carried in the media type specification has a specific
mapping to fields in the Session Description Protocol (SDP) [RFC 4566],
which is commonly used to describe RTP sessions. When SDP is used to
specify sessions employing the IP-MR codec, the mapping is as follows:

    o The media type ("audio") goes in SDP "m=" as the media name.

    o The media subtype (payload format name) goes in SDP "a=rtpmap"
    as the encoding name. The RTP clock rate in "a=rtpmap" MUST 16000.

    o The parameter "ptime" goes in the SDP "a=ptime" attributes.

Any remaining parameters go in the SDP "a=fmtp" attribute by copying
them directly from the media type parameter string as a semicolon-
separated list of parameter=value pairs.

Note that the payload format (encoding) names are commonly shown in
upper case. Media subtypes are commonly shown in lower case. These
names are case-insensitive in both places.

6. Security Considerations

RTP packets using the payload format defined in this specification
are subject to the security considerations discussed in the RTP
specification [RFC 3550] and in any applicable RTP profile. The main
security considerations for the RTP packet carrying the RTP payload
format defined within this memo are confidentiality, integrity, and
source authenticity. Confidentiality is achieved by encryption of the
RTP payload. Integrity of the RTP packets is achieved through a suitable
cryptographic integrity protection mechanism. Such a cryptographic
system may also allow the authentication of the source of the payload.

A suitable security mechanism for this RTP payload format should
provide confidentiality, integrity protection, and at least source
authentication capable of determining if an RTP packet is from a
member of the RTP session.

Note that the appropriate mechanism to provide security to RTP and
payloads following this memo may vary. It is dependent on the
application, the transport, and the signaling protocol employed.
Therefore, a single mechanism is not sufficient, although if suitable,
usage of the Secure Real-time Transport Protocol (SRTP) [RFC 3711] is
recommended.  Other mechanisms that may be used are IPsec [RFC 4301]
and Transport Layer Security (TLS) [RFC 5246] (RTP over TCP); other
alternatives may exist.

This payload format does not exhibit any significant non-uniformity in
the receiver side computational complexity for packet processing, and
thus is unlikely to pose a denial-of-service threat due to the receipt
of pathological data.

7. Congestion Control

The general congestion control considerations for transporting RTP data
apply; see RTP [RFC 3550] and any applicable RTP profile like AVP
[RFC 3551]. However, the multi-rate capability of IP-MR speech coding
provides a mechanism that may help to control congestion, since the
bandwidth demand can be adjusted by selecting a different encoding mode.

The number of frames encapsulated in each RTP payload highly
influences the overall bandwidth of the RTP stream due to header
overhead constraints. Packetizing more frames in each RTP payload
can reduce the number of packets sent and hence the overhead from
IP/UDP/RTP headers, at the expense of increased delay.

If in-band redundancy scheme is used to protect against packet loss,
the amount of introduced redundancy will need to be regulated so that
the use of redundancy itself does not cause a congestion problem. In
other words, a sender SHALL NOT increase the total bitrate when adding
redundancy in response to packet loss, and needs instead to adjust it
down in accordance to the congestion control algorithm being run. Thus,
when adding redundancy, the media bitrate will need to be reduced to
provide room for the redundancy.

8. IANA Considerations

One media type has been defined and needs registration in the media
types registry.

9. Normative References

  [RFC 2119] Bradner, S., "Key words for use in RFCs to Indicate
             Requirement Levels", BCP 14, RFC 2119, March 1997.

  [RFC 3550] Schulzrinne, H., Casner, S., Frederick, R., and
             V. Jacobson, "RTP: A Transport Protocol for Real-Time
             Applications", STD 64, RFC 3550, July 2003.

  [RFC 3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio
             and Video Conferences with Minimal Control", STD 65,
             RFC 3551, July 2003.

  [RFC 4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
             Description Protocol", RFC 4566, July 2006.

  [RFC 3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., Norrman,
             K., "The Secure Real-Time Transport Protocol (SRTP)", RFC
             3711, March 2004.

  [RFC 5246] Dierks, T. and E. Rescorla, "The Transport Layer
             Security (TLS) Protocol Version 1.2", RFC 5246,
             August 2008.

  [RFC 4301] Kent, S. and K. Seo, "Security Architecture for the
             Internet Protocol", RFC 4301, December 2005.

10. Author(s) Information:

Sergey Ikonin Morzeev

Russia 109004
Building 27, A. Solgenizyn street
Tel: +7 495 661-2178
Fax: +7 495 912-6786

11. Disclaimer

This document may contain material from IETF Documents or IETF
Contributions published or made publicly available before November 10,
2008. The person(s) controlling the copyright in some of this material
may not have granted the IETF Trust the right to allow modifications of
such material outside the IETF Standards Process. Without obtaining an
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materials, this document may not be modified outside the IETF Standards
Process, and derivative works of it may not be created outside the IETF
Standards Process, except to format it for publication as an RFC or to
translate it into languages other than English.

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This appendix contains the c-code for implementation of frame parsing
function. This function extracts information about coded frame including
frame size, number of layers, size of each layer and size of perceptual
sensitive classes.

A.1. get_frame_info.c



  Retrieving frame information for IP-MR Speech Codec


#define RATES_NUM       6   // number of codec rates
#define SENSE_CLASSES   6   // number of sensitivity classes (A..F)

// frame types
#define FT_DTX_SPEECH   0   // active speech in DTX mode
#define FT_DTX_SID      1   // silence insertion descriptor
#define FT_NO_DTX       2   // no DTX frame

// get specified bit from coded data
int GetBit(unsigned char *data, int curBit)
  return ((data[curBit >> 3] >> (curBit % 8)) & 1);

// retrieve frame information
int GetFrameInfo(           // o: frame size in bits
  short rate,               // i: encoding rate (0..5)
  short base_rate,          // i: base (core) layer rate,
                            //    if base_rate > rate, then assumed
                            //    that base_rate = rate.
  short allow_DTX,          // i: flag of DTX mode
  unsigned char *pCoded,    // i: coded bit frame
  short pLayerBits          // o: number of bits in layers
  short pSenseBits          // o: number of bits in sensitivity classes
  short *nLayers            // o: number of layers
  static const short Bits_1[4]    = {0, 9, 9, 15};
  static const short Bits_2[16]   = { 43,50,36,31,46,48,40,44,47,43,44,
  static const short Bits_3[2][6] = {{13, 11, 23, 33, 36, 31},
                                     {25,  0, 23, 32, 36, 31},};

  int FrType;
  int i,nBits;

  if (rate < 0 || rate > 5) {
    return 0; // incorrect stream

  for(i = 0; i < SENSE_CLASSES; i++) {
    pSenseBits[i] = 0;

  nBits = 0;
  // extract frame type bit if required
  if (allow_DTX) {
    FrType = GetBit(pCoded, nBits++) ? FT_DTX_SPEECH : FT_DTX_SID;
  } else {
    FrType = FT_NO_DTX;
    int cw_0;
    int b[14];

    // extract meaning bits
    for(i = 0 ; i < 14; i++) {
        b[i] = GetBit(pCoded, nBits++);

    // parse
    if(FrType == FT_DTX_SID) {
      cw_0 = (b[0]<<0)|(b[1]<<1)|(b[2]<<2)|(b[3]<<3);
      rate = 0;
      pSenseBits[0] = 10 + Bits_2[cw_0];
    } else {

      int i, idx;
      int nFlag_1, nFlag_2, cw_1, cw_2;

      nFlag_1 = b[0] + b[2] + b[4] + b[6];
      cw_1 = (cw_1 << 1) | b[0];
      cw_1 = (cw_1 << 1) | b[2];
      cw_1 = (cw_1 << 1) | b[4];
      cw_1 = (cw_1 << 1) | b[6];

      nFlag_2 = b[1] + b[3] + b[5] + b[7];
      cw_2 = (cw_2 << 1) | b[1];
      cw_2 = (cw_2 << 1) | b[3];
      cw_2 = (cw_2 << 1) | b[5];
      cw_2 = (cw_2 << 1) | b[7];

      cw_0 = (b[10]<<0)|(b[11]<<1)|(b[12]<<2)|(b[13]<<3);
      if (base_rate < 0)    base_rate = 0;
      if (base_rate > rate) base_rate = rate;
      idx = base_rate == 0 ? 0 : 1;

      pSenseBits[0] = (FrType == FT_DTX_SPEECH ? 1:0)+14+Bits_2[cw_0];
      pSenseBits[1] = Bits_1[(cw_1 >> 0)&0x3] + Bits_1[(cw_1>>2)&0x3];
      pSenseBits[2] = nFlag_1*5;
      pSenseBits[3] = nFlag_2*30;
      pSenseBits[5] = (4 - nFlag_2)*(Bits_3[idx][0]);

      for (i = 1; i < rate+1; i++) {
        pLayerBits[i] = 4*(Bits_3[idx][i]);

    pLayerBits[0] = 0;
    for (i = 0; i < SENSE_CLASSES; i++) {
        pLayerBits[0] += pSenseBits[i];

    *nLayers = rate+1;

    // count total frame size
    int payloadBitCount = 0;
    for (i = 0; i < *nLayers; i++) {
      payloadBitCount += pLayerBits[i];
    return payloadBitCount;

Authors' Addresses

   Building 27, A. Solgenizyn street
   109004, Moscow, RUSSIA

   Tel: +7 495 661-2178
   Fax: +7 495 912-6786