| Index: webrtc/modules/audio_coding/codecs/opus/opus/src/doc/draft-ietf-payload-rtp-opus.xml
|
| diff --git a/webrtc/modules/audio_coding/codecs/opus/opus/src/doc/draft-ietf-payload-rtp-opus.xml b/webrtc/modules/audio_coding/codecs/opus/opus/src/doc/draft-ietf-payload-rtp-opus.xml
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..c4eb21077b7224ef1443f6d4565832604f47dd38
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_coding/codecs/opus/opus/src/doc/draft-ietf-payload-rtp-opus.xml
|
| @@ -0,0 +1,960 @@
|
| +<?xml version="1.0" encoding="UTF-8"?>
|
| +<!DOCTYPE rfc SYSTEM "rfc2629.dtd" [
|
| +<!ENTITY rfc2119 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml'>
|
| +<!ENTITY rfc3389 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3389.xml'>
|
| +<!ENTITY rfc3550 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml'>
|
| +<!ENTITY rfc3711 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3711.xml'>
|
| +<!ENTITY rfc3551 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3551.xml'>
|
| +<!ENTITY rfc6838 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6838.xml'>
|
| +<!ENTITY rfc4855 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4855.xml'>
|
| +<!ENTITY rfc4566 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4566.xml'>
|
| +<!ENTITY rfc4585 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4585.xml'>
|
| +<!ENTITY rfc3264 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3264.xml'>
|
| +<!ENTITY rfc2974 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2974.xml'>
|
| +<!ENTITY rfc2326 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2326.xml'>
|
| +<!ENTITY rfc3555 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3555.xml'>
|
| +<!ENTITY rfc5124 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5124.xml'>
|
| +<!ENTITY rfc5405 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5405.xml'>
|
| +<!ENTITY rfc5576 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5576.xml'>
|
| +<!ENTITY rfc6562 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6562.xml'>
|
| +<!ENTITY rfc6716 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml'>
|
| +<!ENTITY rfc7202 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.7202.xml'>
|
| +<!ENTITY nbsp " ">
|
| + ]>
|
| +
|
| + <rfc category="std" ipr="trust200902" docName="draft-ietf-payload-rtp-opus-11">
|
| +<?xml-stylesheet type='text/xsl' href='rfc2629.xslt' ?>
|
| +
|
| +<?rfc strict="yes" ?>
|
| +<?rfc toc="yes" ?>
|
| +<?rfc tocdepth="3" ?>
|
| +<?rfc tocappendix='no' ?>
|
| +<?rfc tocindent='yes' ?>
|
| +<?rfc symrefs="yes" ?>
|
| +<?rfc sortrefs="yes" ?>
|
| +<?rfc compact="no" ?>
|
| +<?rfc subcompact="yes" ?>
|
| +<?rfc iprnotified="yes" ?>
|
| +
|
| + <front>
|
| + <title abbrev="RTP Payload Format for Opus">
|
| + RTP Payload Format for the Opus Speech and Audio Codec
|
| + </title>
|
| +
|
| + <author fullname="Julian Spittka" initials="J." surname="Spittka">
|
| + <address>
|
| + <email>jspittka@gmail.com</email>
|
| + </address>
|
| + </author>
|
| +
|
| + <author initials='K.' surname='Vos' fullname='Koen Vos'>
|
| + <organization>vocTone</organization>
|
| + <address>
|
| + <postal>
|
| + <street></street>
|
| + <code></code>
|
| + <city></city>
|
| + <region></region>
|
| + <country></country>
|
| + </postal>
|
| + <email>koenvos74@gmail.com</email>
|
| + </address>
|
| + </author>
|
| +
|
| + <author initials="JM" surname="Valin" fullname="Jean-Marc Valin">
|
| + <organization>Mozilla</organization>
|
| + <address>
|
| + <postal>
|
| + <street>331 E. Evelyn Avenue</street>
|
| + <city>Mountain View</city>
|
| + <region>CA</region>
|
| + <code>94041</code>
|
| + <country>USA</country>
|
| + </postal>
|
| + <email>jmvalin@jmvalin.ca</email>
|
| + </address>
|
| + </author>
|
| +
|
| + <date day='14' month='April' year='2015' />
|
| +
|
| + <abstract>
|
| + <t>
|
| + This document defines the Real-time Transport Protocol (RTP) payload
|
| + format for packetization of Opus encoded
|
| + speech and audio data necessary to integrate the codec in the
|
| + most compatible way. It also provides an applicability statement
|
| + for the use of Opus over RTP. Further, it describes media type registrations
|
| + for the RTP payload format.
|
| + </t>
|
| + </abstract>
|
| + </front>
|
| +
|
| + <middle>
|
| + <section title='Introduction'>
|
| + <t>
|
| + Opus <xref target="RFC6716"/> is a speech and audio codec developed within the
|
| + IETF Internet Wideband Audio Codec working group. The codec
|
| + has a very low algorithmic delay and it
|
| + is highly scalable in terms of audio bandwidth, bitrate, and
|
| + complexity. Further, it provides different modes to efficiently encode speech signals
|
| + as well as music signals, thus making it the codec of choice for
|
| + various applications using the Internet or similar networks.
|
| + </t>
|
| + <t>
|
| + This document defines the Real-time Transport Protocol (RTP)
|
| + <xref target="RFC3550"/> payload format for packetization
|
| + of Opus encoded speech and audio data necessary to
|
| + integrate Opus in the
|
| + most compatible way. It also provides an applicability statement
|
| + for the use of Opus over RTP.
|
| + Further, it describes media type registrations for
|
| + the RTP payload format.
|
| + </t>
|
| + </section>
|
| +
|
| + <section title='Conventions, Definitions and Acronyms used in this document'>
|
| + <t>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 <xref target="RFC2119"/>.</t>
|
| + <t>
|
| + <list style='hanging'>
|
| + <t hangText="audio bandwidth:"> The range of audio frequecies being coded</t>
|
| + <t hangText="CBR:"> Constant bitrate</t>
|
| + <t hangText="CPU:"> Central Processing Unit</t>
|
| + <t hangText="DTX:"> Discontinuous transmission</t>
|
| + <t hangText="FEC:"> Forward error correction</t>
|
| + <t hangText="IP:"> Internet Protocol</t>
|
| + <t hangText="samples:"> Speech or audio samples (per channel)</t>
|
| + <t hangText="SDP:"> Session Description Protocol</t>
|
| + <t hangText="VBR:"> Variable bitrate</t>
|
| + </list>
|
| + </t>
|
| + <t>
|
| + Throughout this document, we refer to the following definitions:
|
| + </t>
|
| + <texttable anchor='bandwidth_definitions'>
|
| + <ttcol align='center'>Abbreviation</ttcol>
|
| + <ttcol align='center'>Name</ttcol>
|
| + <ttcol align='center'>Audio Bandwidth (Hz)</ttcol>
|
| + <ttcol align='center'>Sampling Rate (Hz)</ttcol>
|
| + <c>NB</c>
|
| + <c>Narrowband</c>
|
| + <c>0 - 4000</c>
|
| + <c>8000</c>
|
| +
|
| + <c>MB</c>
|
| + <c>Mediumband</c>
|
| + <c>0 - 6000</c>
|
| + <c>12000</c>
|
| +
|
| + <c>WB</c>
|
| + <c>Wideband</c>
|
| + <c>0 - 8000</c>
|
| + <c>16000</c>
|
| +
|
| + <c>SWB</c>
|
| + <c>Super-wideband</c>
|
| + <c>0 - 12000</c>
|
| + <c>24000</c>
|
| +
|
| + <c>FB</c>
|
| + <c>Fullband</c>
|
| + <c>0 - 20000</c>
|
| + <c>48000</c>
|
| +
|
| + <postamble>
|
| + Audio bandwidth naming
|
| + </postamble>
|
| + </texttable>
|
| + </section>
|
| +
|
| + <section title='Opus Codec'>
|
| + <t>
|
| + Opus encodes speech
|
| + signals as well as general audio signals. Two different modes can be
|
| + chosen, a voice mode or an audio mode, to allow the most efficient coding
|
| + depending on the type of the input signal, the sampling frequency of the
|
| + input signal, and the intended application.
|
| + </t>
|
| +
|
| + <t>
|
| + The voice mode allows efficient encoding of voice signals at lower bit
|
| + rates while the audio mode is optimized for general audio signals at medium and
|
| + higher bitrates.
|
| + </t>
|
| +
|
| + <t>
|
| + Opus is highly scalable in terms of audio
|
| + bandwidth, bitrate, and complexity. Further, Opus allows
|
| + transmitting stereo signals with in-band signaling in the bit-stream.
|
| + </t>
|
| +
|
| + <section title='Network Bandwidth'>
|
| + <t>
|
| + Opus supports bitrates from 6 kb/s to 510 kb/s.
|
| + The bitrate can be changed dynamically within that range.
|
| + All
|
| + other parameters being
|
| + equal, higher bitrates result in higher audio quality.
|
| + </t>
|
| + <section title='Recommended Bitrate' anchor='bitrate_by_bandwidth'>
|
| + <t>
|
| + For a frame size of
|
| + 20 ms, these
|
| + are the bitrate "sweet spots" for Opus in various configurations:
|
| +
|
| + <list style="symbols">
|
| + <t>8-12 kb/s for NB speech,</t>
|
| + <t>16-20 kb/s for WB speech,</t>
|
| + <t>28-40 kb/s for FB speech,</t>
|
| + <t>48-64 kb/s for FB mono music, and</t>
|
| + <t>64-128 kb/s for FB stereo music.</t>
|
| + </list>
|
| + </t>
|
| + </section>
|
| + <section title='Variable versus Constant Bitrate' anchor='variable-vs-constant-bitrate'>
|
| + <t>
|
| + For the same average bitrate, variable bitrate (VBR) can achieve higher audio quality
|
| + than constant bitrate (CBR). For the majority of voice transmission applications, VBR
|
| + is the best choice. One reason for choosing CBR is the potential
|
| + information leak that <spanx style='emph'>might</spanx> occur when encrypting the
|
| + compressed stream. See <xref target="RFC6562"/> for guidelines on when VBR is
|
| + appropriate for encrypted audio communications. In the case where an existing
|
| + VBR stream needs to be converted to CBR for security reasons, then the Opus padding
|
| + mechanism described in <xref target="RFC6716"/> is the RECOMMENDED way to achieve padding
|
| + because the RTP padding bit is unencrypted.</t>
|
| +
|
| + <t>
|
| + The bitrate can be adjusted at any point in time. To avoid congestion,
|
| + the average bitrate SHOULD NOT exceed the available
|
| + network bandwidth. If no target bitrate is specified, the bitrates specified in
|
| + <xref target='bitrate_by_bandwidth'/> are RECOMMENDED.
|
| + </t>
|
| +
|
| + </section>
|
| +
|
| + <section title='Discontinuous Transmission (DTX)'>
|
| +
|
| + <t>
|
| + Opus can, as described in <xref target='variable-vs-constant-bitrate'/>,
|
| + be operated with a variable bitrate. In that case, the encoder will
|
| + automatically reduce the bitrate for certain input signals, like periods
|
| + of silence. When using continuous transmission, it will reduce the
|
| + bitrate when the characteristics of the input signal permit, but
|
| + will never interrupt the transmission to the receiver. Therefore, the
|
| + received signal will maintain the same high level of audio quality over the
|
| + full duration of a transmission while minimizing the average bit
|
| + rate over time.
|
| + </t>
|
| +
|
| + <t>
|
| + In cases where the bitrate of Opus needs to be reduced even
|
| + further or in cases where only constant bitrate is available,
|
| + the Opus encoder can use discontinuous
|
| + transmission (DTX), where parts of the encoded signal that
|
| + correspond to periods of silence in the input speech or audio signal
|
| + are not transmitted to the receiver. A receiver can distinguish
|
| + between DTX and packet loss by looking for gaps in the sequence
|
| + number, as described by Section 4.1
|
| + of <xref target="RFC3551"/>.
|
| + </t>
|
| +
|
| + <t>
|
| + On the receiving side, the non-transmitted parts will be handled by a
|
| + frame loss concealment unit in the Opus decoder which generates a
|
| + comfort noise signal to replace the non transmitted parts of the
|
| + speech or audio signal. Use of <xref target="RFC3389"/> Comfort
|
| + Noise (CN) with Opus is discouraged.
|
| + The transmitter MUST drop whole frames only,
|
| + based on the size of the last transmitted frame,
|
| + to ensure successive RTP timestamps differ by a multiple of 120 and
|
| + to allow the receiver to use whole frames for concealment.
|
| + </t>
|
| +
|
| + <t>
|
| + DTX can be used with both variable and constant bitrate.
|
| + It will have a slightly lower speech or audio
|
| + quality than continuous transmission. Therefore, using continuous
|
| + transmission is RECOMMENDED unless constraints on available network bandwidth
|
| + are severe.
|
| + </t>
|
| +
|
| + </section>
|
| +
|
| + </section>
|
| +
|
| + <section title='Complexity'>
|
| +
|
| + <t>
|
| + Complexity of the encoder can be scaled to optimize for CPU resources in real-time, mostly as
|
| + a trade-off between audio quality and bitrate. Also, different modes of Opus have different complexity.
|
| + </t>
|
| +
|
| + </section>
|
| +
|
| + <section title="Forward Error Correction (FEC)">
|
| +
|
| + <t>
|
| + The voice mode of Opus allows for embedding "in-band" forward error correction (FEC)
|
| + data into the Opus bit stream. This FEC scheme adds
|
| + redundant information about the previous packet (N-1) to the current
|
| + output packet N. For
|
| + each frame, the encoder decides whether to use FEC based on (1) an
|
| + externally-provided estimate of the channel's packet loss rate; (2) an
|
| + externally-provided estimate of the channel's capacity; (3) the
|
| + sensitivity of the audio or speech signal to packet loss; (4) whether
|
| + the receiving decoder has indicated it can take advantage of "in-band"
|
| + FEC information. The decision to send "in-band" FEC information is
|
| + entirely controlled by the encoder and therefore no special precautions
|
| + for the payload have to be taken.
|
| + </t>
|
| +
|
| + <t>
|
| + On the receiving side, the decoder can take advantage of this
|
| + additional information when it loses a packet and the next packet
|
| + is available. In order to use the FEC data, the jitter buffer needs
|
| + to provide access to payloads with the FEC data.
|
| + Instead of performing loss concealment for a missing packet, the
|
| + receiver can then configure its decoder to decode the FEC data from the next packet.
|
| + </t>
|
| +
|
| + <t>
|
| + Any compliant Opus decoder is capable of ignoring
|
| + FEC information when it is not needed, so encoding with FEC cannot cause
|
| + interoperability problems.
|
| + However, if FEC cannot be used on the receiving side, then FEC
|
| + SHOULD NOT be used, as it leads to an inefficient usage of network
|
| + resources. Decoder support for FEC SHOULD be indicated at the time a
|
| + session is set up.
|
| + </t>
|
| +
|
| + </section>
|
| +
|
| + <section title='Stereo Operation'>
|
| +
|
| + <t>
|
| + Opus allows for transmission of stereo audio signals. This operation
|
| + is signaled in-band in the Opus bit-stream and no special arrangement
|
| + is needed in the payload format. An
|
| + Opus decoder is capable of handling a stereo encoding, but an
|
| + application might only be capable of consuming a single audio
|
| + channel.
|
| + </t>
|
| + <t>
|
| + If a decoder cannot take advantage of the benefits of a stereo signal
|
| + this SHOULD be indicated at the time a session is set up. In that case
|
| + the sending side SHOULD NOT send stereo signals as it leads to an
|
| + inefficient usage of network resources.
|
| + </t>
|
| +
|
| + </section>
|
| +
|
| + </section>
|
| +
|
| + <section title='Opus RTP Payload Format' anchor='opus-rtp-payload-format'>
|
| + <t>The payload format for Opus consists of the RTP header and Opus payload
|
| + data.</t>
|
| + <section title='RTP Header Usage'>
|
| + <t>The format of the RTP header is specified in <xref target="RFC3550"/>.
|
| + The use of the fields of the RTP header by the Opus payload format is
|
| + consistent with that specification.</t>
|
| +
|
| + <t>The payload length of Opus is an integer number of octets and
|
| + therefore no padding is necessary. The payload MAY be padded by an
|
| + integer number of octets according to <xref target="RFC3550"/>,
|
| + although the Opus internal padding is preferred.</t>
|
| +
|
| + <t>The timestamp, sequence number, and marker bit (M) of the RTP header
|
| + are used in accordance with Section 4.1
|
| + of <xref target="RFC3551"/>.</t>
|
| +
|
| + <t>The RTP payload type for Opus is to be assigned dynamically.</t>
|
| +
|
| + <t>The receiving side MUST be prepared to receive duplicate RTP
|
| + packets. The receiver MUST provide at most one of those payloads to the
|
| + Opus decoder for decoding, and MUST discard the others.</t>
|
| +
|
| + <t>Opus supports 5 different audio bandwidths, which can be adjusted during
|
| + a stream.
|
| + The RTP timestamp is incremented with a 48000 Hz clock rate
|
| + for all modes of Opus and all sampling rates.
|
| + The unit
|
| + for the timestamp is samples per single (mono) channel. The RTP timestamp corresponds to the
|
| + sample time of the first encoded sample in the encoded frame.
|
| + For data encoded with sampling rates other than 48000 Hz,
|
| + the sampling rate has to be adjusted to 48000 Hz.</t>
|
| +
|
| + </section>
|
| +
|
| + <section title='Payload Structure'>
|
| + <t>
|
| + The Opus encoder can output encoded frames representing 2.5, 5, 10, 20,
|
| + 40, or 60 ms of speech or audio data. Further, an arbitrary number of frames can be
|
| + combined into a packet, up to a maximum packet duration representing
|
| + 120 ms of speech or audio data. The grouping of one or more Opus
|
| + frames into a single Opus packet is defined in Section 3 of
|
| + <xref target="RFC6716"/>. An RTP payload MUST contain exactly one
|
| + Opus packet as defined by that document.
|
| + </t>
|
| +
|
| + <t><xref target='payload-structure'/> shows the structure combined with the RTP header.</t>
|
| +
|
| + <figure anchor="payload-structure"
|
| + title="Packet structure with RTP header">
|
| + <artwork align="center">
|
| + <![CDATA[
|
| ++----------+--------------+
|
| +|RTP Header| Opus Payload |
|
| ++----------+--------------+
|
| + ]]>
|
| + </artwork>
|
| + </figure>
|
| +
|
| + <t>
|
| + <xref target='opus-packetization'/> shows supported frame sizes in
|
| + milliseconds of encoded speech or audio data for the speech and audio modes
|
| + (Mode) and sampling rates (fs) of Opus and shows how the timestamp is
|
| + incremented for packetization (ts incr). If the Opus encoder
|
| + outputs multiple encoded frames into a single packet, the timestamp
|
| + increment is the sum of the increments for the individual frames.
|
| + </t>
|
| +
|
| + <texttable anchor='opus-packetization' title="Supported Opus frame
|
| + sizes and timestamp increments marked with an o. Unsupported marked with an x.">
|
| + <ttcol align='center'>Mode</ttcol>
|
| + <ttcol align='center'>fs</ttcol>
|
| + <ttcol align='center'>2.5</ttcol>
|
| + <ttcol align='center'>5</ttcol>
|
| + <ttcol align='center'>10</ttcol>
|
| + <ttcol align='center'>20</ttcol>
|
| + <ttcol align='center'>40</ttcol>
|
| + <ttcol align='center'>60</ttcol>
|
| + <c>ts incr</c>
|
| + <c>all</c>
|
| + <c>120</c>
|
| + <c>240</c>
|
| + <c>480</c>
|
| + <c>960</c>
|
| + <c>1920</c>
|
| + <c>2880</c>
|
| + <c>voice</c>
|
| + <c>NB/MB/WB/SWB/FB</c>
|
| + <c>x</c>
|
| + <c>x</c>
|
| + <c>o</c>
|
| + <c>o</c>
|
| + <c>o</c>
|
| + <c>o</c>
|
| + <c>audio</c>
|
| + <c>NB/WB/SWB/FB</c>
|
| + <c>o</c>
|
| + <c>o</c>
|
| + <c>o</c>
|
| + <c>o</c>
|
| + <c>x</c>
|
| + <c>x</c>
|
| + </texttable>
|
| +
|
| + </section>
|
| +
|
| + </section>
|
| +
|
| + <section title='Congestion Control'>
|
| +
|
| + <t>The target bitrate of Opus can be adjusted at any point in time, thus
|
| + allowing efficient congestion control. Furthermore, the amount
|
| + of encoded speech or audio data encoded in a
|
| + single packet can be used for congestion control, since the transmission
|
| + rate is inversely proportional to the packet duration. A lower packet
|
| + transmission rate reduces the amount of header overhead, but at the same
|
| + time increases latency and loss sensitivity, so it ought to be used with
|
| + care.</t>
|
| +
|
| + <t>Since UDP does not provide congestion control, applications that use
|
| + RTP over UDP SHOULD implement their own congestion control above the
|
| + UDP layer <xref target="RFC5405"/>. Work in the rmcat working group
|
| + <xref target="rmcat"/> describes the
|
| + interactions and conceptual interfaces necessary between the application
|
| + components that relate to congestion control, including the RTP layer,
|
| + the higher-level media codec control layer, and the lower-level
|
| + transport interface, as well as components dedicated to congestion
|
| + control functions.</t>
|
| + </section>
|
| +
|
| + <section title='IANA Considerations'>
|
| + <t>One media subtype (audio/opus) has been defined and registered as
|
| + described in the following section.</t>
|
| +
|
| + <section title='Opus Media Type Registration'>
|
| + <t>Media type registration is done according to <xref
|
| + target="RFC6838"/> and <xref target="RFC4855"/>.<vspace
|
| + blankLines='1'/></t>
|
| +
|
| + <t>Type name: audio<vspace blankLines='1'/></t>
|
| + <t>Subtype name: opus<vspace blankLines='1'/></t>
|
| +
|
| + <t>Required parameters:</t>
|
| + <t><list style="hanging">
|
| + <t hangText="rate:"> the RTP timestamp is incremented with a
|
| + 48000 Hz clock rate for all modes of Opus and all sampling
|
| + rates. For data encoded with sampling rates other than 48000 Hz,
|
| + the sampling rate has to be adjusted to 48000 Hz.
|
| + </t>
|
| + </list></t>
|
| +
|
| + <t>Optional parameters:</t>
|
| +
|
| + <t><list style="hanging">
|
| + <t hangText="maxplaybackrate:">
|
| + a hint about the maximum output sampling rate that the receiver is
|
| + capable of rendering in Hz.
|
| + The decoder MUST be capable of decoding
|
| + any audio bandwidth but due to hardware limitations only signals
|
| + up to the specified sampling rate can be played back. Sending signals
|
| + with higher audio bandwidth results in higher than necessary network
|
| + usage and encoding complexity, so an encoder SHOULD NOT encode
|
| + frequencies above the audio bandwidth specified by maxplaybackrate.
|
| + This parameter can take any value between 8000 and 48000, although
|
| + commonly the value will match one of the Opus bandwidths
|
| + (<xref target="bandwidth_definitions"/>).
|
| + By default, the receiver is assumed to have no limitations, i.e. 48000.
|
| + <vspace blankLines='1'/>
|
| + </t>
|
| +
|
| + <t hangText="sprop-maxcapturerate:">
|
| + a hint about the maximum input sampling rate that the sender is likely to produce.
|
| + This is not a guarantee that the sender will never send any higher bandwidth
|
| + (e.g. it could send a pre-recorded prompt that uses a higher bandwidth), but it
|
| + indicates to the receiver that frequencies above this maximum can safely be discarded.
|
| + This parameter is useful to avoid wasting receiver resources by operating the audio
|
| + processing pipeline (e.g. echo cancellation) at a higher rate than necessary.
|
| + This parameter can take any value between 8000 and 48000, although
|
| + commonly the value will match one of the Opus bandwidths
|
| + (<xref target="bandwidth_definitions"/>).
|
| + By default, the sender is assumed to have no limitations, i.e. 48000.
|
| + <vspace blankLines='1'/>
|
| + </t>
|
| +
|
| + <t hangText="maxptime:"> the maximum duration of media represented
|
| + by a packet (according to Section 6 of
|
| + <xref target="RFC4566"/>) that a decoder wants to receive, in
|
| + milliseconds rounded up to the next full integer value.
|
| + Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
|
| + multiple of an Opus frame size rounded up to the next full integer
|
| + value, up to a maximum value of 120, as
|
| + defined in <xref target='opus-rtp-payload-format'/>. If no value is
|
| + specified, the default is 120.
|
| + <vspace blankLines='1'/></t>
|
| +
|
| + <t hangText="ptime:"> the preferred duration of media represented
|
| + by a packet (according to Section 6 of
|
| + <xref target="RFC4566"/>) that a decoder wants to receive, in
|
| + milliseconds rounded up to the next full integer value.
|
| + Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
|
| + multiple of an Opus frame size rounded up to the next full integer
|
| + value, up to a maximum value of 120, as defined in <xref
|
| + target='opus-rtp-payload-format'/>. If no value is
|
| + specified, the default is 20.
|
| + <vspace blankLines='1'/></t>
|
| +
|
| + <t hangText="maxaveragebitrate:"> specifies the maximum average
|
| + receive bitrate of a session in bits per second (b/s). The actual
|
| + value of the bitrate can vary, as it is dependent on the
|
| + characteristics of the media in a packet. Note that the maximum
|
| + average bitrate MAY be modified dynamically during a session. Any
|
| + positive integer is allowed, but values outside the range
|
| + 6000 to 510000 SHOULD be ignored. If no value is specified, the
|
| + maximum value specified in <xref target='bitrate_by_bandwidth'/>
|
| + for the corresponding mode of Opus and corresponding maxplaybackrate
|
| + is the default.<vspace blankLines='1'/></t>
|
| +
|
| + <t hangText="stereo:">
|
| + specifies whether the decoder prefers receiving stereo or mono signals.
|
| + Possible values are 1 and 0 where 1 specifies that stereo signals are preferred,
|
| + and 0 specifies that only mono signals are preferred.
|
| + Independent of the stereo parameter every receiver MUST be able to receive and
|
| + decode stereo signals but sending stereo signals to a receiver that signaled a
|
| + preference for mono signals may result in higher than necessary network
|
| + utilization and encoding complexity. If no value is specified,
|
| + the default is 0 (mono).<vspace blankLines='1'/>
|
| + </t>
|
| +
|
| + <t hangText="sprop-stereo:">
|
| + specifies whether the sender is likely to produce stereo audio.
|
| + Possible values are 1 and 0, where 1 specifies that stereo signals are likely to
|
| + be sent, and 0 specifies that the sender will likely only send mono.
|
| + This is not a guarantee that the sender will never send stereo audio
|
| + (e.g. it could send a pre-recorded prompt that uses stereo), but it
|
| + indicates to the receiver that the received signal can be safely downmixed to mono.
|
| + This parameter is useful to avoid wasting receiver resources by operating the audio
|
| + processing pipeline (e.g. echo cancellation) in stereo when not necessary.
|
| + If no value is specified, the default is 0
|
| + (mono).<vspace blankLines='1'/>
|
| + </t>
|
| +
|
| + <t hangText="cbr:">
|
| + specifies if the decoder prefers the use of a constant bitrate versus
|
| + variable bitrate. Possible values are 1 and 0, where 1 specifies constant
|
| + bitrate and 0 specifies variable bitrate. If no value is specified,
|
| + the default is 0 (vbr). When cbr is 1, the maximum average bitrate can still
|
| + change, e.g. to adapt to changing network conditions.<vspace blankLines='1'/>
|
| + </t>
|
| +
|
| + <t hangText="useinbandfec:"> specifies that the decoder has the capability to
|
| + take advantage of the Opus in-band FEC. Possible values are 1 and 0.
|
| + Providing 0 when FEC cannot be used on the receiving side is
|
| + RECOMMENDED. If no
|
| + value is specified, useinbandfec is assumed to be 0.
|
| + This parameter is only a preference and the receiver MUST be able to process
|
| + packets that include FEC information, even if it means the FEC part is discarded.
|
| + <vspace blankLines='1'/></t>
|
| +
|
| + <t hangText="usedtx:"> specifies if the decoder prefers the use of
|
| + DTX. Possible values are 1 and 0. If no value is specified, the
|
| + default is 0.<vspace blankLines='1'/></t>
|
| + </list></t>
|
| +
|
| + <t>Encoding considerations:<vspace blankLines='1'/></t>
|
| + <t><list style="hanging">
|
| + <t>The Opus media type is framed and consists of binary data according
|
| + to Section 4.8 in <xref target="RFC6838"/>.</t>
|
| + </list></t>
|
| +
|
| + <t>Security considerations: </t>
|
| + <t><list style="hanging">
|
| + <t>See <xref target='security-considerations'/> of this document.</t>
|
| + </list></t>
|
| +
|
| + <t>Interoperability considerations: none<vspace blankLines='1'/></t>
|
| + <t>Published specification: RFC [XXXX]</t>
|
| + <t>Note to the RFC Editor: Replace [XXXX] with the number of the published
|
| + RFC.<vspace blankLines='1'/></t>
|
| +
|
| + <t>Applications that use this media type: </t>
|
| + <t><list style="hanging">
|
| + <t>Any application that requires the transport of
|
| + speech or audio data can use this media type. Some examples are,
|
| + but not limited to, audio and video conferencing, Voice over IP,
|
| + media streaming.</t>
|
| + </list></t>
|
| +
|
| + <t>Fragment identifier considerations: N/A<vspace blankLines='1'/></t>
|
| +
|
| + <t>Person & email address to contact for further information:</t>
|
| + <t><list style="hanging">
|
| + <t>SILK Support silksupport@skype.net</t>
|
| + <t>Jean-Marc Valin jmvalin@jmvalin.ca</t>
|
| + </list></t>
|
| +
|
| + <t>Intended usage: COMMON<vspace blankLines='1'/></t>
|
| +
|
| + <t>Restrictions on usage:<vspace blankLines='1'/></t>
|
| +
|
| + <t><list style="hanging">
|
| + <t>For transfer over RTP, the RTP payload format (<xref
|
| + target='opus-rtp-payload-format'/> of this document) SHALL be
|
| + used.</t>
|
| + </list></t>
|
| +
|
| + <t>Author:</t>
|
| + <t><list style="hanging">
|
| + <t>Julian Spittka jspittka@gmail.com<vspace blankLines='1'/></t>
|
| + <t>Koen Vos koenvos74@gmail.com<vspace blankLines='1'/></t>
|
| + <t>Jean-Marc Valin jmvalin@jmvalin.ca<vspace blankLines='1'/></t>
|
| + </list></t>
|
| +
|
| + <t> Change controller: IETF Payload Working Group delegated from the IESG</t>
|
| + </section>
|
| + </section>
|
| +
|
| + <section title='SDP Considerations'>
|
| + <t>The information described in the media type specification has a
|
| + specific mapping to fields in the Session Description Protocol (SDP)
|
| + <xref target="RFC4566"/>, which is commonly used to describe RTP
|
| + sessions. When SDP is used to specify sessions employing Opus,
|
| + the mapping is as follows:</t>
|
| +
|
| + <t>
|
| + <list style="symbols">
|
| + <t>The media type ("audio") goes in SDP "m=" as the media name.</t>
|
| +
|
| + <t>The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding
|
| + name. The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of
|
| + channels MUST be 2.</t>
|
| +
|
| + <t>The OPTIONAL media type parameters "ptime" and "maxptime" are
|
| + mapped to "a=ptime" and "a=maxptime" attributes, respectively, in the
|
| + SDP.</t>
|
| +
|
| + <t>The OPTIONAL media type parameters "maxaveragebitrate",
|
| + "maxplaybackrate", "stereo", "cbr", "useinbandfec", and
|
| + "usedtx", when present, MUST be included in the "a=fmtp" attribute
|
| + in the SDP, expressed as a media type string in the form of a
|
| + semicolon-separated list of parameter=value pairs (e.g.,
|
| + maxplaybackrate=48000). They MUST NOT be specified in an
|
| + SSRC-specific "fmtp" source-level attribute (as defined in
|
| + Section 6.3 of <xref target="RFC5576"/>).</t>
|
| +
|
| + <t>The OPTIONAL media type parameters "sprop-maxcapturerate",
|
| + and "sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by
|
| + copying them directly from the media type parameter string as part
|
| + of the semicolon-separated list of parameter=value pairs (e.g.,
|
| + sprop-stereo=1). These same OPTIONAL media type parameters MAY also
|
| + be specified using an SSRC-specific "fmtp" source-level attribute
|
| + as described in Section 6.3 of <xref target="RFC5576"/>.
|
| + They MAY be specified in both places, in which case the parameter
|
| + in the source-level attribute overrides the one found on the
|
| + "a=fmtp" line. The value of any parameter which is not specified in
|
| + a source-level source attribute MUST be taken from the "a=fmtp"
|
| + line, if it is present there.</t>
|
| +
|
| + </list>
|
| + </t>
|
| +
|
| + <t>Below are some examples of SDP session descriptions for Opus:</t>
|
| +
|
| + <t>Example 1: Standard mono session with 48000 Hz clock rate</t>
|
| + <figure>
|
| + <artwork>
|
| + <![CDATA[
|
| + m=audio 54312 RTP/AVP 101
|
| + a=rtpmap:101 opus/48000/2
|
| + ]]>
|
| + </artwork>
|
| + </figure>
|
| +
|
| +
|
| + <t>Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
|
| + recommended packet size of 40 ms, maximum average bitrate of 20000 bps,
|
| + prefers to receive stereo but only plans to send mono, FEC is desired,
|
| + DTX is not desired</t>
|
| +
|
| + <figure>
|
| + <artwork>
|
| + <![CDATA[
|
| + m=audio 54312 RTP/AVP 101
|
| + a=rtpmap:101 opus/48000/2
|
| + a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000;
|
| + maxaveragebitrate=20000; stereo=1; useinbandfec=1; usedtx=0
|
| + a=ptime:40
|
| + a=maxptime:40
|
| + ]]>
|
| + </artwork>
|
| + </figure>
|
| +
|
| + <t>Example 3: Two-way full-band stereo preferred</t>
|
| +
|
| + <figure>
|
| + <artwork>
|
| + <![CDATA[
|
| + m=audio 54312 RTP/AVP 101
|
| + a=rtpmap:101 opus/48000/2
|
| + a=fmtp:101 stereo=1; sprop-stereo=1
|
| + ]]>
|
| + </artwork>
|
| + </figure>
|
| +
|
| +
|
| + <section title='SDP Offer/Answer Considerations'>
|
| +
|
| + <t>When using the offer-answer procedure described in <xref
|
| + target="RFC3264"/> to negotiate the use of Opus, the following
|
| + considerations apply:</t>
|
| +
|
| + <t><list style="symbols">
|
| +
|
| + <t>Opus supports several clock rates. For signaling purposes only
|
| + the highest, i.e. 48000, is used. The actual clock rate of the
|
| + corresponding media is signaled inside the payload and is not
|
| + restricted by this payload format description. The decoder MUST be
|
| + capable of decoding every received clock rate. An example
|
| + is shown below:
|
| +
|
| + <figure>
|
| + <artwork>
|
| + <![CDATA[
|
| + m=audio 54312 RTP/AVP 100
|
| + a=rtpmap:100 opus/48000/2
|
| + ]]>
|
| + </artwork>
|
| + </figure>
|
| + </t>
|
| +
|
| + <t>The "ptime" and "maxptime" parameters are unidirectional
|
| + receive-only parameters and typically will not compromise
|
| + interoperability; however, some values might cause application
|
| + performance to suffer. <xref
|
| + target="RFC3264"/> defines the SDP offer-answer handling of the
|
| + "ptime" parameter. The "maxptime" parameter MUST be handled in the
|
| + same way.</t>
|
| +
|
| + <t>
|
| + The "maxplaybackrate" parameter is a unidirectional receive-only
|
| + parameter that reflects limitations of the local receiver. When
|
| + sending to a single destination, a sender MUST NOT use an audio
|
| + bandwidth higher than necessary to make full use of audio sampled at
|
| + a sampling rate of "maxplaybackrate". Gateways or senders that
|
| + are sending the same encoded audio to multiple destinations
|
| + SHOULD NOT use an audio bandwidth higher than necessary to
|
| + represent audio sampled at "maxplaybackrate", as this would lead
|
| + to inefficient use of network resources.
|
| + The "maxplaybackrate" parameter does not
|
| + affect interoperability. Also, this parameter SHOULD NOT be used
|
| + to adjust the audio bandwidth as a function of the bitrate, as this
|
| + is the responsibility of the Opus encoder implementation.
|
| + </t>
|
| +
|
| + <t>The "maxaveragebitrate" parameter is a unidirectional receive-only
|
| + parameter that reflects limitations of the local receiver. The sender
|
| + of the other side MUST NOT send with an average bitrate higher than
|
| + "maxaveragebitrate" as it might overload the network and/or
|
| + receiver. The "maxaveragebitrate" parameter typically will not
|
| + compromise interoperability; however, some values might cause
|
| + application performance to suffer, and ought to be set with
|
| + care.</t>
|
| +
|
| + <t>The "sprop-maxcapturerate" and "sprop-stereo" parameters are
|
| + unidirectional sender-only parameters that reflect limitations of
|
| + the sender side.
|
| + They allow the receiver to set up a reduced-complexity audio
|
| + processing pipeline if the sender is not planning to use the full
|
| + range of Opus's capabilities.
|
| + Neither "sprop-maxcapturerate" nor "sprop-stereo" affect
|
| + interoperability and the receiver MUST be capable of receiving any signal.
|
| + </t>
|
| +
|
| + <t>
|
| + The "stereo" parameter is a unidirectional receive-only
|
| + parameter. When sending to a single destination, a sender MUST
|
| + NOT use stereo when "stereo" is 0. Gateways or senders that are
|
| + sending the same encoded audio to multiple destinations SHOULD
|
| + NOT use stereo when "stereo" is 0, as this would lead to
|
| + inefficient use of network resources. The "stereo" parameter does
|
| + not affect interoperability.
|
| + </t>
|
| +
|
| + <t>
|
| + The "cbr" parameter is a unidirectional receive-only
|
| + parameter.
|
| + </t>
|
| +
|
| + <t>The "useinbandfec" parameter is a unidirectional receive-only
|
| + parameter.</t>
|
| +
|
| + <t>The "usedtx" parameter is a unidirectional receive-only
|
| + parameter.</t>
|
| +
|
| + <t>Any unknown parameter in an offer MUST be ignored by the receiver
|
| + and MUST be removed from the answer.</t>
|
| +
|
| + </list></t>
|
| +
|
| + <t>
|
| + The Opus parameters in an SDP Offer/Answer exchange are completely
|
| + orthogonal, and there is no relationship between the SDP Offer and
|
| + the Answer.
|
| + </t>
|
| + </section>
|
| +
|
| + <section title='Declarative SDP Considerations for Opus'>
|
| +
|
| + <t>For declarative use of SDP such as in Session Announcement Protocol
|
| + (SAP), <xref target="RFC2974"/>, and RTSP, <xref target="RFC2326"/>, for
|
| + Opus, the following needs to be considered:</t>
|
| +
|
| + <t><list style="symbols">
|
| +
|
| + <t>The values for "maxptime", "ptime", "maxplaybackrate", and
|
| + "maxaveragebitrate" ought to be selected carefully to ensure that a
|
| + reasonable performance can be achieved for the participants of a session.</t>
|
| +
|
| + <t>
|
| + The values for "maxptime", "ptime", and of the payload
|
| + format configuration are recommendations by the decoding side to ensure
|
| + the best performance for the decoder.
|
| + </t>
|
| +
|
| + <t>All other parameters of the payload format configuration are declarative
|
| + and a participant MUST use the configurations that are provided for
|
| + the session. More than one configuration can be provided if necessary
|
| + by declaring multiple RTP payload types; however, the number of types
|
| + ought to be kept small.</t>
|
| + </list></t>
|
| + </section>
|
| + </section>
|
| +
|
| + <section title='Security Considerations' anchor='security-considerations'>
|
| +
|
| + <t>Use of variable bitrate (VBR) is subject to the security considerations in
|
| + <xref target="RFC6562"/>.</t>
|
| +
|
| + <t>RTP packets using the payload format defined in this specification
|
| + are subject to the security considerations discussed in the RTP
|
| + specification <xref target="RFC3550"/>, and in any applicable RTP profile such as
|
| + RTP/AVP <xref target="RFC3551"/>, RTP/AVPF <xref target="RFC4585"/>,
|
| + RTP/SAVP <xref target="RFC3711"/> or RTP/SAVPF <xref target="RFC5124"/>.
|
| + However, as "Securing the RTP Protocol Framework:
|
| + Why RTP Does Not Mandate a Single Media Security Solution"
|
| + <xref target="RFC7202"/> discusses, it is not an RTP payload
|
| + format's responsibility to discuss or mandate what solutions are used
|
| + to meet the basic security goals like confidentiality, integrity and
|
| + source authenticity for RTP in general. This responsibility lays on
|
| + anyone using RTP in an application. They can find guidance on
|
| + available security mechanisms and important considerations in Options
|
| + for Securing RTP Sessions [I-D.ietf-avtcore-rtp-security-options].
|
| + Applications SHOULD use one or more appropriate strong security
|
| + mechanisms.</t>
|
| +
|
| + <t>This payload format and the Opus encoding do not exhibit any
|
| + significant non-uniformity in the receiver-end computational load and thus
|
| + are unlikely to pose a denial-of-service threat due to the receipt of
|
| + pathological datagrams.</t>
|
| + </section>
|
| +
|
| + <section title='Acknowledgements'>
|
| + <t>Many people have made useful comments and suggestions contributing to this document.
|
| + In particular, we would like to thank
|
| + Tina le Grand, Cullen Jennings, Jonathan Lennox, Gregory Maxwell, Colin Perkins, Jan Skoglund,
|
| + Timothy B. Terriberry, Martin Thompson, Justin Uberti, Magnus Westerlund, and Mo Zanaty.</t>
|
| + </section>
|
| + </middle>
|
| +
|
| + <back>
|
| + <references title="Normative References">
|
| + &rfc2119;
|
| + &rfc3389;
|
| + &rfc3550;
|
| + &rfc3711;
|
| + &rfc3551;
|
| + &rfc6838;
|
| + &rfc4855;
|
| + &rfc4566;
|
| + &rfc3264;
|
| + &rfc2326;
|
| + &rfc5576;
|
| + &rfc6562;
|
| + &rfc6716;
|
| + </references>
|
| +
|
| + <references title="Informative References">
|
| + &rfc2974;
|
| + &rfc4585;
|
| + &rfc5124;
|
| + &rfc5405;
|
| + &rfc7202;
|
| +
|
| + <reference anchor='rmcat' target='https://datatracker.ietf.org/wg/rmcat/documents/'>
|
| + <front>
|
| + <title>rmcat documents</title>
|
| + <author/>
|
| + <date/>
|
| + <abstract>
|
| + <t></t>
|
| + </abstract></front>
|
| + </reference>
|
| +
|
| +
|
| + </references>
|
| +
|
| + </back>
|
| +</rfc>
|
|
|
Chromium Code Reviews
Issue 1612443002:
Create local copy of Opus v1.1.2
Base URL: https://chromium.googlesource.com/external/webrtc.git@master