codecs

This document describes the bitstream format of Advanced Professional Video (APV) and its decoding process. APV is a professional video codec providing visually lossless compression mainly for recording and post production.

This document defines the Free Lossless Audio Codec (FLAC) format and its streamable subset. FLAC is designed to reduce the amount of computer storage space needed to store digital audio signals. It does this losslessly, i.e., it does so without losing information. FLAC is free in the sense that its specification is open and its reference implementation is open source. Compared to other lossless audio coding formats, FLAC is a format with low complexity and can be encoded and decoded with little computing resources. Decoding of FLAC has been implemented independently for many different platforms, and both encoding and decoding can be implemented without needing floating-point arithmetic.

This document describes the RTP payload format of the Secure Communication Interoperability Protocol (SCIP). SCIP is an application-layer protocol that provides end-to-end session establishment, payload encryption, packetization and de-packetization of media, and reliable transport. This document provides a globally available reference that can be used for the development of network equipment and procurement of services that support SCIP traffic. The intended audience is network security policymakers; network administrators, architects, and original equipment manufacturers (OEMs); procurement personnel; and government agency and commercial industry representatives.

This document describes the RTP payload format for the Tactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) speech coder. TSVCIS is a scalable narrowband voice coder supporting varying encoder data rates and fallbacks. It is implemented as an augmentation to the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder by conveying additional speech coder parameters to enhance voice quality. TSVCIS augmented speech data is processed in conjunction with its temporally matched Mixed Excitation Linear Prediction (MELP) 2400 speech data. The RTP packetization of TSVCIS and MELPe speech coder data is described in detail.

This document provides requirements for a video codec designed mainly for use over the Internet. In addition, this document describes an evaluation methodology for measuring the compression efficiency to determine whether or not the stated requirements have been fulfilled.

This document describes the RTP payload format for the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder. MELPe's three different speech encoding rates and sample frame sizes are supported. Comfort noise procedures and packet loss concealment are described in detail.

This document updates RFC 5104 by fixing a shortcoming in the specification language of the Codec Control Message Full Intra Request (FIR) description when using it with layered codecs. In particular, a decoder refresh point needs to be sent by a media sender when a FIR is received on any layer of the layered bitstream, regardless of whether those layers are being sent in a single or in multiple RTP flows. The other payload-specific feedback messages defined in RFC 5104 and RFC 4585 (which was updated by RFC 5506) have also been analyzed, and no corresponding shortcomings have been found.

This document outlines the audio codec and processing requirements for WebRTC endpoints.

This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network. It specifies the video processing that is required as well as video codecs and their parameters.

This document specifies Real-time Transport Protocol (RTP) payload formats to be used for the Enhanced Variable Rate Narrowband-Wideband Codec (EVRC-NW). Three media type registrations are included for EVRC-NW RTP payload formats. In addition, a file format is specified for transport of EVRC-NW speech data in storage mode applications such as email.

Several MIME type/subtype combinations exist that can contain different media formats. A receiving agent thus needs to examine the details of such media content to determine if the specific elements can be rendered given an available set of codecs. Especially when the end system has limited resources, or the connection to the end system has limited bandwidth, it is helpful to know from the Content- Type alone if the content can be rendered.

This document specifies two parameters, 'codecs' and 'profiles', that are used with various MIME types or type/subtype combinations to allow for unambiguous specification of the codecs employed by the media formats contained within, or the profile(s) of the overall container format. This document obsoletes RFC 4281; RFC 4281 defines the 'codecs' parameter, which this document retains in a backwards compatible manner with minor clarifications; the 'profiles' parameter is added by this document.

By labeling content with the specific codecs indicated to render the contained media, receiving systems can determine if the codecs are supported by the end system, and if not, can take appropriate action (such as rejecting the content, sending notification of the situation, transcoding the content to a supported type, fetching and installing the required codecs, further inspection to determine if it will be sufficient to support a subset of the indicated codecs, etc.).

Similarly, the profiles can provide an overall indication, to the receiver, of the specifications with which the content complies. This is an indication of the compatibility of the container format and its contents to some specification. The receiver may be able to work out the extent to which it can handle and render the content by examining to see which of the declared profiles it supports, and what they mean. [STANDARDS-TRACK]

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 packet and introduces redundancy for robustness against packet loss and bit errors. [STANDARDS-TRACK]

This document specifies the payload format for packetization of Global System for Mobile Communications Half Rate (GSM-HR) speech codec data into the Real-time Transport Protocol (RTP). The payload format supports transmission of multiple frames per payload and packet loss robustness methods using redundancy. [STANDARDS-TRACK]

This document describes the RTP payload format of a mU-law EMbedded Coder for Low-delay IP communication (UEMCLIP), an enhanced speech codec of ITU-T G.711. The bitstream has a scalable structure with an embedded u-law bitstream, also known as PCMU, thus providing a handy transcoding operation between narrowband and wideband speech. [STANDARDS-TRACK]

This document specifies the payload format for packetization of the G.719 full-band codec encoded audio signals into the Real-time Transport Protocol (RTP). The payload format supports transmission of multiple channels, multiple frames per payload, and interleaving. [STANDARDS-TRACK]

This document specifies Real-time Transport Protocol (RTP) payload formats to be used for the Enhanced Variable Rate Wideband Codec (EVRC-WB) and updates the media type registrations for EVRC-B codec. Several media type registrations are included for EVRC-WB RTP payload formats. In addition, a file format is specified for transport of EVRC-WB speech data in storage mode applications such as email. [STANDARDS-TRACK]

This document specifies a few extensions to the messages defined in the Audio-Visual Profile with Feedback (AVPF). They are helpful primarily in conversational multimedia scenarios where centralized multipoint functionalities are in use. However, some are also usable in smaller multicast environments and point-to-point calls.

The extensions discussed are messages related to the ITU-T Rec. H.271 Video Back Channel, Full Intra Request, Temporary Maximum Media Stream Bit Rate, and Temporal-Spatial Trade-off. [STANDARDS-TRACK]

This document updates the Enhanced Variable Rate Codec (EVRC) RTP payload formats defined in RFC 3558 with several enhancements and extensions. In particular, it defines support for the header-free and interleaved/bundled packet formats for the EVRC-B codec, a new compact bundled format for the EVRC and EVRC-B codecs, as well as discontinuous transmission (DTX) support for EVRC and EVRC-B-encoded speech transported via RTP. Voice over IP (VoIP) applications operating over low bandwidth dial-up and wireless networks require such enhancements for efficient use of the bandwidth. [STANDARDS-TRACK]

This document is an addendum to RFC 4348, which specifies the RTP payload format for the Variable-Rate Multimode Wideband (VMR-WB) speech codec. This document specifies some updates in RFC 4348 to enable support for the new operating mode of VMR-WB standard (i.e., VMR-WB mode 4). These updates do not affect the existing modes of VMR-WB already specified in RFC 4348.

The payload formats and their associated parameters, as well as all provisions, restrictions, use cases, features, etc., that are specified in RFC 4348 are applicable to the new operating mode with no exception. [STANDARDS-TRACK]

This memo specifies an RTP payload format for encapsulating Video Codec 1 (VC-1) compressed bit streams, as defined by the Society of Motion Picture and Television Engineers (SMPTE) standard, SMPTE 421M. SMPTE is the main standardizing body in the motion imaging industry, and the SMPTE 421M standard defines a compressed video bit stream format and decoding process for television. [STANDARDS-TRACK]

This document specifies a real-time transport protocol (RTP) payload format to be used for the Variable-Rate Multimode Wideband (VMR-WB) speech codec. The payload format is designed to be able to interoperate with existing VMR-WB transport formats on non-IP networks. A media type registration is included for VMR-WB RTP payload format.

VMR-WB is a variable-rate multimode wideband speech codec that has a number of operating modes, one of which is interoperable with AMR-WB (i.e., RFC 3267) audio codec at certain rates. Therefore, provisions have been made in this document to facilitate and simplify data packet exchange between VMR-WB and AMR-WB in the interoperable mode with no transcoding function involved. [STANDARDS-TRACK]

This document describes an Asynchronous Transfer Mode (ATM) package for the Media Gateway Control Protocol (MGCP). This package includes new Local Connection Options, ATM-specific events and signals, and ATM connection parameters. Also included is a description of codec and profile negotiation. It extends the MGCP that is currently being deployed in a number of products. Implementers should be aware of developments in the IETF Megaco Working Group and ITU SG16, which are currently working on a potential successor to this protocol. This memo provides information for the Internet community.

The purpose of this paper is to establish a mechanism by which codecs registered within Microsoft's WAVE and AVI Registries may be referenced within the IANA Namespace by Internet applications. This memo provides information for the Internet community. It does not specify an Internet standard of any kind.

The purpose of the Voice File Interchange Protocol (VFIP) is to permit the interchange of various types of speech files between different systems in the ARPA-Internet community. Suggestions for improvement are encouraged.