From a7455ca26f725255857942ad7fbb88322ab16ee4 Mon Sep 17 00:00:00 2001 From: Jean-Marc Valin Date: Fri, 24 Sep 2010 20:23:39 -0400 Subject: Name changed to "Harmony" --- doc/draft-ietf-codec-description.xml | 391 ----------------------------------- doc/draft-ietf-codec-harmony.xml | 391 +++++++++++++++++++++++++++++++++++ 2 files changed, 391 insertions(+), 391 deletions(-) delete mode 100644 doc/draft-ietf-codec-description.xml create mode 100644 doc/draft-ietf-codec-harmony.xml (limited to 'doc') diff --git a/doc/draft-ietf-codec-description.xml b/doc/draft-ietf-codec-description.xml deleted file mode 100644 index 27203810..00000000 --- a/doc/draft-ietf-codec-description.xml +++ /dev/null @@ -1,391 +0,0 @@ - - - - - - - -Definition of the IETF Interactive Audio Codec - - - -Octasic Inc. -
- -4101, Molson Street -Montreal -Quebec - -Canada - -+1 514 282-8858 -jean-marc.valin@octasic.com -
- - - -Skype Technologies S.A. -
- -Stadsgaarden 6 -Stockholm - -11645 -SE - -+46 855 921 989 -koen.vos@skype.net -
- - - - - -General - - - - - -This document describes a codec designed for interactive speech and audio -transmission over the Internet. - - - - - - -
- -We propose a hybrid codec based on a linear prediction layer (LP) and an -MDCT-based enhancement layer. The main idea behind the proposal is that -the speech low frequencies are usually more efficiently coded using -linear prediction codecs (such as CELP variants), while the higher frequencies -are more efficiently coded in the transform domain (e.g. MDCT). For low -sampling rates, the MDCT layer is not useful and only the LP-based layer is -used. On the other hand, non-speech signals are not always adequately coded -using linear prediction, so for music only the MDCT-based layer is used. - - - -In this proposed prototype, the LP layer is based on the -SILK codec - and the MDCT layer is based on the -CELT codec - . - - -This is a work in progress. -
- -
- - -In hybrid mode, each frame is coded first by the LP layer and then by the MDCT -layer. In the current prototype, the cutoff frequency is 8 kHz. In the MDCT -layer, all bands below 8 kHz are discarded, such that there is no coding -redundancy between the two layers. Also both layers use the same instance of -the range coder to encode the signal, which ensures that no "padding bits" are -wasted. The hybrid approach makes it easy to support both constant bit-rate -(CBR) and varaible bit-rate (VBR) coding. Although the SILK layer used is VBR, -it is easy to make the bit allocation of the CELT layer produce a final stream -that is CBR by using all the bits left unused by the SILK layer. - - -The implementation of SILK-based LP layer is similar to the description in -the SILK Internet-Draft with the main exception that -SILK was modified to -use the same range coder as CELT. The implementation of the CELT-based MDCT -layer is available from the CELT website and is a more recent version (0.8.1) -of the CELT Internet-Draft. -The main changes -include better support for 20 ms frames as well as the ability to encode -only the higher bands using a range coder partially filled by the SILK layer. - - -In addition to their frame size, the SILK and CELT codecs require -a look-ahead of 5.2 ms and 2.5 ms, respectively. SILK's look-ahead is due to -noise shaping estimation (5 ms) and the internal resampling (0.2 ms), while -CELT's look-ahead is due to the overlapping MDCT windows. To compensate for the -difference, the CELT encoder input is delayed by 2.7 ms. This ensures that low -frequencies and high frequencies arrive at the same time. - - - -
- -The source code is currently available in a -Git repository -which references two other -repositories (for SILK and CELT). Some snapshots are provided for -convenience at along -with sample files. -Although the build system is very primitive, some instructions are provided -in the toplevel README file. -This is very early development so both the quality and feature set should -greatly improve over time. In the current version, only 48 kHz audio is -supported, but support for all configurations listed in - is planned. - -
- -
- -
- -There are three possible operating modes for the proposed prototype: - -A linear prediction (LP) mode for use in low bit-rate connections with up to 8 kHz audio bandwidth (16 kHz sampling rate) -A hybrid (LP+MDCT) mode for full-bandwidth speech at medium bitrates -An MDCT-only mode for very low delay speech transmission as well as music transmission. - -Each of these modes supports a number of difference frame sizes and sampling -rates. In order to distinguish between the various modes and configurations, -we need to define a simple header that can used in the transport layer -(e.g RTP) to signal this information. The following describes the proposed -header. - - - -The LP mode supports the following configurations (numbered from 00000...01011 in binary): - -8 kHz: 10, 20, 40, 60 ms (00000...00011) -12 kHz: 10, 20, 40, 60 ms (00100...00111) -16 kHz: 10, 20, 40, 60 ms (01000...01011) - -for a total of 12 configurations. - - - -The hybrid mode supports the following configurations (numbered from 01100...01111): - -32 kHz: 10, 20 ms (01100...01101) -48 kHz: 10, 20 ms (01110...01111) - -for a total of 4 configurations. - - - -The MDCT-only mode supports the following configurations (numbered from 10000...11101): - -8 kHz: 2.5, 5, 10, 20 ms (10000...10011) -16 kHz: 2.5, 5, 10, 20 ms (10100...10111) -32 kHz: 2.5, 5, 10, 20 ms (11000...11011) -48 kHz: 2.5, 5, 10, 20 ms (11100...11111) - -for a total of 16 configurations. - - - -There is thus a total of 32 configurations, so 5 bits are necessary to -indicate the mode, frame size and sampling rate (MFS). This leaves 3 bits for the number of frames per packets (codes 0 to 7): - -0-2: 1-3 frames in the packet, each with equal compressed size -3: arbitrary number of frames in the packet, each with equal compressed size (one size needs to be encoded) -4-5: 2-3 frames in the packet, with different compressed sizes, which need to be encoded (except the last one) -6: arbitrary number of frames in the packet, with different compressed sizes, each of which needs to be encoded -7: The first frame has this MFS, but others have different MFS. Each compressed size needs to be encoded. - -When code 7 is used and the last frames of a packet have the same MFS, it is -allowed to switch to another code for them. - - - -The compressed size of the frames (if needed) is indicated -- usually -- with one byte, with the following meaning: - -0: No frame (DTX or lost packet) -1-251: Size of the frame in bytes -252-255: A second byte is needed. The total size is (size[1]*4)+(size[0]%4)+252 - - - - -The maximum size representable is 255*4+3+252=1275 bytes. For 20 ms frames, that -represents a bit-rate of 510 kb/s, which is really the highest rate anyone would want -to use in stereo mode (beyond that point, lossless codecs would be more appropriate). - - -
- -Simplest case: one packet - - - -
- -
- - - -Four frames of the same compressed size: - - - -
- -
- - - -Two frames of different compressed size: - - - -
- -
- - - -Three frames of different durations: - - - - -
- -
- -
- - -
- -
- - -The codec needs to take appropriate security considerations -into account, as outlined in and . -It is extremely important for the decoder to be robust against malicious -payloads. Malicious payloads must not cause the decoder to overrun its -allocated memory or to take much more resources to decode. Although problems -in encoders are typically rarer, the same applies to the encoder. Malicious -audio stream must not cause the encoder to misbehave because this would -allow an attacker to attack transcoding gateways. - - -In its current version, this prototype codec likely does NOT meet these -security considerations, so it should be used with caution. - -
- - -
- -This document has no actions for IANA. - -
- -
- -Thanks to all other developers, including Soeren Skak Jensen, Gregory Maxwell, -Christopher Montgomery, Karsten Vandborg Soerensen, and Timothy Terriberry. - -
- - - - - - - - - -SILK Speech Codec - - - - - - - - - - - - - - - - -Constrained-Energy Lapped Transform (CELT) Codec - - - - - - - - - - - - - - - - - - -Internet Denial-of-Service Considerations - - - - - -IAB - - -This document provides an overview of possible avenues for denial-of-service (DoS) attack on Internet systems. The aim is to encourage protocol designers and network engineers towards designs that are more robust. We discuss partial solutions that reduce the effectiveness of attacks, and how some solutions might inadvertently open up alternative vulnerabilities. This memo provides information for the Internet community. - - - - - - -Guidelines for Writing RFC Text on Security Considerations - - - - - - -All RFCs are required to have a Security Considerations section. Historically, such sections have been relatively weak. This document provides guidelines to RFC authors on how to write a good Security Considerations section. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. - - - - - - - - - - - - diff --git a/doc/draft-ietf-codec-harmony.xml b/doc/draft-ietf-codec-harmony.xml new file mode 100644 index 00000000..2d938ecd --- /dev/null +++ b/doc/draft-ietf-codec-harmony.xml @@ -0,0 +1,391 @@ + + + + + + + +Definition of the Harmony Audio Codec + + + +Octasic Inc. +
+ +4101, Molson Street +Montreal +Quebec + +Canada + ++1 514 282-8858 +jean-marc.valin@octasic.com +
+ + + +Skype Technologies S.A. +
+ +Stadsgaarden 6 +Stockholm + +11645 +SE + ++46 855 921 989 +koen.vos@skype.net +
+ + + + + +General + + + + + +This document describes the Harmony codec, designed for interactive speech and audio +transmission over the Internet. + + + + + + +
+ +We propose the Harmony codec based on a linear prediction layer (LP) and an +MDCT-based enhancement layer. The main idea behind the proposal is that +the speech low frequencies are usually more efficiently coded using +linear prediction codecs (such as CELP variants), while the higher frequencies +are more efficiently coded in the transform domain (e.g. MDCT). For low +sampling rates, the MDCT layer is not useful and only the LP-based layer is +used. On the other hand, non-speech signals are not always adequately coded +using linear prediction, so for music only the MDCT-based layer is used. + + + +In this proposed prototype, the LP layer is based on the +SILK codec + and the MDCT layer is based on the +CELT codec + . + + +This is a work in progress. +
+ +
+ + +In hybrid mode, each frame is coded first by the LP layer and then by the MDCT +layer. In the current prototype, the cutoff frequency is 8 kHz. In the MDCT +layer, all bands below 8 kHz are discarded, such that there is no coding +redundancy between the two layers. Also both layers use the same instance of +the range coder to encode the signal, which ensures that no "padding bits" are +wasted. The hybrid approach makes it easy to support both constant bit-rate +(CBR) and varaible bit-rate (VBR) coding. Although the SILK layer used is VBR, +it is easy to make the bit allocation of the CELT layer produce a final stream +that is CBR by using all the bits left unused by the SILK layer. + + +The implementation of SILK-based LP layer is similar to the description in +the SILK Internet-Draft with the main exception that +SILK was modified to +use the same range coder as CELT. The implementation of the CELT-based MDCT +layer is available from the CELT website and is a more recent version (0.8.1) +of the CELT Internet-Draft. +The main changes +include better support for 20 ms frames as well as the ability to encode +only the higher bands using a range coder partially filled by the SILK layer. + + +In addition to their frame size, the SILK and CELT codecs require +a look-ahead of 5.2 ms and 2.5 ms, respectively. SILK's look-ahead is due to +noise shaping estimation (5 ms) and the internal resampling (0.2 ms), while +CELT's look-ahead is due to the overlapping MDCT windows. To compensate for the +difference, the CELT encoder input is delayed by 2.7 ms. This ensures that low +frequencies and high frequencies arrive at the same time. + + + +
+ +The source code is currently available in a +Git repository +which references two other +repositories (for SILK and CELT). Some snapshots are provided for +convenience at along +with sample files. +Although the build system is very primitive, some instructions are provided +in the toplevel README file. +This is very early development so both the quality and feature set should +greatly improve over time. In the current version, only 48 kHz audio is +supported, but support for all configurations listed in + is planned. + +
+ +
+ +
+ +There are three possible operating modes for the proposed prototype: + +A linear prediction (LP) mode for use in low bit-rate connections with up to 8 kHz audio bandwidth (16 kHz sampling rate) +A hybrid (LP+MDCT) mode for full-bandwidth speech at medium bitrates +An MDCT-only mode for very low delay speech transmission as well as music transmission. + +Each of these modes supports a number of difference frame sizes and sampling +rates. In order to distinguish between the various modes and configurations, +we need to define a simple header that can used in the transport layer +(e.g RTP) to signal this information. The following describes the proposed +header. + + + +The LP mode supports the following configurations (numbered from 00000...01011 in binary): + +8 kHz: 10, 20, 40, 60 ms (00000...00011) +12 kHz: 10, 20, 40, 60 ms (00100...00111) +16 kHz: 10, 20, 40, 60 ms (01000...01011) + +for a total of 12 configurations. + + + +The hybrid mode supports the following configurations (numbered from 01100...01111): + +32 kHz: 10, 20 ms (01100...01101) +48 kHz: 10, 20 ms (01110...01111) + +for a total of 4 configurations. + + + +The MDCT-only mode supports the following configurations (numbered from 10000...11101): + +8 kHz: 2.5, 5, 10, 20 ms (10000...10011) +16 kHz: 2.5, 5, 10, 20 ms (10100...10111) +32 kHz: 2.5, 5, 10, 20 ms (11000...11011) +48 kHz: 2.5, 5, 10, 20 ms (11100...11111) + +for a total of 16 configurations. + + + +There is thus a total of 32 configurations, so 5 bits are necessary to +indicate the mode, frame size and sampling rate (MFS). This leaves 3 bits for the number of frames per packets (codes 0 to 7): + +0-2: 1-3 frames in the packet, each with equal compressed size +3: arbitrary number of frames in the packet, each with equal compressed size (one size needs to be encoded) +4-5: 2-3 frames in the packet, with different compressed sizes, which need to be encoded (except the last one) +6: arbitrary number of frames in the packet, with different compressed sizes, each of which needs to be encoded +7: The first frame has this MFS, but others have different MFS. Each compressed size needs to be encoded. + +When code 7 is used and the last frames of a packet have the same MFS, it is +allowed to switch to another code for them. + + + +The compressed size of the frames (if needed) is indicated -- usually -- with one byte, with the following meaning: + +0: No frame (DTX or lost packet) +1-251: Size of the frame in bytes +252-255: A second byte is needed. The total size is (size[1]*4)+(size[0]%4)+252 + + + + +The maximum size representable is 255*4+3+252=1275 bytes. For 20 ms frames, that +represents a bit-rate of 510 kb/s, which is really the highest rate anyone would want +to use in stereo mode (beyond that point, lossless codecs would be more appropriate). + + +
+ +Simplest case: one packet + + + +
+ +
+ + + +Four frames of the same compressed size: + + + +
+ +
+ + + +Two frames of different compressed size: + + + +
+ +
+ + + +Three frames of different durations: + + + + +
+ +
+ +
+ + +
+ +
+ + +The codec needs to take appropriate security considerations +into account, as outlined in and . +It is extremely important for the decoder to be robust against malicious +payloads. Malicious payloads must not cause the decoder to overrun its +allocated memory or to take much more resources to decode. Although problems +in encoders are typically rarer, the same applies to the encoder. Malicious +audio stream must not cause the encoder to misbehave because this would +allow an attacker to attack transcoding gateways. + + +In its current version, the Harmony codec likely does NOT meet these +security considerations, so it should be used with caution. + +
+ + +
+ +This document has no actions for IANA. + +
+ +
+ +Thanks to all other developers, including Soeren Skak Jensen, Gregory Maxwell, +Christopher Montgomery, Karsten Vandborg Soerensen, and Timothy Terriberry. + +
+ + + + + + + + + +SILK Speech Codec + + + + + + + + + + + + + + + + +Constrained-Energy Lapped Transform (CELT) Codec + + + + + + + + + + + + + + + + + + +Internet Denial-of-Service Considerations + + + + + +IAB + + +This document provides an overview of possible avenues for denial-of-service (DoS) attack on Internet systems. The aim is to encourage protocol designers and network engineers towards designs that are more robust. We discuss partial solutions that reduce the effectiveness of attacks, and how some solutions might inadvertently open up alternative vulnerabilities. This memo provides information for the Internet community. + + + + + + +Guidelines for Writing RFC Text on Security Considerations + + + + + + +All RFCs are required to have a Security Considerations section. Historically, such sections have been relatively weak. This document provides guidelines to RFC authors on how to write a good Security Considerations section. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. + + + + + + + + + + + + -- cgit v1.2.3