Internet Engineering Task Force (IETF) T. Terriberry Request for Comments: 7845 Mozilla Corporation Updates: 5334 R. Lee Category: Standards Track Voicetronix ISSN: 2070-1721 R. Giles Mozilla Corporation April 2016 Ogg Encapsulation for the Opus Audio Codec Abstract This document defines the Ogg encapsulation for the Opus interactive speech and audio codec. This allows data encoded in the Opus format to be stored in an Ogg logical bitstream. Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 5741. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc7845. Copyright Notice Copyright (c) 2016 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Packet Organization . . . . . . . . . . . . . . . . . . . . . 4 4. Granule Position . . . . . . . . . . . . . . . . . . . . . . 6 4.1. Repairing Gaps in Real-Time Streams . . . . . . . . . . . 7 4.2. Pre-skip . . . . . . . . . . . . . . . . . . . . . . . . 9 4.3. PCM Sample Position . . . . . . . . . . . . . . . . . . . 9 4.4. End Trimming . . . . . . . . . . . . . . . . . . . . . . 10 4.5. Restrictions on the Initial Granule Position . . . . . . 10 4.6. Seeking and Pre-roll . . . . . . . . . . . . . . . . . . 11 5. Header Packets . . . . . . . . . . . . . . . . . . . . . . . 12 5.1. Identification Header . . . . . . . . . . . . . . . . . . 12 5.1.1. Channel Mapping . . . . . . . . . . . . . . . . . . . 16 5.2. Comment Header . . . . . . . . . . . . . . . . . . . . . 22 5.2.1. Tag Definitions . . . . . . . . . . . . . . . . . . . 25 6. Packet Size Limits . . . . . . . . . . . . . . . . . . . . . 26 7. Encoder Guidelines . . . . . . . . . . . . . . . . . . . . . 27 7.1. LPC Extrapolation . . . . . . . . . . . . . . . . . . . . 28 7.2. Continuous Chaining . . . . . . . . . . . . . . . . . . . 28 8. Security Considerations . . . . . . . . . . . . . . . . . . . 29 9. Content Type . . . . . . . . . . . . . . . . . . . . . . . . 30 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 32 11.1. Normative References . . . . . . . . . . . . . . . . . . 32 11.2. Informative References . . . . . . . . . . . . . . . . . 33 Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 34 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35 1. Introduction The IETF Opus codec is a low-latency audio codec optimized for both voice and general-purpose audio. See [RFC6716] for technical details. This document defines the encapsulation of Opus in a continuous, logical Ogg bitstream [RFC3533]. Ogg encapsulation provides Opus with a long-term storage format supporting all of the essential features, including metadata, fast and accurate seeking, corruption detection, recapture after errors, low overhead, and the ability to multiplex Opus with other codecs (including video) with minimal buffering. It also provides a live streamable format capable of delivery over a reliable stream-oriented transport, without requiring all the data (or even the total length of the data) up-front, in a form that is identical to the on-disk storage format. Ogg bitstreams are made up of a series of "pages", each of which contains data from one or more "packets". Pages are the fundamental unit of multiplexing in an Ogg stream. Each page is associated with
a particular logical stream and contains a capture pattern and checksum, flags to mark the beginning and end of the logical stream, and a "granule position" that represents an absolute position in the stream, to aid seeking. A single page can contain up to 65,025 octets of packet data from up to 255 different packets. Packets can be split arbitrarily across pages and continued from one page to the next (allowing packets much larger than would fit on a single page). Each page contains "lacing values" that indicate how the data is partitioned into packets, allowing a demultiplexer (demuxer) to recover the packet boundaries without examining the encoded data. A packet is said to "complete" on a page when the page contains the final lacing value corresponding to that packet. This encapsulation defines the contents of the packet data, including the necessary headers, the organization of those packets into a logical stream, and the interpretation of the codec-specific granule position field. It does not attempt to describe or specify the existing Ogg container format. Readers unfamiliar with the basic concepts mentioned above are encouraged to review the details in [RFC3533]. 2. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119].
3. Packet Organization An Ogg Opus stream is organized as follows (see Figure 1 for an example). Page 0 Pages 1 ... n Pages (n+1) ... +------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +-- | | | | | | | | | | | | | |+----------+| |+-----------------+| |+-------------------+ +----- |||ID Header|| || Comment Header || ||Audio Data Packet 1| | ... |+----------+| |+-----------------+| |+-------------------+ +----- | | | | | | | | | | | | | +------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +-- ^ ^ ^ | | | | | Mandatory Page Break | | | ID header is contained on a single page | 'Beginning Of Stream' Figure 1: Example Packet Organization for a Logical Ogg Opus Stream There are two mandatory header packets. The first packet in the logical Ogg bitstream MUST contain the identification (ID) header, which uniquely identifies a stream as Opus audio. The format of this header is defined in Section 5.1. It is placed alone (without any other packet data) on the first page of the logical Ogg bitstream and completes on that page. This page has its 'beginning of stream' flag set. The second packet in the logical Ogg bitstream MUST contain the comment header, which contains user-supplied metadata. The format of this header is defined in Section 5.2. It MAY span multiple pages, beginning on the second page of the logical stream. However many pages it spans, the comment header packet MUST finish the page on which it completes. All subsequent pages are audio data pages, and the Ogg packets they contain are audio data packets. Each audio data packet contains one Opus packet for each of N different streams, where N is typically one for mono or stereo, but MAY be greater than one for multichannel audio. The value N is specified in the ID header (see Section 5.1.1), and is fixed over the entire length of the logical Ogg bitstream.
The first (N - 1) Opus packets, if any, are packed one after another into the Ogg packet, using the self-delimiting framing from Appendix B of [RFC6716]. The remaining Opus packet is packed at the end of the Ogg packet using the regular, undelimited framing from Section 3 of [RFC6716]. All of the Opus packets in a single Ogg packet MUST be constrained to have the same duration. An implementation of this specification SHOULD treat any Opus packet whose duration is different from that of the first Opus packet in an Ogg packet as if it were a malformed Opus packet with an invalid Table Of Contents (TOC) sequence. The TOC sequence at the beginning of each Opus packet indicates the coding mode, audio bandwidth, channel count, duration (frame size), and number of frames per packet, as described in Section 3.1 of [RFC6716]. The coding mode is one of SILK, Hybrid, or Constrained Energy Lapped Transform (CELT). The combination of coding mode, audio bandwidth, and frame size is referred to as the configuration of an Opus packet. Packets are placed into Ogg pages in order until the end of stream. Audio data packets might span page boundaries. The first audio data page could have the 'continued packet' flag set (indicating the first audio data packet is continued from a previous page) if, for example, it was a live stream joined mid-broadcast, with the headers pasted on the front. If a page has the 'continued packet' flag set and one of the following conditions is also true: o the previous page with packet data does not end in a continued packet (does not end with a lacing value of 255) OR o the page sequence numbers are not consecutive, then a demuxer MUST NOT attempt to decode the data for the first packet on the page unless the demuxer has some special knowledge that would allow it to interpret this data despite the missing pieces. An implementation MUST treat a zero-octet audio data packet as if it were a malformed Opus packet as described in Section 3.4 of [RFC6716]. A logical stream ends with a page with the 'end of stream' flag set, but implementations need to be prepared to deal with truncated streams that do not have a page marked 'end of stream'. There is no reason for the final packet on the last page to be a continued packet, i.e., for the final lacing value to be 255. However, demuxers might encounter such streams, possibly as the result of a transfer that did not complete or of corruption. If a packet
continues onto a subsequent page (i.e., when the page ends with a lacing value of 255) and one of the following conditions is also true: o the next page with packet data does not have the 'continued packet' flag set, OR o there is no next page with packet data, OR o the page sequence numbers are not consecutive, then a demuxer MUST NOT attempt to decode the data from that packet unless the demuxer has some special knowledge that would allow it to interpret this data despite the missing pieces. There MUST NOT be any more pages in an Opus logical bitstream after a page marked 'end of stream'. 4. Granule Position The granule position MUST be zero for the ID header page and the page where the comment header completes. That is, the first page in the logical stream and the last header page before the first audio data page both have a granule position of zero. The granule position of an audio data page encodes the total number of PCM samples in the stream up to and including the last fully decodable sample from the last packet completed on that page. The granule position of the first audio data page will usually be larger than zero, as described in Section 4.5. A page that is entirely spanned by a single packet (that completes on a subsequent page) has no granule position, and the granule position field is set to the special value '-1' in two's complement. The granule position of an audio data page is in units of PCM audio samples at a fixed rate of 48 kHz (per channel; a stereo stream's granule position does not increment at twice the speed of a mono stream). It is possible to run an Opus decoder at other sampling rates, but all Opus packets encode samples at a sampling rate that evenly divides 48 kHz. Therefore, the value in the granule position field always counts samples assuming a 48 kHz decoding rate, and the rest of this specification makes the same assumption. The duration of an Opus packet as defined in [RFC6716] can be any multiple of 2.5 ms, up to a maximum of 120 ms. This duration is encoded in the TOC sequence at the beginning of each packet. The number of samples returned by a decoder corresponds to this duration exactly, even for the first few packets. For example, a 20 ms packet
fed to a decoder running at 48 kHz will always return 960 samples. A demuxer can parse the TOC sequence at the beginning of each Ogg packet to work backwards or forwards from a packet with a known granule position (i.e., the last packet completed on some page) in order to assign granule positions to every packet, or even every individual sample. The one exception is the last page in the stream, as described below. All other pages with completed packets after the first MUST have a granule position equal to the number of samples contained in packets that complete on that page plus the granule position of the most recent page with completed packets. This guarantees that a demuxer can assign individual packets the same granule position when working forwards as when working backwards. For this to work, there cannot be any gaps. 4.1. Repairing Gaps in Real-Time Streams In order to support capturing a real-time stream that has lost or not transmitted packets, a multiplexer (muxer) SHOULD emit packets that explicitly request the use of Packet Loss Concealment (PLC) in place of the missing packets. Implementations that fail to do so still MUST NOT increment the granule position for a page by anything other than the number of samples contained in packets that actually complete on that page. Only gaps that are a multiple of 2.5 ms are repairable, as these are the only durations that can be created by packet loss or discontinuous transmission. Muxers need not handle other gap sizes. Creating the necessary packets involves synthesizing a TOC byte (defined in Section 3.1 of [RFC6716]) -- and whatever additional internal framing is needed -- to indicate the packet duration for each stream. The actual length of each missing Opus frame inside the packet is zero bytes, as defined in Section 3.2.1 of [RFC6716]. Zero-byte frames MAY be packed into packets using any of codes 0, 1, 2, or 3. When successive frames have the same configuration, the higher code packings reduce overhead. Likewise, if the TOC configuration matches, the muxer MAY further combine the empty frames with previous or subsequent nonzero-length frames (using code 2 or variable bitrate (VBR) code 3). [RFC6716] does not impose any requirements on the PLC, but this section outlines choices that are expected to have a positive influence on most PLC implementations, including the reference implementation. Synthesized TOC sequences SHOULD maintain the same mode, audio bandwidth, channel count, and frame size as the previous packet (if any). This is the simplest and usually the most well-
tested case for the PLC to handle and it covers all losses that do not include a configuration switch, as defined in Section 4.5 of [RFC6716]. When a previous packet is available, keeping the audio bandwidth and channel count the same allows the PLC to provide maximum continuity in the concealment data it generates. However, if the size of the gap is not a multiple of the most recent frame size, then the frame size will have to change for at least some frames. Such changes SHOULD be delayed as long as possible to simplify things for PLC implementations. As an example, a 95 ms gap could be encoded as nineteen 5 ms frames in two bytes with a single constant bitrate (CBR) code 3 packet. If the previous frame size was 20 ms, using four 20 ms frames followed by three 5 ms frames requires 4 bytes (plus an extra byte of Ogg lacing overhead), but allows the PLC to use its well-tested steady state behavior for as long as possible. The total bitrate of the latter approach, including Ogg overhead, is about 0.4 kbps, so the impact on file size is minimal. Changing modes is discouraged, since this causes some decoder implementations to reset their PLC state. However, SILK and Hybrid mode frames cannot fill gaps that are not a multiple of 10 ms. If switching to CELT mode is needed to match the gap size, a muxer SHOULD do so at the end of the gap to allow the PLC to function for as long as possible. In the example above, if the previous frame was a 20 ms SILK mode frame, the better solution is to synthesize a packet describing four 20 ms SILK frames, followed by a packet with a single 10 ms SILK frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms gap. This also requires four bytes to describe the synthesized packet data (two bytes for a CBR code 3 and one byte each for two code 0 packets) but three bytes of Ogg lacing overhead are needed to mark the packet boundaries. At 0.6 kbps, this is still a minimal bitrate impact over a naive, low-quality solution. Since medium-band audio is an option only in the SILK mode, wideband frames SHOULD be generated if switching from that configuration to CELT mode, to ensure that any PLC implementation that does try to migrate state between the modes will be able to preserve all of the available audio bandwidth.
4.2. Pre-skip There is some amount of latency introduced during the decoding process, to allow for overlap in the CELT mode, stereo mixing in the SILK mode, and resampling. The encoder might have introduced additional latency through its own resampling and analysis (though the exact amount is not specified). Therefore, the first few samples produced by the decoder do not correspond to real input audio, but are instead composed of padding inserted by the encoder to compensate for this latency. These samples need to be stored and decoded, as Opus is an asymptotically convergent predictive codec, meaning the decoded contents of each frame depend on the recent history of decoder inputs. However, a player will want to skip these samples after decoding them. A 'pre-skip' field in the ID header (see Section 5.1) signals the number of samples that SHOULD be skipped (decoded but discarded) at the beginning of the stream, though some specific applications might have a reason for looking at that data. This amount need not be a multiple of 2.5 ms, MAY be smaller than a single packet, or MAY span the contents of several packets. These samples are not valid audio. For example, if the first Opus frame uses the CELT mode, it will always produce 120 samples of windowed overlap-add data. However, the overlap data is initially all zeros (since there is no prior frame), meaning this cannot, in general, accurately represent the original audio. The SILK mode requires additional delay to account for its analysis and resampling latency. The encoder delays the original audio to avoid this problem. The 'pre-skip' field MAY also be used to perform sample-accurate cropping of already encoded streams. In this case, a value of at least 3840 samples (80 ms) provides sufficient history to the decoder that it will have converged before the stream's output begins. 4.3. PCM Sample Position The PCM sample position is determined from the granule position using the following formula: 'PCM sample position' = 'granule position' - 'pre-skip' For example, if the granule position of the first audio data page is 59,971, and the pre-skip is 11,971, then the PCM sample position of the last decoded sample from that page is 48,000.
This can be converted into a playback time using the following formula: 'PCM sample position' 'playback time' = --------------------- 48000.0 The initial PCM sample position before any samples are played is normally '0'. In this case, the PCM sample position of the first audio sample to be played starts at '1', because it marks the time on the clock _after_ that sample has been played, and a stream that is exactly one second long has a final PCM sample position of '48000', as in the example here. Vorbis streams use a granule position smaller than the number of audio samples contained in the first audio data page to indicate that some of those samples are trimmed from the output (see [VORBIS-TRIM]). However, to do so, Vorbis requires that the first audio data page contains exactly two packets, in order to allow the decoder to perform PCM position adjustments before needing to return any PCM data. Opus uses the pre-skip mechanism for this purpose instead, since the encoder might introduce more than a single packet's worth of latency, and since very large packets in streams with a very large number of channels might not fit on a single page. 4.4. End Trimming The page with the 'end of stream' flag set MAY have a granule position that indicates the page contains less audio data than would normally be returned by decoding up through the final packet. This is used to end the stream somewhere other than an even frame boundary. The granule position of the most recent audio data page with completed packets is used to make this determination, or '0' is used if there were no previous audio data pages with a completed packet. The difference between these granule positions indicates how many samples to keep after decoding the packets that completed on the final page. The remaining samples are discarded. The number of discarded samples SHOULD be no larger than the number decoded from the last packet. 4.5. Restrictions on the Initial Granule Position The granule position of the first audio data page with a completed packet MAY be larger than the number of samples contained in packets that complete on that page. However, it MUST NOT be smaller, unless that page has the 'end of stream' flag set. Allowing a granule position larger than the number of samples allows the beginning of a stream to be cropped or a live stream to be joined without rewriting
the granule position of all the remaining pages. This means that the PCM sample position just before the first sample to be played MAY be larger than '0'. Synchronization when multiplexing with other logical streams still uses the PCM sample position relative to '0' to compute sample times. This does not affect the behavior of pre-skip: exactly 'pre-skip' samples SHOULD be skipped from the beginning of the decoded output, even if the initial PCM sample position is greater than zero. On the other hand, a granule position that is smaller than the number of decoded samples prevents a demuxer from working backwards to assign each packet or each individual sample a valid granule position, since granule positions are non-negative. An implementation MUST treat any stream as invalid if the granule position is smaller than the number of samples contained in packets that complete on the first audio data page with a completed packet, unless that page has the 'end of stream' flag set. It MAY defer this action until it decodes the last packet completed on that page. If that page has the 'end of stream' flag set, a demuxer MUST treat any stream as invalid if its granule position is smaller than the 'pre-skip' amount. This would indicate that there are more samples to be skipped from the initial decoded output than exist in the stream. If the granule position is smaller than the number of decoded samples produced by the packets that complete on that page, then a demuxer MUST use an initial granule position of '0', and can work forwards from '0' to timestamp individual packets. If the granule position is larger than the number of decoded samples available, then the demuxer MUST still work backwards as described above, even if the 'end of stream' flag is set, to determine the initial granule position, and thus the initial PCM sample position. Both of these will be greater than '0' in this case. 4.6. Seeking and Pre-roll Seeking in Ogg files is best performed using a bisection search for a page whose granule position corresponds to a PCM position at or before the seek target. With appropriately weighted bisection, accurate seeking can be performed in just one or two bisections on average, even in multi-gigabyte files. See [SEEKING] for an example of general implementation guidance. When seeking within an Ogg Opus stream, an implementation SHOULD start decoding (and discarding the output) at least 3840 samples (80 ms) prior to the seek target in order to ensure that the output audio is correct by the time it reaches the seek target. This "pre-roll" is separate from, and unrelated to, the pre-skip used at the beginning of the stream. If the point 80 ms prior to the seek
target comes before the initial PCM sample position, an implementation SHOULD start decoding from the beginning of the stream, applying pre-skip as normal, regardless of whether the pre- skip is larger or smaller than 80 ms, and then continue to discard samples to reach the seek target (if any).