Vorbis Packet Block Size 35

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Rayna Himenez

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Dec 5, 2023, 6:11:43 PM12/5/23
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The WebVTT cues are stored as the data portion of Block elements in the track,per the formatting described below. All WebVTT data stored within a WebM BlockMUST be encoded as UTF-8. The timestamp of the WebM block and its duration aresynthesized from the start and end times specified on the timestamp of theWebVTT cue. A BlockGroup element (not a SimpleBlock) MUST be used to containthe Block element, in order to also use a BlockDuration element, which isnecessary to losslessly encode the original timestamp of the WebVTT cue.

The Opus format is based on a combination of the full-bandwidth CELT format and the speech-oriented SILK format, both heavily modified: CELT is based on the modified discrete cosine transform (MDCT) that most music codecs use, using CELP techniques in the frequency domain for better prediction, while SILK uses linear predictive coding (LPC) and an optional Long-Term Prediction filter to model speech. In Opus, both were modified to support more frame sizes, as well as further algorithmic improvements and integration, such as using CELT's range encoder for both types. To minimize overhead at low bitrates, if latency is not as pressing, SILK has support for packing multiple 20 ms frames together, sharing context and headers; SILK also allows Low Bit-Rate Redundancy (LBRR) frames, allowing low-quality packet loss recovery. CELT includes both spectral replication and noise generation, similar to AAC's SBR and PNS, and can further save bits by filtering out all harmonics of tonal sounds entirely, then replicating them in the decoder.[16] Better tone detection is an ongoing project to improve quality.

vorbis packet block size 35


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SILK supports frame sizes of 10, 20, 40 and 60 ms. CELT supports frame sizes of 2.5, 5, 10 and 20 ms. Thus, hybrid mode only supports frame sizes of 10 and 20 ms; frames shorter than 10 ms will always use CELT mode. A typical Opus packet contains a single frame, but packets of up to 120 ms are produced by combining multiple frames per packet. Opus can transparently switch between modes, frame sizes, bandwidths, and channel counts on a per-packet basis, although specific applications may choose to limit this.

Decoding is done in a way similar to libvorbis. First, initialize a kate_info and akate_comment structure. Then, read headers by calling kate_decode_headerin. Onceall headers have been read, a kate_state is initialized for decoding using kate_decode_init,and kate_decode_packetin is called repeatedly with data packets. Events (eg, text) can beretrieved via kate_decode_eventout.

Encoding is also done in a way similar to libvorbis. First initialize a kate_infoand a kate_comment structure, and fill them out as needed. kate_encode_headers willcreate ogg packets from those. Then, kate_encode_text is called repeatedly for allthe text events to add. When done, calling kate_encode_finish will create an end ofstream packet.

I understand that an Ogg page comprises several Vorbis packets (up to
a maximum of 255), and I didn't mean to suggest otherwise. My claim
that "WebM easily beats Ogg" stands, however, because you can store as
many Vorbis audio frames on a Webm audio block (that uses lacing) as
you can Vorbis audio packets on an Ogg page. WebM is more efficient
because the amortized cost of cluster element overhead approaches 0 as
more audio blocks are put on the same cluster.

> My claim
> that "WebM easily beats Ogg" stands, however, because you can store as
> many Vorbis audio frames on a Webm audio block (that uses lacing) as
> you can Vorbis audio packets on an Ogg page. WebM is more efficient
> because the amortized cost of cluster element overhead approaches 0 as
> more audio blocks are put on the same cluster.

OK, 5s cluster size, so to be fair, 5s Ogg page size.64 kbps Vorbis is 186 bytes per Vorbis packet. Both Ogg and Matroska
are going to use Xiph lacing for this because it's most efficient in
this case.That's 43 packets per second, so 215 packets for 5s. That's 28+215
bytes overhead in the Ogg muxing, or 243 bytes for 5 second, or 389
bits overhead per second for .6% overhead.(I realize this is unrelated to the raw bit overhead argument, but
that 389 bits per second gets you error, capture, gap and truncation
detection, because, again, those aren't optional in Ogg and I don't
recall if WebM allows that now. It didn't in the early drafts. I
agree that checksums are of only marginal use in video, but they're
essential in audio. Audio bit errors can and do damage equipment and
hearing).I'll let you do the math for WebM since it's true I haven't looked at
the draft spec since summer last year. I'm well versed in Matroska
itself, but I'm not sure I've kept up with the WebM restrictions.Monty

That could be investigated, because right now one of the big problem
in WebM is the size of the Vorbis init packet. It would be very to add
an extension to Matroska to have it compressed. In fact being a codec
private it would just take calling the codec "A_VORBIS2" and treating
the codec private as zlib compressed rather than raw. Adding support
in all OSS players will be very fast and easy.

> That could be investigated, because right now one of the big problem
> in WebM is the size of the Vorbis init packet. It would be very to add
> an extension to Matroska to have it compressed. In fact being a codec
> private it would just take calling the codec "A_VORBIS2" and treating
> the codec private as zlib compressed rather than raw. Adding support
> in all OSS players will be very fast and easy.

A processing time control in CADWorker has been added. This prevents a DMU from not being imported due to a referenced file looping. The processing max time is based on the size of the file to process, and its format. Only the blocked file is canceled. This can be disabled with cvar r.CADTranslator.EnableTimeControl=0.

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