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Hey everyone, I've written a FLAC plug-ing for Adobe Premiere. It's actually part of the Ogg Vorbis plug-in I originally set out to write. You can see it here on GitHub: https://github.com/fnordware/AdobeOgg I just finished the first beta a few hours ago. You can download binary plug-ins from here: http://www.fnordware.com/downloads/Ogg_v0.5b1_mac.zip http://www.fnordware.com/downloads/Ogg_v0.5b1_win.zip If you don't have Premiere, you can always download the 30-trial. If you don...

Hey everyone, I've written a FLAC plug-ing for Adobe Premiere. It's actually part of the Ogg Vorbis plug-in I originally set out to write. You can see it here on GitHub: https://github.com/fnordware/AdobeOgg I just finished the first beta a few hours ago. You can download binary plug-ins from here: http://www.fnordware.com/downloads/Ogg_v0.5b1_mac.zip http://www.fnordware.com/downloads/Ogg_v0.5b1_win.zip If you don't have Premiere, you can always download the 30-day trial. If you do...

Hey everyone, I've written an Ogg Vorbis plug-in for Adobe Premiere. You can see it here on GitHub: https://github.com/fnordware/AdobeOgg I just finished the first beta a few hours ago. You can download binary plug-ins from here: http://www.fnordware.com/downloads/Ogg_v0.5b1_mac.zip http://www.fnordware.com/downloads/Ogg_v0.5b1_win.zip If you don't have Premiere, you can always download the 30-trial. If you don...

Hey everyone, I've written an Ogg Vorbis plug-in for Adobe Premiere. You can see it here on GitHub: https://github.com/fnordware/AdobeOgg I just finished the first beta a few hours ago. You can download binary plug-ins from here: http://www.fnordware.com/downloads/Ogg_v0.5b1_mac.zip http://www.fnordware.com/downloads/Ogg_v0.5b1_win.zip If you don't have Premiere, you can always download the 30-trial. If you don...

What got me experimenting with Opus is that I heard it's going to be a supported codec in WebM, which I also have a Premiere plug-in for: http://github.com/fnordware/AdobeWebM I just posted a new beta that includes Opus support. Naturally, I *think* I'm doing it right, but I'd love for someone more knowledgeable about Opus or WebM to take a look. The closest thing to a standards document for putting Opus into WebM is here: http://wiki.xiph.org/Mat...

Hey everyone, I've added Ogg Opus support to my Adobe Premiere plug-in here: http://github.com/fnordware/AdobeOgg Now I'll add Opus support to me WebM plug-in too. I've got this Opus stuff mostly figured out, but I have a few questions. Here's one: What do the numbers in mapping mean? I see that opus.h refers to the Vorbis channel mapping order, so does mapping in Opus take that ord...

Yes, it's basically an intrinsic aspect of the mathematics. If you seek into the middle of an Opus stream, that's about how long it takes before the audio converges to be correct. On Jan 17, 2014 2:13 PM, "Brendan Bolles" <brendan at fnordware.com> wrote: > On Jan 7, 2014, at 11:11 AM, Brendan Bolles wrote: > > > The closest thing to a standards document for putting Opus into WebM is > here: > > > > http://wiki.xiph.org/MatroskaOpus > > > I'm curious about the part there where it says SeekPreRo...

...s from scratch and importing the FLAC source right into my project and I get the same errors. If someone would be so kind as to take a look, I have a branch set up, ready to show you the problem (after you download the Premiere SDK). I've posted this as a GitHub issue here: http://github.com/fnordware/AdobeOgg/issues/4 Help me, Obi-Wan Kanobi! Brendan

On 01-07-13 16:11, Brendan Bolles wrote: > Hey everyone, I've written a FLAC plug-ing for Adobe Premiere. It's actually part of the Ogg Vorbis plug-in I originally set out to write. You can see it here on GitHub: > > https://github.com/fnordware/AdobeOgg Hi Brendan, Nice work! I'm not an Adobe Premiere user, but for what I know it has quite a lot of users so I'm pretty sure this will gain the FLAC format quite a few more users. > The reason I wanted to post here is that I wasn't able to get FLAC working on Windows. The...

On 2014-01-06 10:24 AM, Brendan Bolles wrote: > http://github.com/fnordware/AdobeOgg Cool! > What do the numbers in mapping mean? These numbers tell you how to map the decoded opus channels to particular speakers. > For example, opus_multistream_surround_encoder_create() sends out mapping[] = {0, 4, 1, 2, 3, 5}. > For 6 channels the Vorbis order is front left,...

On 06/01/14 06:21 PM, Ralph Giles wrote: > It's written from the point of view of the decoder, so it seems > backwards if you're thinking about encoding. It may help to think of it > as opus_multistream_surround_encouder_create() returning what you have > to pass to the decoder to understand the stream configuration. Not only that, but it's designed so that the same signal

...the frame data. When you combine a starting frame with the subsequent frame, the data you get back is from the starting timestamp forward. Because Opus needs to preroll a larger amount, that trick wouldn't work with Opus. Monty On Tue, Jan 21, 2014 at 1:00 PM, Brendan Bolles <brendan at fnordware.com>wrote: > On Jan 17, 2014, at 2:19 PM, Benjamin Schwartz wrote: > > > Yes, it's basically an intrinsic aspect of the mathematics. If you seek > into the middle of an Opus stream, that's about how long it takes before > the audio converges to be correct. > > &...

Hi everyone, I'm working on a WebM plug-in for Premiere. You probably already know that WebM is a Matroska container that uses Vorbis for audio compression. I'm pretty new to the world of audio compression and need some help. You can see my code here: http://github.com/fnordware/AdobeWebM First of all, I wonder if it's possible to seek to an exact audio sample in WebM. Whereas an Ogg page provides you with the granule position, Matroska provides a much less precise timecode, measured in milliseconds. So the second frame at time 41.6666ms gets rounded to 42ms in Mat...

Hey everyone, I'm new to the world of audio, so can you please drop some very basic audio knowledge on me? What's the correct way to convert from float audio samples to integer ones like FLAC uses? If I'm converting to 8-bit, that has a range of -128 to 127, right? Do I multiply my float input by 127 or 128? Brendan

On Jan 7, 2014, at 3:12 AM, lvqcl wrote: > It looks like the linker cannot find libFLAC_static.lib and libFLAC++_static.lib. > By default FLAC moves them to objs/release/lib/ (or objs/debug/lib/) folder. Unfortunately, I don't think that's it. Visual Studio gives me a specific error when it can't find a certain library file. I think these are the errors you get when a

I get reliable crashes if I create a 6-channel encoder using opus_multistream_surround_encoder_create(). If I use opus_multistream_encoder_create() instead and pass in the same parameters that opus_multistream_surround_encoder_create() sends out (streams = 4, coupled_streams = 2, mapping = {0, 4, 1, 2, 3, 5}) I don't get the crashes. I notice that opus_demo.c uses

On Jan 6, 2014, at 11:54 AM, Ralph Giles wrote: > The first entry, mapping[0], > describes front left and contains the value '0', so that speaker is the > left channel of the first coupled stream. mapping[1] is '4', so the > front centre speaker is given the first mono stream. mapping[2] is '1' > so the front right speaker gets the right channel of the first

On Jan 7, 2014, at 11:11 AM, Brendan Bolles wrote: > The closest thing to a standards document for putting Opus into WebM is here: > > http://wiki.xiph.org/MatroskaOpus I'm curious about the part there where it says SeekPreRoll should be set to 80000000. I believe those are in nanoseconds, so that's 0.08 seconds, 3840 samples at 48kHz. Here's a page explaining that it

On Jan 17, 2014, at 2:19 PM, Benjamin Schwartz wrote: > Yes, it's basically an intrinsic aspect of the mathematics. If you seek into the middle of an Opus stream, that's about how long it takes before the audio converges to be correct. I see. Thanks for the info. Any idea about the other Matroska tags? I guess CodecDelay will be the value you get from OPUS_GET_LOOKAHEAD()

On Jan 17, 2014, at 3:00 PM, Timothy B. Terriberry wrote: > The intent was not to allow gaps in the file like this (that would > confuse the encoder, compress badly, and complicate decoders, plus be > unrepresentable in Ogg). The intent was to allow trimming from the > _last_ frame in a stream to allow sample-accurate total duration. Ahhh, that makes sense. So you'd add some