similar to: Controlling the vorbis encoder precisely

Ok, related to the replys from Eric and Monty there are 3 possible solutions - 1. use lossless encoding like FLAC - 2. insert large chunks of silence between the wave sounds - 3. use completely seperate chained streams -> 1. Flac can't be used because of disk space and bandwidth limitations. -> 2. I also tried this variant but in that case my sounds don't start immediately

Hello, > (1) Split the content into chain segments > (2) extract the first and last data packet from each segment, for all tracks > (3) recover / calculate the granulepos values for the first and last data packets (*) > (4) convert the granulepos values from the first and last data packets into time values > (5) take the largest difference between start and end time as the total

Hi Ralph, thanks for your posting. Yes, the standard example player in the theora distribution could also do it when theora would not need so much time. I removed the frame dropping from the example because it's based on some audio stuff under Linux that isn't available on Windows. Currently I've no frame dropping handling build in because the first goal is a good raw performance of

4) Subtract 1 from the keyframe, then repeat step 3). 5) Begin reading from the frame discovered in step 4. Drop any packets which are output on the first page. Count down until we reach the keyframe, dropping packets until then. 6) Continue counting down until we reach the target frame, we are now decoding each frame/packet. At the target frame produce the YUV output. You can see this in code

4) Subtract 1 from the keyframe, then repeat step 3). 5) Begin reading from the frame discovered in step 4. Drop any packets which are output on the first page. Count down until we reach the keyframe, dropping packets until then. 6) Continue counting down until we reach the target frame, we are now decoding each frame/packet. At the target frame produce the YUV output. You can see this in code

Dear theora developer community, currently I'm working on a simple Theora player for Windows. But the code in the player_example.c seems not to have the performance of other implementations like the Direct Show filters by illuminate. In the example player, all important things are done in one thread: decoding the next vorbis or theora packet(s) and reading from the physical stream (+ split

Where can I download stable RC2 source code? I've downloaded the nightly CVS, but it seems unusable right now. The Windows SDK does not appear to include source code or Borland compatible libraries. One thing I've noticed about the nightly CVS code is that the nested alloca() calls seem to confuse Borland's compiler. I had to change code in mapping0_forward() from:

Hi, I have now played around with seeking a bit. Currently I can seek into a file position in splayer and resynchronize playing (via ogg_sync_pageseek) from there. I did not take chained streams into account yet, as I'm not sure if it would make splayer far to complex for a sample application. Maybe later. Before I implement seeking to a specified time (ala libvorbisfile), can you answer

Hi, Currently in Godot, we need some help to fix audio pops during looping. We decode it ourselves and don't use vorbisfile. We strongly guess that we're sometimes not sample accurate enough during seeking and missing something. Here's a pull request featuring some internal fixes: https://github.com/godotengine/godot/pull/80452/files I think it also lists the most vital files in

Hi, I have been trying to fit an un-penalised gam in mgcv (in order to get more reliable p-values for hypothesis testing), but I am struggling to get the model to fit sucessfully when I add in a te() interaction. The model I am trying to fit is: gam(count~ s(x1, bs = "ts", k = 4, fx = TRUE) + s(x2, bs = "ts", k = 4, fx = TRUE) + te(x2, x3, bs =

Hello, I'm just totally confused. In my theora streams encoded using ffmpeg2theora (but also when using my own encoder) I have packets with a granulepos of -1 so I can't identify the packet during a seeking operation correctly. I can also see those strange value when I just print the packet granulepos before sending it to the Theora decoder. I know why there are PAGES with granularpos of

Also noticed that right now there are two directories in the CVS source tree: doc and docs doc is empty and could probably be marked for delete? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the

Using libvorbis (1.3.5) I wish to extract the raw vorbis packets.  I've built some simple code on the excellent libvorbis API overview on the xiph.org site, but the example relies on the ogg_packet struct for final output and input to decoder, and shows now examples on how to do without it. Taking a look at the vorbis_bitstream_flush() function, which in the overview is the last step before

It seems to me that running ogg_sync_pageout doesn't automatically advance the page. This is good if you haven't worked with the given page, makes coding somewhat easier. However, when does a page advance. Is it after a call to ogg_stream_pagein? On a side note, I need to do seeking on top of libvorbis, I'd love to use vorbisfile but sadly I can't. Is there a reasonable way to

Background on setup Here is some basic background of my setup I have 3 domains setup running over a IPSEC tunnel over the Internet with one PDC BDC and some file servers at each location all running samba (I am sorry I do not have the version number in front of me for samba) from SUSE enterprise 10 using the LDAP backend. There is a trust setup between all the domains and the IPSEC tunnels is

On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.mail at gmail.com> wrote: > Jose Pablo Carballo <jose.carballo at ridgerun.com> wrote: > >> - channels = 2; >> - bps = 16; >> + channels = ((unsigned)buffer[23] << 8) | buffer[22]; >> + bps = ((unsigned)buffer[35] << 8) | buffer[34]; >> total_samples = (((((((unsigned)buffer[43] << 8)

Hi, In the last days I've been taking as reference the example found in examples/c/encode/file/main.c. With it I've been able to encode a 2ch, 16 bps, 44100 sample rate input WAV file to a FLAC file. Now I've been trying to modify this example to encode a 2ch, 24 bps, 96000 sample rate WAV file. I have to say I'm a bit lost on how I should read the input file in this case, and

I am looking for any application for windows or qnx that can encode a wave file. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the list

Hi Guys, I've just faced a wrong warning trying to encode a 24 bits WAV file if(wFormatTag == 1) { if(bps != 8 && bps != 16) { if(bps == 24 || bps == 32) { /* let these slide with a warning since they're unambiguous */ flac__utils_printf(stderr, 1, "%s: WARNING: legacy WAVE file has

Hello all, I've been writing a basic encoder using libFLAC++. It seems to work fine, as in the resultant file is playable in any FLAC compatible player, however the compression ratio is very small - the main example being a 60MB file compressing to 54MB, i guess there's an option i've missed, but i cannot track it down. I was hoping someone could point it out for me: