search for: qualiti

Wow, gcc 3.1.1 generates some superior code indeed. I've included my new results. The encoding rate results WRT -q setting haven't really changed that much. The interesting thing was that gcc 3.1.1 gave me way better than 10% increase in speed in some places. All tests are conducted with a wav file being read from a RAID0 array of two disks and written to a third disk as ogg to reduce

I'm a bit confused about VBR with Opus. I see that the default is VBR, but I don't see any way to specify a quality setting. I can set a target bitrate, but that's definitely not what I want; I want a constant quality level, like "-q" in oggenc, and for the encoder to select bitrates based on the desired quality. For example, for playback through earbuds or laptop

I am planning to develop an R package to manage all aspects of data quality. I am very experienced in data quality, but fairly new to R. I have tried to find a suitable data quality package, and am surprised not to find much to suit my requirements. Developing the package would be an ambitious effort, involving several contributors (that I have already identified, and who also do not have much R

Hi there, I'm using libvorbis in my program and need to encode to target bitrate. I know libvorbis prefer to use quality scale but I can't use it. I've found something at faq http://vorbis.com/faq/#quality *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly 160kbps, and 10 gives about 400kbps. Most people seeking very-near-CD-quality audio encode at a quality

Hi everyone, I am trying to make some publication-quality plots for use in Microsoft Word, but I am having trouble creating high-quality plots that are supported by Microsoft Word. If I use the R plot function to create the figure, the lines are jagged, and the picture is not of high quality (same with JPEG(), TIFF(), and PNG() functions). I have tried using the Cairo package, but it distorts

When i use VBR do i need to set VBR quality or speex quality? Is the speex quality referred to the overall quality and instead VBR quality only if i use VBR? Do these qualities override each other? Thanks. --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070616/28f25b...

Sounds like you'll be reinventing square wheels. Searching "data quality package" on rseek.org brought up many hits. Cheers, Bert Bert Gunter "The trouble with having an open mind is that people keep coming along and sticking things into it." -- Opus (aka Berkeley Breathed in his "Bloom County" comic strip ) On Fri, Aug 4, 2017 at 2:56 AM, Architector Data

Hi, I just built a sample application with speex resampler in linux and I tried to resample 8K sine wave tone mono to 48k using speex_resample_process_int. I am using a tool called EAQUAL for audio quality. I find the quality of Speex resampler to be decreasing when I increase the quality q of the resampler init function. Can some one give me pointers regarding this?? As per the API, if the

The ffmpeg2theora chapter[1] of the Theora Cookbook says that the only advantage of two-pass encoding is that you can hit a specific target size. But isn't it so that you can reserve bits for difficult parts of a video, which improves the perceived quality? It may be true that Theora does quite alright without two passes, but it seems like a no-brainer to me that two passes will always be

Hi, I was wondering if anyone has experimented with Speex's wideband (16kHz) mode at lower quality settings. In particular I have been using quality 3, and with wideband input files the resultant frequency spectrum is limited to about an upper end around 3.5kHz (almost telephony quality bandwidth). Has anyone tried increasing the spectral bandwidth at the expense of lowering the

I am trying to dynamically set the compression or quality of an image when displaying it. I have seen in the RMagick documentation how to set the quality only when writing the file to disk. How can I set the quality dynamically or set the compression? The method "image.quality=" does not exist and "image.compression_type=" is not in the documentation. I have tried

Hi All, Actually, the question is right on the topic. Is it possible to change the quality while encoding ? If the answer is yes, does the jitter buffer provides info about latency, e.g. : if latency is too high, we lower down the quality, if the latency is okay, we are raising the quality until latency rise up again... Thank you for your help. Morgan

Hi all, I am running a series of experiments where after manipulating my data I run several clustering algorithms (agnes, diana and a clustering method of my own) on the data. I wanted to determine which clustering method did the best job, so therefore I had defined my own quality measure using two criteria: compactness of the data within the clusters themselves and the amount of seperation

> All you say look normal to me. Okay, clarification requires less code to be written than a bug. :) > > Does that makes sense? I would have thought that --dtx alone would have = > doen it, > > but maybe it needs perfect digital silence to work? Or only uses the voi= > ce activation > > to know when there's silence? > > DTX requires the VAD to know when

I have looked at the codec's available but I don't know how get the highest quality recorded message. If a user calls in over the normal telephone network is this limited to the carriers codec or the codec at the asterisk side? Would I get a higher quality result using VoIP rather than the normal network? Any help would be appreciated. Thanks Fats.

On 27.11.2017 02:00, YIRAN LI wrote: > I'm using libvorbis in my program and need to encode to target bitrate. I > know libvorbis prefer to use quality scale but I can't use it. > > I've found something at faq http://vorbis.com/faq/#quality > > *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly > 160kbps, and 10 gives about 400kbps. Most

Hi, I'm testing Asterisk installation on my Slackware Linux Box, I am making calls from kphone. I've noticed that standard sample voice files attached to * distro (*.gdm files) havn't good quality. What can be done to improve it? I'm using alsa drivers. BR, Corvin

hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice

To close debian bug #271052, http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=271052 I went ahead and did the following: $ diff -u ../original/speex-1.1.6/libspeex/nb_celp.c libspeex/nb_celp.c --- ../original/speex-1.1.6/libspeex/nb_celp.c 2004-07-15 01:16:52.000000000 -0400 +++ libspeex/nb_celp.c 2005-02-27 08:24:49.000000000 -0500 @@ -1746,6 +1746,21 @@ st->submodeSelect =

Hi Jean-Marc, I thought at quality 3 (wideband) - wb_submode1 that the 4-8k band was not using a codebook table. From the code I can see that some sort of "lsp" encoding is performed. What exactly does this encode? (I assume lsp means line-spectral pairs) The reason I am asking is I'm comparing the "effective" spectral bandwidth of Speex against the