similar to: CVBR (constrained variable bitrate)

I'm sorry if this question has been asked before; I've looked through the archives and haven't seen anything. The problem I'm seeing is that oggenc's VBR encoding doesn't seem to pay attention to any sort of bitrate limitation, either the -m or bitrate_hard_min settings. It isn't that it temporarily dips below the minimum; the average for the whole (in this case,

Hi There, I have two OPUS handset clients say A & B A is 8 KHz, 12.2kbps cvbr supported OPUS client B is 8 KHz, 16kbps cvbr supported OPUS client When i try to encode a same voice frame(20ms sample frame) at different time intervals(not parallel encoding) for both A & B using same encoder handle by changing only bit rate. Issue here is, some noise is heard for B OPUS client, Ex:

Hi! So here is what I got with different encoder settings; still not sure what the "best" setting is. 6kbps seems to add distortions, so I tried 12kbps. MP3-Original: LAME 3.99r, 120kbps, 44100Hz, Stereo, VBR V5 (22:23, 19.8MB) Opus (--raw-rate 44100 --bitrate 56 --vbr --comp 5): (44:45?, 23.4MB): Broken Opus (--bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 12.1 MB, 74kbps)

Hi Vince, Normally, Opus should work fine on the CC3220, though real-time encoding may be a bit tight depending on what configuration you choose. You'll probably want to configure with --enable-fixed-point and --disable-float-api (or define FIXED_POINT and DISABLE_FLOAT_API if you don't use autoconf). The encoder has a bunch of run-time settings that you may want to set, depending on

On 02/20/2018 06:48 PM, Rodriguez, Vince wrote: > I am trying to get the library to build without using the pseudostack > define, and use either VAR_ARRAYS or ALLOC, but it seems the global > stack is not defined. Not sure what you mean here. If your compiler supports C99 VLAs, then by all means just define VAR_ARRAYS and you're good to go. If not, the second best option is alloca(),

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

Jean-Marc, Thanks for the response and the helpful info. I am trying to get the library to build without using the pseudostack define, and use either VAR_ARRAYS or ALLOC, but it seems the global stack is not defined. Where do can I define this in my example? VR -----Original Message----- From: Jean-Marc Valin [mailto:jmvalin at jmvalin.ca] Sent: Tuesday, February 20, 2018 5:40 PM To:

Jean-Marc and others, We are using Speex inside our Video Conferencing application and have some questions about the trade-off of complexity vs bitrate. A little background information is needed: 1. For us the actual bit-rate isn't that big a deal. Speex running at 14kbps or 8 kbps is minor compared to the video traffic running at 128, 256 or 384 kbps. 2. We care a little about

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis/attachments/20041014/25f999a2/attachment.htm -------------- next part -------------- ? dear All, Iam working on OGG VORBIS TREMOR code to port and optimize on ARM processor. Can any one make me clear regarding the relation ship between this 1. What is the relation ship between Quality factor and Bitrates?

Michael, > See the examples. If you initialise a managed mode (which this is), you > MUST use vorbis_bitrate_addblock() and vorbis_bitrate_flushpacket(). Thanks for the tip. I added the calls, and it works now. > You SHOULD also give an option to set min/max, since they're now used, > and to use a VBR mode rather than the managed modes here. But when streaming, IMHO it is

I suspect the main difference is due to time alignment of the samples and the exact trimming at the beginning and at the end of the encoded files. I also note that there's a missing "--" in front of "bitrate" and only one "-" instead of two for cvbr in your opusenc command line. Make sure both are also linked to the same libopus (opusenc could be using the system

Hi, I am interested in this part of the answer: "Make sure both are also linked to the same libopus (opusenc could be using the system libopus 1.0.x while opus_demo is from git)" I dowloaded a tarball distrib of libopus (1.1-beta),build and installed it,and afterwards did the same with opus-tools (0.1.7),using the former installed libopus library (also using libogg 1.3.1). When I execute

On 2010-04-11 10:09, Randy Yates wrote: > If I specify VBR mode via > > speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_VBR,&isTrue); Try: speex_encoder_ctl(pSpeexEncoder, SPEEX_SET_ABR, &desiredRate);

With this patch, ogginfo will display bitrate information (upper/lower/nominal/average bitrate). It is patched against today's CVS code (main branch). I am also working on true VBR support in oggenc (the user can specify the minimum & maximum bitrate) and will send a patch when it's done. <HR NOSHADE> <UL> <LI>application/x-gzip attachment: ogginfo-diff.gz

Hi, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

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 have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Hi. Sorry if this is the wrong list - a pointer to the right would be appreciated. I have a number of Ogg files, that have a VBR bitrate of approx. 256 kbps. I have bought a portabel Ogg player (iRiver iFP 795) that unfortunately only support bitrates up to approx. 225 kbps. So my question is, if there is a tool to change the bitrate down to e.g. 220 kbps. I know I can do this with sox, or

> vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); > > which by all reasons should generate a 96 kb/s stream, as all > max_bitrate, nominal_bitrate and min_bitrate are set to 96. Strangely > enough, the generated stream's bitrate (according to XMMS and WinAmp) > varies between 49 and 59. Currently min and max are ignored by the library, since bounded bitrates

Hi all, I used vorbis several years ago because of its freeness and VBR, but after playing ha.org and getting an mp3-capable discman, I switched to APS and API mp3 encoding. Now, I am liking musepack a lot, but the flexibility is more "there" for the vorbis format, including players, tagging comments (lyrics in the comments - killer idea!), etc.... However, I am a bit worried