similar to: cbr/vbr decoding - supported sample rates

Hi, Regarding to my previous question, I think I have found out from oggenc source code that the correct way to initialize encoding is: vorbis_encode_init_vbr (or vorbis_encode_setup_managed for cbr etc.) vorbis_encode_ctl (to turn off or use avg bitrate management) vorbis_encode_setup_init By this way, there is no problem with encoded stream. However, there is no documentation of the three

Hi, I am trying to encode PCM to OGG based on example code in OggVorbis SDK Win 1.0. There is no problem with "vorbis_encode_init_vbr(&vi, 1, 44100, 0.4f);" setting. However, using "vorbis_encode_init(&vi, 1, 44100, 96000, 96000, 96000);" setting gives such a warning by ogginfo: "Warning: EOS not set on stream 1" I have modified ogginfo to display exact

Hi, How can I find more information about vorbis enc and ogg api ? sdk seems to have not enough information. Thanks in advance. --- Mete BALCI Senior Game Programmer Momentum DMT Istanbul, TURKEY -------------------------------------------------------------------------- Confidentiality Notice: This e-mail may contain confidential and/or privileged information that is intended only for the

Fredag, 31 januar 2003, skrev du: <p>>Next thing is, that I would like to clarify the difference between a codec's >latency (which I would die to know in exact figures for the vorbis codec one >day - as well as block sizes, but that's another story) and the (bandwidth) >"smoothing window" latency - call it buffering or whatever. One thing is the codec

Hi, Maybe this is documented somewhere, if so, please send me a link to the documentation. My question is: how to set up different (VBR, CBR, ABR) modes when calling the Ogg Vorbis API? Currently I do: CBR: ret = vorbis_encode_setup_managed( &vorbisInfo, getInChannel(), getOutSampleRate(),

On 01/21/2015 07:51 PM, Daniel K wrote: > Hi guys, > > I'm using Opus for speech in wide-band mode (sampling rate 16000) and > 20ms frames with signal type set to SIGNAL_VOICE. > > I have a few questions here: > > 1. > When I choose VBR mode, the codec seems to choose the bitrate on its own. > However, that seems to be an issue on mobile devices. In some cases,

Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr => true)? 2) If there relation between setting the vbr => true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value?

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

Just in case my quotation causeed misunderstnaing... Forget about what I quoted before. Probably, by CBR, Monty meant ABR since CBR is stupidly patented and this is the reason why Vorbis has no CBR option. If you know this, it cannot be a problem but I see there is a question asking what is ABR in the mailing list. >On Thu 28 Jun 2001, 23:27:41 PDT, Monty (xiphmont@xiph.org)Wrote: >then

Hi, I try to use the video tag to post videos on my blog I'm using ffmeg2theora to transcode videos in theora Resolution of 320x240 pixels and constant bitrate (CBR) of ~400 kbps seems to be ok for publishing on my blog. I've read that CBR is a better choice than VBR for web publishing regarding the way TCP/IP works. ("In TCP it takes a while for a connection to increase its

fwiw, there are definitely applications that would use Vorbis but can't until there is a _strict_ CBR mode. Meaning, architecturally the datarate simply cannot exceed X no matter what happens to the quality. Send silence if you have to, but limit the datarate. -dbm -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef

#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 | Milestone: Component: Opus |

#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 | Milestone: Component: Opus |

Greetings Fellow Vorbis Gurus! How can I encode, using CDex, at a CBR (constant bit rate) rather than the VBR? Also, can one take a 128kbs MP3 file and re-encode it to ogg? If so, please explain the procedure. By the way, I'm using a Windows based system. Thank you! /jp/ dojpaul@coffey.com --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

opusdec -V opusdec opus-tools f2a2e88 (using libopus unknown) I've got an opus file encoded from a .wav off a cd, 44100Hz: opusinfo 2-24-Overture_in_C_\(In_Memoriam\).opus Processing file "2-24-Overture_in_C_(In_Memoriam).opus"... New logical stream (#1, serial: 38134f1f): type opus Encoded with libopus unknown User comments section follows... ENCODER=opusenc from opus-tools

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

Hi Everyone, I am getting the following error message Error: cannot allocate vector of size 2.6 Gb In addition: Warning messages: 1: In dim(res$res) = dim(bi) : Reached total allocation of 8122Mb: see help(memory.size) 2: In dim(res$res) = dim(bi) : Reached total allocation of 8122Mb: see help(memory.size) 3: In dim(res$res) = dim(bi) : Reached total allocation of 8122Mb: see

I'm writing a voice communication application, and I've got a few issues that I'd like to get ironed out, but I don't know enough about the speex implementation. First of all, this application is mainly used for conferencing - many people are in a room and only 1-2 are ever talking at a time. So, always encoding and transmitting everyone's audio stream would be rather

Hii all I bought te110p card. I configured zaptel.com ; span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 and zapata.conf is; switchtype=euroisdn signalling=pri_net group=1 context=nortel dchannel => 16 channel => 1-15,17-31 I receving the following error on my gentoo; May 18 13:48:44 WARNING[8755]: Ignoring canpark May 18 13:48:44 WARNING[8755]: Ignoring dchannel May 18 13:48:44

I don't know why you're getting sound carrying over to the next time you encode - that doesn't sound normal to me. Have you tried saving and examining the raw audio you're feeding to the encoder? Have you tried encoding and decoding that using speexenc/speexdec? I use Speex in VBR mode for a VoIP app. I'm always recording and running the audio through the preprocessor