similar to: Can i ask about speex codec ?

Hi,I'm having performance issues encoding with speex on an iphone4 + iOS5. The same code yields excellent results on iPad2 + iOS5, Mac OS X 10.6 as well as Windows 7. The call to speex_encode_int is takes 0.0065 to complete. Because encoding is so latent the receiver incurs underruns. Here is my speex setup: ? ? unsigned val = 0; ? ? /**? ? ?* Initialize the encoder in narrow band mode.? ?

alabama_tuck71 wrote: > Dammit :( It would have a great work distraction...thank you, though! Second, the iPad doesn't have an x86 CPU in it. It has an ARM CPU in it. Wine barely even supports ARM natively. It certainly doesn't support running x86 programs--like World of Warcraft--on ARM hardware. Long story short: there is no way to run the Windows version of WoW on an iPad

Hi Conrad, Thanks a lot for the info. This is exactly what I needed. We want to use speex in our IP telephony client and I was indeed trying to put together a GUI configuration screen. Best regards, Chris > On Tue, May 20, 2003 at 06:04:39PM +0200, Chris Flerackers wrote: >> Hi, >> >> Is there are document where the interaction between all the configuration >> options

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

On May 23, 2011, at 11:35, Dennis Brunnenmeyer wrote: > I'm well aware how compression works. But images and document files > do not depend on the relative timing of the data to reproduce > themselves. They are in essence only two-dimensional in space, > whereas the data in a sound file is time-dependent. Images are three-dimensional or maybe five-dimensional,

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,

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

> > So there should be some equivalency between --abr values and --quality > > values? Or do they effect the rate in different ways. > > Actually, --abr adjusts the quality in real-time so that the average > bit-rate meets a certain target. with just --quality, the final average > bit-rate is dependent on the data, so that's why --abr is there. Is there some time

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(),

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

> 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 recently installed icecast and iceS and all seems to be well. I have spent four days trying to understand and use darkice and finally gave up. Using darkice configuration: [general] duration = 0 # do it forever bufferSecs = 10 [input] #device=hw:0,0 device = /dev/dsp sampleRate = 22050 # 44100 or 22050 or 11025... bitsPerSample = 16 #16 bit samples channel = 2 # 1 for mono, 2 means stereo.

On Tue, Jan 17, 2006 at 02:19:17PM -0500, me@sbooth.org wrote: > Hi all, > > I have recently added Speex support to my app (http://sbooth.org/Max/) and > am in the process of creating a GUI for the user to control the codec > parametere. I am new to Speex and as such I am not really sure which > parameters are more important than others, and deserve prominent > placement,

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

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

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

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

Brian... You've been both polite and helpful. Thanks. I do understand the dimensional nature of images and sound, though I admittedly glossed over the details while trying to draw attention to time rather than spatial artifacts. What I was looking for was confirmation that a properly designed application would decode FLAC without temporal issues. I believe you've made that perfectly

Hi. I'd like to ask a little about where Ogg stands today when it comes to streaming audio. The VBR has been the main argument in the incompatibility for streaming but has this changed somehow or will it? Is there something else besides quick and large scaled bandwith usage variations and the difficulty of buffering, that could cause problems? CBR could be used but some people say that it

There is a big, big difference between cbr and vbr, btw. The machine in question is a Pentium 3 700 oc'ed to 800, 512MB ram, win nt 5.1 (2k and xp being nt) etc In an earlier version the oddcast dsp ate up so much cpu time when set at say 64kbps (for examples sake) 44.1kHz, stereo that the machine became totally unusable and the winamp task had to be (slowly) ended with task manager. Now,