similar to: Speex settings and jitter

Hi, I'm trying to cover all Speex configuration options to set quality/mode/bitrate in my application and I was wondering if the following all makes sense: - abr_bitrate - use average bitrate and set it using bitrate parameter - SPEEX_SET_ABR - cbr_bitrate - use constant bitrate and set it using bitrate parameter - SPEEX_SET_BITRATE - cbr_mode - use constant bitrate and

Hi, Is there are document where the interaction between all the configuration options of speex is explained? Speex seems to have CBR, VBR and ABR. You can also use SPEEX_SET_QUALITY (SPEEX_SET_VBR_QUALITY) and SPEEX_SET_BITRATE which I suppose can't be set at the same time. Is there a list of possible combinations somewhere? I also have another question related to jitter. To minimize jitter,

The audio frame speex generates sounds pretty terrible most of the time, and I don't use it for jitter correction instead I just use it for dropped packets - so I usually drop the late packet. It sounds acceptable as long as I drop less than 5% of traffic (dropping 2 in a row makes a bad robot noise, so I reset the stream in that case). The good news is that on an unsaturated DSL line jitter

Right - and I deal with that on the receiver end based on an approximation of sender's and receiver's responsiveness - the minimum latency I've been able to get into the system is about 150 ms. Of that, jitter buffering is about 40-100ms. I'd love to figure out how to get that down without killing myself on thread switching or Win32 kernel calls, but ms has to actually implement

In my experience most of the jitter related issues are because people are using too small of audio buffer sizes that match the framing size of Speex - particularly in Windows. This isn't a problem with Speex, but as a programmer you should collect and append a few frames to match the size of your output audio frame buffer before attempting to play the sound. -----Original Message----- From:

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.? ?

hi, im a student from a local university in our place. i'm not sure if this is the place to post this question so please pardon me if im wrong. my questions is that when i transfer speex encoded samples through a rtp session wire, upon receiving the samples i immediately decode and play them, but i get a crackling sound when i decode and play each sample received. 1) i used 160*5 as my wave

Hi, On http://www.speex.org/docs/manual/speex-manual/node11.html#SECTION001140000000000000000 the bitrate for Speex Wideband mode 9 and 10 is 9 34,400 10 42,400 On http://tools.ietf.org/html/draft-ietf-avt-rtp-speex-03 the bitrate for Speex Wideband mode 9 and 10 is 9 34.2 kbit/s 10 42.2 kbit/s I believe the draft contains the right values: E.g. mode 10 Narrowband mode 7 -> 492

Hi, No, it only happens with speex_wb_mode. But like I said, it is because in wideband mode, the function sb_encoder_init is called and this function initializes the sampling rate using speex_*de*coder_ctl instead of speex_*en*coder_ctl In narrowband mode, the sampling rate is just initialized to 8000 st->sampling_rate=8000; BTW, I saw that my email was added to an existing thread instead of

I'm using speex to encode wav files. How does the --bitrate n option (use bit-rate n or lower) afects the quality and time of the encoding? If I don't use that option how speex determines the bitrate to use in encoding? Thanks in advance Miguel Gomes -----Mensagem original----- De: owner-speex-dev@xiph.org em nome de Chris Flerackers Enviada: qui 18-03-2004

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

Jean-Marc Valin wrote: >>In particular, (I'm not really sure, because I don't thorougly >>understand it yet) I don't think your jitterbuffer handles: >> >>DTX: discontinuous transmission. >> >> > >That is dealt with by the codec, at least for Speex. When it stops >receiving packets, it already knows whether it's in DTX/CNG mode.

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> In particular, (I'm not really sure, because I don't thorougly > understand it yet) I don't think your jitterbuffer handles: > > DTX: discontinuous transmission. That is dealt with by the codec, at least for Speex. When it stops receiving packets, it already knows whether it's in DTX/CNG mode. > clock skew: (see discussion, though) Clock skew is one of the main

Hello all, I am currently investigating the JitterBuffer struct provided in the Speex library, and I am actually thinking about using it with two different codecs: namely, Speex-NB and AMR-NB. From looking at the code, it seems that JitterBuffer is capable of working for any codec. Both Speex-NB and AMR-NB (and probably also other narrowband codecs) produce 20 ms frames and the sampling frequency

Hi, It could (and should) definitely be done, but it's somewhere (unfortunately not on top) on my TODO list. I think the simplest way (until I really integrate the preprocessor with the codec, which will take a while) to do it is to add a speex_encoder_ctl() call as you proposed. I'd call it SPEEX_SET_EXT_VAD_STATUS and I'm willing to merge a patch :-) Jean-Marc Le samedi 03

> Hi, > > It could (and should) definitely be done, but it's somewhere > (unfortunately not on top) on my TODO list. I think the simplest way > (until I really integrate the preprocessor with the codec, which will > take a while) to do it is to add a speex_encoder_ctl() call as you > proposed. I'd call it SPEEX_SET_EXT_VAD_STATUS and I'm willing to merge > a

> Hello, Hi :) First off, could you try to set your email client to break long lines before transmitting? In my (somewhat outdated) pine, the lines appear as VERY long lines when I try to reply, making it hard to read :) Minor detail though, I should probably fix pine. Some day. > The way you describe how the jitter buffer should be implemented makes me > wonder: How does the

Could a method be added to use the VAD from the preprocessor to send the packet with the 'dtx_enable' flag in it, so the decoder on the other end knows it's in dtx mode? (and SPEEX_GET_DTX_STATUS returns 1) At the moment, I can hack around it by doing something like: if (VAD for prev frame && ! VAD for this frame) { iArg = 0;

Hello, I'm new in using speex and programming with it. So, please help me to be more clear about it. I'm trying to write a voice decoder and encoder program using libspeex and using its variable bit-rates feature. At the encoder section I set the bit-rate of the output stream by setting "SPEEX_SET_ABR" option using this command: