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2011 Dec 06
2
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two thread send frames to only encoder at same time,two thread will use one buffer .it will work wrong if not Handling thread synchronization. if not use same state, it need more memory, but no need to consider thread safe. ------------------ ???? ------------------ ???: "Jean-Marc Valin"<jmvalin at jmvalin.ca>; ????: 2011?12?6?(???) ??3:24 ???:
2011 Dec 07
1
回复: 回复: 回复: (no subject)
Hi, i think it will work correct.when one thread use encode,the others wait for. in java, i think we can do like this: synchronized (Encode) { Encode.encode(short[] in, byte[] out); } In the doc of speex: it says the speex is not thread-safe. ------------------ ???? ------------------ ???: "????? ?????????"<romius99 at mail.ru>; ????: 2011?12?7?(???) ??4:00 ???:
2011 Dec 06
1
回复: (no subject)
yes? you are right.but its not one instance of codec. ------------------ ???? ------------------ ???: "Jean-Marc Valin"<jmvalin at jmvalin.ca>; ????: 2011?12?6?(???) ??12:52 ???: "speex-dev"<speex-dev at xiph.org>; ??: Re: [Speex-dev] (no subject) Actually, it *is* thread safe as long as you don't use the *same* state at the same time from two calls.
2011 Dec 07
0
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I'll try to explain. I want to create dll with only one function "Encode" that will encode file to speex format. In my application I will use this function from 16 threads, if it will work correct? 2011/12/6 ???. <xialonghua at vip.qq.com> > > ** > two thread send frames to only encoder at same time,two thread will use > one buffer > .it will work wrong if not
2004 Aug 06
1
Java bindings
Hi, I have writen a simple JNI wrapper for speex so i could use it from Java, so far i managed to record from a raw pcm file and from the javasound inputLine and to decode the output in narrowband CBR mode and play it back still using javasound. And it sounds quite good. I haven't tried anything else because my project has no use for wideband and ultra wideband. VBR would be nice but i have
2005 Jul 06
2
click at end of playback
Thank you very much for your immediate attention! I've exported my wav files to raw files before encoding and the click is now gone. That sure beats discarding the last packet, which was the workaround "solution" I was going to implement... Best regards, David Resnick -----Original Message----- From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] Sent: Thursday, July 07,
2004 Aug 06
1
LIB usage
Hello, I'm trying to use the speexlib but I just can't do it. I've made just like the example (on the site) says. Some times I get SIGFAULT. Also, I'm reading the input_frames from /dev/dsp (directly). What is missing here? speex_bits_init(&bits); enc_state = speex_encoder_init(&speex_nb_mode); speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frame_size);
2005 Jul 06
3
click at end of playback
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2005 Jan 04
3
different output for opt/non-opt on win32 platform
hi all, thanks Jean for the codec. I am new to speech coding. Am interested to implement speex on a TI dsp, say C54. just downloaded speex v 1.1.6. when I removed optimisation flags for win32 in libspeex.dsp, I get a slightly different output for decoding using speexdec.exe (debug version). no prob for speexenc.exe except for some differences in the ogg header. for anyone interested, i
2005 Jul 06
0
click at end of playback
If you have a couple minutes, it probably wouldn't be hard to see where the bug is (I just don't have time this week). It would have to do with the granulepos computation in speexenc. Jean-Marc Le jeudi 07 juillet 2005 ? 08:41 +0300, David Resnick a ?crit : > Thank you very much for your immediate attention! > > I've exported my wav files to raw files before encoding and
2005 Jul 06
0
click at end of playback
Actually, I just tried again with the wav file directly and it just seems to be a bug with the way the speexenc utility handles wav files. The problem doesn't occur with the equivalent raw file (which I used for testing). It's probably really stupid anyway. Jean-Marc Le mercredi 06 juillet 2005 ? 14:00 +0300, David Resnick a ?crit : > I am using speexenc/speexdec to convert WAV files
2004 Aug 06
2
production code
hi. i joined the list a week ago. i'm using speexlib as the codec for GameSpy's new voice SDK [for voice chat for video games]. everything works well. kudos. i'd like to make a contribution to the codebase. i am bothered by the spew of warnings ( about 300 ) from my compiler. i typically frown on warnings anywhere in my codebase. COMPILER: Visual C++ .NET 2003 version 7.1.3088
2007 Dec 21
2
(no subject)
> the speex files cannot be played by "Windows" (I assume you mean the > media player) natively. speex codec will have to be supported as a > playload type. > > to do that, you have to install a Speex ACM. > take a look at > http://www.free-codecs.com/download/Speex_Voice_ACM_Codec.htm Actually, I tend to recommend using Ogg files instead of wav. Those can be
2007 Dec 21
2
(no subject)
Hi Jean, This is my last attempt at trying to use speex for a project we are working on. I have contacted you before, but i have had no joy what so ever trying to find a working example in c++ that shows how to add a header to a raw speex file, so that it an be played using windows. If you do know of a working example can you please supply a link. As this is a commercial project maybe you could
2009 Jul 20
0
No subject
faced this exact same problem a few times on more than one servers and it was 1) dialplan issue which was not hanging up the zap channels correctly 2) using more than 8 spans on a server. Asterisk can't handle more than 96 zap channels on T1s. FXO/FXS combinations can vary the number of spans but if you know what I mean by spans, in production don't use more than 6 spans. On 2010-03-17
2007 May 23
1
(no subject)
Hello Sir/Madam, I wish to use the speex codec on eclipse(c/C++ environment). The aim of my project is to encode speech data using speex, the drop some packets randomly and then decode the data and the see the effects of dropping the packets. Could you please letme know how I can setup the source code on eclipse. Thanks Nihar
2004 Aug 06
1
(no subject)
Hello... I'm developing Embedder VoIP as a project in my university and I'm thinking that I can use Speex as my codec of choice... I just wanted to know if any of you has done realtime Speex encoding/decoding with a dsp in an VoIP enviroment -----BEGIN GEEK CODE BLOCK----- Version 3.1 GED/J d-- s:++>: a-- C++(++++) ULU++ P+ L++ E---- W+(-) N+++ o+ K+++ w--- GE/CS/MU d-- s:- a-- C++(++)
2007 Jul 12
0
No subject
Use the astDB. Add entries like: /vipMap/custNumber: Target number eg. /vipMap/441364698123: 07712191046 In this case, 441364123123 is my VIP customers number, and 07712191046 is my mobile number.. You can do a little bit of webby stuff to maintain this via the manager interface. If you can't find any, drop me an email - I have some I modified after finding some ideas online. (I
2015 Jan 15
1
speex
How are you: speex开源真好,我早想使用了,但我发现speex不支持64位编译,我真痛心,由于apple发表了一片“不支持64位的app一律不能上架”,希望我能够使用这个‍ Speex open source is really good, I want to use the early, but I find speex does not support 64 - bit compiled, I really painful, because apple issued a "does not support 64 - bit app is not on", I hope I can use this‍ -------------- next part -------------- An HTML attachment was
2007 Jul 19
2
noise reduction capabilities
Hi, I am considering to use Speex in a particular environment and I want to ask if you thought it realistic. I am going to use Wireless SIP (G.711) handsets in an environment that is typcially noisey (background), connecting to an Asterisk PBX. I am considering converting the G.711 to Speex so I can "massage" out the background noise. Is this realistic? Any input would be great.