similar to: Corruption with speex 1.1.x

It's not playing at half speed... It's playing at 2x speed. -Dragon Jean-Marc Valin wrote: > If it plays at half the speed, it means that either you're setting the > sampling rate wrong or (if you're using Linux) the machine at the other > end is using the broken i810 OSS audio driver which can't set itself to > mono. About writing code properly, just look at

What do you mean by write my code properly? With the way it's written right now it is encoding writing out to a stream and decoding from that stream but i'd like to use VBR but when i try it my app fails. and also on a different note I've written some test apps where it recordes from the mic encodes decodes then playes but when i try streaming it over the internet it has long

Dear developers, I am not able to compile speexenc and speexdec projects of speex1.0.5 on windows because of ogg.h file (and related file) are not included with the source code. So can you please include ogg.h and other file which should included in ogg folder in source code so that i can compile it and use this source. Thank you

Could someone tell me what list/newsgroup I should consult for help in implementing a simple (libspeex) decoder with my MFC/C++ waveOut? I just finished implementing the ogg vorbis equivalent, and I do have the sample code in the Speex Codec Manual. Seems like I have to dig awfuly hard to find windows sample code. Thank you very much. -------------- next part -------------- An HTML attachment

The following patch (based off of the current CVS tree) is necessary to compile Speex on IRIX (or any other system that doesn't include getopt_long in the system libraries). I would suggest further that src/getopt_win.h be renamed to getopt_long.h since it's not just Windows that lacks getopt_long. Michael Index: configure.in

Hi, to make in/output on STDIN/STDOUT work on OS/2 this small patch is needed: ---cut--- diff -ruN o:src/speexdec.c src/speexdec.c --- o:src/speexdec.c 2007-12-08 05:01:08.000000000 +0100 +++ src/speexdec.c 2008-01-12 00:48:00.000000000 +0100 @@ -231,6 +231,8 @@ { #if defined WIN32 || defined _WIN32 _setmode(_fileno(stdout), _O_BINARY); +#elif defined OS2 +

On 01/04/2008, Erwan A <mout551 at hotmail.fr> wrote: > > Hi, > > Yes i agree with you. You don't have to delete these files. > > But if i cross compile with ogg header files, i have the following error : > > > /usr/lib/libogg.so: could not read symbols: Invalid operation > collect2: ld returned 1 exit status > make[2]: *** [speexenc] Erreur 1

this is not really a development question, but i didn't find another mailing list to ask it. any idea why size of wav file doubles when it is encoded to speex and back to wav: $ ls -ls testi.wav 40 -rw-r--r-- 1 foo foo 40674 May 4 14:38 testi.wav $ speexenc --denoise --agc --quality 10 testi.wav testi.spx Encoding 8000 Hz audio using narrowband mode (mono) $ ls -ls testi.spx 20

I am using speex for encoding and decoding speech files in a speech synthesis program. I am using concatenative approach for speech synthesis using diphones (small speech chunks) as basic unit. Now in my application, the order of diphones to be joined after decoding is determined at runtime. Therefore I have to encode each diphone independent of others. I find out that when I encode diphones

Hi all, I'm trying to use libspeex 1.1.10 on an ARM926EJ-Sid(wb) rev 3 (v5l). I executed the speexenc and speexdec test files and they can encode and decode. But I'm getting 95% of cpu utilization on the codification and 44% on the decodification. I saw in the post: http://lists.xiph.org/pipermail/speex-dev/2005-June/003485.html that this version of speex works fine on ARM

Hi, I just started using speex recently and I've been having problem with Speex ver. 1.1.10 decoder. I used Speex to encode a wav file to spx then decode it back to wav file to see the voice quality that results from different speex options. Somehow, I couldn't get the original wav file back. I used both Speex ver 1.1.6 and ver 1.1.10 to both encode and decode, despite the same output

Hi, I have done some code cleanup of my speexenc patch for skeleton support. Also a warning message have been added to let the users know that some decoders may fail with skeleton output enabled. The speexdec patch enables speexdec to play skeleton enabled speex file. Thanks in advance for the feedbacks. --tahseen -------------- next part -------------- A non-text attachment was scrubbed...

Dear sir, I've a problem that the .wav file couldn't play after calling "sampleenc male.wav|sampledec male_speex_15.wav".I found that the new file male_speex_15.wav is smaller than the original file in size.I implemented the test on Linux system.The original file male.wav is 96044 bytes,while the new file male_speex_15.wav is 96000 bytes.I'm eager to know the reason.Thankyou!

Hi, What kind of latency is expected using 8,16,32 khz? I am trying to do a realtime stream server, and I am having latency above >500ms ( I capture the sound using the mic, encode it and send it to the client). I am using ALSA system, a SB128 PCI and a 800Mhz P3 . What can I do to lower the latency? I tried a test : arecord -t raw | speexenc - - | speexdec - and I found that I also have a 500

How can I reduce latency on the following? arecord -r22050 -c2 -fs16_le | speexenc - - | speexdec - I have no coding experience in C, so can't write my own stdin/stdout wrapper. this is just an echo test, but it's severely lagged. the mailing list posts on this subject didn't give a solution. I used this over ssh, and managed to have a usable conversation with someone, accept for the

Hi all, i've to encode voice from a voicemodem. I choose speex 1.0.5 for its quality in voice encoding. I've tried to implement an encoder but unsuccesfully. Here's my code: /* ============ SPEEX stream ENCODER ============================================ */ int SPEEX_EncodePCM(struct _IDA_ClientSocket *IDA,char *buffer,unsigned char *PCM,int num_samples) { /* buffer point to the

Hello Bernard, Friday, December 13, 2002, 7:22:54 AM, you wrote: Bernard> I've one small request - an option on speexenc that allows you to Bernard> specify a speex file to append to, allowing you to concatenate Bernard> streams without losing quality by decoding & encoding. Ideally, it Bernard> would: But you can `cat speex1.ogg speex2.ogg> unionspeex.og` and still have

Hi, I've enocded and decoded a file using speexencexe/speexdec.exe respectively. However when i use the example code to encode a file and use speexdec.exe to decode it; not only do i get a file that is 3kb larger than if i had encoded it with the .exe but also I get a "this doesn't look like a speex file" message, and it is unable to decode it. I've checked around and it

Hello, I encode a file with speex so i get a .spx file but i want to decode the same file i get only one second of my file and nothing Some idea why? Thanks Damien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070430/d5a95090/attachment.html

Hi All, I would like to communicate speech over a network compressed using speex. However, I do not want to communicate a whole Ogg-formatted file. I am interested in only the speech frames. I invoked: speexenc input-file-name - > raw-speech This, I am thinking gives the speech frames only. To play this back at the receiving end, do I need to format it into Ogg formatted file before I can