similar to: Programming questions

Hi there. I am having a little trouble understanding the frame sizes chosen by the codec. testenc_uwb.c from the speex-1.0 source distribution has a framesize of 640 hardcoded and makes use of this value exclusively. However, a mode query on the actual codec returns 320 as a framesize for this mode. int tmp; speex_mode_query(&speex_uwb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);

> you are better off using the vogg orbis codec. speex is meant > specifically for telephonic voice. it takes a single human voice and > compresses it well. it cannot handle muliple voices or music very well. That part is true, so of course it depends on the application. I guess I should have added that for most applications, 16 kHz is recommended instead of 44.1 kHz. > For

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

Hi All, I am using Speex for encoding/decoding the audio stream for my streaming application. I have used almost the same code In the sampleenc.c and sampledec.c, except that I have used The buffers from the mic as input for encoder and encoded audio As input for decoder, instead of files input. My Problem is the audio played on the receiving side after decoding Is only a "hush" or it

Hi all, I have just started playing with speex, and come up with the following code, which just encode a frame of 160 shorts, and the decode it. For some reason the decoded sample is completely different than the original one. is my code wrong? If so what? Or is it a reasonable which depends of values that weren't correctly set? Thanks, Andre #include <stdio.h> #include

For a streaming application like VOIP, you collect 20 ms of samples, feed this through the encoder, stick it in an RTP packet, and send if over the network. On the receive side you feed packets through a jitter buffer to the decoder, and then copy the output audio to your output device. Speex runs efficiently on most compilers, so the real-time requirement is not a big deal, as long as you

First let me introduce myself: Sr Computer Scientist 20+ yrs SW/HW development Audio Rookie < 60 days Intel based computer Soundblaster 128 PCI Linux Fedora Core 2 ALSA - Advanced Linux Sound Architecture ALSA provides the audio and MIDI functionality to the Linux operating system as of Fedora Core 2 and is the future of Linux (so I have been told) http://www.alsa-project.org/ I have read

Hi all Sorry if this is a dumb question: does the input_frame passed to speex_encode_int *have* to be frame_size samples long? e.g., If I only have 100 samples left to encode (which is less than the frame_size of 160 samples), can I just use an array that contains 100 samples, or do I need to create an array containing the 100 "real" samples plus 60 null samples at the end?

Hi, I am new to speex and I am trying to use the sample program given in the speex document. I have made some modification to that sample, so that input can be read from a file and the output can be re-directed to the file. I tried to encode an audio file using the sampleenc and decode the same by sampledec.c I am able to do it successfully, but when i try to play the output file from the

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

I have successfully DECODED speex on TI C5509: #define TESTENC_BYTES_PER_FRAME 20 /* 8kbps */ #define TESTENC_QUALITY 4 /* 8kbps */ I am trying to generate the files I need with speexenc.exe: speexenc -n --quality 4 -V male.wav male.spx But I can't decode the files on C5x. Yes, I have seen that speexenc.exe adds Ogg header and

> I have compiled the sample sampleenc.c w/o any errors. then I tried to > compress a sound file which is in WAV and RAW format but the file size is from > 200K to 80byte. I think there is somehing wrong but no idea why will be like > that. any idea? > > I used > 1. sampleenc test.wav > test.spx > 2. sampleenc test.raw > test.spx No wonder it's not very

Hi, I want to take a .wav and generate a bytestream that I can feed direct to speex_decode(). The speexenc program appears to produce it wrapped up in an ogg stream, and it doesn't have any sort of "raw" mode. I've tried hacking on it to remove the ogg parts, but I must have got it wrong, as my decode dies after decoding 160 bytes (which I believe is one frame) with an

Thank you for the quick response Thorvald, but I think that's not the problem here :[ I know how to capture the buffer and how to play it in the output buffer of the DS. The problem is (probably) with same kind of short/floats/bytes error in conversion/copying that the coder doesn't get. I can have my buffer locked during the compression, it not the problem at the moment. What I really

Changing from using floats to shorts did fix the high pitched tone problem. I'm having other problems but I'll look into it more first. SteveK wrote: > > On May 21, 2006, at 6:33 PM, Kevin Jenkins wrote: > >> When I just copy the microphone input buffer to the output buffer the >> sound plays OK. But if I encode and decode the buffer through Speex I >>

Hi everyone, Our team is working on a realtime voice communication application. We are using openAL 1.1 to capture the samples and then encode them using speex. On the remote machine we decode the samples and play them using openAL again. Capture and play formats for the samples is AL_FORMAT_MONO16 and frequency is 22050. we are using wide band encoding/decoding. The encoding sample

I'm getting garbled playback with decoded fragments and I'm hoping someone here can point me in the right direction to correcting the problem. Essentially I'm capturing audio from the microphone. I stream it over the net, but for testing purposes with this API I'm just grabbing the whole chunk and encoding / decoding it right away and then updating the sound buffer for

> For encoding into and decoding from "raw Speex stream" (if you can call > it that), I suggest you start with the sampleenc.c and sampledec.c > example code in Appendix B of the Speex Manual. There is no such a thing as a "raw Speex format". Also, sampleenc.c and sampledec.c are good for learning how to use the API, but the compressed format shouldn't be used in

Hi All, In sampleenc.c and sampledec.c, if I change the FRAME_SIZE to any other value, I get very garbled speech. Can anyone tell me if I need to set something else if I would like to change the frame size ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070502/35ea524b/attachment.htm

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!