similar to: Decoding multiple Speex frames

On 21/04/2008, Danny Peled <Danny.Peled at nice.com> wrote: > > Hello, Hi, > Could you please explain or send an example for decoding a packet that > contains multiple Speex frames. > > According to the API document the speex_decode_int() decodes only one frame > of speech. the frames are just concatenated together in an Ogg Speex packet. Simply call

On Fri, Apr 4, 2008 at 12:19 AM, Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > Jahn, Ray (R.) a ?crit : > > > Dear Speex codec community: > > > > I am working on conversion of voice files. I could not figure out how to use speexdec.exe 1.2.3 in piped mode in order to avoid the creation of the potentially large intermediate *.wav or *.pcm files. Any

I have an audio file compressed by JSpeex that I'm trying to decompress on the iPhone. I've been trying to work from the speexdec example in an attempt to port the core logic to ObjC. I have a class that does the decode and passes the result back to another class. (currently the other class is a unit test.) My other class is writing the converted output to disk. I compare the results of my

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

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

Hi, I was wonder how Speex handles (if at all ) lost packets when decoding the bit-stream . 1. Does Speex has any mechanism to handle lost packets (like AMR has for example)? 2. I am using Speex in a constant frame size (160 bytes), narrowband, VBR off, SPEEX_SET_QUALITY = 3 and SPEEX_SET_COMPLEXITY =1 . I encode speech stream and sending the encoded packets to the

I tried feeding in the 3 encoded frame in ONE BLOCK, and calling speex_decode() 3 times in a roll. Only the 1st frames came out perfectly. For the other 2, I got "corrupt" frame warning. I was supposed to get 38 bytes consumed each frame (narrow-band, VBR off). I tried speex_bits_remaining() to peek on the # of bits consumed, and got variable (clearly wrong)#s returned. But if I

I'm having trouble with a Speex VOIP effort ... my first use of Speex. My development platform is Windows XP, but I intend to bring this up on OS X as well. At the moment, I send across speex frames (say 4 per packet... a configurable setting), and the settings of both encoder and decoder are simply fixed in code at the moment (wideband, short format input). I am fairly confident that I

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear

Dear Jean-Marc and Peter: Thank you both very much for your time and advice. I did not realize that Lame MP3 code has a -r option (without reading its code). I have tried the suggested command lines verbatim with the added -r option, along with other combination of option settings. I could not figure out how to eliminate the distortion in the result, like voice turning either high or low

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

Jean-Marc Valin wrote: > > > The decoder will return -1 if it can't decode anything else, so you just > do something like: > > while(1) { > err = speex_decode_int(state, bits, pcm); > if (err==-1) > break; > output_data(pcm); > } I think I still didn't fully understand. As a test I encoded some frames (wideband, with quality set to 8),

Hi, I've been looking at using speex over RTP. Section 3.3 of draft-ietf-avt-rtp-speex-05 states that: "For the purposes of packetizing the bit stream in RTP, it is only necessary to consider the sequence of bits as output by the Speex encoder [speex_manual], and present the same sequence to the decoder." So, I assume that I just pass the whole contents of the RTP packet (less the

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Hi, I am new to speex library. I am doing project in VC++. My project details are, I have to capture voice from mic and encode it then decode and play. I am capturing the voice and storing in a buffer from there i am reading frame by frame ( 160 bytes) . These bytes i am converting to short int then calling speex_encode-int(). The output buffer i am passing to doder speex_decode_int() function.

Hi, I got my hands on the LCA 2003 CD today and I tried to listen to it on my powerbook running Linux. Instead of getting speech, I got an earfull of noise! Attempting the same experiment on an i386 with the same CD produced understandable speech. It seems that the output routine in speexdec converts the output data to little endian short format, which is incorrect on the ppc architecture. The

Hello, I would like to run a script over a speex file which determines, how long the speech contained in the speex file is. So I run through the ogg file, skip the first two ogg _pages_ (speex header and comment header, both contain IMHO only 1 ogg _packet_), and then count in any ogg page the number of the ogg _frames_ (which, unless I used -nframes while encoding, which I did not), and multiply

Hi Jean, > > > Well, just trace it, how many times are you > calling > > > speex_decode_int()? > > > > Maximum is 51 times per second. Will this cause > any > > CPU high utilization? > > That's normal... What CPU are you using? If it's a > fixed-point CPU, then > the reason is probably just the fact that the packet > loss

Hi all, I've been hitting the odd floating point exception errors using the new alpha encoders. They're encoding a live stream so I can't reproduce the crash on demand, and it might only happen once every few days so it's been fun trying to get a backtrace. The crash happens in a fairly scary block of code, I wouldn't have a clue where to begin trying to fix it!