similar to: input_frame format

The API reference says it is "+-2^16 range." Which is right? I've yet to successfully encode or decode anything. Jean-Marc Valin wrote: >It is documented and the range is +-32767 > > Jean-Marc > >Le mardi 07 juin 2005 ? 11:59 -0500, Matt DeMoss a ?crit : > > >>/input_frame/ is a /(float *)/ pointing to the beginning of a speech frame >>

Thanks, it makes more sense to me now. Does speex_bit_read_from also append in that manner? Jean-Marc Valin wrote: > speex_bits_reset is before encoding, not decoding. If you don't call it, > then everything you encode will be appended so it will grow without end. > As for the point of the SpeexBits struct, it's there as a sort of "array > of bits" to it's easy

The example in the documentation demonstrates using speex_bits_reset before encoding and decoding each frame. " speex_bits_reset(&bits); speex_encode(enc_state, input_frame, &bits); nbBytes = speex_bits_write(&bits, byte_ptr, MAX_NB_BYTES); " What happens if this is not done? I think I'm missing the point of having a SpeexBits object.

I'm confused about the format of the PCM/WAV data for speex_encode & speex_decode speex_encode(enc_state, input_frame, &bits); In the manual, it says "input_frame is a (float *) pointing to the beginning of a speech frame" (for encode) But I've found that in speexenc.c and testenc.c, short* is used instead of float* So, isn't it signed 16 bit samples(if

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);

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?

Hello all, I am trying out the echo cancellation of Speex (1.2beta2) on Windows XP. I use the Windows Media API (i.e. waveInOpen, waveOutOpen etc) for playback and capture, but I cannot make echo cancellation works in my use case. Here is a description of what I tried: (the sound is in 8kHz and 16 bits per sample, the frame size is 240, tail_length is long enough to cover the far end to near end

Speex 1.0.3 uses floats, but the unstable version 1.1.x uses short. Jean-Marc Le ven 16/04/2004 16:04, Kenji Chan a crit : > Im confused about the format of the PCM/WAV data for speex_encode & > speex_decode > > speex_encode(enc_state, input_frame, &bits); > > In the manual, it says input_frame is a (float *) pointing to the > beginning of a speech

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 get a high pitched constant tone in the background. I actually do hear my voice speaking when I talk, but it's faint and much quieter than the tone. Here's what my data looks like: Input is the first 5 floats of each input buffer. Output is

Hi Jean-Marc, For the residue in the echo_cancel function I pass in a pointer to an array of floats. First I used an array the size of the framesize, but I discovered this causes crashes because the echo_cancel funtion writes framesize+1 floats to the pointer. I find this plus one strange, since almost all things are exactly the framesize. Is this an error in the echo_cancel function or simply

I have (finally) posted my Speex wrapper classes. They are at: http://www.grandgent.com/spx/tgAudioCodec.zip I followed your recommendations and they worked fine with 1.1.0. However, I'm still having the same problem with 1.1.7 that I had the last time I tried to upgrade. I'm using the same code with both versions, except for calling speex_encode_int instead of speex_encode, and

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> That is a dumb question :)<br> <br> The encoder expects SPEEX_GET_FRAME_SIZE at all times. If you are

Jean-Marc, Ok, I understand. It's floating point but I don't have a mechanism in place yet to capture the source data. I'll do that, reproduce the problem, and put together a minimal sample program that can reproduce it given a data file. Thanks, Tom Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > > Hi Tom, > > In order to trace the problem,

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 >>

Hello, i'm adding echo cancellation to my voip application, but when i call speex_echo_capture() or speex_echo_playback() causes an assertion failed, it seens that there is a delete in any where. i've encapsulated echo canceller in a class that storage the state and simply call the functions: void EchoCanceller::Playback(short *echo_frame) { speex_echo_playback(echo_state,

-------Original Message------- From: Eid Date: 10/09/05 14:07:44 To: speex-dev@xiph.org Subject: problem in encoder/decoder hi all, I'm developing an app to broadcast voice over web. i tried to use speex and download ver 1.04 documentation and use it to develop encoding and decoding but after decoding voice is damaged i attach my encode/decode function. BOOL

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

Hi All, I am working on my master project to port the Speex Acousic Canceller to FPGA. But I am not able to get a satisfied result in its performance whenI integrated it to a SIP softphone as my first step. The SIP softphone calls to a PSTN gateway and lands to my land line phone. I can hear the echo of my every word while I talks on the land line phone. I am using G.711 codec, frame_size 80

Ok, this is a silly question, but what does the jitter buffer do? I'm really new to audio, so please bear with me. From what I gather (primarily from the list archive), the jitter buffer is a wrapper around the Speex decoder. I give it the packets I receive, in whatever order I receive them, and then it gives me back a clean stream of audio samples. But what I don't entirely

Ah, I'm sorry, I have read the manual and believe I have a reasonably good grasp on how to use the Speex encoder and decoder altogether. In fact I've been using it with great success in my P2P SIP/RTP VoIP application for almost a year now; it's been working wonderfully and I can't thank you enough. However, the manual makes no mention of the jitter buffer, nor does it (so