similar to: About "Convert 8->16 bits"

Good day, a quick bug while encoding speexes in speexenc. While reading a .wav in function read_samples, size can get to negative values, as long as fread is able to go forward with reading the file. This can happen with Protools-generated wavs, where they put additional information after the "data" fourcc. I created a very quick hack by adding: int tentative_read =

Hi everybody, I have been examining on how I can use speex completely so I have one more question. I looked at examples and source code and it goes on. speex_bits_insert_terminator is called in speex_bits_write function. Shall I call this method one more time anywhere? Because in speexenc project, firstly speex_bits_insert_terminator is called then speex_bits_write is called when we assume that

Hi everybody, I need clarification about speex_bits_write function usage. There is an example usage of it. nbBytes = speex_bits_write(&bits, cbits, MAX_FRAME_BYTES); I examine the content of this function and which value should I set to MAX_FRAME_BYTE. In speexenc project, it is 2000 and in manual it is 200. For example, I set it to 200 but the length of bits->chars is 250

Hi all and Jean-Marc, Firstly thanks a lot to you for helps, at last I accomplished to use speex-1.2beta1 in my speech application. I try to understand speex source code and improve speech quality. The problem in my application (like many people in the list says) is noise. So I've added VAD in preprocessor as following. VAD in preprocessor is very good, there is not noise outside speech. I

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, 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 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

/input_frame/ is a /(float *)/ pointing to the beginning of a speech frame What is the range on the floats in the input_frame that speex_encode() needs? The library that I'm using to get the sound gives me floats that are between -1.0 and +1.0. I was assuming that this is also the case for speex, but I don't see it documented. I'm thinking now that maybe this is not the case for

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 >>

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

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 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,

Thanks a lot for your explanations. Is it right I understand? if I've set SPEEX_SET_BITRATE to 16000(bps) this means that output is 16000/8=2000 bytes and I should set MAX_FRAME_BYTES and the length of cbits to 2000. Is it right? Maybe it is simple question but your answer is important for me, actually thanks a lot for your helps. Yasemin The last argument should simply be the size of your

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.

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

Hi Jean Marc, We are running the speex echo canceller code on a Windows Mobile smartphone using a TI OMAP 201MHz 850. We are making voip calls and audio is working without echo canceller code. Simply adding the following 2 functions whether the echo cancellation is wired in (ie we use the output frame ) or not performance becomes terrible. Note 2 separate threads are used to handle audio.

Hi, Speex-dev I've been able to get Speex to compile on SM400M(MIPSII) with problems. ------ Rebuild All started: Project: speexenc, Configuration: Debug SM400M (MIPSII) ------ Deleting intermediate and output files for project 'speexenc', configuration 'Debug|SM400M (MIPSII)' Compiling... getopt.c getopt1.c skeleton.c speexenc.c .\speexenc.c(501) : warning

Hello Is there a way to restrict access to Dovecot server during mailboxes convertion ? I need dovecot to be running to convert mailboxes from a format to another, but I would like to block all our LAN clients until the end of convertion. If there is no internal Dovecot mechanism I will use some firewall rule on the server. Thanks

Can you first try the svn version (or apply https://trac.xiph.org/changeset/11882). As I mentioned in a previous email, there's a bad bug in the 1.2beta1 echo canceller. Jean-Marc Coffey, Michael a ?crit : > 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