similar to: Compress ratio

I ran some further tests on mdf and here are the results: 1. reduced tail length to 100ms, aligned mic and speaker signals to within 10ms - almost no echo attenuation 2. aligned mic and speaker signals to within 5 samples - still almost no echo attenuation 3. ran testecho using the same file for mic and speaker - very good echo cancellation (of course this is expected, but I needed to do a sanity

Thanks for alerting me to the new changes. I just tried the latest code from SVN, but unfortunately I still have just about the same results. The estimated echo that gets subtracted from the actual echo is such a small signal that it doesn't really result in any noticeable echo attenuation. I currently have my filter size set to 2 seconds even though the echo in my microphone file is only

It seems that speex (speex 1.0.4 on OSX 10.3.9) is writing over the input buffer (with what looks like a sine wave). Included below is a short program that demonstrates the problem. Is this a bug, or am I doing something wrong? I don't see this behavior mentioned in the docs. I don't really need the input buffer, but I'm guessing this problem is related to a high pitched sound

I'm pretty much sure of it. When I test inverting the inputs, my output is pretty much the same as my speaker signal. Whereas the way that I normally test the output is my mic signal with very little attenuation. If you are interested I can send my test files; they are about 94KB each. -Jason --- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Are you sure you're

Hi, I have a module sampling raw PCM data on Win CE as 10ms time slice (160 bytes), mono, 8000HZ, 16 bits per sample. Does anyone know what is the mflops for using fixed point on a Win CE compared to using floating point? Looking at the manual, "In practice, frame_size will correspond to 20 ms when using 8, 16, or 32 kHz sampling rate." for a 8 kHz sampling, the framesize should be

Hi all, Please a moment to look my source code, this is very similar to example of the documentation. I added FIXED_POINT flag. My source code compile and build but not decode correctly (the error may be in the while). I work with eVC 4.0 and pocket pc 2003. Someone know what is the error? Thanks. Rodrigo. #include "speex.h" #define FRAME_SIZE 160 void CPlayerDlg::OnButton3() {

Had a try. The reason why a simple delay is not that good is mainly due to the initialization of the filter parameter that still takes a few seconds (if they are perfectly in sync, you sort of get lucky). Otherwise, you real recording seems to have something odd in it. Are you sampling from a different card then the one that's playing the sound? or maybe the mic (or something else) in the room

Hello, I would like to test the echo cancel algorithm available in unstable version 1.1.4. This echo canceller can be used with other codecs like G711? Somebody could send me some documentation or sample, or explain the next functions: SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); void speex_echo_state_destroy(SpeexEchoState *st); void

Hi, My problem is at the second fread function, fread(&nbBytes, sizeof(int), 1, fin); //the first fread fread(cbits,1, nbBytes, fin);// the second fread. When I'm debugging always nbytes is greater than cbits. This give me a execution error. Thanks. Rodrigo. <p><p><p>-----Mensaje original----- De: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org] En nombre

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

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

Hello, When I decode a recoded sound encoded by speex, it has too much noise. The noise is "slightly" reduced if I set the quality to 10. How do I get rid of this noise? The code is as follows. For encoding . state = speex_encoder_init ( &speex_nb_mode ); tmp = 7; speex_encoder_ctl ( state, SPEEX_SET_MODE, &tmp); // set quality tmp = 10; speex_encoder_ctl ( state,

Hi Jean, Thank you for your reply. Ok... I'm gonna use 'samples per channel' everywhere I see 'samples'... but what about the 'speex_echo_playback' function ? it does the following loop: ... for (i=0;i<st->frame_size;i++) st->play_buf[st->play_buf_pos+i] = play[i]; ... So... if frame size is 'samples per channel' it will copy only half the

Hi, I'm using the echo cancellation api and I would like to clarify the 'frame_size' parameter used in speex_echo_state_init(frame_size,filter_length). In the 'docs' it says: "...where frame_size is the amount of data (in samples)you want to process at once..." So... here are my questions: if I use stereo input/output do I have to put the samples doubled ? For

Since r12660, the speex_jitter_get with high latency doesn?t works, I have no sound. Before this release, the speex_jitter_get works in all conditions. speex_jitter_get return void, then I cannot know the reason of this problem. Regards Ouss -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: dimanche 18 mars 2007 23:07 To: Ouss Cc:

I've tried some further debugging to see what mdf is actually doing. Instead of sending: tmp_out = (float)ref[i] - st->y[i+st->frame_size] to the output, I just sent st->y[i+st->frame_size] to see what was being subtracted from the microphone input. When I open this in Audacity, I see a very small signal at about -40dBm. The actual echo in my sample has a power closer to -20dBm.

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

Attached is the amended second patch. It now extends the multistream API as well to 80/100/120 ms and incorporates changes based on Mark's comments. Thanks, Felicia On Mon, Jun 13, 2016 at 4:21 PM Felicia Lim <flim at google.com> wrote: > Hi Mark, Jean-Marc, > > Thanks for your comments. > > On Sun, Jun 12, 2016 at 6:34 AM Mark Harris <mark.hsj at gmail.com>

Please provide the input file that produces this with opus_demo. On 16/04/15 03:24 AM, Suresh Thiriveedi wrote: > Hi Jean-Marc, > > Could you please update if you got a chance to look into. As I > mentioned, I don't see the same issue in 1.1.1, but I don't see any > difference in 1.1.1 other than optimization based on the architecture. > This optimization could have

Hi Felicia, A few comments: > - /* CELT can only support up to 20 ms */ > subframe_size = st->Fs/50; > - nb_subframes = frame_size > st->Fs/25 ? 3 : 2; > + nb_subframes = frame_size/subframe_size; This will use six 20ms frames to make a 120ms packet, even for SILK-only mode where frames can be up to 60ms. For SILK, two 60ms frames would be a more