similar to: problems with speex_echo_cancel()

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of

Hi Julia, Version 1.2-beta1 has a bug in the echo canceller. Try either svn or 1.1.12 -- or apply this patch to 1.2beta1: https://trac.xiph.org/changeset/11882 Jean-Marc julia rg a ?crit : > Hello Jean-Marc, > > > > I solved the variable delay problem, but I still have trouble with > speex_echo_cancel(). When i try testecho.c with clean speech for far-end input > and

Hello Jean-Marc and all, I?ve got a problem with speex_echo_cancel usage. I only managed to get an attenuation of 5-8 db using testecho.c and it depends heavily on the playback/capture delay. I used clear speech as far-end input and same speech with a bit of reverb, attenuation and delay as mic nput. I also tried using played and recorded speech for mic input. I tried different constant delays

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

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

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

Hi, A couple things you may want to check: - set sampling rate to 8 kHz (at least for now) - make sure the far end signal in the playback signal is always a bit in advance (never late) compared to the mic signal. - Set the tail length to something around 100 ms. Also, if you're using two different soundcards (as I understand) for the playback and the capture, you're *never* going to get

Hi, I've attached a patch against SVN r14645 which adds GCC visibility information to all symbols exported from libspeex.so and libspeexdsp.so. It includes a configure.ac change to test that both the compiler flags and __attribute__((visibility)) works, and if so will #define EXPORT __attribute__((visibility("default"))) and if not #define EXPORT I've attached a diff output

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call

Hi, I've attached a patch against SVN r14645 which adds GCC visibility information to all symbols exported from libspeex.so and libspeexdsp.so. It includes a configure.ac change to test that both the compiler flags and __attribute__((visibility)) works, and if so will #define EXPORT __attribute__((visibility("default"))) and if not #define EXPORT I've attached a diff output

Terrific! The patch fixes my problem quite decisively. I will want to use this with a 16K sample rate. Is it expected to work well with just another slight patch, or will we need to modify it more deeply? Also, are there any tunable parameters besides tail length? I am wondering about the aggressiveness of adaptation and I know you are an expert in that area. Thanks so much, Michael

Hello, Need some help using the speex_echo_cancel. I've read the documentation about the speex_echo_cancellation function: speex_echo_cancellation(echo_state, input_frame, echo_frame, output_frame); (in) echo_state => speex internal state. (in) input_frame => audio captured by mic. (in) echo_frame => the signal that was played in the speaker. (out) output_frame => the

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

For some reason the attachment did not go through. Re-attaching. From: Mukund Raman Sent: Wednesday, April 01, 2015 6:12 PM To: 'opus at xiph.org' Subject: Opus multi-stream/surround: Audio corruption on decoded content Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally,

Hi, I implemented a small patch that allows the internal convergence state of the echo canceller to be saved in a file for later use, especially after a process restart or machine reboot. This enables immediate echo cancellation the second time the AEC is run. Of course this works only if the acoustic environment of the device doesn't change and if the soundcard latency is constant. To use

> I did a bit more testing, and ended up creating a set of testsamples, as > follows: > > ideal.pcm => me saying "This is what I'd like to hear" > junk.pcm => me saying "With a bit of luck, this is gone" > mixdown.pcm => ideal.pcm, with junk.pcm started 7 ms later and 3 dB lower. The problem with that is that it's nowhere near real

Dear all I have a problem with splitting up a data frame called ReVerb: ?? str(ReVerb) `data.frame': 92713 obs. of 16 variables: $ CHILD : Factor w/ 7 levels "ABE","ADA","EVE",..: 1 1 1 1 1 1 1 1 1 1 ... $ AGE : Factor w/ 484 levels "1;06.00","1;06.16",..: 43 43 43 99 99 99 99 99 99 99 ... $ AGE_Q : num 2.0 2.0 2.0 2.4 2.4

> - set sampling rate to 8 kHz (at least for now) > - make sure the far end signal in the playback signal is always a bit in > advance (never late) compared to the mic signal. > - Set the tail length to something around 100 ms. > > Also, if you're using two different soundcards (as I understand) for the > playback and the capture, you're *never* going to get echo