similar to: speex DTX chore

Hi guys I have tried compiling this attached code, I made all the buffers 320, there is no trace of a 160 buffer, but I get a " SpeexEncoder requires 320 samples to process a Frame, not 160" error. Maybe there's something I'm missing, here's my code: import java.io.IOException; import java.io.FileOutputStream; import java.io.File; import

hi, i did some tests about performance in jspeex ... uls@mainframe:~/eclipse/avrelay$ java TestClient Diff1: 0 10 Diff2: 141 started. uls@mainframe:~/eclipse/avrelay$ <p>{ SpeexEncoder e1=new SpeexEncoder(); e1.init(0,1,44100,1); byte[] by1=new byte[320]; long l1=System.currentTimeMillis(); long l2=System.currentTimeMillis(); System.out.println("Diff1:

I done it speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_START, &g.s.VADstart); speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_CONTINUE, &g.s.VADcontin); /*char *lisDebugCh = (char*) malloc(20); int decimalFcvt, signFcvt; static int firstDebug = 1; lisDebugCh = gcvt(g.s.VADstart, 20, lisDebugCh); if(firstDebug == 1) { ::MessageBoxA(NULL,

I am not sure if this is right place to ask for help on jspeex or can some one suggest. I have tried on jspeex sourceforge.net page. I am trying to use Pcm2SpeexAudioInputStream by creating it from tdl which received pcm_signed audio data as follows: AudioFormat format = new AudioFormat (AudioFormat.Encoding.PCM_SIGNED, 8000, 16, 1, 2, 8000, true); DataLine.Info targetInfo = new DataLine.Info

Hi all, First of all, thanks to Speex developper for the all the job. I am trying to implement my own Speex DirectShow fitlers for VoIP following the documentation and sample code's. I am facing audio glitches when encoding - decoding PCM data. The encoder and decoder procedures are copied below. What about the lookahead size ? how shouw we apply it in encoding stage ? thanks in advance

Hello, I'm having a problem using Speex - I'm getting white noise/static from the following process: (Load File) -> (Encode) -> (Decode) -> (Play) I know there's no issue with the loading or playing because (Load File)-> (Play) plays the file properly. I believe the problem lies with the speex_bits_write and speex_bits_read_from functions, because if I change my encode

hi, i'm using jspeex to encode signed little endian pcm 16bits data to speex, so i can then publish it to a flash client. On rtmp spec is said that is used network byte order, so in a speex frame consisted in more than 1 byte, i guess i have to send it in big endian order. By the source code of jspeex i see that when i pass pcm data as a parameter to SpeexEncoder.processData they use it as

Hey there =] I intend to develop a voip application using the Jspeex (Java implementation of Speex) library. At this point, the progrm can capture the voice from the microphone to an "ByteArrayOutputStream" object and play it back. I`m willing to add the encode\decode features of Jspeex but I can't make it to work, I`m probably missing somethnig. The code (Part of it) : -----

Hi Sanjiv, There does indeed appear to be some trouble with the Javasound SPI encoder (the SPI decoder works well). I am working on it, and hope to have a fix soon. In the mean time, you can use the encoder outside of JavaSound, that one works well (see the command line encoder as example code of how to implement it). sincerely Marc Gimpel Head of research Wimba <p>On

When encoding using the Opus multistream API, is it possible for opus_multistream_encode() to return 0? For example, what happens if multiple streams are being encoded, each with DTX enabled and all streams emit DTX packets at the same time? What about the case of a single stream with DTX enabled? Thanks Kevin O'Connor -------------- next part -------------- An HTML attachment was

Hi Gonzalo, Thanks for reporting this DTX issue. It should now be fixed in git. Can you confirm this it is indeed the case? Cheers, Jean-Marc On 19/05/14 10:20 AM, Gonzalo Mariano (gmariano) wrote: > Hello: > > > > We noticed that opus reconstructed noise is pulsing with a 400ms pattern > when dtx is enabled in silk mode. This is independent of the background > noise

The email might have been blocked due to the attachment size, here?s a link to it. https://www.dropbox.com/s/r9j7l3cbb02ob7i/opusDtx_noiseFix1_inOut.zip Pascal On 26/06/14 19:02, "Pascal Huart (phuart)" <phuart at cisco.com> wrote: >Hi Jean-Marc, >Thanks for the fix, the reconstructed signal is now much better. >Unfortunately there are still discontinuities which show

Tested DTX feature in Opus 1.0.2 and 1.1 for 20, 40, and 60ms frame size. They all generated only one encoded frame every 400ms during silence period as expected. However, for 10 ms frame size, the DTX period was 200ms instead. Has anyone else experienced the same? Thanks, Chris Wang -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm working on an opus codec for the asterisk voip app. Googling DTX found: https://bugs.chromium.org/p/webrtc/issues/detail?id=4559 To paraphrase, if there's silence long enough, opus enters Celt-only mode, which has no DTX. Is this still correct ? And is the suggested work-around effective? sean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I want use opus DTX to save bandwidth in my half duplex VOIP project, so i require help for below questions 1)what is opus DTX and documents to study DTX. 2)if opus DTX is enable do i need send comfort noise or completly drop silence detected frame. 3)How extactly OPUS DTX works with RTP protocol. 4) how to test with OPUS demo binary Thanks & Regards Vittalprasad B R 8722397247

Hi, I changed the VAD sensitivity by varying VAD_SNR_FACTOR_Q16 and now it shows the difference in silence frames. Thanks Suresh On 4 October 2016 at 17:57, Suresh Thiriveedi <sthiriveedi at gmail.com> wrote: > Hi, > > When we pass around 9K samples of only ambient noise (no voice), the > encoder which is enabled FEC+DTX is detecting only some 140 frames as > non-voice

Hi. I found that DTX is always unavailable when DISABLE_FLAOT_API is not defined in OPUS_AUTO mode. As you know analysis_info.valid is alway true except NaN case and is_silence is alway true except digital zero signal case. In general, following condition will be alway true except exceptional case. (analysis_info.valid || is_silence) But in a code, there is a NOT expression in front of above

Hi. I experienced same issue reported in https://gitlab.xiph.org/xiph/opus/issues/2026 The problem is that there are periodical(400ms) noise during DTX when the noise changed dramatically. I have found that the VAD value of silk does not affects to the DTX decision. If the background noise suddenly changes(by excessive echo cancellation, mute on/off operation, etc.) the silk generate a voice

On 2019-04-08 4:55, Gustaf Ullberg wrote: > Thank you Mark. > > I agree and have now updated the pull request with a new commit, > addressing your comments. > Please take a look. > > /Gustaf I think you will also need to check the mode of the previous frame (st->prev_mode) before using internal SILK encoder state. It could have been in SILK DTX some time ago, but then

On 2019-04-10 3:27, Gustaf Ullberg wrote: > Yes, good point. I added the checking of prev_mode for Silk DTX to avoid > using stale data from the Silk state. > The PR is updated, and I'm attaching an updated patch. > > /Gustaf Thanks. This looks good to me. - Mark