search for: unvoiced

...sted in any opportunity to accelerate this. Thanks, Peter ________________________________ From: Jean-Marc Valin <jmvalin at mozilla.com> Sent: Wednesday, June 7, 2017 2:46:52 AM To: Freshman, Peter; opus at xiph.org Subject: [EXTERNAL] Re: [opus] Submitting a patch that exposes VAD voiced/unvoiced signal type Hi Peter, There's two main issues with a patch like the one you're proposing. First, the data is only valid when SILK is being used and is essentially undefined in CELT mode. The second issue is that by exposing internals, it makes it impossible to improve these algorithms sin...

...de: error = opus_encoder_get_internal_flags(vad->opus, &internalflags); if (error != OPUS_OK) { return OPUSVAD_OPUS_ERROR; } cur_signal_type = internalflags.prevSignalType; if ((vad->cur_state == STATE_NO_STATE) && (cur_signal_type == TYPE_UNVOICED || cur_signal_type == TYPE_VOICED) && (vad->prev_signal_type == TYPE_NO_VOICE_ACTIVITY)) { ... Our library uses this information to apply end pointing on voice-based audio. For example, we work with many customers implementing speech-enabled TV set-top-box solutions. We offer a sma...

...------------------------------------------------------------ > *From:* Jean-Marc Valin <jmvalin at mozilla.com> > *Sent:* Wednesday, June 7, 2017 2:46:52 AM > *To:* Freshman, Peter; opus at xiph.org > *Subject:* [EXTERNAL] Re: [opus] Submitting a patch that exposes VAD > voiced/unvoiced signal type > > Hi Peter, > > There's two main issues with a patch like the one you're proposing. > First, the data is only valid when SILK is being used and is essentially > undefined in CELT mode. The second issue is that by exposing internals, > it makes it imposs...

Hi, I'm reaching out because we'd like to contribute back to the project a patch that exposes the signal type of the audio packet when encoding the PCM audio to OPUS. We've found the Opus VAD algorithm to be exceptional in this regard and have written a library that leverages this information for audio end-pointing. Attached is the patch. Please let us know if you'd be willing to

Hi Peter, There's two main issues with a patch like the one you're proposing. First, the data is only valid when SILK is being used and is essentially undefined in CELT mode. The second issue is that by exposing internals, it makes it impossible to improve these algorithms since it would break API compatibility. I'm not fundamentally against trying to expose some information, but

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc

...he framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a variable frame size implementation based on some voiced/unvoiced part? I was also trying to add pitch prediction in WB/UWB part i.e. in high band coding, but found it really difficult to do it. Tell me if I can use the direct functions of pitch prediction in NB ? and HOW ? dev *************************************************************************** Devi L...

...I don't know of any voice specific coder that even attempts to capture energy above 10kHz. SBR just isn't relevent. Most wideband speech coding captures only 7kHz to 8kHz bandwidth. The key improvement that gives over the 3kHz to 4kHz most mainstream voice coders capture is to clean up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear almost the same at telephone bandwidth. At 7kHz bandwidth they have enough character to make them more distinguishable. The basic intelligibility improvement you get is usually small. However, the voice is rather more pleasant and less tiri...

...in time by just skipping packages, reset by: speex_decoder_ctl(state, SPEEX_RESET_STATE, NULL); and starting decoding, I see that the quality of decoding varies depending on the initial frame characteristics. It seems that the signal gets close to the original quite fast when the first frames are unvoiced. If I start decoding in the middle of a voiced chunk then decoding quality is comperatively much lower, more frames are needed until the signal gets closer to the original. This was the feedback part. So the obvious question is: do you know an easy way to get rid of this problem? Thanks. Regards,...

...zu4i3dmg5f91tv/20msAudioModeQuestion.zip?dl=0 <https://www.dropbox.com/s/bzu4i3dmg5f91tv/20msAudioModeQuestion.zip?dl=0> If there is one single thing to listen to, it would be ar3_20_audio.wav, loop the section "china hit" starting t=0.6s and listen for artifacts in the unvoiced speech. reference is ar3.wav. and by comparison ar2_10_audio.wav ( same segment, sounds more like the reference ar3.wav) Here is a cat of the README.txt. Thanks very much! 16bit, 16kHz input wav files (ar1, ar2, ar3), content from ~50Hz to near 8kHz. All .pcm files are 16kHz, 16...

...any voice specific coder that even attempts to capture > energy above 10kHz. SBR just isn't relevent. Most wideband speech coding > captures only 7kHz to 8kHz bandwidth. The key improvement that gives > over the 3kHz to 4kHz most mainstream voice coders capture is to clean > up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear > almost the same at telephone bandwidth. At 7kHz bandwidth they have > enough character to make them more distinguishable. The basic > intelligibility improvement you get is usually small. However, the voice > is rather more pl...

...ks, retrain the pitch codebooks, retrain the innovation codebooks, adjust just about every other buffer size and then figure out why Speex not crashes on you :-) > How to change the buffer size and how its affected. > can we have a variable frame size implementation based on some > voiced/unvoiced part? Sure, that'll be about 1000-2000 lines of code, plus the retraining of all codebooks. Any that would gain you what exactly? > I was also trying to add pitch prediction in WB/UWB part i.e. in high > band coding, but found it really difficult to do it. Tell me if I can > use the...

...ks, retrain the pitch codebooks, retrain the innovation codebooks, adjust just about every other buffer size and then figure out why Speex not crashes on you :-) > How to change the buffer size and how its affected. > can we have a variable frame size implementation based on some > voiced/unvoiced part? Sure, that'll be about 1000-2000 lines of code, plus the retraining of all codebooks. Any that would gain you what exactly? > I was also trying to add pitch prediction in WB/UWB part i.e. in high > band coding, but found it really difficult to do it. Tell me if I can > use the...

hi, thanks for answers to my previous qns. have some more of them. hope it's ok to ask questions on the basics of CELP here. let me know if it's not. 1) synthesized filter stability after interpolation of LSFs I read from some resources that if LSF representation of LPCs is used for interpolation, stability of the synthesis filter is guaranteed. from another source, i read that the

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

I have a CentOS 6 machine that was initially installed as CentOS 6.4 in May of 2013. It's /boot filesystem is 200M which, IIRC, was the default /boot size at the time. The most recent kernel update (2.6.32-573.18.1.el6) fails because of lack of space in /boot. The workaround is edit /etc/yum.conf, reduce installonly_limit from 5 to something lower (I used 3), remove the oldest kernel via

...pus_int offset_Q10; + opus_int32 RDmin_Q10, Gain_Q10; + VARDECL( opus_int32, x_sc_Q10 ); + VARDECL( opus_int32, delayedGain_Q10 ); + VARDECL( NSQ_del_decs_struct, psDelDec ); + int32x4_t t_s32x4; + SAVE_STACK; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + silk_assert( MAX_DEL_DEC_STATES == 4 ); + silk_assert( DECISION_DELAY == 40 ); /* Table next_smpl_buf_idx_table[] and mode_DECISION_DELAY_tabl...

NSQ_LPC_BUF_LENGTH is independent of DECISION_DELAY. --- silk/define.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/silk/define.h b/silk/define.h index 781cfdc..1286048 100644 --- a/silk/define.h +++ b/silk/define.h @@ -173,11 +173,7 @@ extern "C" #define MAX_MATRIX_SIZE MAX_LPC_ORDER /* Max of LPC Order and LTP order */ -#if( MAX_LPC_ORDER >