search for: l16

Hello, I have added a new -l16 "extreme" compression quality setting to the NHW Project. This new -l16 quality setting is surprisingly good actually, because I find it better on some images than -l15 quality setting whereas files size is smaller.I notably find this new -l16 quality setting better than x265 (HEVC). Mo...

I thought - what about the software - maybe it needs updated. After doing so I get a list: Audio codecs PCMU (8000 Hz) PCMA (8000 Hz) opus (48000 Hz) L16/16000 (16000 Hz) G.726-32 (8000 Hz) L16/8000 (8000 Hz) speex/16000 (16000 Hz) speex/8000 (8000 Hz) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200617/cb8d2959/attachment.html>

...L5 24 -0.090884932234919 5 L6 25 -2.09088493223492 6 L7 26 3.36473558331119 7 L9 27 -0.559327664097794 8 L10 2 0.181304809992027 9 L12 4 -3.05932766409779 10 L13 5 3.94067233590221 11 L16 7 1.68130480999203 12 L17 8 -1.31869519000797 13 L18 9 -2.05932766409779 14 L19 10 2.44067233590221 15 L21 13 -1.66682168482594 16 L22 14 0.0925458410842417 17 L23 15 -2.90745415891576 18 L28...

Hi, May be this helps: m1<- as.matrix(read.table(text=" y1 g24 y1 0 1 g24 1 0 ",sep="",header=TRUE)) m2<-as.matrix(read.table(text="y1 c1 c2 l17 ?y1 0 1 1 1 ?c1 1 0 1 1 ?c2 1 1 0 1 ?l17 1 1 1 0",sep="",header=TRUE)) m3<- as.matrix(read.table(text="y1 h4??? s2???? s30 ?y1 0 1 1 1 ?h4 1 0 1 1 ?s2 1 1 0 1 ?s30 1 1 1

...d the only difference are below, so looks like nothing extra gets compiled when the with-ssl is used... With --with-ssl: [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject --disable-speex-codec --disable-speex-aec --disable-bcg729 --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec --disable-g722-codec --disable-g7221-codec --disable-opencore-amr --disable-silk --disable-opus --disable-video --disable-v4l2 --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc --without-external-pa --witho...

...oking much better. Jerry On Wed, Jun 17, 2020 at 4:01 PM Jerry Geis <jerry.geis at gmail.com> wrote: > I thought - what about the software - maybe it needs updated. > After doing so I get a list: > > Audio codecs > PCMU (8000 Hz) > PCMA (8000 Hz) > opus (48000 Hz) > L16/16000 (16000 Hz) > G.726-32 (8000 Hz) > L16/8000 (8000 Hz) > speex/16000 (16000 Hz) > speex/8000 (8000 Hz) > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200617/041b0064/attachment.h...

I see this device : Axis C8033 Audio Bridge Quick Specs: Communications Protocol: SIP. Ethernet Ports: 1x 10/100. PoE: 802.3af/at Type 1 Class 2. Additional Interfaces: Audio: one-way/two-way, mono. Audio Codecs: G.711, G.726, WAV, MP3. Edge Storage: microSD, microSDHC, microSDXC. Operating Temperature: 4°F - 122°F. What is Codec WAV and MP3 to asterisk ??? Jerry -------------- next part

...below, so looks like > nothing extra gets compiled when the with-ssl is used... > With --with-ssl: > [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject > --disable-speex-codec --disable-speex-aec --disable-bcg729 > --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec > --disable-g722-codec --disable-g7221-codec --disable-opencore-amr > --disable-silk --disable-opus --disable-video --disable-v4l2 > --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv > --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc > --wi...

...IP4 217.10.79.55 t=0 0 m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:110 speex/8000 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=silenceSupp:off - - - - a=direction:active a=nortpproxy:yes 17 headers, 20 lines Using latest request as basis request Sending to 217.10.79.219 : 5060 (non-NAT) Found peer 'SipGate' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.10.79....

...the pertinent wine output streams. First run (no systematic crash): fixme:win:EnumDisplayDevicesW ((null),0,0x33e27c,0x00000000), stub! fixme:d3d:swapchain_init Add OpenGL context recreation support to context_validate_onscreen_formats fixme:d3d:debug_d3dformat Unrecognized 909200416 (as fourcc: L16) WINED3DFORMAT! fixme:d3d:getFormatDescEntry Can't find format unrecognized(909200416) in the format lookup table fixme:d3d:debug_d3dformat Unrecognized 909200449 (as fourcc: AL16) WINED3DFORMAT! fixme:d3d:getFormatDescEntry Can't find format unrecognized(909200449) in the format lookup tab...

...ed example works with bfd-ld and with gold, but not with lld: $ cat rodatareloc.s ... .align 16 leaq .JTab(%rip), %r10 jmp *(%r10, %rdx, 8) ... .section .rodata .JTab: .quad .L00, .L01, .L02, .L03, .L04, .L05, .L06, .L07 .quad .L08, .L09, .L10, .L11, .L12, .L13, .L14, .L15, .L16 $ gcc -o rodatareloc.s.o -c rodatareloc.s $ lld -o rodatareloc.so -shared rodatareloc.s.o ld: error: rodatareloc.s.o:(.rodata+0x0): can't create dynamic relocation R_X86_64_64 against local symbol in readonly segment defined in rodatareloc.s.o Changing the section from .rodata to .data fixe...

...Content-Type: application/sdp Content-Length: 595 v=0 o=MxSIP 0 1 IN IP4 192.168.2.18 s=SIP Call c=IN IP4 192.168.2.18 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=pt...

...: application/sdp Content-Length: 698 v=0 o=MxSIP 0 0 IN IP4 192.168.96.141 s=SIP Call c=IN IP4 192.168.96.141 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 4 4 98 97 115 96 9 108 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:108 G7221/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:101...

...ype: application/sdp Content-Length: 461 v=0 o=root 284 284 IN IP4 munged s=session c=IN IP4 munged t=0 0 m=audio 14466 RTP/AVP 0 8 4 3 2 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - This Retransmits several times and then the call is scheduled for destruction. The "CANCEL" sip messages seem to fail...

...nt-Length: 384 v=0 o=root 21604 21604 IN IP4 192.0.0.0 s=session c=IN IP4 192.0.0.0 t=0 0 m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:14 MPA/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 (no NAT) to 66.33.146.12:5060 -- Called 1800XXXXXXX@net2phone Sip read: SIP/2.0 407 Unauthorized Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2 From: "111111111111@net2phone.com" <sip:111111111111@net2p...

...nt-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71 s=SIP Call t=0 0 m=audio 10032 RTP/AVP 0 8 18 102 9 116 101 c=IN IP4 10.168.154.71 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:102 L16/16000 a=rtpmap:9 G722/8000 a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 10034 RTP/AVP 97 c=IN IP4 10.168.154.71 b=TIAS:2000000 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;ma...

...DP 2130706430 2001:123:ab:123::2 14385 typ host a=candidate:Hcbb5ed22 2 UDP 2130706430 fe80::21f:c6ff:fec4:926a 14385 typ host a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrat...

...e >> nothing extra gets compiled when the with-ssl is used... >> With --with-ssl: >> [pjproject] Configuring with *--enable-ssl *--prefix=/opt/pjproject >> --disable-speex-codec --disable-speex-aec --disable-bcg729 >> --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec >> --disable-g722-codec --disable-g7221-codec --disable-opencore-amr >> --disable-silk --disable-opus --disable-video --disable-v4l2 >> --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv >> --disable-ffmpeg --disable-openh264 --disable-ipp --disable-lib...

I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...101 0-15 Sent to remote server by * v=0 o=root 4205 4205 IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio 16798 RTP/AVP 4 3 0 8 2 5 10 7 18 110 97 101 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 192.246.69.223:5060 Received from remote server v=0 o=root 9755 9756 IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio 10066 RTP/AVP...