similar to: OPUS Packet Size

Hi everyone, We finally made it! Opus is now standardized by the IETF as RFC 6716 (http://tools.ietf.org/html/rfc6716). See the announcements at: https://hacks.mozilla.org/2012/09/its-opus-it-rocks-and-now-its-an-audio-codec-standard/ http://www.xiph.org/press/2012/rfc-6716/ Feel free to spread those around :-) We're also releasing both 1.0.0 (same code as the RFC) and 1.0.1, which is a

Hi everyone, We finally made it! Opus is now standardized by the IETF as RFC 6716 (http://tools.ietf.org/html/rfc6716). See the announcements at: https://hacks.mozilla.org/2012/09/its-opus-it-rocks-and-now-its-an-audio-codec-standard/ http://www.xiph.org/press/2012/rfc-6716/ Feel free to spread those around :-) We're also releasing both 1.0.0 (same code as the RFC) and 1.0.1, which is a

Hello Tim and others, Thanks for your help explaining this process on IRC. I wrote out a first draft in the RFC xml format. I have attached the xml (labeled as xml.txt so it will appear inline) and the rendered txt files. Please let me know where I can make improvements. I will upload this draft to the IETF datatracker and send it out to codec@ after addressing your comments. -------------- next

On Fri, Dec 19, 2014 at 9:03 PM, Timothy B. Terriberry <tterribe at xiph.org> wrote: > Daniel K wrote: >> true, then why stop at 120ms and not support longer packets? > > https://tools.ietf.org/html/rfc6716#section-3.2.5 > > "...the audio duration contained within a packet MUST NOT exceed 120 ms..." If you're wondering why the limit exists, there are a

Hello All When sending the Opus Packet in RTP Payload, the compressed frame is the output of the encoder? Also the config value as given in the RFC6716, 16...19 | CELT-only | NB | 2.5, 5, 10, 20 ms 16 corresponds to 2.5 ms 17 corresponds to 5 ms 18 corresponds to 10 ms 19 corresponds to 20 ms Is this correct representation of the data? Also in the RFC3551 the payload

Tim, Would you mind giving me a more specific example of the sort of document that you think this should look like? I'd like to write up something that is somewhat final. On Mon, May 2, 2016 at 9:30 PM, Michael Graczyk <mgraczyk at google.com> wrote: > On Fri, Apr 29, 2016 at 4:32 PM, Timothy B. Terriberry > <tterribe at xiph.org> wrote: > > As a general point,

Hello Jean-Marc, Thanks for the quick reply and comments. On Thu, May 26, 2016 at 5:41 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Hi Michael, > > Here's some more minor comments below. As long as you address the two > comments from my previous email (254 -> 2 and the draft name), the draft > is good for submitting as initial version on the IETF website (even

Dear all, I have been working on samba 3.6 fileserver with LDAP backend. when i add user from ldap, user will be added to samba server but i could not enable users sambaAcctFlags from [NU ] to [U ], with "smbpasswd -en user" command. root at samba:~# smbpasswd -en aabajobir Enabled user aabajobir. root at samba:~# pdbedit -Lw |grep aabajobir aabajobir:10010340:NO

Hello sir, i have server with linux mandrek and i have clients with windows 98 . can you help me to do samba configuration . thank you for that . Best Regards, ENG.Hani M. AL-Zaid; computer engineer Computer and Electronics Research Institute (CERI); King Abdulaziz City for Science and Technology (KACST); P.O.Box 6086, Riyadh 11442; Phone: +966 - 1 - 481-3762; Fax: +966 - 1 - 481-3764

Hello, I understand the encoder provides an option for FEC to provide some protection against packet loss, but I don't understand the details of this arrangement. I'd appreciate answers to the following: * Adding FEC seems to change the encoded audio bit-stream itself, i.e., it doesn't just add additional protection bits, but also changes the encoded bits. This is easy to show by

Is there a reason why opus_repacketier_cat is limited to 120ms packets? It seems 120ms frames can only be combination frames, that is, only constructed by combining smaller frames of the same config. If this is true, then why stop at 120ms and not support longer packets? Regards, -- Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have application, where I am reciving the RTP packets, which has OPUS payload. >From the RTP packets I got following information: (12 byes Header) tells about the version, payload time, time stamp, srsc, etc. The rest of the packet is OPUS payload (raw format), The TOC byte from OPUS payload tells its 20ms frame, even the time stamp different of 960 means 20 msec frame. Questions: 1)

Hi all, I am a developer for embedded system and totally new to Opus and open source audio codec. I'm now using Microchip dsPIC33 to develop a decoder for Opus. I am now using sources of libopus. The first question is for calling opus_decode(), do I need to skip the header bytes of an opus file before I can input the file data to opus_decode()? Or will libopus handle this automatically?

Hi Jean-Marc, A simpler question. How does opus_encode() generate packets of 20ms (SILK-only or Hybrid)? Concatenating two 10ms frames or doing it straight with just one 20ms frame?

Hello everybody, I am having trouble with samba ldap authentication Here are some of the outputs... root at samba:~# smbclient -L //localhost -U justin Enter justin's password: session setup failed: NT_STATUS_LOGON_FAILURE Log out puts: root at samba:~# tail /var/log/samba/log.smbd [2015/11/14 09:02:16.560186, 0] passdb/lookup_sid.c:1684(get_primary_group_sid) [2015/11/14

Hi All, I have taken ITU standard 48KHz female voice file and Encoded using opus_demo by giving follwoing arguments. opus_demo -e audio 48000 1 32000 female1.pcm female1frame.opus And using repacketiser tried to pack 6frames/packet(6*20msec=120msec). While packing I observed below error for 3 times. Error: opus_repacketizer_cat() failed: corrupted stream when i inspect the encoded file, it

ly or Hybrid frames for 40 or 60ms packet, respectively. That is based on = concatenating 20ms frames, right? Is 60ms the largest packet opus_encode() can generate? In order to get pac= kets of up to 120 ms by combining multiple frames as described in RFC6716 c= lause 2.1.4 one would need to use the "repacketizer". That is if I want to= have a 120 ms packet, I would need to take

Hi, I'm working on porting libopus and I'd like to get information how to test Opus Encoder properly. There is information in RFC6716, that Opus Decoder's test files (http://opus-codec.org/testvectors/opus_testvectors.tar.gz) were created specifically to exercise all aspects of the decoder. Using these files, a custom implementation of libopus can be verified to be sure the Opus

Daniel K wrote: > true, then why stop at 120ms and not support longer packets? https://tools.ietf.org/html/rfc6716#section-3.2.5 "...the audio duration contained within a packet MUST NOT exceed 120 ms..."

Dear all, in chapter 2 of http://jmvalin.ca/papers/aes135_opus_celt.pdf it is mentioned, that encoder and decoder has an algorithmic delay of 6.5 ms (lookahead and resampling delay). In Table 54 of https://tools.ietf.org/html/rfc6716 different delays for the resampler of the SILK encoder are given. However, I measure the same round trip delay for opus encoder->decoder regardless which