similar to: the length is error

On 10/26/2013 01:11 PM, Wang, Chris wrote: > 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? Just one 20 ms frame. It always returns a single frame except when it just can't (e.g. 60 ms CELT). > From your explanations below, opus_encode() will concatenate

Hi Everbody, i am not so good with codecs and a lot of terms in RFC 6716 of OPUS are not so familiar for me. I work in VoIP domain (Internet telephony) and try to support OPUS codec throughout our network. Therefore, I am trying actully to calculate or estimate the biggest possible size of the RTP OPUS Packet in case of WB or FB. Unfortunately, The "Frame Length Coding" paragraph of

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

Thanks Jean-Marc and Benjamin for the answers. One follow-up question. If I use a repacketizer as Jean-Marc suggested by combining two 60ms frames to form a 120ms frame, without extracting individual frames and using a new TOC, I would need to have a "de-packetizer" that does the exact opposite of repacketizer. De-packetizer would need to separate this 120ms frame into two 60ms frames

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

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

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

On 2016-06-17 10:50 AM, Rizwan Ishaq wrote: > 1) does 960 means number of samples in the (20msec frame)? (fs=48KHz) That's correct. > 2) The raw opus packet (i.e OPUS payload), how can I convert it to an > playable format? You can pass the payload section of the RTP packets directly to an opus decoder to convert to PCM float or integer samples. To save it in a file other audio

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

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 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

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

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

commit 8b38694a2dc8b18374310df50174f1e4376d6824 vhost/net: virtio 1.0 byte swap had this chunk: - heads[headcount - 1].len += datalen; + heads[headcount - 1].len = cpu_to_vhost32(vq, len - datalen); This adds datalen with the wrong sign, causing guest panics. Fixes: 8b38694a2dc8b18374310df50174f1e4376d6824 Reported-by: Alex Williamson <alex.williamson at redhat.com>

commit 8b38694a2dc8b18374310df50174f1e4376d6824 vhost/net: virtio 1.0 byte swap had this chunk: - heads[headcount - 1].len += datalen; + heads[headcount - 1].len = cpu_to_vhost32(vq, len - datalen); This adds datalen with the wrong sign, causing guest panics. Fixes: 8b38694a2dc8b18374310df50174f1e4376d6824 Reported-by: Alex Williamson <alex.williamson at redhat.com>

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..."

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?

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?

(Erm, sorry about that...) I was talking about errors of the type: Got 0 bytes. The problem was sporatic, and seemed to be load/speed based. That is, the faster the Linux box, the more errors we had. A look through clitar.c led me to a solution. Currently, clitar.c is written to pad a file with zeros if a file read error occurs. This is normal. All of our errors were of the 0 byte length