similar to: Delays of encoder / decoder

Quoting John Ridges <jridges at masque.com>: > Speex tells me that the decoder is always 5 ms, but it says that the > encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is > there an extra frame of delay in the encoder that isn't otherwise > accounted for? Oh, delay = frame_size + lookahead If you have a frame size of 20 ms, then there's no choice but

Hi Folks, thanks for the great work, not sure if this is the right list for this type of quesiton. We are looking to use only Opus as "one codec for all", with VoIP-out obviously we want to tune it. I am planning to use fmtp in SDP to control server/client Opus settings. Something like : - *maxplaybackrate*: a hint about the maximum output sampling rate that the receiver is

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

I believe if you add the *decoder* lookahead time to the encoder lookahead time you are already referencing, you will get the numbers that you have calculated. --John On 4/11/2012 1:00 PM, speex-dev-request at xiph.org wrote: > Message: 1 > Date: 10 Apr 2012 22:07:20 +0200 > From: Thilo K?hler<koehlerthilo at gmx.de> > Subject: [Speex-dev] Speex Codec Delay Problem > To:

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

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

JM, I also need to know the precise delays from Speex but I used the SPEEX_GET_LOOKAHEAD control requests to determine them (plus the "speex_resampler_get_output_latency" function from the resampler). The returned values from the Speex lookahead request don't seem to match with the values you gave Alexander. Am I doing this wrong? Thanks, John Ridges speex-dev-request at

Hello R experts, I have two questions related to the nnls library (http://www.inside-r.org/packages/cran/nnls), and more broadly to linear regression with positive coefficients. Sample code is below the Qs. Q1: Regular regression (with lm) gives me the significance of each variable. How do I get variable significance with nnls? If there's no ready function, any easy way to manually derive

Speex tells me that the decoder is always 5 ms, but it says that the encoder is 5 ms for NB, 8.9375 ms for WB, and 10.90625 ms for UWB. Is there an extra frame of delay in the encoder that isn't otherwise accounted for? John Ridges Jean-Marc Valin wrote: > Quoting John Ridges <jridges at masque.com>: > >> I also need to know the precise delays from Speex but I used

Hello All! SPEEX introduces an additional delay to the audio data, I found out by reverse enginiering (it is NOT the lookahead time): narrow band : delay = 200 - framesize + lookahead = 200 - 160 + 40 = 80 samples wide band : delay = 400 - framesize + lookahead = 400 - 320 + 143 = 223 samples uwide band : delay = 800 - framesize + lookahead = 800 - 640 + 349 = 509 samples To get the

1. What to do with the last frame that is smaller then frame size? During encoding, the last frame is often smaller than the required frame size. In the sample code, proper number of zeros are padded at the end. So if I don't want those padded zero after decoding, I assume that it is up to me to keep track of the number of zeros. Is it right? 2. What does SPEEX_GET_LOOKAHEAD do? How to

Hello, Is FEC supposed to work in NARROWBAND mode ?(with maxaveragebitrate=12000; maxplaybackrate=8000 ) ?I am having some confusing results, it appears that FEC is enabled in the encoder, but the decoder cannot find any packet with FEC. I am also wondering if this piece of code is correct (webrtc): /* The following is to parse the LBRR flags. */? if (opus_packet_parse(payload,

On 2015-12-18 02:29, Simon Urbanek wrote: > Michael, > > I got access to PDP AIX so I can try to replicate your problem. Can you, please, share exactly your setup - AIX version and well as how exactly you installed the compilers (=where from)? I can then try to replicate it. AFAICS there is no official binary for gfortran nor gcc 4.7 so it must be some 3rd party - which could also be a

Jia Pu a ?crit : > 1. What to do with the last frame that is smaller then frame size? > During encoding, the last frame is often smaller than the required frame > size. In the sample code, proper number of zeros are padded at the end. > So if I don't want those padded zero after decoding, I assume that it is > up to me to keep track of the number of zeros. Is it right? Right.

Hello! I am encoding small snippets of audio (e.g. 100ms) that contain important audio from the first sample till the last one. (means they dont start/end silent). Doing so raised a couple of questions I couldnt solve by reading the docu/faq/internet search. 1. Does the "complexity" parameter influence only the speed of the encoder or also the speed of the decoder? (I need fast

> >> 2. What does SPEEX_GET_LOOKAHEAD do? How to use it? >> In speexenc.c, there is following code: >> >> ... >> speex_encoder_ctl(st, SPEEX_GET_LOOKAHEAD, &lookahead); >> ... >> nb_encoded = -lookahead; >> >> Can someone explain what this means? > > The lookahead is the number of samples you need to discard at the > start.

First patch fixes two typos in manual. Second patch adds few notes about latency, as promised. Hopefully I haven't added any typos and grammar errors :) On Wed, Jul 1, 2009 at 8:02 PM, Alexander Chemeris<Alexander.Chemeris at sipez.com> wrote: > Thank you for clarification. I attached a patch which (IMHO) > makes it more clear for everyone who just read Doxygen > docs.

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 the paper section describes how CELT and SILK had implemented a look/up head delay of (2.5ms), (5ms) respectively , could you explain me> 1) those values are fixed. 1) how you guarantee those values independent of the architecture's performance. Thanks so much On Tue, Apr 18, 2017 at 10:05 AM, Diego Alejandro Parra Guzman < daparrag at correo.udistrital.edu.co>

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