similar to: RFC for Opus Packet in RTP Payload

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

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 Giovanni, I am running both Opus Encoder and Decoder on a TI Cortex M4 core without any issues. What is the specific issue that you are facing? Can you explain? Regards Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20160105/0bc0bc2c/attachment.htm

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

FYI, the Opus RTP payload format is now RFC7587: https://tools.ietf.org/html/rfc7587 Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [payload] RFC 7587 on RTP Payload Format for the Opus Speech and Audio Codec Date: Tue, 30 Jun 2015 16:33:17 -0700 (PDT) From: rfc-editor at rfc-editor.org To: ietf-announce at ietf.org, rfc-dist at rfc-editor.org CC: drafts-update-ref at

Hello Tim, Jean-Marc Thanks for the clarification. Let me study the sample OPUS file and see if my understanding is now clarified. Regards Amit On Mon, May 9, 2016 at 2:02 PM, Timothy B. Terriberry <tterribe at xiph.org> wrote: > Amit Ashara wrote: > >> I am referring to the following file >> >> >>

Hi, I am new to Opus, I want to decode a RTP packet which has opus payload. Following can describe my intention: I have RTP-packet----------> rtp_header+ opus_payload(20ms/960/48000/1channel) I have now opus_palyload after removing the rtp_header, what I want, a simple code using the opus_decoder which can give me the 20 msec opus paylaod consists of 960 samples. Can anybody put a simple

Hello Jean-Marc, As an example, I am using the output of opus encoder to store the file as the following format and read back the same during decode process, without having much overhead. (Thought it would be useful to put a picture rather than running text) [image: Inline image 2] Regards Amit On Thu, May 12, 2016 at 10:47 AM, Amit Ashara <ashara.amit at gmail.com> wrote: > Hello

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

On 05/12/2016 10:35 AM, Amit Ashara wrote: > For HMI panels, except for the capture pattern and a single page segment > entry, other fields are not important, and which results in almost 7% > overhead for a 20ms raw frame encoded with Opus. I'm not sure how you get a 7% overhead. In most uses I've seen, the overhead is more around 1%. > At the same time the > file

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)

The overhead of Ogg (in file size) is pretty small and it's efficient enough for most applications (and uses far less CPU than the codec anyway). If anything, you might want to look at optimizing the existing Ogg implementation (e.g. like Tremor did in the context of Vorbis). Of course, you're always free to design a new container, but I doubt it's worth it and it's a lot of work

Hi there, Im trying to understand the structure of an opus UDP payload. Recently i capture a test audio containing opus codec. I notice that some packets contains the same number of payload block.(e.g.58 bytes) When i look at the UDP data i see the first few four bytes were having these hex representation, 0x0d 0x80 0x5d 0xeb Can someone explain to me the meaning of the first four bytes in the

>>> Amit Ashara <ashara.amit at gmail.com> schrieb am 11.05.2016 um 19:32 in Nachricht <CAEyg9sjvTWMBMMCJ8HQcYmbv1BtNt54CgpqWaGNm02MWrKcxaQ at mail.gmail.com>: > Hello Jean-Marc, > > So for the moment we can assume that this method is also OK to use? > > On Embedded Systems, both SRAM and Flash can be a restricting factor > besides the compute time. To

>>> Amit Ashara <ashara.amit at gmail.com> schrieb am 12.05.2016 um 17:47 in Nachricht <CAEyg9sgjbsxQY-=VnhQrKiGeTcFSRr1wxOPUhNyCQF8Piuahow at mail.gmail.com>: > Hello Jean-Marc, > > Assuming that a 48KHz, 20ms 8-bit linear PCM data which is 960 bytes is > compressed to 64 bytes (for assumption). The with the Oggs header (4 byte) Actually what I don't

Amit Ashara wrote: > 1. Since the stream I am working with is a mono channel, what should be > the advised page_segments to use. I am using an embedded system so > keeping the flash and SRAM usage are vital for the development. The number of channels has no impact on this at all. > 2. In the OpusTag the is the libopus a mandatory field? Yes.

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 Tim I am referring to the following file https://en.wikipedia.org/wiki/File:Sample_of_%22Another_Day_in_Paradise%22.ogg I opened the file in a HEX editor. I do not see the string OpusHead in the packet. It starts with Oggs. Also checking the occurrence of Oggs, I see that the first packet has BOS, the next all (except last) have 00 (which is not defined in the RFC as continuation) and

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: