similar to: Opus Raw Pakcets

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 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

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 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

Dear authors of Opus, currently, the manpages that come with opus-tools are written in the traditional man(7) markup language. I am proposing to rewrite them into the semantic markup of the mdoc(7) language. I am willing to do the work. Both the man(7) and mdoc(7) languages have been around for decades, and are supported by the prevalent formatters: groff(1) on most Linuxes and mandoc(1) on the

I configure LLVM build with this command cd llvm-8.0.0.src & md buildsys-x64-MT & cd buildsys-x64-MT & cmake -G "Ninja" -DLLVM_TARGETS_TO_BUILD=X86 -DCMAKE_BUILD_TYPE=Release -DLLVM_USE_CRT_RELEASE=MT -DLLVM_ENABLE_RTTI=1 -DLLVM_ENABLE_TERMINFO=OFF -DCMAKE_INSTALL_PREFIX=../x64-MT .. The number of build objects grew with every major release to 1660 for LLVM 7 and 1761 for

opus_decode() produces 16-bit signed linear PCM, and opus_decode_float() produces 32-bit floating point PCM that is useful when you want a higher bit depth. If you need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed integers for 8-bit linear PCM. (It uses signed for

Hi everyone, I'm trying to use Opus in an Embedded Project which I'm developing in a STM32F429IIT6 microcontroller. Currently, I'm playing wav files in it. I ported all the files from OpusLib 1.1.4 needed to compile the project, but when I try to create and OpusDecode state, it returns the error -7 (OPUS_ALLOC_FAIL). I'm compiling it with the USE_ALLOCA flag. I also tried to

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

Hi Please fix it so that opus-tools from git can be cross-compiled with mingw. Look here ---> http://lists.xiph.org/pipermail/opus/2012-October/001800.html

Hello, I've been using Opus on an STM32 M4 platform for speech coding in mono mode. I thought I'd try stereo for grins to see if I can handle the CPU load, and I'm getting a return code of -1 from opus_decode_float (using CBR and 40ms frames). I decided to try the opusenc and opusdec tools to just see how the command line apps would behave. I am getting decoded audio, but I am

On Sun, Jun 7, 2020 at 5:53 PM <luke-tierney at uiowa.edu> wrote: > > On Sun, 7 Jun 2020, peter dalgaard wrote: > > > So this wasn't tested for a month? > > > > Anyways, Free() is just free() with a check that we're not freeing a null pointer, followed by setting the pointer to NULL. At that point of tcltk.c, we have > > > > for (objc = i = 0;

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Hi I'm having a problem compiling opus-tools-0.1.7. Version opus-tools-0.1.6 seems to compile OK. I've tried with opus-1.0.3 and opus-1.1-beta. The errors are like this:- "undefined reference to `sqrtf'" etc. This OS is Peppermint Three, similar to Ubuntu 12.04. It uses:- gcc --version gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3 Google says it's maybe something to do

Hi, I had found a Opus TODO list in https://wiki.xiph.org/index.php?title=OpusTodo&mobileaction=toggle_view_desktop . The item "Opus-tools" interest me. And here is my question about "Port opusdec to libopusfile/libopusurl": 1. Does this function have achieved? On one hand, I think not literally. On the other hand, after briefly reviewing opusdec.c of opus-tools, I found

I''m trying to setup a Merb cluster + attachment_fu to handle uploads for a client''s Rails app. I''m using Flash to upload the files and I realized that Merb was giving me an error for each upload. It turns out that Flash doesn''t seem to send an Accept header and Merb''s responder crashes because it''s trying to split the missing Accept header.

Hi all, i am trying convert pcm (16bit pcm) stereo file to mono pcm file using opus_encode and opus_decode, i am able do this but i have doubt about TOC byte after opus encode. below is how encoder and decoder structures are used to do encode and decode file opus_encoder_create(8000, 2, OPUS_APPLICATION_AUDIO, &err); opus_decoder_create(8000, 1, &err); after opus encode bits looks like

Dear Opus developers. This is the first time I write here, so hello to everybody! Sorry to disturb you but I would like to ask you something I could not answer by googling and by looking at this mailing list archive. I have just started investigating this new and promising codec for real-time audio transport over the internet for industrial applications. I was previously experimenting with RTP

Hello Mark, The resulting 8 bit file has a lot of squelching noise compared to the 16 bit output from OPUS decoder. During encode I am using popi16fmtBuffer[ui32Loop] = (opus_int16)pcRdBuf[ui32Loop]; And during decode since the data is in the lower 8 bit I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara

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