similar to: opus demo program

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

Hi, I have downloaded the latest stable version *1.0.2 opus audio codec* from the following link: http://www.opus-codec.org/downloads/ *Details of OS & CPU:* OS : Microsoft Windows XP CPU : intel core 2 Duo cpu I am using Microsoft visual C++ 2010 Express to build the codec (opus.vcxproj available in the package downloaded). I am able to build it & could generate the application

I wonder if anybody try to compress a pure sine wave using OPUS codec. When I compressed the mono 1KHz, 16bits 48000 samples per sec. audio stream using the 'opus_demo' utility: opus_demo -e audio 48000 1 2min_1kHz_Sine_16bit_48kHz.wave 2min_1kHz_Sine_16bit_48kHz.opus_raw I had the output stream that is shown below. 00 00 00 01 00 00 00 00 78 00 00 00 01 00 00 00 00 78 00 00 00 01 00

Hi Jean The problem is that for using this package I need to use all the libs like ogg, even Speex?, no, and also where i can find an easy example like opus_demo.c to create the .opus file Greetings Toni 2013/10/28 Jean-Marc Valin <jmvalin at jmvalin.ca> > Hi Toni, > > The package you want is opus-tools. You can get it from the download > section. For file distribution,

Hello: We noticed that opus reconstructed noise is pulsing with a 400ms pattern when dtx is enabled in silk mode. This is independent of the background noise level and is found with speech + non-speech period test files as well as variable level noise-only test files. This issue can be reproduced with opus v1.1 using this command: ./opus_demo voip 16000 1 25000 -dtx input.bin

Hi all! Using the code in opus_demo.c I write a c program for encode and decode the audio from one microphone, I can encode and decode raw audio into .opus file, but really is not an opus file because i need to add the "OpusHead" and the "OpusTags". In order that any opus player can play that file. So I only need in my c program how to add this streams to my file an how to

Hi, I was wondering if someone could help me. I'm having a bit of a problem with opus_demo.c. I created a new library and refactored the code to only do a decode. I then encoded a wav file using the Opus tool OpusEncode.exe. When I try to decode the file encoded with OpusEncode.exe using the code refactored from opus_demo.c, the code appears to be reading from the wrong point in

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Hi I used opus_demo to clarify how opus works, I try this command: opus_demo -e" audio 8000 1 8000 -cbr /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus And I get the file test.opus, then comes the first problem, this file hasn't header so i can't reproduce it using an external tool, how can i add a header to this file Then when i trie to run opus_demo -d 8000 1

Hi Sakharam, I see 2 potential issues with what you are doing. 1. ./opus_demo -e voip 48000 2 16 music_orig.wav testcase30.opus in above command, "16" for bits/sec seems too low. I'm no audio expert, but just cant convince myself you can get any reasonable audio data with 16 bits/sec. FYI, I was able to encode and decode with 16 bits/sec, but when I played the decode file with

I didn't expect this, is it normal? When you use inbandfec, and have packet losses, you end up with more audio samples than you started with. With short frame (2.5), the FEC isn't supposed to do anything but it shouldn't do that should it? and 20mS frame, should do 'something' but? add audio? I'm using opus_demo for this. Here's the steps (I have my test audio in

Dear Opus community, I have a question about the SILK inband FEC. I am using the opus_demo (opus-1.2-alpha) for testing opus codec, and when I choose the "-inbandfec" option I get the exactly the same results when I don't add it to the command line. Are these results logical? Thank you in advance. Best regards, Ayoub. *--------------------BOUZIANE AyoubPhone : (+212) 633 092 157*

HI, I'm trying to simulate an audio conference where each leg can be with a different bit rate. This needs to encode the source PCM to to different bit rates back to back and store and send respective encoded frames/packet to the respective channel. For this I changed the opus_demo as below. But the output of the second encoded frames is completely garbled. Appreciate if anyone can suggest

On Thu, Feb 22, 2018 at 9:53 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 02/22/2018 09:34 PM, Stepan Salenikovich wrote: > > Its unexpected because the decoder continues to output all samples > > of -32768 even when the microphone input is silence or near silence, so > > I would expect the decoded values to be at or near 0. > > Oh, if the output is

On 21 November 2014 at 18:06, Timothy B. Terriberry <tterribe at xiph.org> wrote: > > Viswanath Puttagunta wrote: >> >> a. Simplest use case to validate this optimization for correctness. >> b. Simplest use case to validate this optimization for performance. >> >> Would prefer something like opusdec that can be executed on command >> line. > >

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

Hello! I encoded a voice file (48kHz) with opusbin\opusenc.exe with the standard settings and decoded it. The output was amazing. I could not hear any loss at all. Then i encoded the same file with opus_demo.exe and standard settings and then decoded it. The output had a sizzling noise, even when I used full bandwidth. I think I have played around with any of the settings in opus_demo.exe,

--- src/opus_demo.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/src/opus_demo.c b/src/opus_demo.c index 72506b5..5a75679 100644 --- a/src/opus_demo.c +++ b/src/opus_demo.c @@ -245,14 +245,14 @@ int main(int argc, char *argv[]) double bits=0.0, bits_max=0.0, bits_act=0.0, bits2=0.0, nrg; double tot_samples=0; opus_uint64 tot_in, tot_out; - int

On Thu, Feb 22, 2018 at 8:34 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Hi Stepan, > > I would need more information to be able to investigate further. It's > legal for the decoder to output -32768, so it would be good if you could > explain how this is unexpected. Its unexpected because the decoder continues to output all samples of -32768 even when the

>> >> a. Simplest use case to validate this optimization for correctness. >> >> b. Simplest use case to validate this optimization for performance. >> >> >> >> Would prefer something like opusdec that can be executed on command >> >> line. >> > >> > >> > The easiest thing to use is probably opus_demo (opusdec