similar to: Can't play opus files after encoding

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

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

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-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but I hear

Hi, I am interested in this part of the answer: "Make sure both are also linked to the same libopus (opusenc could be using the system libopus 1.0.x while opus_demo is from git)" I dowloaded a tarball distrib of libopus (1.1-beta),build and installed it,and afterwards did the same with opus-tools (0.1.7),using the former installed libopus library (also using libogg 1.3.1). When I execute

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

I suspect the main difference is due to time alignment of the samples and the exact trimming at the beginning and at the end of the encoded files. I also note that there's a missing "--" in front of "bitrate" and only one "-" instead of two for cvbr in your opusenc command line. Make sure both are also linked to the same libopus (opusenc could be using the system

Thanks a lot for your tips. What I have to done to debug: got the raw wav-file, compressed/decompressed it on my PC and saved OPUS packets as an array. Moved this array to the embedded platform and made sure that the frames decoded from the OPUS packets are the same. However, the high noise level was the same, but it wasn`t caused by OPUS. The problem was with my sound bus between an embedded

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

Still can't reproduce. What OS and compiler version? Jean-Marc On 21/04/15 02:48 AM, Suresh Thiriveedi wrote: > Hi, > > There is no change in the compiler flags. I'm using as it is from the > original code. No change in the Makefile and I believe it is using the > floating point only by default. > > We are using 8k samples and mono so the commands is as follows.

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

Hi I am using opus_demo binary to test 48KHz Stereo file, i used below arguments to encode and decode. ./opus_demo -e audio 48000 2 48k_stereo.pcm stereo.opus ./opus_demo -d 48000 2 stereo.opus stereo.pcm After decoding i found that both left and right channels are identical, even input stereo file channels are differential in gain. 1. Is this the expected behavior of OPUS CODEC, if yes then

Hi Looks like 1.1 version is sensitive to the system architecture/compiler/kernel version. Below is my observation in various linux system I have. As you mentioned, you are not observed the crash in your system, can you please share your system details. And also please comment on the below table/observations. *Machine IP* *optimization flags* *RHEL version* *kernel version* *gcc version*

I just tried decoding with v1.1: ./opus_demo -d 48000 2 opus_encoded_crash.opus out.pcm and I see no issue (including with valgrind). Does the same command-line create problems for you? What compile flags did you use? fixed-point or float, any assembly, ...? Could be assembly here, or even a compiler bug wouldn't be unheard of. Cheers, Jean-Marc On 20/04/15 07:27 AM, Suresh Thiriveedi

Hello everyone, as I did not find a solution on my own, perhaps you can help me: For my app ( Qt 5.2.1 with C++, Windows 8, Mingw32) I need to continously convert raw pcm data to opus. I have been trying hard to link or compile opus, but yet did not succeed. From the docs I already know how to call the functions. But I can't figure out how to access them. Please, tell me what files I need to

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

On 2016-07-12 9:34 AM, Aleksandr Mikoff wrote: > If I encode each frame, decode it and send it back to the speakers I > hear a very noisy signal - it is so large, so it is almost impossible to > distinguish words in my speech... This effect can be caused by some data type issues: - If you are using opus_encode() and opus_decode(), make sure to use 16-bit signed integer samples in the

Hi, We are able to reproduce the issue with the 1.1 opus_demo (sample file). We captured the frames in our server just before the opus_decode and fed the file to opus_demo (1.1) and it is crashing. Same file is tested with 1.1.1 and it is fine. So this is in line with our server testing observation and I think here we can conclude that the 1.1 library is crashing while handling a specific mode

On 10/28/2013 01:37 PM, Antonio Juan wrote: > The problem is that for using this package I need to use all the libs > like ogg, even Speex?, You do not need Speex. You do need libogg because .opus files *are* Ogg files. > no, and also where i can find an easy example > like opus_demo.c to create the .opus file There's an easy example called opusenc in the opus-tools package :-)

Hi, There is no change in the compiler flags. I'm using as it is from the original code. No change in the Makefile and I believe it is using the floating point only by default. We are using 8k samples and mono so the commands is as follows. [root at MEDIA opus-1.1]# ./opus_demo -d 8000 1 opus_encoded_crash.opus opus_encoded_crash.pcm *And segmentation is as below..*. ............ Calling