similar to: a problem with bandwidth configuration

Hi, I try to run the opus_demo with input sample rate 16Khz and bandwidth ? FB, I want the encoder to convert the rate, which I thought it would. "./opus_demo ?e audio 16000 1 50000 ?bandwidth FB?." But when I look at the frames created, the Toc ? config has values of 16Khz. For example "0xB8": "Config" = 0x17 = 23, which means CELT only WB 16Khz 20ms

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

--- 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 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. > >

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

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

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

>> >> 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

Hi, I'm looking a porting the Opus encoder (encoder only) to a Cypress PSoC 4 (a 32-bit ARM Cortex-M0 with very limited resources), and would like to know the feasibility of stripping down the encoder to fit. I've been playing with the encoder configured with "--enable-fixed-point --disable-intrinsics". The end result I'm trying to get to is encoding a single 16kHz 16-bit

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

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,

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, 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

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,

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

Red Hat Enterprise Linux Server release 6.4 (Santiago) gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) We see the issue in all our Intel based Linux servers. Thanks Suresh On 21 April 2015 at 12:41, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Still can't reproduce. What OS and compiler version? > > Jean-Marc > > On 21/04/15 02:48 AM, Suresh Thiriveedi

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, Apologies if this is a known issues, but I have found what I believe is a bug in the fixed point implementation of the Silk codec and could not find any mention on this in the archives. The bug can be easily reproduced with the fixed point demo program (./configure ?enable-fixed-point ?disable-float-api && make) using the following command: ./opus_demo voip 16000 1 23000

Hi, I'm looking a porting the Opus encoder (encoder only) to a Cypress PSoC 4 (a 32-bit ARM Cortex-M0 with very limited resources), and would like to know the feasibility of stripping down the encoder to fit. I've been playing with the encoder configured with "--enable-fixed-point --disable-intrinsics". The end result I'm trying to get to is encoding a single 16kHz 16-bit

To be decodable by opus_demo, you'll have to add the 8-byte "header". Just put in the length of the packet followed by "0" for the encoder range (0 means "not present"). That being said, from previous experience, the most likely cause of the crash is a bug in your software causing a corruption in Opus. So it's safe to assume that if you can't reproduce