similar to: OPUS_STEREO

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

Hi All, I have taken ITU standard 48KHz female voice file and Encoded using opus_demo by giving follwoing arguments. opus_demo -e audio 48000 1 32000 female1.pcm female1frame.opus And using repacketiser tried to pack 6frames/packet(6*20msec=120msec). While packing I observed below error for 3 times. Error: opus_repacketizer_cat() failed: corrupted stream when i inspect the encoded file, it

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. > >

>> >> 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 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 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

On 24 November 2014 at 14:53, Viswanath Puttagunta <viswanath.puttagunta at linaro.org> wrote: > > 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

On 24 November 2014 at 17:48, Peter Robinson <pbrobinson at gmail.com> 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 >>> >>

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

--- 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

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, 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 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

Hello, I received feedback from engineers working on NE10 [1] that it would be better to use NE10 [1] for FFT optimizations for opus use cases. However, these FFT patches are currently in review and haven't been integrated into NE10 yet. While the FFT functions in NE10 are getting baked, I wanted to optimize the celt_pitch_xcorr (floating point only) and use it to introduce ARM NEON

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. The easiest thing to use is probably opus_demo (opusdec does a bunch of extra things, plus for interactive use we care about both the

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

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,

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

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

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