similar to: Questions regarding OPUS_APPLICATION_AUDIO vs OPUS_BANDWIDTH_NARROWBAND

Can you explain what you mean by "center point"? Jean-Marc On 24/02/15 11:30 AM, Tony wrote: > I have an audio device whose 'driver' gives me Opus encoded frames > using OPUS_APPLICATION_AUDIO and max bandwidth set to > OPUS_BANDWIDTH_NARROWBAND. How does Opus encoder decide the center > point of the 4K bandpass filter? Is it done frame by frame? > > --

Hi All. I am seeing the following unexpected behavior with OPUS_SET_MAX_BANDWIDTH. I expect that setting this to OPUS_BANDWIDTH_NARROWBAND would give similar results to passing an 8Khz sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

I'm using opus to encode some music (classical lute, if it makes a difference). How do I use these encoder ctl's ? They seem to be doing the same thing. How are they different ? Are they different ? Should they both be set ? What happens if just one or the other is set ? Thanks for any help. sean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, Is there any way to bandpass filter in R thanks nuncio -- Nuncio.M Research Scientist National Center for Antarctic and Ocean research Head land Sada Vasco da Gamma Goa-403804 [[alternative HTML version deleted]]

Hi, does anybody have any ideas regarding the easiest and most efficient way of implementing a bandpass filter in R ? any help would be truly appreciated. cheers, Pat Johnston -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or

I'm new using OPUS and I've implemented the OPUS lib under PIC32MZ, using the MIPS configuration. It compiles correctly and it seems that all the procedures invoked returns no error. However, when I excite the encoder with a pure 1 kHz tone, the encoding/decoding procedure returns al the samples to silence (the buffer is filled with 0x8001 or 0x7fff). The configuration is 48000 sps, 64kHz

Thanks for the reply. For my application I unfortunately need a better signal reconstruction. It's not necessarily a problem that the constant DC voltage is removed, but the audio signal will need to be more exact because it's also studied visually. To be more exact, I need to record and transmit audio data recorded from animals (mostly birds). Are there ways to achieve a more similar

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

Hello, I have a few questions about the block size in ogg vorbis 1. The allowed blocksizes are powers of two between 64 and 8192 Samples. As I understood , there are fixed sizes for long and short blocks . The encoder can pick any allowed value for long blocks and onother value, that must be smaller or equal to the first one, for short blocks. On which base does the encoder choose the size for

Which one of the various libraries on xiph.org allow me to create an Opus file? The 3 or so libraries on the Opus download page all seem to be for reading files, converting and/or encode/decode streams. I have device that outputs a stream of Opus 'frames' and I need to save them into an Opus file. -- Tony -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I noticed some odd behavior with the Opus encoder, and I was wondering if this is a known issue. First, note that this problem occurs when the Opus encoder is created with OPUS_APPLICATION_VOIP, there is no issue if the encoder is created with OPUS_APPLICATION_AUDIO. If compiled for a 32 bit architecture (i386), the encoder experiences significant slowdowns when regular audio is followed

Hello Tony, opusenc from opus-tools works for me.. Just tried it successfully on my x86_64 Ubuntu Trusty 14.04 box. I was just able to do $ sudo apt-get install opus-tools $ opusenc music_48kbps.wav music_48kbps.opus I remember also being able to compile opus-tools (git://git.xiph.org/opus-tools.git) some time ago. Regards, Vish On 23 February 2015 at 12:30, Tony <yellowjacketlite at

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

Hi. I'm wondering if anyone out there can give me a little help trying to figure something out: I presently am running CentOS4.1 with GNOME as my desktop (It runs quite well, I must say!). I just have a little curiosity: I have a few audio files (.wav, .mp3 and .ogg) and I like to "preview" them. When run my mouse over the .wav files (and mp3 files) the

Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came

Dr. Williams, I ran across your inquiry on one of the R-help mailing lists regarding digital filter design and implementation. I found no response to your email in the archives and was wondering if you were able to find anything. Thanks, Israel -- Israel Christie, Ph.D. Email: ichristie at gmail.com Phone: 865.766.0214 Mobile: 865.406.4615

Hi Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading