similar to: OPUS_SET_MAX_BANDWIDTH does not have expected results

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? -- Tony -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

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? > > --

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

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Hi,? Anyone could provide more information on the level of complexity (OPUS_SET_COMPLEXITY(x) , x between 0 and 10 ?) which is recommended for generic Android devices ? How about the iPhone devices ??Is there a way to choose this complexity level according to the device model ? Any hints ? We've seen they set 5 in webrtc :http://code.google.com/p/webrtc/issues/detail?id=3093Freeswitch is

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension.

Hey, I am having trouble with the size of the encoded bytes by Opus. I am also having issue with the Bandwidth ctl. Here is the scenario. If I encode 16khz sampled audio: opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND)) opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&x)) = 1102 opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&x)) = 1103 average encoded size = 120 bytes if I

For some reason the attachment did not go through. Re-attaching. From: Mukund Raman Sent: Wednesday, April 01, 2015 6:12 PM To: 'opus at xiph.org' Subject: Opus multi-stream/surround: Audio corruption on decoded content Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I

Hi all ! If someone could give me a hint on how to proceed with the following i'd be very happy: I have a test setup on an nrf52832 (Cortex M4) in which I receive audio from a PDM microphone (64 sample frame) and pass it directly to an I2S device i.e. from ISR to ISR. With uncompressed audio this works just fine. Now I try to insert OPUS1.3 in the path but cannot make it work. The

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

Hello all, after FOMS I decided to take a look at the opus library and I found that I could improve a bit the build system and cleanup the code a little bit. Most of the changes to the code has been suggested by my two tools cowstats and missingstatic (part of the ruby-elf gem if you care). HTH, Diego

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

Hi, Can anyone let me know if this is a bug? Thank you, Chandrakala ----- Original Message ----- From: "Logan Stromberg" <loganstromberg at gmail.com> To: "Chandrakala Madhira" <chandrakala.madhira at soctronics.com> Cc: opus at xiph.org Sent: Wednesday, November 22, 2017 12:12:39 PM Subject: Re: [opus] Reg an issue with smoothing factor in VAD

Jean-Marc, Thanks for the response and the helpful info. I am trying to get the library to build without using the pseudostack define, and use either VAR_ARRAYS or ALLOC, but it seems the global stack is not defined. Where do can I define this in my example? VR -----Original Message----- From: Jean-Marc Valin [mailto:jmvalin at jmvalin.ca] Sent: Tuesday, February 20, 2018 5:40 PM To:

Hi All. Does Opus support mid-steam sample rate changes? For example if I have an audio stream which consists of 1 minute of 8Khz audio, then 10 seconds of 32Khz audio, then 2 minutes of 8Khz audio, can this be encoded in a single Opus stream which the decoder will understand? If not, are there any plans for implementing this? Cheers Philip

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

---------- Forwarded message --------- From: Peter Svensson <petersvenss85 at gmail.com> Date: tis 19 feb. 2019 kl 20:43 Subject: Re: [opus] Custom mode To: Emily Bowman Hi Emily ! Thank you for responding. I think my problem is not (yet) with OPUS itself. Encoding at complexity 0 takes 1.6ms ( 4.342ms at complexity 10 !) and decoding takes 1.9ms. 3.5ms, out of my 4.096ms budget, is

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