similar to: OPUS Encoder Bitrate setting

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

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);

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

Hi Folks, thanks for the great work, not sure if this is the right list for this type of quesiton. We are looking to use only Opus as "one codec for all", with VoIP-out obviously we want to tune it. I am planning to use fmtp in SDP to control server/client Opus settings. Something like : - *maxplaybackrate*: a hint about the maximum output sampling rate that the receiver is

Hi all, Please guide me How to change frame_size of opus codec at run-time (20ms, 40ms, 60ms) I'm stucking in this case: 1. init codec width default config (frame_size =20ms, bandwidth=48KHz, bitrate = 48kbps...), then in runtime changing: - bitrate = 24, 16, 6kbps: sound is OK - frame_size = 40ms, 60ms: Not OK, sound is distort so bad 2. init codec with frame_size = 40ms , others is

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

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 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

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, 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

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

Hi, folks, I've been hunting down some strange bugs in audio I've been doing. While hunting my bugs down, I tripped across what appears to be an Opus bug, but it's not clear where it's coming from. Note that the optimization choices differ between the two in the config.log below. How can I force them to be the same? Presumably I need to force the android version toward the

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

Hi everyone, Some time ago, I sent a pull request <https://github.com/xiph/opus/pull/107> to the Opus github page. Jean-Marc asked me to post it to the mailing list so everyone can have a look at it. You can find the description and code changes below. Please let me know if you have any questions or concerns. Best regards Gustaf Ullberg In WebRTC, we would like to be able to

On 2019-04-01 3:37, Gustaf Ullberg wrote: > Hi everyone, > > Some time ago, I sent a pull request > <https://github.com/xiph/opus/pull/107> to the Opus github page. > Jean-Marc asked me to post it to the mailing list so everyone can have a > look at it. > > You can find the description and code changes below. Please let me know > if you have any questions or

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 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

Hi, When I use the encode tool with the -vbr switch OR the -cvbr switch I do not see any change in the file size. There is no difference in file size between the above 2 options when I specify a bitrate using the -bitrate option either. My question is when I enter a bitrate it does a good job of meeting my request, so what is the purpose of the -cvbr switch ? What does it do that

Thank you Mark. I agree and have now updated the pull request with a new commit, addressing your comments. Please take a look. /Gustaf On Fri, 5 Apr 2019 at 11:41, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-01 3:37, Gustaf Ullberg wrote: > > Hi everyone, > > > > Some time ago, I sent a pull request > > <https://github.com/xiph/opus/pull/107>

Hi, I want use opus DTX to save bandwidth in my half duplex VOIP project, so i require help for below questions 1)what is opus DTX and documents to study DTX. 2)if opus DTX is enable do i need send comfort noise or completly drop silence detected frame. 3)How extactly OPUS DTX works with RTP protocol. 4) how to test with OPUS demo binary Thanks & Regards Vittalprasad B R 8722397247