similar to: Encoding OPUS with difference bitrates

If you encode with int iByteSizeEncoded = opus_encode(m_enc, m_ShortsInput, (1920/sizeof(short)), m_EncodedBytes, (960*6)); ... the byte size is 120. I thought that an odd value like 359 has to be a fault on my side. I thought it would be a multiple of something. So 359 is really correct??? Thank you for clearing this up. Am 06.10.2013 16:50, schrieb Jean-Marc Valin: > Why

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

--- 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

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);

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

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

Change-Id: I16bd4cd990d8ad5888c9e30016218ac25242ecb5 --- src/opus_multistream_encoder.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/opus_multistream_encoder.c b/src/opus_multistream_encoder.c index 30cafe1..c07132f 100644 --- a/src/opus_multistream_encoder.c +++ b/src/opus_multistream_encoder.c @@ -747,7 +747,8 @@ static void ambisonics_rate_allocation( if

On Wed, Jun 4, 2014 at 4:31 PM, Timothy B. Terriberry <tterribe at xiph.org> wrote: > Alpha Thinktink wrote: >> max_data_bytes=-11 > > That value is passed in by you. I also don't think passing such a value > would have worked in earlier releases. It indicates the size of the > buffer you are passing to the encoder to receive the encoded output. Actually, I'm

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, I use SpeexEcho in my own softphone using MGCP protocol register to a PBX under WindowsXP in my laptop. I make a call: softphone ---> PBX ----> MediaGateWay ---PotsLine---> Traditional Phone. In the other side(TraditionalPhone), I did not here any echo voice, SpeexEcho works well. But, if I let laptop to make some sounds, such as play music, I heard all

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

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:

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

<style>table.customTableClassName {margin-bottom: 10px;border-collapse: collapse;display: table;}.customTableClassName td, .customTableClassName th {border: 1px solid #ddd;}</style><div id="write-custom-write" tabindex="0" style="font-size: 12px; font-family: 宋体; outline: medium none currentcolor;"><p style="margin:0px;">Dear

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

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 There, I have two OPUS handset clients say A & B A is 8 KHz, 12.2kbps cvbr supported OPUS client B is 8 KHz, 16kbps cvbr supported OPUS client When i try to encode a same voice frame(20ms sample frame) at different time intervals(not parallel encoding) for both A & B using same encoder handle by changing only bit rate. Issue here is, some noise is heard for B OPUS client, Ex:

Note also that the packet size you give the encoder also acts as an absolute max on the bitrate. For example, if you ask for 32 kb/s VBR but give a max packet size of 120 bytes, then you're absolutely certain the bitrate will never go over 48 kb/s. Jean-Marc On 05/12/2018 12:42 PM, Albin Stigö wrote: > Just a follow up... I guess I was a bit confused about the VBR > setting. I realise

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

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