similar to: Control encoded packet size

On 01/21/2015 07:51 PM, Daniel K wrote: > Hi guys, > > I'm using Opus for speech in wide-band mode (sampling rate 16000) and > 20ms frames with signal type set to SIGNAL_VOICE. > > I have a few questions here: > > 1. > When I choose VBR mode, the codec seems to choose the bitrate on its own. > However, that seems to be an issue on mobile devices. In some cases,

Hey Kevin! Once you start dropping under (I think) 32kbps, the codec naturally starts losing stereo separation for the benefit of quality. If you really want perfect stereo at 16kbps CBR (this is horrifically low in my opinion) you might be better off splitting the audio and encoding each part at 8kbps CBR. Or, there may be a magic "enforce 100% stereo" one could pass? (doesn't look

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

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

Hello, I've been using Opus on an STM32 M4 platform for speech coding in mono mode. I thought I'd try stereo for grins to see if I can handle the CPU load, and I'm getting a return code of -1 from opus_decode_float (using CBR and 40ms frames). I decided to try the opusenc and opusdec tools to just see how the command line apps would behave. I am getting decoded audio, but I am

Hi, Opus team I'm Huang Hao and trouble you again. When I used the OpusRepacketizer as Jean-Marc mentioned, I got a result (maybe problem I'm not sure) that perplexes me, please give me some comments. The result be described as below: First(all init which are needed I have done), I used opus_encode() to get the encoded data successfully. The encoder control parameters are set: VBR=0,

#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 | Milestone: Component: Opus |

On 02/20/2018 06:48 PM, Rodriguez, Vince wrote: > 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. Not sure what you mean here. If your compiler supports C99 VLAs, then by all means just define VAR_ARRAYS and you're good to go. If not, the second best option is alloca(),

Hi Vince, Normally, Opus should work fine on the CC3220, though real-time encoding may be a bit tight depending on what configuration you choose. You'll probably want to configure with --enable-fixed-point and --disable-float-api (or define FIXED_POINT and DISABLE_FLOAT_API if you don't use autoconf). The encoder has a bunch of run-time settings that you may want to set, depending on

Hi Everbody, i am not so good with codecs and a lot of terms in RFC 6716 of OPUS are not so familiar for me. I work in VoIP domain (Internet telephony) and try to support OPUS codec throughout our network. Therefore, I am trying actully to calculate or estimate the biggest possible size of the RTP OPUS Packet in case of WB or FB. Unfortunately, The "Frame Length Coding" paragraph of

Hello, I am trying to implement a realtime encoding then streaming solution using the vorbis codec (would be wrapped in a "7F" type for minimal framing overhead) for sending audio over a low bit rate wireless link. since we need a constant bittrate (i.e. constant packet size) for this solution we want to run the alg in constant bitrate mode; for evaluation of the codec I followed the

#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 | Milestone: Component: Opus |

Hi all, I try to enable FEC in the encoder using the macro OPUS_SET_INBAND_FEC and I set the packet loss percentage to a constant value of 30%, using the macro OPUS_SET_PACKET_LOSS_PERC. Please find my encoder settings below: opus: encoder fmtp (maxplaybackrate=8000;maxaveragebitrate=24000;sprop-stereo=1;cbr=1;useinbandfec=1;usedtx=1) opus: encode bw=narrow bitrate=24000 fch=auto vbr=0 fec=1

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, FYI, I'm proposing the following update to the Opus spec to improve low-bitrate hybrid quality. If you'd like to comment, you can reply to this email: https://www.ietf.org/mail-archive/web/codec/current/msg03247.html Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [codec] Last proposal for the update draft Date: Fri, 17 Jun 2016 08:26:04 -0400 From: Jean-Marc Valin

Dear Christian van Bijleveld, You can use any of the below DSPs of Texas Instruments 1. TMS320C674x - This supports floating point implementation of opus 2. TMS320C66x - This supports both floating and fixed point implementations 3. TMS320C64x - This supports only fixed point implementation Regards, Mahantesh On Mon, May 13, 2013 at 10:12 PM, <opus-request at xiph.org> wrote: >

Hi Mark, Jean-Marc, Thanks for your comments. On Sun, Jun 12, 2016 at 6:34 AM Mark Harris <mark.hsj at gmail.com> wrote: > Hi Felicia, > > A few comments: > > > - /* CELT can only support up to 20 ms */ > > subframe_size = st->Fs/50; > > - nb_subframes = frame_size > st->Fs/25 ? 3 : 2; > > + nb_subframes =

Hi I used opus_demo to clarify how opus works, I try this command: opus_demo -e" audio 8000 1 8000 -cbr /somedirectory/pcm_mono_ochomil.raw /somedirectory/test.opus And I get the file test.opus, then comes the first problem, this file hasn't header so i can't reproduce it using an external tool, how can i add a header to this file Then when i trie to run opus_demo -d 8000 1

Just a follow up... I guess I was a bit confused about the VBR setting. I realise now that packets tend to stay very close to OPUS_SET_BITRATE so that solves my problem. --Albin On Sat, May 12, 2018 at 6:19 PM, Albin Stigö <albin.stigo at gmail.com> wrote: > Thanks for the input! > > --Albin > > On Sat, May 12, 2018 at 6:00 PM, Orestes Zoupanos > <oresteszoupanos at

Hi Albin! There may be some details at: https://tools.ietf.org/html/rfc7845#section-6 Otherwise, I hope someone else on this list might be able to give a better formula for estimating packet size. Regards, Orestes On 12/05/2018 16:31, Albin Stigö wrote: Hello, I'm working on an Opus encoder block for GNUradio (a signal processing toolkit). I was wondering if there's some