similar to: running at 44.1K but with standard frame sizes

Hi Jean-Marc, On Jun 15, 2013, at 2:23 AMEDT, Jean-Marc Valin wrote: >> I'm looking at how to run Opus at 44.1K. I have flexibility in the >> frame sizes of the unencoded audio, and packet sizes on the RF link. > > You should probably consider resampling. It's not that expensive and it > would make things easy. But otherwise, see below. Yes, considering your and

> I'm looking at how to run Opus at 44.1K. I have flexibility in the > frame sizes of the unencoded audio, and packet sizes on the RF link. You should probably consider resampling. It's not that expensive and it would make things easy. But otherwise, see below. On 06/14/2013 06:23 PM, Marc Lindahl wrote: > So, I was digging through the code, and I didn't see any attempt to

Thanks for the answers Benjamin? On Jun 14, 2013, at 8:05 PMEDT, Benjamin Schwartz wrote: > I have flexibility in the frame sizes of the unencoded audio, and packet sizes on the RF link. > > This implies that you don't have a very tight latency constraint, so you can afford to run a resampler. > I assume the resample costs CPU cycles? the RX is battery powered, I'd just as

Hi Jean-Marc, On Jun 15, 2013, at 12:20 PMEDT, Jean-Marc Valin wrote: > > >> So I still wonder, if you set up a custom mode, but then had all the >> settings the same as a normal mode, would the codec perform worse, or >> the same? > > You'll have to try normal vs custom modes and choose. The only thing I'm > telling you is don't run a 48 kHz

On 06/15/2013 12:00 PM, Marc Lindahl wrote: >> Do not do that, ever. Everything is calibrated for 48 kHz and you >> will likely cause audible noise. > > How would it cause audible noise, I don't understand that part? > After all the frequency calculations are off by 8%, that's not too > extreme... Well, the encoder is designed to follow the ear's response and

I saw the custom API, but nothing explicitly says "CELT-only" just "custom sample rate and frame size". I'll dig further now that you've pointed me in a direction. Thanks, -a On 7/6/15, 6:18 PM, Jean-Marc Valin wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > I believe what you want is called Opus custom (OPUS_CUSTOM in the > code).

It's my understanding that the CELT layer of Opus has a maximum input sample rate of 48k, and frequencies above 20k are effectively not encoded. I've been trying to get up to speed on the specification, and studying its operation, but as far as I can infer, there is a fixed set of 21 bands distributed logarithmically to encode DC to 20k. If I were inclined to encode at say, 96k, and pass

Hi Marcus, Thanks for the fixes. I did some more cmake build testing and encountered a few issues: The option -DFORTIFY_SOURCE=2 should be -D_FORTIFY_SOURCE=2, as the macro has a leading underscore. In the autotools build it defines this if it is not already defined (m4/ax_add_fortify_source.m4). When custom modes are not enabled, the cmake build is nevertheless installing the include file

I have a video conference like application that I've been working on for a while now, and a recent change is causing some odd problems, and I was wondering if anyone else had seen problems like this. The issue I'm seeing is that when using the sound card for capture, the audio will eventually get about 1-2 seconds out of synch (delayed), from the video. However, if I use USB devices

[ I'm not in the list because I didn't find a digested version; please move the lists to sourceforge.net, and we would have the digested version. I read the replies from the archive. ] Hello. Would it be possible to allow Vorbis use the same codebook for multiple files? I could keep a 650 MB codebook on CD-R and use that for all my audiofiles. If that is possible, how much the

Hey guys. I think Vorbis is pretty cool, but since the current OggEnc only offers 44.1kHz, it limits what I wanted to use it for. So I've been using Lame to get 16kHz mono Vorbis files. I'm curious about whether Lame does Vorbis encoding the "right" way for non-44.1k stuff, or whether it just encodes as it would for 44.1k & changes the sample rate on the output, but I'm

Is there a configuration or compile flag that lets me disable the SILK portion of the codec and use CELT only? I could have sworn that there is something, but I can't seem to find it in the mailing list archives. The application here is that I am attempting to update from the old CELT codec to OPUS. Unfortunately, the CELT codec was running *very* close to the CPU (MIPS32--80MHz) limit

Marc Lindahl wrote: > Of course, I'm setting up a bunch of tests to evaluate these, what I was asking was more along the lines of, > If you set up the same exact, including the sample rate, do you get the same results (e.g. same code path, calculations, etc.?) If you configure a custom mode with the standard parameters (48 kHz sampling rate and a frame size of 120, 240, 480, or 960),

I have a doubt reg. RESIDUAL_CODING_METHOD_PARTITIONED_RICE2 http://flac.sourceforge.net/format.html#partitioned_rice2 tells that 5 bits are used to specify the Rice parameter. And "Escape code, meaning the partition is in unencoded binary form using n bits per sample; n follows as a 5-bit number" Question: What is the use of the "escape code" if we have to specify the Rice

John, Thanks for the reply! You mentioned output sample rates should be 44100 or 48000, should I worry about input (Mic) Sample rates as well? (Currently I was requesting the sample rate on both ends to be 16000 samplesPerSecond, for ease of passing into the codec) Also, do you recommend any particular resampler that I should use, or are any of the ones out there probably okay, or should

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

Hi all, I'd like to modify rsync to add a flag to compare the MD5 signature of the unencoded audio data in the header of a FLAC file to determine whether or not to transfer a file. The reason being that I've got a large number of FLAC files, many of which are corrupted in the destination volume, but many of which are valid but the tags have been modified. The sizes of both the source and

Hello, You know, I always worry about the precision and the float constants... and there is a large lookup table in the floor1.c ... and I figure out a way to keep the code size and speed, but to improve the precision at this lookup table. (the difference is small, but audible) Here is the modifications in the floor1.c: tatic unsigned long FLOOR_fromdB_LOOKUP[256]={

Hi I have gone through the document available in the net regarding the Vorbis encoder /Decoder. Based on that i have prepared a understanding document on the encoder/decoder block. I would like to know whether my understanding of the coder is OK. If there are any other additional block /information pl. provide me with the same. Thanks and regards S.Padmashri <HR NOSHADE> <UL>

For the last couple months, Nuance has performed extensive testing on how the Opus codec performs in the speech recognition task. I'm hoping to publish a full report in the coming months, but until then all I have is a teaser. Opus performed within about 1% of the WER (Word Error Rate) of unencoded audio. This is compared to about 5% for Speex, which was the previous codec of choice. Well