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The Opus resampler is actually a bugfixed version of the Speex one. Same interface/API, but with the bugs removed. It's why I recommended it :-) Otherwise I would have recommended something entirely different, like SoXR. On 07/05/13 22:12, Hermann Weber wrote: > Dear Nikos, > > thank you very much for your advice! > > I am not a little bit new to C++, and experiments

Hello! I would like to know if it is possible to request encoding to a higher samplerate than the original data. For example my uncompressed data was 22 kHz, and I would now like to decode the compressed data to 48 kHz (to feed it to a filter which expects a high "input" format). Is this possible with Speex? Thank you. Hermie

I'm not using Opus at all. I'm just including its resampler in my own sources. It's not even a DLL; it's directly compiled together with the rest of my code. You need these sources from the opus-tools package (http://www.opus-codec.org/downloads/): arch.h resample.c resample_sse.h speex_resampler.h stack_alloc.h In your project file, define these macros: #define

Dear Nikos, thank you very much for your advice! I am not a little bit new to C++, and experiments (like using another third party library) always turn out to be days of work for me. That is why I would first like to try Speex's new internal resampling function. I am currently still using the old 1.0.4 version. I have now downloaded the new 1.2rc1 version. Unfortunately it does not want

Hello Jean-Marc, I did some experiments with the fixed-point version of the resampler in SVN. Basically it works very well, great work! But I experienced a problem when upsampling audio data with slight clipping. This seems to cause an overflow somewhere, resulting in "cracks" in the output. I'm aware that the resampler hasn't been released yet, but I wanted to mention it. :-) Daniel

...n that directional transforms will bring marginal improvement, not significant in the case of my compression codec.This also confirms what I have read from the Rududu author who tested directional wavelets in his codec without luck. I have potentially found a very good candidate for the YUV420 2x2 upsampling.I haven't studied the impact of this processing on the codec, but I believe that better downsampling and upsampling algorithms for U and V components will bring better quality to the codec.Here is the upsampling I'm thinking of, the demo page is very impressive: https://kdepepo.wordpress...

...n with heavy clipping. So using 14 fractional bits internally instead of 15 and doing the saturation right before converting to the 16 bit output sample value could help. But certainly there are more sophisticated solutions to this problem. :-) During the tests I encountered another problem. While upsampling a continuous ?6 dB sine from 8000 Hz to 11025 Hz I heard that the resampler produces a click every 5 seconds. After changing the output sample rate to 11000 Hz the clicking disappeared. Daniel

Dear Nikos, thanks! But you use Opus only for resampling, not for entirely replacing Speex, don't you? Greetings! Hermie Am 07.05.2013 22:53, schrieb Nikos Chantziaras: > The Opus resampler is actually a bugfixed version of the Speex one. Same > interface/API, but with the bugs removed. It's why I recommended it :-) > Otherwise I would have recommended something entirely

...> > my understanding is that the speex modes are tuned for particular > sample rates, so using the 32kHz mode with 44.1kHz data will not yield > as good quality as using the 32kHz mode with 32kHz data. ok, it's strange, by nature, such compression algorithm are able to make down or upsampling very well... Do there is an impact on CPU load ? Vincent

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

...cond, is it acceptable that a codec is better than another in 60% of the cases but perform quite badly in 5% of the cases? I would be very interested in your opinion on this subject. Very quickly to finish, my codec really need improvements at some stages.It really needs to have better YUV420 down/upsampling algorithms (don't know about image Super-Resolution upsampling with machine learning or convolutional neural network, they're excellent but too slow for image compression I fear..., there are also other algorithms that are very good)... For the lower quality settings (-l1,-l2 mainly, don...

...t of the training set against the actual values using: pmat<- plot(y[1:400]) lines(c$fitted.values, col="blue", lwd=2) It seems as though neural net is not learning the negative values. I have tried running nnet several times, but each time I get the same problem. I have also tried upsampling, but no joy. I suspect that I am not using nnet correctly. Can anyone provide any hints/solutions? Any help appreciated. Kind Regards, Samuel Kemp.

...quency with the newly added time points being filled in with NA, 0, or perhaps interpolated. My data might be regularly or irregularly spaced. For example, I might have quarterly data that I would like to handle as a monthly time series with NAs filled in for the missing months. RSiteSearch("upsample") gave one link to a function in the "waveslim" package that I'm not familiar with. It seems to me that this would be a fairly common time series task and thus am hoping to find something in the more common time series packages/classes such as ts, zoo, tseries, etc... I will n...

...dering if there was any interest in extending the flac container format to support sample rates above the 655k current limit. Please note this is not for using ridiculously high sample rate sourced audio for some imaginary audible benefit. I've been involved in some experimentation with offline upsampling in software prior to delivery to an external DAC as a way to bypass the DAC's internal oversampling and found that wavpak is the only compressed container format that supports the 705/768 sample rates but it is very poorly supported and extremely slow during decompression, and fully uncompre...

...des are tuned for particular > >> sample rates, so using the 32kHz mode with 44.1kHz data will not yield > >> as good quality as using the 32kHz mode with 32kHz data. > > > > ok, it's strange, by nature, such compression algorithm are able to make > > down or upsampling very well... > > Do there is an impact on CPU load ? > > Nothing strange if you ever worked with speech compression before. yes, also you could add : if i had any clue in audio domain ! :-) > Speech compression algorithms always are tunned to particular freq, > else they woul...

...val16 *oldE, int nbEBands, /** Apply window and compute the MDCT for all sub-frames and all channels in a frame */ static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS_RESTRICT in, - celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample) + celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample, + int arch) { const int overlap = mode->overlap; int N; @@ -435,7 +436,9 @@ static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS fo...

...val16 *oldE, int nbEBands, /** Apply window and compute the MDCT for all sub-frames and all channels in a frame */ static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS_RESTRICT in, - celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample) + celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample, + int arch) { const int overlap = mode->overlap; int N; @@ -435,7 +436,9 @@ static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS fo...

...val16 *oldE, int nbEBands, /** Apply window and compute the MDCT for all sub-frames and all channels in a frame */ static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS_RESTRICT in, - celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample) + celt_sig * OPUS_RESTRICT out, int C, int CC, int LM, int upsample, + int arch) { const int overlap = mode->overlap; int N; @@ -435,7 +436,9 @@ static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS fo...

Changes from RFC PATCH v3 - Just rebased on tip - For all else, please see notes from RFC PATCH v3 at http://lists.xiph.org/pipermail/opus/2015-March/002902.html - latest wip opus tree/branch https://git.linaro.org/people/viswanath.puttagunta/Ne10.git branch: rfcv4_final_fft_ne10 Viswanath Puttagunta (1): armv7(float): Optimize encode usecase using NE10 library Makefile.am

Changes from RFC PATCH v1: - passing arch parameter explicitly - reduced stack usage by ~3.5K by using scaled NE10 fft version - moved all optimization array functions to arm_celt_map.c - Other cleanups pointed out by Timothy Phil, As you mentioned earlier, could you please address all compile and linker errors/warnings coming out of Ne10 library? You can find my working Ne10 repo at [1] You