search for: 48khz

I have been challenged by someone that opusfile's 48kHz most of the time forced resampling but I didn't have good arguments to counter. I've checked the opus-codec.org pages and Monty's pages without finding good arguments. If you encode files to .opus with opusenc and decode with opusfile's api the sampling rate is constantly 48kHz ig...

Hi, I was just trying to use speex for sampling frequency >48KHz. In the original Speex-1.0.5 its restricted only upto 48KHz. I tired to modify it by changing the boundary conditions( the error conditions, i.e. if sampling freq >48KHz, it gives error) in /src/speexenc.c and then it atleast doesnt give the error, there is flow in decoding or encoding(i think)....

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with the only option to re-sample even if the internal sample rate was set to 8kHz. This may not seem like a b...

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > The encoder and decoder can handle,...

...with amplitude 0x6000 and frequency about 1150 Hz) E. encode the PCM data and decode it immediately, count the CPU usages. F. repeat until reach the duration time (1000ms or 10000ms) G. The summary of STM32F407 Test Result as below: Mode Sample Chan Freq. Duration Encode + Decode = Total FLOAT 48kHz 2 1150 1000ms 2735ms + 3367ms = 6102ms FIXED 48kHz 2 1150 1000ms 2112ms + 1543ms = 3698ms FIXED 48kHz 1 1150 1000ms 1312ms + 911ms = 2249ms FIXED 24kHz 1 1150 1000ms 1067ms + 783ms = 1872ms FIXED 16kHz 1 1150 1000ms 922ms + 711ms = 1651ms FIXED 12kHz 1...

...formance with 1.1 and the ARM optimizations. Please let me know, if you need more informations. opus-tools version 0.1.7 [root at studio-connect.de audio]# opusenc music_orig.wav music_orig.opus Encoding using libopus 1.0.3 (audio) ----------------------------------------------------- Input: 48kHz 2 channels Output: 2 channels (2 coupled) 20ms packets, 96kbit/sec VBR Preskip: 312 Encoding complete ----------------------------------------------------- Encoded: 1 minute and 30.82 seconds Runtime:...

...xed-point and disable float APIs).</p><p style="margin:0px;">it seems too slow for the VOIP application.</p><p style="margin:0px;"><br></p><p style="margin:0px;">Case 1:<br></p><p style="margin:0px;">48KHz Sampling rate, Stereo, VBR, frame size: 20ms, Bit-rates: 96kbps</p><p style="margin:0px;">Encode cost: 2.11x real time</p><p style="margin:0px;">Decode cost: 1.54x real time</p><p style="margin:0px;">Encode + Decode: 3.65x<br>&...

...up “down pitched” and heavily distorted. Due to the 32MHz base clock (+ prescalers) available it is not possible to get the desired 16kHz sample rate. I am therefore forced to work with 15625Hz. As I understand it the opus encoder does not mind a strange sample rate as it will re-sample it to 48kHz (Correct?). The decoder, however, will always output 48kHz, 24kHz, 16kHz etc (Correct?) which is incompatible with my 15625Hz rate. (However; If my assumptions are correct there could be a possibility to make my I2S device master and, thus, be able to pull the data at 16kHz) Consequently I have...

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful...

...(44.1KHz) if use speex and separate entropy coding. Please suggest me clearly as I have very small time left to wrap up my work to submit. thanking you, Regards, Devilal Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > I was just trying to use speex for sampling frequency >48KHz. In the > original Speex-1.0.5 its restricted only upto 48KHz. I tired to modify > it by changing the boundary conditions( the error conditions, i.e. if > sampling freq >48KHz, it gives error) in /src/speexenc.c and then it > atleast doesnt give the error, there is flow in decoding o...

> I have one doubt again, that is Vorbis use DCT/MDCT based algorithm > and also use psychoacoustic model so this is lossy codec. Speex is also a lossy codec. > And I dont think it ca regenerate a better matching waveform than > speex. At bit-rates above 32 kbps, Vorbis tends to produce better results than Speex, even for speech. The only advantages of Speex over Vorbis at these

Hello, I'm an audiphile. Will opus ever be developed to work with sampling rates higher than 48kHz? (I hope so)

...n.ca> Cc: opus at xiph.org Sent: Tuesday, March 15, 2016 1:18 PM Subject: Re: [opus] Question on opus_decoder output sampling rate Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: The encoder and decoder can handle, 8, 1...

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course the encoder sees just WAV (RAW) with possible MP3 decoding artefacts, but anyway: Is there a way to handle this? Maybe applying a low-pass filter before sending to...

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi Basil, I am going to make the app free to download and eventually release it as an open source project, yes. Thank you for the hint on the 48kHz. I'm planning on making an unofficial download link for the .APK on coolmic.net as soon as I get a couple of annoying bugs squashed, I'll keep the list updated. Cheers, Jordan On 01/17/2015 04:08 PM, Basil Mohamed Gohar wrote: > Jordan, Really great app idea. I am definitely intereste...

Hi: I guess this is mainly directed at Monty. I see from the CVS list that a lot of changes have been going in lately, especially in the area of 44.1 and 48khz. I know I was certainly under the impression that this area was more or less complete. My question is, apart from the experimental 48kbps stereo mode that went in today, are the changes for 44.1 and 48khz going to result in quality changes or is it just simpler code? Geoff. -- Geoff Shang &lt...

...rs, Jean-Marc On 02/04/15 10:42 AM, Julien Chavanton wrote: > Hi, is there any way to tell the decoder the output sampling Fz we want ? > > opus_decoder_create = Sampling rate of input signal (Hz) > > Considering this example (VoIP-out from WebRTC/RTP) > > MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with > internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with > 48kHz)] >> 48kHz(?) >> G.711(8kHz) > > This leaves us with the only option to re-sample even if the internal > sample rate was set to 8kHz. &g...

...wer frequencies had been denied): It should move linearly on foobar's logarithmic scale (as indicated in the file's comment). Actually I don't know the exact definition of logarithmic in Audacity, but it seems it met my needs (using linear sweep is somewhat ridiculous for the ears). On 48kHz on Opus: I'm unsure. I thought Opus can't handle 96kHz and it downsamples automatically. Why using 96kHz in the original: AFAIK Vorbis and Opus both use frequency components to encode the file. With higher sampling frequencies in the original, I was expecting to reduce the aliasing effect...

I'm rebroadcasting a realplayer stream and there are two problems. The icecast2 sounds tinny and its delayed something like 3secs. Any advice to perhaps use speex as it's talk radio? Can I cut the delay? Here is my ices2 config: http://static.natalian.org/2006-06-21/ices-alsa.xml Here is details of the feed, so you can take a listen: