similar to: High noise after encoding/decoding procedure

Thanks a lot for your tips. What I have to done to debug: got the raw wav-file, compressed/decompressed it on my PC and saved OPUS packets as an array. Moved this array to the embedded platform and made sure that the frames decoded from the OPUS packets are the same. However, the high noise level was the same, but it wasn`t caused by OPUS. The problem was with my sound bus between an embedded

On 2016-07-12 9:34 AM, Aleksandr Mikoff wrote: > If I encode each frame, decode it and send it back to the speakers I > hear a very noisy signal - it is so large, so it is almost impossible to > distinguish words in my speech... This effect can be caused by some data type issues: - If you are using opus_encode() and opus_decode(), make sure to use 16-bit signed integer samples in the

I need to extract SILK only FIXED POINT code. I have a couple of questions in this regard. 1. Is it enough to enable compile time flag (FIXED_POINT) in the config.h, include silk_fixed library and exclude silk_float in the opus_demo project. I am working in the MSVC framework. Anyone has tried this before? 2. It seems there is no compile time flag to enable SILK only code, the core

Hello everyone. I would like to use the opus codec in the following scenario: the PCM audio signal, coming from bluetooth module is captured by STM32F446 MCU and then decoded by opus codec. However, I don't know where to start to build the opuslib for ARM Cortex-M4. Could someone provide any hints or give any brief guide on this? WBR, Aleksandr -------------- next part -------------- An

Hello everyone. I would like to use the opus codec in the following scenario: the PCM audio signal, coming from bluetooth module is captured by STM32F446 MCU and then decoded by opus codec. However, I don't know where to start to build the opuslib for ARM Cortex-M4. Could someone provide any hints or give any brief guide on this? WBR, Aleksandr -------------- next part -------------- An

Hello Ralph, > Likewise opus_encode() takes 16 bit samples, so you need to extend each > sample from an 8 bit source before encoding. Two questions 1. In opusenc.c which API does the extending the 8-bit to 16-bit? 2. If that is the case then how will 24 bit PCM sample work? Regards Amit On Thu, Jan 7, 2016 at 12:21 PM, Ralph Giles <giles at thaumas.net> wrote: > On 07/01/16

Hello All I have successfully run the Opus Decoder for 16-bit WAV files. However when doing the same on 8-bit, the decoder produces noise, but on 16 bit data the output is working. Both the 8 and 16 bit files are from the same source and hence except for some loss of quality on 8 bit, they are identical in total play back duration. For both 8 and 16 bit data I have used the following parameters

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

Using File.size(myfile) I can get the size of the uploaded file in _bytes_. Does anybody know of a function to convert bytes into a more human readable format? If I had a wish list for a "file_size_in_words()" function, it would do this: 10752 bytes becomes "10.5 Kilobytes". 2213814 bytes becomes "2.1 Megabytes". 238 bytes becomes "Less Than 1 Kilobyte".

Hi All. I have an application where the sample rate of the sender and receiver can vary by a small margin and the latency needs to be maintained within bounds and can't drift significantly and the system has to be able to cope with clock mismatches up to 0.5%. For example, the sender may have a clock rate of 48.1kHz and the receiver may have a clock rate of 47.9kHz. Unfortunately the clock

Hello, I have an p2p voice chat application using WAVE PCM (winmm). Now i am trying to add opus encoding to it to send it over the TCP/IP and then decoding it back to play - but without success (without opus it works ok) Here is an example of my code. I get message from input device then encode with opus then decode it back to output wave header and play. Doing it i hear only noise in my

In R's maptools package, is there a built-in function to test if a given point is "inside" a given polygon on the map? The map was loaded from an ESRI Shapefile. The point's latitude and longitude are known. Thank you! Aleks -- ------------------------ Aleksandr Andreev Fulbright Fellow Graduate School of Management St Petersburg State University

Hi all, i am trying convert pcm (16bit pcm) stereo file to mono pcm file using opus_encode and opus_decode, i am able do this but i have doubt about TOC byte after opus encode. below is how encoder and decoder structures are used to do encode and decode file opus_encoder_create(8000, 2, OPUS_APPLICATION_AUDIO, &err); opus_decoder_create(8000, 1, &err); after opus encode bits looks like

A few years back, I wrote some code to plot maps with the maptools package. Now I am trying to reproduce my results, only to find out that maptools has been updated and my code no longer works. I've been able to fix the first part of it by forcing spb <- maptools:::read.shape("/home/sasha/Documents/maps/spb.shp") (as read.shape has been deprecated) but now when I do plot(spb,

opus_decode() produces 16-bit signed linear PCM, and opus_decode_float() produces 32-bit floating point PCM that is useful when you want a higher bit depth. If you need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed integers for 8-bit linear PCM. (It uses signed for

Please provide the input file that produces this with opus_demo. On 16/04/15 03:24 AM, Suresh Thiriveedi wrote: > Hi Jean-Marc, > > Could you please update if you got a chance to look into. As I > mentioned, I don't see the same issue in 1.1.1, but I don't see any > difference in 1.1.1 other than optimization based on the architecture. > This optimization could have

To be decodable by opus_demo, you'll have to add the 8-byte "header". Just put in the length of the packet followed by "0" for the encoder range (0 means "not present"). That being said, from previous experience, the most likely cause of the crash is a bug in your software causing a corruption in Opus. So it's safe to assume that if you can't reproduce

--- On Mon, 3/12/12, aleksandr shfets <a_shfets at mail.ru> wrote: > From: aleksandr shfets <a_shfets at mail.ru> > Subject: Fwd: Re[2]: [R] B-spline/smooth.basis derivative matrices > To: "Vassily Shvets" <shv736 at yahoo.com> > Received: Monday, March 12, 2012, 5:15 PM > > > > -------- ???????????? ????????? > -------- > ?? ????:

Hi. I made a webrtc relay with recording and dumped the SDP requests and RTP packets into files. Then I made a java decoder based on jitsi. Although the files contain all the needed info: encription keys, codec info, timestamps, etc., I could only decode one side in one of 2 conversations. For example, the RTP payload is decrypted successfully, but opus_packet_get_nb_samples() or opus_decode()

Hi all, I am a developer for embedded system and totally new to Opus and open source audio codec. I'm now using Microchip dsPIC33 to develop a decoder for Opus. I am now using sources of libopus. The first question is for calling opus_decode(), do I need to skip the header bytes of an opus file before I can input the file data to opus_decode()? Or will libopus handle this automatically?