similar to: Opus, a little confusing !

Hi everybody I wrote a couple of months ago about implementing the opus codec on a DK52 from Nordic (nrf52832 MCU, Cortex M4). In the meantime I got a bit further and would again like to ask some questions. My settings: I included all the opus files into my project (opus-1.2.1) Defined: FIXED POINT, VAR_ARRAYS, DISABLE FLOAT API, OPUS_BUILD, CELT_MODE_ONLY I am able to allocate enough memory

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 Benjamin, The original WAV file I have is linear 8-bit PCM. I want to ensure that original linear formats are kept as is. Later I will add support for ulaw. Regards Amit On Fri, Jan 8, 2016 at 5:34 PM, Benjamin Schwartz < benjamin.m.schwartz at gmail.com> wrote: > Do you really need linear 8-bit PCM or do you want ulaw? Linear 8-bit is > ... pretty rare. > > On Thu,

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

On 07/01/16 10:04 AM, Amit Ashara wrote: > opus_decoder_ctl(sOpusDec, OPUS_SET_LSB_DEPTH(ui32BitsPerSample)); OPUS_SET_LSB_DEPTH only affects the encoder. If you check the return value here you should get OPUS_UNIMPLEMENTED. > output_samples = opus_decode(sOpusDec, (const unsigned char > *)&pcRdBuf[0], len, opi16_out, (ui32SizeOfWrBuf/ui8ScaleFactor), 0); I suspect the issue is

Maybe i found the source of the problem. The basestation is running several threads. One per sending client and one per receiving client. In each thread there is either an opus_encode() or an opus_decode() but there is in total only one OpusEncoder and one OpusDecoder defined which where used in every thread. Now every thread has it own OpusEncoder or OpusDecoder. After 20minutes of audiostreaming

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

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'm new using OPUS and I've implemented the OPUS lib under PIC32MZ, using the MIPS configuration. It compiles correctly and it seems that all the procedures invoked returns no error. However, when I excite the encoder with a pure 1 kHz tone, the encoding/decoding procedure returns al the samples to silence (the buffer is filled with 0x8001 or 0x7fff). The configuration is 48000 sps, 64kHz

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

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

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

On 10/26/2013 01:11 PM, Wang, Chris wrote: > A simpler question. How does opus_encode() generate packets of 20ms > (SILK-only or Hybrid)? Concatenating two 10ms frames or doing it > straight with just one 20ms frame? Just one 20 ms frame. It always returns a single frame except when it just can't (e.g. 60 ms CELT). > From your explanations below, opus_encode() will concatenate

Thanks Jean-Marc and Benjamin for the answers. One follow-up question. If I use a repacketizer as Jean-Marc suggested by combining two 60ms frames to form a 120ms frame, without extracting individual frames and using a new TOC, I would need to have a "de-packetizer" that does the exact opposite of repacketizer. De-packetizer would need to separate this 120ms frame into two 60ms frames

ly or Hybrid frames for 40 or 60ms packet, respectively. That is based on = concatenating 20ms frames, right? Is 60ms the largest packet opus_encode() can generate? In order to get pac= kets of up to 120 ms by combining multiple frames as described in RFC6716 c= lause 2.1.4 one would need to use the "repacketizer". That is if I want to= have a 120 ms packet, I would need to take

Hi all, i try to set up something like a mixer and soundbridge. There are several sender clients which sends opus encoded audiostreams. One basestation is decoding the streams and mixing them all together. After that, the resulting stream is encoded with opus again and send to one or two receiving clients. Opus is set up to encode 10ms packets in stereo with 16bit per sample and 44kHz. So,

HI, I'm trying to simulate an audio conference where each leg can be with a different bit rate. This needs to encode the source PCM to to different bit rates back to back and store and send respective encoded frames/packet to the respective channel. For this I changed the opus_demo as below. But the output of the second encoded frames is completely garbled. Appreciate if anyone can suggest

having a hard time communicating on IRC, thank you gmaxwell, very informative. anyway, we were discussing the proper implementation of FEC on the decoder side. well, encoder side is just a boolean thing so that's alright. i gave an example where the receiver lost 5 rtp packets, 1 2 3 4 and 5 during which, we call opus_decode with a null pointer and fec=0 for every packet lost. now, when it

Hi All, We are using libopus 1.0.2 lib version currently and observing one crash issue in opus_encode() function some times. ------------------------------------------- (gdb) bt #0 0x00007fcdf7c90ea2 in opus_encode () from /usr/lib64/libopus.so.0 -------------------------------------------- Anybody faced this problem earlier and have any idea whether its solved in the latest version libopus