similar to: Opus and sender and receiver sample rate drift.

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?

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

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

Yes, I parse the RTP header, decrypt the payload and then feed the decrypted data to opus. Besides, opus_decoder_get_nb_samples() on 1st packet returns reasonable number 960, but then opus_decode() fails. On Fri, Nov 29, 2013 at 11:39 PM, Jean-Marc Valin <jmvalin at jmvalin.ca>wrote: > On 29/11/13 01:49 PM, Ilya Basin wrote: > > For example, the RTP payload is decrypted

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

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. 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, We are able to reproduce the issue with the 1.1 opus_demo (sample file). We captured the frames in our server just before the opus_decode and fed the file to opus_demo (1.1) and it is crashing. Same file is tested with 1.1.1 and it is fine. So this is in line with our server testing observation and I think here we can conclude that the 1.1 library is crashing while handling a specific mode

Hello Mark, The resulting 8 bit file has a lot of squelching noise compared to the 16 bit output from OPUS decoder. During encode I am using popi16fmtBuffer[ui32Loop] = (opus_int16)pcRdBuf[ui32Loop]; And during decode since the data is in the lower 8 bit I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara

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

On 29/11/13 01:49 PM, Ilya Basin wrote: > For example, the RTP payload is decrypted successfully, but > opus_packet_get_nb_samples() or opus_decode() return > OPUS_INVALID_PACKET. Are you sure you didn't forget to strip the RTP header before decoding it? Jean-Marc

Hi, There is no change in the compiler flags. I'm using as it is from the original code. No change in the Makefile and I believe it is using the floating point only by default. We are using 8k samples and mono so the commands is as follows. [root at MEDIA opus-1.1]# ./opus_demo -d 8000 1 opus_encoded_crash.opus opus_encoded_crash.pcm *And segmentation is as below..*. ............ Calling

Hi Jean-Marc, Thanks for your response. Please find the details as below. *Backtrace we got for this crash:* #0 0x0000000000800c54 in opus_decode_frame (st=0x38906b8f99d09c5, data=0xf0aa10b4ef1008ae <Address 0xf0aa10b4ef1008ae out of bounds>, len=-188613428, pcm=0x6e80016085efd57, frame_size=44037315, decode_fec=58716895) at src/opus_decoder.c:384 #1 0x00000000008009c0 in

Red Hat Enterprise Linux Server release 6.4 (Santiago) gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) We see the issue in all our Intel based Linux servers. Thanks Suresh On 21 April 2015 at 12:41, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Still can't reproduce. What OS and compiler version? > > Jean-Marc > > On 21/04/15 02:48 AM, Suresh Thiriveedi

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

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

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

When I run some large optimizations in R under Windows it puts a heavy load on the CPU, which makes other applications that run at the same time very slow. Is it possible to instruct R to be less aggressive in its use of computer resources, so I can use my computer for other purposes while running R (probably considerably slower) in the background? Han de Vries -------------------- This email

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

Beste tinc-developer, Ik moet van mijn baas een aantal VPN's aanleggen en ben nu thuis bezig om te testen. Ik heb 2 computers in een thuisnetwerkje (totaal 6 computers) geinstalleerd met SuSe 7.3, daar de klanten dit ook hebben. De computers zijn verbonden met een switch. Dit is de inhoud van mijn bestanden: Op computer 1 genaamd suse met ip 192.168.0.99 netmask 255.255.255.0