search for: opus_decod

...ic locations can be sending to different channel on this device at the same time. I am using the OPUS codec in mono celt mode with a data rate of 32,000 bits per second. What is the best practice for dropping a frame? Will there be significant audio artefacts if a frame is just dropped, that is opus_decode (n-1) and then opus_decode (n+1)? What is the best practice for adding a frame to the jitter buffer? Will there be significant audio artefacts if a packet is added by calling opus_decode with a null input playload, that is opus_decode(n), opus_decode(null) and followed by opus_decode(n+1)? Than...

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? Actually I have browsed through the site, but there is nothing detail talking about the header bytes of opus file. When I compare two opus files by h...

...Suresh Thiriveedi <sthiriveedi at gmail.com > <mailto:sthiriveedi at gmail.com>> wrote: > > 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 0x00000000008009...

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 opus_decode_frame (st=0x712357d0, data=0x7eff...

...live call between webRTC browser client and > another legacy client. Our server is there in between and transcoding > from opus to another codec and this is observed while decoding the opus. > > Anyway, I'll try to capture/dump the packets in the server before > feeding to the opus_decode and share with you. But this will not have > the first 8 bytes (length+enc range) to directly feed to the sample > binary. Please let me know if this is fine. > > Thanks > Suresh > > On 16 April 2015 at 17:36, Jean-Marc Valin <jmvalin at jmvalin.ca > <mailto:jmvalin...

Dear Sir? I need opus_decoder open source(not include opus_encoder) to transplat into ARM platform. Thank you for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20131203/de84b10c/attachment-0001.htm

...ld perform similar to Speex if you'd have to have it resample to 44.1 khz. Cheers,Dragos From: Julien Chavanton <jchavanton at gmail.com> To: Jean-Marc Valin <jmvalin at jmvalin.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 a...

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 frame as explained in my earlier mail. He...

...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 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, opi1...

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 for...

...tes other than 8/12/16/24/48, then you'll have to do resampling. Have a look at the speexdsp resampler if you don't already have one. Cheers, 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...

This fixes a tiny copy/paste error in a comment, which confused me for a moment when first reading the code: https://github.com/xiph/opus/pull/60 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170824/efa597ed/attachment-0001.html>

This is observed on a live call between webRTC browser client and another legacy client. Our server is there in between and transcoding from opus to another codec and this is observed while decoding the opus. Anyway, I'll try to capture/dump the packets in the server before feeding to the opus_decode and share with you. But this will not have the first 8 bytes (length+enc range) to directly feed to the sample binary. Please let me know if this is fine. Thanks Suresh On 16 April 2015 at 17:36, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Please provide the input file that produc...

...ange 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 opus_decode123. len[toggle]:79, output_samples:96000 data[0] = 78 data[0] = 78 78 87 46 18 4f fe a6 be 7d 8 6 33 e2 79 ee e4 71 55 a7 3a 8 c9 48 d6 a7 20 3b 7 95 18 b8 4b 8f 24 fa a6 50 87 97 9c d7 13 d0 b2 c3 c4 6d 2f 8b 6c 13 6f bb 16 cc 20 85 4e c7 5d 2e 90 41 ae 47 8b 3e 36 eb c7 c8 28 94 3 c3 f9 52...

.....? Could be assembly here, or even a compiler bug wouldn't be unheard of. Cheers, Jean-Marc On 20/04/15 07:27 AM, Suresh Thiriveedi wrote: > 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 frame as explained in...

...on. For both 8 and 16 bit data I have used the following parameters ui32SamplingRate = 8000 ui32Channel = 1 FRAME_SIZE_IN_MS is 20 MAX_PACKET is 1500 ui8ScaleFactor = 1 for 8-bit and 2 for 16-bit data ui32BitsPerSample = 8 for 8-bit data and 16 for 16-bit data The code is as follows sOpusDec = opus_decoder_create(ui32SamplingRate, ui32Channel, &i32error); if (i32error != OPUS_OK) { return((int)i32error); } opus_decoder_ctl(sOpusDec, OPUS_SET_LSB_DEPTH(ui32BitsPerSample)); ui32SizeOfWrBuf = (ui32SamplingRate*ui32Channel*FRAME_SIZE_IN_MS*ui8ScaleFactor)/1000; opi16_out = (int16_t*)calloc...

...ue in some specific cases? Thanks Suresh On 13 April 2015 at 12:39, Suresh Thiriveedi <sthiriveedi at gmail.com> wrote: > 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 opus_decode_fr...

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...

Hi, I'm curious to know when would be the 1.1.1 stable version available. In 1.1, we are facing crash when opus library is trying to decode the CELT-only, full band and 20 ms. So we tried with 1.1.1 beta and it looks to be fine. Is there any open issue regarding this in 1.1 version? Thanks Suresh -------------- next part -------------- An HTML attachment was scrubbed... URL:

...gt; resampling. Have a look at the speexdsp resampler if you don't already > have one. > > Cheers, > > 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 >...