search for: opus_decoder

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

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?

...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=0x7effff9ab72d > "~?`\\?K\005??y?w+g~?S2\025?\036T?\002x??h!???\220\233\066s?\030#gb > \rn?rF\005Q?\213;?`\207$O?(m\222=9??/h??t??E?w?\237\"\206z\005 > \213?...

...d 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=0x7effff9ab72d "~?`\\?K\005??y?w+g~?S2\025?\036T?\002x??h!??? \220\233\066s?\030#gb\rn?rF\005Q?\213;?`\207$O?(m\222=9??/h??t??E?w? \237\"\206z\005\213?u at e", len=88, pcm=0x7effff9a6a80, frame_size=640, de...

...us_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=0x7effff9ab72d > > > "~?`\\?K\005??y?w+g~?S2\025?\036T?\002x??h!???\220\233\066s?\030#gb > > \rn?rF\005Q?\...

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 am...

...99d09c5, > > > > > > 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=0x7effff9ab72d > > > > > "~?`\\?K\005??y?w+g~?S2\025?\036T?\002x??h!???\220\233\066s?\030#gb...

...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, opi16_...

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

...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>

...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=0x7effff9ab72d > > "~?`\\?K\005??y?w+g~?S2\025?\036T?\002x??h!???\220\233\066s?\030#gb > > \rn?rF\005Q?\213;?`\207$O?(m\222=9??/h??t??E?w?\23...

...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=0x7effff9ab72d > > > > > > > > >...

...gt; > > 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=0x7effff9ab72d > > > > > > "~?`\\?K\005??y?w+g~?S2\025?\0...

...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((...

...ace 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=0x7effff9ab72d "~?`\\?K\005??y?w+g~?S2\025?\036T?\002x??h!??? > \220\233\066s?\030#gb\rn?rF\005Q?\213;?`\207$O?(m\222=9??/h??t??E?w? > \237\"\206z\005\213?u at e", len=88, pcm=0x...

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 l...

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 > &...