search for: opus_application_audio

I have an audio device whose 'driver' gives me Opus encoded frames using OPUS_APPLICATION_AUDIO and max bandwidth set to OPUS_BANDWIDTH_NARROWBAND. How does Opus encoder decide the center point of the 4K bandpass filter? Is it done frame by frame? -- Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20150224...

Can you explain what you mean by "center point"? Jean-Marc On 24/02/15 11:30 AM, Tony wrote: > I have an audio device whose 'driver' gives me Opus encoded frames > using OPUS_APPLICATION_AUDIO and max bandwidth set to > OPUS_BANDWIDTH_NARROWBAND. How does Opus encoder decide the center > point of the 4K bandpass filter? Is it done frame by frame? > > -- > Tony > > > _______________________________________________ > opus mailing list > opus at xiph.org...

I'm using opus to encode some music (classical lute, if it makes a difference). How do I use these encoder ctl's ? They seem to be doing the same thing. How are they different ? Are they different ? Should they both be set ? What happens if just one or the other is set ? Thanks for any help. sean -------------- next part -------------- An HTML attachment was scrubbed... URL:

...with occasional dips and peaks, I still do not get a reasonably good visual signal reconstruction. Also, I'm not quite sure if my opus implementation works as it should yet. I can run the encoder with OPUS_APPLICATION_RESTRICTED_LOWDELAY, but If I try to run it with OPUS_APPLICATION_VOIP or OPUS_APPLICATION_AUDIO the opus_encode function returns just 2 (size of encoded samples) and the opus decode just returns 0's. I would imagine to get the best audio reconstruction with OPUS_APPLICATION_AUDIO. What could be the reason for OPUS_APPLICATION_AUDIO not working? Best, Nick ______________________________...

...ly 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 bandwidth, two channels, OPUS_APPLICATION_AUDIO application. The lib version is 1.1.3. The initialization procedure is as follows: _encoder = opus_encoder_create(48000, 2, OPUS_APPLICATION_AUDIO, &error); error = opus_encoder_ctl(_encoder, OPUS_SET_BITRATE(64000)); _decoder = opus_decoder...

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 this 00 00 00 12 31 69 52 10 08 doubts are 1)i know 12(hexa) is size of encoded frame and 08 is TOC byte, here i have doubt i am using stereo input and 20ms frame then TOC should be 0C, and 2)also about the 4 by...

Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came up in encoding. Chandrakala, are these the encoding parameters that you believe should trigger the error? - Logan Hi, > > We are looking at the VAD im...

...sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl options: OPUS_SET_VBR=1, OPUS_SET_VBR_CONSTRAINT=unconstrained, OPUS_SET_COMPLEXITY=10, OPUS_APPLICATION_AUDIO, frame=60ms I compress 2 separate audio streams which only differ in the sample rate. One is 8Khz and the other is 16Khz. Both contain identical 4Khz bandwidth audio (the 16Khz file is a properly up-sampled copy of the 8Khz file). When compressing these, the 16Khz input generates about dou...

Hi all, I noticed some odd behavior with the Opus encoder, and I was wondering if this is a known issue. First, note that this problem occurs when the Opus encoder is created with OPUS_APPLICATION_VOIP, there is no issue if the encoder is created with OPUS_APPLICATION_AUDIO. If compiled for a 32 bit architecture (i386), the encoder experiences significant slowdowns when regular audio is followed by digital silence. The size of the encoded frames is also far too big: 128 byes for 20ms of pure silence. The problem does not occur if the library was compiled for a 64 bit...

...an issue with smoothing factor in VAD implementation Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came up in encoding. Chandrakala, are these the encoding parameters that you believe should trigger the error? - Logan <blockquote> Hi, We...

...tailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err); opus_encoder_ctl(enc, OPUS_SET_BITRATE(fs * 2)); opus_encoder_ctl(enc, OPUS_SET_BANDWIDTH(OPUS_AUTO)); opus_encoder_ctl(enc, OPUS_SET_VBR(1)); opus_encoder_ctl(enc, OPUS_SET_VBR_CONSTRAINT(0)); opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(0)); opus_encoder_ctl(enc, OPUS_SET_INBA...

Hi Drew, I think the libopusenc patch is better, but there's still a few issues left: 1) The static MAX_PACKET_BUFFER_SIZE value is still problematic because if you link libopusenc with a new version of libopus that supports higher order projection or just more projection channels for order 3, then you will overflow the buffer. I think what you'd want is a _ope_opus_header_get_size() call

...>> 3); waveform.nAvgBytesPerSec = waveform.nSamplesPerSec * waveform.nBlockAlign; #define BUFSIZE 512 ... case MM_WIM_DATA: { WAVEHDR *Hdr = (WAVEHDR *) msg.lParam; int opusRes; m_opusEncoder = opus_encoder_create(48000, 2, OPUS_APPLICATION_AUDIO, &opusRes); m_opusDecoder = opus_decoder_create(48000, 2, &opusRes); unsigned char data[BUFSIZE]; opus_int32 dataLength = opus_encode(m_opusEncoder, (short*)Hdr->lpData, 80, data, BUFSIZE); dataLength = opus_decode...

...d behavior with the Opus encoder, and I was wondering > > if this is a known issue. > > > > > > > > First, note that this problem occurs when the Opus encoder is created > > with OPUS_APPLICATION_VOIP, there is no issue if the encoder is created > > with OPUS_APPLICATION_AUDIO. > > > > > > > > If compiled for a 32 bit architecture (i386), the encoder experiences > > significant slowdowns when regular audio is followed by digital silence. > > The size of the encoded frames is also far too big: 128 byes for 20ms of > > pure sile...

...family != 255) return OPE_UNIMPLEMENTED; else if (streams <= 0 || streams>255 || coupled_streams<0 || coupled_streams >= 128 || streams+coupled_streams > 255) return OPE_BAD_ARG; - st=opus_multistream_encoder_create(48000, enc->channels, streams, coupled_streams, mapping, OPUS_APPLICATION_AUDIO, &ret); + st=_oge_create(48000, enc->channels, streams, coupled_streams, mapping, OPUS_APPLICATION_AUDIO, &ret); This code is allowing family 253 for a deferred init, but does not create a projection encoder in that case, so it looks like it will fail when writing the id header since...

...or fun, I tried to reproduce such an overflow. I turned on > all debug macros, assertions, and checked arithmetic and then > encoded 2 hours of mixed speech/audio with these parameters: > > Sample rate = 48000 > Channels = 1 > Application = OPUS_APPLICATION_AUDIO > Bitrate = 24 KB/s > Force Mode = MODE_SILK_ONLY > Signal Type = OPUS_SIGNAL_AUTO > Complexity = 10 > Frame size = 480 samples (10ms) > > No errors came up in encoding. Chandrakala, are these the > encoding parameters...

...VAD > implementation > > Just for fun, I tried to reproduce such an overflow. I turned on all debug > macros, assertions, and checked arithmetic and then encoded 2 hours of > mixed speech/audio with these parameters: > > Sample rate = 48000 > Channels = 1 > Application = OPUS_APPLICATION_AUDIO > Bitrate = 24 KB/s > Force Mode = MODE_SILK_ONLY > Signal Type = OPUS_SIGNAL_AUTO > Complexity = 10 > Frame size = 480 samples (10ms) > > No errors came up in encoding. Chandrakala, are these the encoding > parameters that you believe should trigger the error? > > - L...

Hi Opus list, Please forgive me if this has been asked before. I find that Opus encoder created in mode OPUS_APPLICATION_AUDIO (as opposed to _VOIP) is using a lot of bits to encode silent periods of speech. This is relevant to a voip application for which good quality music is desirable, and in which I add a minimal comfort noise (order of few bits loud, e.g. MLS signal of amplitude 1 or 2) to outgoing frames (towards op...

...; >> Just for fun, I tried to reproduce such an overflow. I turned on all >> debug macros, assertions, and checked arithmetic and then encoded 2 hours >> of mixed speech/audio with these parameters: >> >> Sample rate = 48000 >> Channels = 1 >> Application = OPUS_APPLICATION_AUDIO >> Bitrate = 24 KB/s >> Force Mode = MODE_SILK_ONLY >> Signal Type = OPUS_SIGNAL_AUTO >> Complexity = 10 >> Frame size = 480 samples (10ms) >> >> No errors came up in encoding. Chandrakala, are these the encoding >> parameters that you believe should...

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