similar to: How to change codec frame_size at runtime

Hello Thomas and Amit, Thanks for your notice and the detailed 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);

Hello! You have errors there: silk_warped_autocorrelation_FLP MAX_SHAPE_LPC_ORDER was changed from 16 to 24, so array initialization is incorrect test_opus_api opus_encoder_ctl( enc, OPUS_SET_EXPERT_FRAME_DURATION ... ) returned status is incorrect. Best regards, Dmitriy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

"frame_size = PCM audio in signed 16-bit format (native endian). There must be exactly frame_size samples per channel." Sorry for such a simple request, but could you please clarify the definition of frame_size when using stereo audio channels? When you say "frame_size samples" does this mean that, if my frame_size is defined as 128, there are 64 left samples and 64 right

In stereo mode, you need to use the init_mc() call and consider the number of samples *per channel*. Also, more than 20 ms frames are a bad idea. Jean-Marc danflu at uninet.com.br a ?crit : > Hi, > > I'm using the echo cancellation api and I would like to > clarify the 'frame_size' parameter used in > speex_echo_state_init(frame_size,filter_length). > > >

Hi, I'm using the echo cancellation api and I would like to clarify the 'frame_size' parameter used in speex_echo_state_init(frame_size,filter_length). In the 'docs' it says: "...where frame_size is the amount of data (in samples)you want to process at once..." So... here are my questions: if I use stereo input/output do I have to put the samples doubled ? For

Hi Jean, Thank you for your reply. Ok... I'm gonna use 'samples per channel' everywhere I see 'samples'... but what about the 'speex_echo_playback' function ? it does the following loop: ... for (i=0;i<st->frame_size;i++) st->play_buf[st->play_buf_pos+i] = play[i]; ... So... if frame size is 'samples per channel' it will copy only half the

Jean-Marc, Yes, fixed-point is enabled (#define FIXED_POINT in config.h). I haven't tried anything lower than 4kbps; like I mentioned, 6kbps sounds fine so it's very strange. The only code I changed to get 4kbps was setting quality to 1 so that is the only difference in my test code between using 4kbps and 6kbps. It seems to be in the encoder. On the ARM side, I encoded a 8Khz 16-bit

Hey, I am having trouble with the size of the encoded bytes by Opus. I am also having issue with the Bandwidth ctl. Here is the scenario. If I encode 16khz sampled audio: opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND)) opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&x)) = 1102 opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&x)) = 1103 average encoded size = 120 bytes if I

Tried your fixed_generic.h change but that didn't help. Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Sent: Wednesday, December 12, 2007 4:13:35 PM Subject: Re: [Speex-dev] 4kbps sounds robotic on TMS320C64 Jean-Marc, Yes, fixed-point is enabled (#define FIXED_POINT in

The following code doesn''t work in Safari, though its working well on FF/IE. We just recently started migrating and we are find issues with Safari 3.X var SipIxView = Class.create({ initialize: function() { } }); var CdrView = Class.create(SipIxView,{ view: function() { } }); I get the following error in Safari. var cdrView = new CdrView(); TypeError:

Hi, I'm currently working on the TI DaVinci processor, which has an ARM926 processor and a TMS320C64+ DSP in it. I have been using Speex 6kbps (narrowband, quality=2, complexity=1, perceptual_enh=0) on the DSP C64+ side without any issues; the voice sounds very good for such a low bandwidth requirement. However, recently, I tried out the 4kbps mode (changing quality to 1) on the DSP and it

Hi Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth

Hello Joshua, I understand. My main aim to first understand the processing overhead when decisions have to be made for a fixed audio in signal, reduce it so that I can then begin making an update to reduce the memory (on chip flash and RAM) footprint. Thanks Amit On Thu, Dec 31, 2015 at 7:58 PM, Joshua Bowman <silverbacknet at gmail.com> wrote: > Opus will always use whatever mode it

Hi There, I have two OPUS handset clients say A & B A is 8 KHz, 12.2kbps cvbr supported OPUS client B is 8 KHz, 16kbps cvbr supported OPUS client When i try to encode a same voice frame(20ms sample frame) at different time intervals(not parallel encoding) for both A & B using same encoder handle by changing only bit rate. Issue here is, some noise is heard for B OPUS client, Ex:

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

Hi, I’m wondering how can I get the speech/music classification result when encoding the audio in opus 1.3.1? I found in the file opus_encoder.c, there is a opus_encoder_ctl request as OPUS_GET_VOICE_RATIO_REQUEST, so I wrote in my program the below code: #define OPUS_GET_VOICE_RATIO(x) 11019, __opus_check_int_ptr(x) int32_t voiceRatio; opus_encoder_ctl(encoder,

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

Hi! So here is what I got with different encoder settings; still not sure what the "best" setting is. 6kbps seems to add distortions, so I tried 12kbps. MP3-Original: LAME 3.99r, 120kbps, 44100Hz, Stereo, VBR V5 (22:23, 19.8MB) Opus (--raw-rate 44100 --bitrate 56 --vbr --comp 5): (44:45?, 23.4MB): Broken Opus (--bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 12.1 MB, 74kbps)

Hi, I noticed that speex.org has a banner that mentions that Opus is better than Speex in all aspects. The supported bitrate range for Speex seems to be as low as 2kbps though but Opus can only go as low as 6kbps. Is this one aspect where Speex is still preferred? (I understand that it's not a very common scenario though). Thanks, Manpreet. -------------- next part -------------- An HTML