similar to: opus Digest, Vol 109, Issue 8

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

On 02/23/2018 10:47 AM, Rodriguez, Vince wrote: > 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... Tried running a profiler? > I've also noticed if I change my encoder to silk only, I get a Heap > overflow... I have dynamically allocated a heap for the codec,

Jean-Marc, Thanks for the response and the helpful info. I am trying to get the library to build without using the pseudostack define, and use either VAR_ARRAYS or ALLOC, but it seems the global stack is not defined. Where do can I define this in my example? VR -----Original Message----- From: Jean-Marc Valin [mailto:jmvalin at jmvalin.ca] Sent: Tuesday, February 20, 2018 5:40 PM To:

On 02/20/2018 06:48 PM, Rodriguez, Vince wrote: > I am trying to get the library to build without using the pseudostack > define, and use either VAR_ARRAYS or ALLOC, but it seems the global > stack is not defined. Not sure what you mean here. If your compiler supports C99 VLAs, then by all means just define VAR_ARRAYS and you're good to go. If not, the second best option is alloca(),

Hi Vince, Normally, Opus should work fine on the CC3220, though real-time encoding may be a bit tight depending on what configuration you choose. You'll probably want to configure with --enable-fixed-point and --disable-float-api (or define FIXED_POINT and DISABLE_FLOAT_API if you don't use autoconf). The encoder has a bunch of run-time settings that you may want to set, depending on

Hello All, I'm looking to try to port a Opus Decoder and Encoder onto the TI CC3220SF (http://www.ti.com/product/CC3220) device. Currently, I have successfully been able to get the decode working for a .ogg file saved locally to the serial flash. My end goal is bidirectional audio using OPUS between two devices. While looking into the documentation for the Encoder, I was not sure what the

Marc Lindahl wrote: > Of course, I'm setting up a bunch of tests to evaluate these, what I was asking was more along the lines of, > If you set up the same exact, including the sample rate, do you get the same results (e.g. same code path, calculations, etc.?) If you configure a custom mode with the standard parameters (48 kHz sampling rate and a frame size of 120, 240, 480, or 960),

Hi, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

Thanks for the reply. For my application I unfortunately need a better signal reconstruction. It's not necessarily a problem that the constant DC voltage is removed, but the audio signal will need to be more exact because it's also studied visually. To be more exact, I need to record and transmit audio data recorded from animals (mostly birds). Are there ways to achieve a more similar

Hello Thomas, I use the following configure command to link against Ne10 Eg: configure --host=arm-linux-gnueabihf --enable-intrinsics --with-NE10-libraries=${BUILD_NE10_LIB} --with-NE10-includes=${BUILD_NE10_INC}" So, in my normal testing, I explicitly specify where the NE10 header files are installed and where the NE10 libraries are installed. Looking back at configure.ac

ffmpeg should be able to add it with a second -i parameter containing the thumbnail, however I think coverart embedding was only added very recently and have not tried it myself. Make sure it doesn't generate a Theora track! Alternatively there is this tool: https://github.com/Martchus/tageditor On 04/18/2018 04:38 PM, Sarah Szabo wrote: > > Adding the title and author worked

Just realized I sent this direct to Thomas and not to the list, whoops. Bad gmail. -map_metadata is the command to automatically apply the first input's metadata to the output, along with many parameters to integrate multiple source's metadata. I wish it was default. An explicit -metadata will override any tag map_metadata produces. On Wed, Apr 18, 2018 at 1:12 PM Thomas Daede <tdaede

Hi, I'm trying to make OPUS run on TI55, at the moment on the evm 5515 board. I'm compiling with FIXED_POINT, DISABLE_FLOAT_API, TI_C5X_ASM, CONFIG_TI_C55X. I'm actually interested only in the CELT part, so I'm also using OPUS_APPLICATION_RESTRICTED_LOWDELAY. It compiles but I'm still having few problems. I was wondering if anyone succeeded in making it run on this device, or

Hi everybody I wrote a couple of months ago about implementing the opus codec on a DK52 from Nordic (nrf52832 MCU, Cortex M4). In the meantime I got a bit further and would again like to ask some questions. My settings: I included all the opus files into my project (opus-1.2.1) Defined: FIXED POINT, VAR_ARRAYS, DISABLE FLOAT API, OPUS_BUILD, CELT_MODE_ONLY I am able to allocate enough memory

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

For some reason the attachment did not go through. Re-attaching. From: Mukund Raman Sent: Wednesday, April 01, 2015 6:12 PM To: 'opus at xiph.org' Subject: Opus multi-stream/surround: Audio corruption on decoded content Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I

On Sat, 2015-10-24 at 22:16 -0700, Thomas Daede wrote: > Everything above 96kbps on that table is speculative, as the highest > multi-participant listening testing done was at 96kbps. Here's the > results from that test, if you're curious: > > http://listening-test.coresv.net/results.htm > > As you can see, at that rate Opus ranged from slightly perceptible to >

Hi Jean-Marc, On Jun 15, 2013, at 12:20 PMEDT, Jean-Marc Valin wrote: > > >> So I still wonder, if you set up a custom mode, but then had all the >> settings the same as a normal mode, would the codec perform worse, or >> the same? > > You'll have to try normal vs custom modes and choose. The only thing I'm > telling you is don't run a 48 kHz

Created corresponding unit test, and the optimization is bit exact with C function. --- silk/SigProc_FIX.h | 7 ++- silk/arm/arm_silk_map.c | 12 ++++ silk/arm/inner_prod_aligned_arm.h | 58 +++++++++++++++++++ silk/arm/inner_prod_aligned_neon_intr.c | 66 ++++++++++++++++++++++ silk/enc_API.c

--- silk/SigProc_FIX.h | 4 +++ silk/arm/SigProc_FIX_arm64.h | 46 ++++++++++++++++++++++++++++++ silk/arm/macros_arm64.h | 66 ++++++++++++++++++++++++++++++++++++++++++++ silk/macros.h | 4 +++ silk_headers.mk | 2 ++ 5 files changed, 122 insertions(+) create mode 100644 silk/arm/SigProc_FIX_arm64.h create mode 100644 silk/arm/macros_arm64.h diff