g711 | Go package to encode and decode ITU-T G.711 sound data

I try to encode raw PCM sound to G711A and G711U and then decode it, with this codecs everything works fine because I can choose any value for AVCodecContext frame_size for encoding, but in case of Opus codec the AVCodecContext frame_size is equal to 120, so if I understood correctly if my input data array size is bigger than 120 then I need to do some kind of buffering and split my input data into several parts and then sequentially put it to AVFrame->data and pass the AVFrame to encoding.

In result I get a very bad sound and I get this result not only when I use Opus codec but also in G711 if I set it's AVCodecContext frame_size to some value that will be less than size of my input data.

So my question is: what it the correct way to encode input data if it's size if bigger than AVCodecContext frame_size? Do I need to split my input data into some parts that <= AVCodecContext frame_size if so how should I do that?

At this moment my code looks like this:

I'm trying to use NAudio to make a client to simulate a softphone to send telephony RTP packets in g.711 MuLaw format by capturing the local microphone/speaker devices, but this process is missing some steps that don't make sense with the outdated information available. MuLaw is incompatible with MediaFoundationResampler and WdlResampler, the ACM resampler garbles the audio quality completely, and the below code gets me to PCM but from there there's no information on how to go forward. Is there supposed to be a low-pass filter or something added here? Are you supposed to be converting the raw byte data from the WasapiCapture event to 16-bit according to a 2013 article (which is incompatible with the MFR anyway)?

I have no knowledge or experience working with audio, so this entire process is foreign to me and need some direction on where to proceed from here, as the only closest answer didn't actually post how they "solved" it.

I am using NAudio to attempt to convert Linear16 PCM wav files that come out of a 3rd party Text-To-Speech API to G711 8-bit 8-khz MULAW that will work as a telephony prompt. Using techniques found in the library authors documentation and some stack overflow posts and specifically following suggestion to do a 2 step conversion.

I need to convert an audio file (any common format) into a rtp stream saved in a .pcap file with G.722 Codec.

The generated .pcap file will be sent with SIPp using:

I'm recording a wav file using GStreamer receiving G711 flow through a UDP port. Any wav player can play the file, but shows a wrong duration and cannot fast forward. I believe that GStreamer writes the header at the beginning with empty data.

This pipeline can reproduce the issue:

Given a commandline ffmpeg -f lavfi -i "sine=frequency=1000:duration=5" -ar 8000 -c:a FOO pipe:1, ffmpeg might complain it's "unable to find a suitable output format". For some codecs, ffmpeg has a default container format, e.g. -c:a libmp3lame will produce MP3's. For other codecs, you just repeat yourself (sort of): -c:a pcm_alaw -f alaw

But what if I want to stream the raw codec output, and there's no matching -f? Can I just take -f alaw and assume that it doesn't do anything? (G711 alaw is a simple codec which produces a byte stream, so -f alaw presumably just copies that byte stream)

Obviously the other side of the pipe needs to know how the data stream needs to be interpreted, if there's no container info. But assume that I already know the other side of the pipe is expecting an audio stream, sampled at 8 kHz, encoded with -c:a FOO.

I have to write a program that solves the 7-11 problem (if you don't know what this is, Google will explain) but instead of finding the values that add and multiply to make $7.11, I have to find all the unique values of a, b, c and d that add and multiply to make $x.yz between $1.00 and $9.99.

It gets to the if statement where it checks if a, b, c and d multiply to make n but it is never true.

I am writing a program which should get sound from a pulseaudio source. I took pacat.c source file as an example.

Some of functions are declared in /pulsecore/*.h I include them into my source code:

#include

#include

#include

#include

#include

But compiler can't find them. fatal error: pulsecore/log.h: No such file or directory

I build pulseaudio 9.0 with Yocto.

In /$BUILDSDIR/pulseaudio/9.0-r0/image there are no such headers. But they exist in pulseaudio source code and also there /pulseaudio/9.0-r0/package/usr/src/debug/pulseaudio/9.0-r0/pulseaudio-9.0/src/pulsecore/

Is there any special option for pulseaduio to configure to copy headers into image? Or I should update yocto recipe to copy desired files?