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GitHub - libsndfile/libsamplerate: An audio Sample Rate ...
https://github.com/libsndfile/libsamplerate
This is libsamplerate, 0.2.2. libsamplerate (also known as Secret Rabbit Code) is a library for performing sample rate conversion of audio data. The src/ directory contains the source code for library itself.; The docs/ directory contains the libsamplerate documentation.; The examples/ directory contains examples of how to write code using libsamplerate.; The tests/ …
ALSA: What is the right way to capture directly playback ...
https://stackoverflow.com/questions/66878624/alsa-what-is-the-right-way-to-capture-directly-playback-audio-with-low-latency
Record audio on first USB sound card, simultaneously play back this audio on second sound card with low latency. Platform: Raspberry PI. Behavior. I tried many approaches (see below), but I can't get it to work. Searching for alsaloop gives very little or outdated results. What is the modern way to do it?
ALSA audio samplerate converter | PeteParker's Weblog
https://peteparker.wordpress.com/2007/09/28/alsa-audio-samplerate-converter/
hi friends, spent some time to set up audio stuff today. i wanted to try amarok, seems to be a nice GUI audio player by default it uses xine as a backend – but sound stuttered. also when just calling xine binary directl, setting audio.synchronization.force_rate to 48000 in ~/.xine/config helped but seemed not to affect amarok, anyways, after some reading i found …
alsa-plugins-rsound/samplerate.txt at master · …
https://github.com/Themaister/alsa-plugins-rsound/blob/master/doc/samplerate.txt
The plug plugin has also a similar field, "rate_converter". Or, more easily, define a global variable "defaults.pcm.rate_converter", which is used as the default converter type by plug and rate plugins: defaults.pcm.rate_converter "samplerate". Write the above in your ~/.asoundrc or /etc/asound.conf.
Tuning | Music Player Daemon Wiki | Fandom
https://mpd.fandom.com/wiki/Tuning
audio_output { type "alsa" name "my ALSA device" auto_resample "no" } The manual page provides more information about that, and how to choose libsamplerate's algorithm. Taking a look for samplerate_converter in your mpd.conf is good start point, consult manual of mpd.conf for possible values.
Sample Rate Conversion Quality in ALSA
https://alsa-devel.alsa-project.narkive.com/PFijNHd9/sample-rate-conversion-quality-in-alsa
amount of ALSA core will have to be modified to really fix the problem. The main problem is the buffer and period sized as they pass through dmix. For example, if the sound card hardware is running with 1024 samples per period at 48000, and an application wished to use a sample rate of 44100, the application should really get 940.8 samples per ...
AlsaPlayer
http://alsaplayer.sourceforge.net/
AlsaPlayer is a new type of PCM player. It is heavily multi-threaded and tries to excercise the ALSA library and driver quite a bit. It has some very interesting features unique to Linux/Unix players. The goal is to create a fully pluggable framework for playback of all sorts of media with the focus on PCM audio data.
Does anyone have trouble with pulse audio using a shit ton ...
https://www.reddit.com/r/linux/comments/1ogbem/does_anyone_have_trouble_with_pulse_audio_using_a/
When configuring Alsa itself on a card that doesn't mix in hardware, apart from dmix (nowadays on by default), might try routing through libsamplerate - 1 2. You do not need PulseAudio to have multiple apps with sound at once.
[Solved] Thunderbird don't play *.wav files. / Newbie ...
https://bbs.archlinux.org/viewtopic.php?id=125517
sound for mail notifications works. Summary - it must be esound and pulseaudio installed and running. It's necessary to change audio input in Audacious from ALSA to PulseAudio. All works well ;] Thanks. Last edited by Fixxer (2020-09-14 12:10:33)
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