alsa

I am New to audio programming.I want to create small application which is capable of playing and gives volume control . I am using alsa-lib. I want to know what is the purpose of switch (ex.Master Playback switch), enum in mixer elements and what value should i set to those switchs . Please suggest me some tutorial for mixer settings as well as alsa programming . Just collecting some here, that have example code: ALSA Programming HOWTO v.1.0.0 [alsamodular.sourceforge.net] A tutorial on using the ALSA Audio API [equalarea.com] 2002 A close look at ALSA [volkerschatz.com] ALSA API - Sample

I've been trying to look into using the OpenSL ES library that is available for doing native audio with android 2.3, but it appears that the header files and the 600 page pdf of the spec are the only available documentation. Where should I be looking for examples, tutorials, or a brief overview of the capabilities? Phonon There's an example app that comes with the NDK called NativeAudio . It has examples of nearly anything you would ever want to implement: effects, streaming, synthesis, etc. If you look at its code, you'll get a good head start. You could also check victor lazzarini's audio

最近学习用到alsa，所以就准备移植alsa到自己的tiny6410上， 首先需要准备些东西，tiny6410的文件系统，如果用NFS就更好； alsa-lib-1.0.24.1.tar.bz2 alsa-utils-1.0.24.2.tar.bz2 好了，下面开始移植 环境：fedora12 交叉工具位置: /opt/FriendlyARM/toolschain/4.5.1/ 开发板根文件系统位置： /opt/filesystem/rootfs-qt4-2/ 在/opt/alsa下解压源码 tar xjf alsa-lib-1.0.24.1.tar.bz2 tar xjf alsa-utils-1.0.24.2.tar.bz2 1.开始移植alsa-lib 进入alsa-lib源码目 录，输入以下命令 ./configure --host=arm-linux --target=arm-linux --prefix=/opt/FriendlyARM/toolschain/4.5.1/arm-none-linux-gnueabi/lib/arm-alsa --enable-static --enable-shared --disable-python --with-configdir=/usr/share --with-plugindir=/usr/lib/alsa_lib

I have audio conferencing device which has a mic and speaker. Both mic and speaker having same sampling rate set by snd_pcm_hw_params_set_rate_near() say 8000 Hz. Do I need to set period size and number of periods to be same for both (for two such devices communicate each other) ? Or can I choose a different period/period size for speaker than the one configured for mic ? How do I decide on which period and period size to use ? I am using snd_pcm_hw_params_set_period_size_near() and snd_pcm_hw_params_set_periods_near() APIs to set period size and period for both speaker and mic. (I am new to

I'm looking for a way to feed audio data from a file into the microphone so when 3rd party applications (such as arecord or Chromium's "search by voice" feature) use the microphone for audio input, they receive the audio data from the file instead. Here's my scenario : An application I wrote records audio data from the microphone (using ALSA) and saves it to a file (audioFile0.raw). At some unknown point in time in the future, some unknown 3rd party application (as in, something I did not develop so I have no development control over, such as the Chromium web browser's "search by voice"

I'm using ALSA for and audio application on Linux, I found great docs explain how to use it : 1 and this one . although I have some issues to understand this part of the setup : /* Set number of periods. Periods used to be called fragments. */ if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return(-1); } what does mean set a number of period when I'm using the PLAYBACK mode and : /* Set buffer size (in frames). The resulting latency is given by */ /* latency = periodsize * periods / (rate * bytes_per_frame) */ if (snd_pcm

This question is inspired by Can you run GUI apps in a docker container? . The basic idea is to run apps with audio and ui (vlc, firefox, skype, ...) I was searching for docker containers using pulseaudio but all containers I found where using pulseaudio streaming over tcp. (security sandboxing of the applications) https://gist.github.com/hybris42/ce429de428e5af3a344a https://github.com/jlund/docker-chrome-pulseaudio https://github.com/tomparys/docker-skype-pulseaudio In my case I would prefere playing audio from an app inside the container directly to my host pulseaudio. (without ssh