When I'm playing my music with Deadbeef, Audacious or via MPD I get this period and buffer size parameters. Depending on bit-rate of recorded music files, you get some default values I guess. Is there a way to alter this default values with asoundrc or something else? I can do it easily with Jackd server but then you are stuck with one rate until you manually restart jackd and change it. <htpc>[~]# cat /proc/asound/card1/pcm0p/sub0/hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 5513 buffer_size: 22050 <htpc>[~]# cat /proc/asound/card1/pcm0p/sub0/hw_params access: RW_INTERLEAVED format: S24_3LE subformat: STD channels: 2 rate: 96000 (96000/1) period_size: 12000 buffer_size: 48000 I would like to change this default values in some asoundrc file. For example: If you play some music that is 24/92kHz recorded, change period size to 512 and buffer size to 1024. And when playback is back to 44.1kHz rate, set period size to 128 and buffer size to 256. How can I do that? ------------------------------------------------------------------------------ Get 100% visibility into Java/.NET code with AppDynamics Lite! It's a free troubleshooting tool designed for production. Get down to code-level detail for bottlenecks, with <2% overhead. Download for free and get started troubleshooting in minutes. http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clktrk _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
