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| author | Eric Wong <normalperson@yhbt.net> | 2006-07-16 16:52:06 +0000 |
|---|---|---|
| committer | Eric Wong <normalperson@yhbt.net> | 2006-07-16 16:52:06 +0000 |
| commit | 3c8e88b053e901de804a7d38f709056444155161 (patch) | |
| tree | 89399b629b5e0dd6d76239fec764997d4716a7f8 /src/audioOutputs | |
| parent | b59aa757205b2dc330d7864d88381c67beefb254 (diff) | |
| download | mpd-3c8e88b053e901de804a7d38f709056444155161.tar.gz mpd-3c8e88b053e901de804a7d38f709056444155161.tar.xz mpd-3c8e88b053e901de804a7d38f709056444155161.zip | |
audioOutput_alsa: calculate period size from sample rate
... instead of hard-coding it to a ridiculously high value that
makes bandwidth-starved devices unhappy.
libao (in SVN) does the same thing, and this calculation was indeed
taken from it.
Low-bandwidth USB (1.1) sound devices seem to need this to prevent
underrun / broken pipe errors (during hw setup, no less) from being
triggered.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4362 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/audioOutputs')
| -rw-r--r-- | src/audioOutputs/audioOutput_alsa.c | 7 |
1 files changed, 5 insertions, 2 deletions
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c index 976b72d31..0842b9f83 100644 --- a/src/audioOutputs/audioOutput_alsa.c +++ b/src/audioOutputs/audioOutput_alsa.c @@ -26,7 +26,8 @@ #define ALSA_PCM_NEW_SW_PARAMS_API #define MPD_ALSA_BUFFER_TIME 500000 -#define MPD_ALSA_PERIOD_TIME 50000 +#define MPD_ALSA_PERIOD_TIME 0 +#define MPD_ALSA_SAMPLE_XFER 256 #include "../conf.h" #include "../log.h" @@ -211,7 +212,9 @@ static int alsa_openDevice(AudioOutput * audioOutput) err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams, &alsa_buffer_time, 0); if(err < 0) goto error; - + + if (!alsa_period_time && sampleRate > 0) + alsa_period_time = 1000000 * MPD_ALSA_SAMPLE_XFER / sampleRate; cmd = "snd_pcm_hw_params_set_period_time_near"; err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams, &alsa_period_time, 0); |