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

audioOutput_alsa: calculate period size from sample rate
3c8e88b0 · Eric Wong · b59aa757 · 3c8e88b0
Commit 3c8e88b0 authored Jul 16, 2006 by Eric Wong
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audioOutput_alsa.c src/audioOutputs/audioOutput_alsa.c +4 -1

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--- 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"
@@ -212,6 +213,8 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 			&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);