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alsa: use blocking instead of non-blocking write

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The way we used non-blocking mode was HORRIBLE.

It was non-blocking to ALSA, but we end up blocking in a busy
loop that does absolutely NOTHING but retry.  We don't check
for playback cancellation (like we do in decoders) or anything.

This is seriously broken and I can imagine it affects people on
fast CPUs more because we do asynchronous output buffering and
our ALSA device will always have data ready.
This commit is contained in:
Eric Wong 2008-09-09 09:03:08 +02:00 committed by Max Kellermann
parent 37489b1f97
commit 5c81b716e2
1 changed files with 6 additions and 1 deletions
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7
src/audioOutputs/audioOutput_alsa.c
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@ -37,6 +37,9 @@ static const char default_device[] = "default";
#include <alsa/asoundlib.h>
/* #define MPD_SND_PCM_NONBLOCK SND_PCM_NONBLOCK */
#define MPD_SND_PCM_NONBLOCK 0
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size);
@ -157,16 +160,18 @@ static int alsa_openDevice(struct audio_output *audioOutput)
ad->device, audioFormat->bits);
err = snd_pcm_open(&ad->pcmHandle, ad->device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
SND_PCM_STREAM_PLAYBACK, MPD_SND_PCM_NONBLOCK);
if (err < 0) {
ad->pcmHandle = NULL;
goto error;
}
#if MPD_SND_PCM_NONBLOCK == SND_PCM_NONBLOCK
cmd = "snd_pcm_nonblock";
err = snd_pcm_nonblock(ad->pcmHandle, 0);
if (err < 0)
goto error;
#endif /* MPD_SND_PCM_NONBLOCK == SND_PCM_NONBLOCK */
period_time_ro = period_time = ad->period_time;
configure_hw: