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-rw-r--r--doc/mpd.conf.513
-rw-r--r--src/audioOutputs/audioOutput_alsa.c16
2 files changed, 28 insertions, 1 deletions
diff --git a/doc/mpd.conf.5 b/doc/mpd.conf.5
index 5ede7e744..34c194089 100644
--- a/doc/mpd.conf.5
+++ b/doc/mpd.conf.5
@@ -274,6 +274,19 @@ Setting this allows you to use memory-mapped I/O. Certain hardware setups may
benefit from this, but most do not. Most users do not need to set this. The
default is to not use memory-mapped I/O.
.TP
+.B auto_resample <yes or no>
+Setting this to "no" disables ALSA's software resampling, if the
+hardware does not support a specific sample rate. This lets MPD do
+the resampling. "yes" is the default and allows ALSA to resample.
+.TP
+.B auto_channels <yes or no>
+Setting this to "no" disables ALSA's channel conversion, if the
+hardware does not support a specific number of channels. Default: "yes".
+.TP
+.B auto_format <yes or no>
+Setting this to "no" disables ALSA's sample format conversion, if the
+hardware does not support a specific sample format. Default: "yes".
+.TP
.B buffer_time <time in microseconds>
This sets the length of the hardware sample buffer in microseconds. Increasing
it may help to reduce or eliminate skipping on certain setups. Most users do
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index ebe40d04c..e2132b9b9 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -37,6 +37,10 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
typedef struct _AlsaData {
const char *device;
+
+ /** the mode flags passed to snd_pcm_open */
+ int mode;
+
snd_pcm_t *pcmHandle;
alsa_writei_t *writei;
unsigned int buffer_time;
@@ -50,6 +54,7 @@ static AlsaData *newAlsaData(void)
AlsaData *ret = xmalloc(sizeof(AlsaData));
ret->device = default_device;
+ ret->mode = 0;
ret->pcmHandle = NULL;
ret->writei = snd_pcm_writei;
ret->useMmap = 0;
@@ -80,6 +85,15 @@ alsa_configure(AlsaData *ad, ConfigParam *param)
ad->buffer_time = atoi(bp->value);
if ((bp = getBlockParam(param, "period_time")))
ad->period_time = atoi(bp->value);
+
+ if (!getBoolBlockParam(param, "auto_resample", true))
+ ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
+
+ if (!getBoolBlockParam(param, "auto_channels", true))
+ ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
+
+ if (!getBoolBlockParam(param, "auto_format", true))
+ ad->mode |= SND_PCM_NO_AUTO_FORMAT;
}
static void *alsa_initDriver(mpd_unused struct audio_output *ao,
@@ -156,7 +170,7 @@ static int alsa_openDevice(void *data, struct audio_format *audioFormat)
ad->device, audioFormat->bits);
err = snd_pcm_open(&ad->pcmHandle, ad->device,
- SND_PCM_STREAM_PLAYBACK, 0);
+ SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
ad->pcmHandle = NULL;
goto error;