audioOutput_alsa: add use_mmap, period_time, buffer_time options

doc/mpd.conf.5

@@ -236,6 +236,26 @@ separate options.  An example for oss: "dsp=/dev/dsp".  An example for alsa09:
 This specifies how many bytes to write to the audio device at once.  Used only
 by the ao output type.  This parameter is to work around a bug in older
 versions of libao on sound cards with very small buffers.  The default is 1024.
+.SH OPTIONAL ALSA OUTPUT PARAMETERS
+.TP
+.B use_mmap <yes or no>
+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.
+.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 not need to change this.
+The default is 500000 microseconds (0.5 seconds).
+.TP
+.B period_time <time in microseconds>
+This sets the time between hardware sample transfers in microseconds.
+Increasing this can reduce CPU usage while lowering it can reduce underrun
+errors on bandwidth-limited devices.  Some users have reported good results
+with this set to 50000, but not all devices support values this high.
+Most users do not need to change this.
+The default is 256000000 / sample_rate(kHz), or 5804us for CD-quality audio.
 .SH REQUIRED SHOUT OUTPUT PARAMETERS
 .TP
 .B name <name>

src/audioOutputs/audioOutput_alsa.c

@@ -46,6 +46,8 @@ typedef struct _AlsaData {
 	char * device;
 	snd_pcm_t * pcmHandle;
 	alsa_writei_t * writei;
+	unsigned int buffer_time;
+	unsigned int period_time;
 	int sampleSize;
 	int useMmap;
 	int canPause;
@@ -59,6 +61,8 @@ static AlsaData * newAlsaData() {
 	ret->pcmHandle = NULL;
 	ret->writei = snd_pcm_writei;
 	ret->useMmap = 0;
+	ret->buffer_time = MPD_ALSA_BUFFER_TIME;
+	ret->period_time = MPD_ALSA_PERIOD_TIME;
 
 	return ret;
 }
@@ -70,17 +74,23 @@ static void freeAlsaData(AlsaData * ad) {
 }
 
 static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) {
-	BlockParam * bp = NULL;
-	AlsaData * ad;
-	
-	if(param) bp = getBlockParam(param, "device");
-
-	ad = newAlsaData();
-	
+	AlsaData * ad = newAlsaData();
+
+	if (param) {
+		BlockParam * bp = getBlockParam(param, "device");
+		ad->device = bp ? strdup(bp->value) : strdup("default");
+
+		if ((bp = getBlockParam(param, "use_mmap")) &&
+					(!strcasecmp(bp->value, "yes") ||
+					 !strcasecmp(bp->value, "true")))
+			ad->useMmap = 1;
+		if ((bp = getBlockParam(param, "buffer_time")))
+			ad->buffer_time = atoi(bp->value);
+		if ((bp = getBlockParam(param, "period_time")))
+			ad->period_time = atoi(bp->value);
+	}
 	audioOutput->data = ad;
 
-	ad->device = bp ? strdup(bp->value) : strdup("default");
-
 	return 0;
 }
 
@@ -118,8 +128,6 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 	unsigned int channels = audioFormat->channels;
 	snd_pcm_uframes_t alsa_buffer_size;
 	snd_pcm_uframes_t alsa_period_size;
-	unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME;
-	unsigned int alsa_period_time = MPD_ALSA_PERIOD_TIME;
 	int err;
 	char * cmd = NULL;
 
@@ -210,14 +218,14 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 
 	cmd = "snd_pcm_hw_params_set_buffer_time_near";
 	err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
-			&alsa_buffer_time, 0);
+			&ad->buffer_time, 0);
 	if(err < 0) goto error;
 
-	if (!alsa_period_time && sampleRate > 0)
-		alsa_period_time = 1000000 * MPD_ALSA_SAMPLE_XFER / sampleRate;
+	if (!ad->period_time && sampleRate > 0)
+		ad->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);
+			&ad->period_time, 0);
 	if(err < 0) goto error;
 
 	cmd = "snd_pcm_hw_params";