implemented alsa audioOutput plugin, now it needs testing

git-svn-id: https://svn.musicpd.org/mpd/trunk@3008 09075e82-0dd4-0310-85a5-a0d7c8717e4f

4 files changed, 302 insertions, 2 deletions

diff --git a/TODO b/TODO
index d6cfbd027..2d5dc0672 100644
--- a/TODO
+++ b/TODO
@@ -22,6 +22,10 @@
 *) add a method for clearling the audio device's buffer.  This isn't possible 
 	with libao api.
 
+*) add support so that audioOutput plugins can modify the output audio format.
+	(This way, alsa's _near functions can be used to adjust for output
+	 devices on the fly: for channels, bits, and rate)
+
 *) add support for playing aac streams (gee, thanks icecast)
 
 *) implement apev2 and id3v1 tag reader from xmms-musepack plugin
diff --git a/src/Makefile.am b/src/Makefile.am
index 595259c26..ee52e9605 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -2,6 +2,7 @@ bin_PROGRAMS = mpd
 SUBDIRS = $(ID3_SUBDIR) $(MAD_SUBDIR) $(MP4FF_SUBDIR)
 
 mpd_audioOutputs = \
+	audioOutputs/audioOutput_alsa.c \
 	audioOutputs/audioOutput_ao.c \
 	audioOutputs/audioOutput_oss.c \
 	audioOutputs/audioOutput_shout.c
diff --git a/src/audio.c b/src/audio.c
index 02974228a..cf2b54c20 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -56,9 +56,10 @@ int cmpAudioFormat(AudioFormat * f1, AudioFormat * f2) {
 	return memcmp(f1, f2, sizeof(AudioFormat));
 }
 
+extern AudioOutputPlugin alsaPlugin;
 extern AudioOutputPlugin aoPlugin;
-extern AudioOutputPlugin shoutPlugin;
 extern AudioOutputPlugin ossPlugin;
+extern AudioOutputPlugin shoutPlugin;
 
 /* make sure initPlayerData is called before this function!! */
 void initAudioDriver() {
@@ -66,9 +67,10 @@ void initAudioDriver() {
 	int i;
 
 	initAudioOutputPlugins();
+	loadAudioOutputPlugin(&alsaPlugin);
 	loadAudioOutputPlugin(&aoPlugin);
-	loadAudioOutputPlugin(&shoutPlugin);
 	loadAudioOutputPlugin(&ossPlugin);
+	loadAudioOutputPlugin(&shoutPlugin);
 
 	pdAudioDevicesEnabled = (getPlayerData())->audioDeviceEnabled;
 
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
new file mode 100644
index 000000000..63217e9fe
--- /dev/null
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -0,0 +1,293 @@
+/* the Music Player Daemon (MPD)
+ * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "../audioOutput.h"
+
+#include <stdlib.h>
+
+#ifdef HAVE_ALSA
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+#define MPD_ALSA_BUFFER_TIME 500000
+#define MPD_ALSA_PERIOD_TIME 50000
+
+#include "../conf.h"
+#include "../log.h"
+#include "../sig_handlers.h"
+
+#include <string.h>
+#include <assert.h>
+#include <signal.h>
+
+#include <alsa/asoundlib.h>
+
+typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer,
+                                                snd_pcm_uframes_t size);
+
+typedef struct _AlsaData {
+	char * device;
+	snd_pcm_t * pcm_handle;
+	int mmap;
+	alsa_writei_t * writei;
+} AlsaData;
+
+static AlsaData * newAlsaData() {
+	AlsaData * ret = malloc(sizeof(AlsaData));
+
+	ret->device = NULL;
+	ret->pcm_handle = NULL;
+	ret->writei = snd_pcm_writei;
+	ret->mmap = 0;
+
+	return ret;
+}
+
+static void freeAlsaData(AlsaData * ad) {
+	if(ad->device) free(ad->device);
+
+	free(ad);
+}
+
+static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) {
+	BlockParam * bp = getBlockParam(param, "device");
+	AlsaData * ad = newAlsaData();
+	
+	audioOutput->data = ad;
+
+	ad->device = bp ? strdup(bp->value) : strdup("default");
+
+	return 0;
+}
+
+static void alsa_finishDriver(AudioOutput * audioOutput) {
+	AlsaData * ad = audioOutput->data;
+
+	freeAlsaData(ad);
+}
+
+static int alsa_openDevice(AudioOutput * audioOutput) 
+{
+	AlsaData * ad = audioOutput->data;
+	AudioFormat * audioFormat = &audioOutput->outAudioFormat;
+	snd_pcm_format_t bitformat;
+	snd_pcm_hw_params_t * hwparams;
+	snd_pcm_sw_params_t * swparams;
+	unsigned int sampleRate = audioFormat->sampleRate;
+	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;
+
+	switch(audioFormat->bits) {
+	case 8:
+		bitformat = SND_PCM_FORMAT_S8;
+		break;
+	case 16:
+		bitformat = SND_PCM_FORMAT_S16;
+		break;
+	case 24:
+		bitformat = SND_PCM_FORMAT_S16;
+		break;
+	case 32:
+		bitformat = SND_PCM_FORMAT_S16;
+		break;
+	default:
+		ERROR("Alsa device \"%s\" doesn't support %i bit audio\n",
+				ad->device, audioFormat->bits);
+		return -1;
+	}
+
+	err = snd_pcm_open(&ad->pcm_handle, ad->device, 
+			SND_PCM_STREAM_PLAYBACK, 0);
+	if(err < 0) {
+		ad->pcm_handle = NULL;
+		goto error;
+	}
+
+	err = snd_pcm_nonblock(ad->pcm_handle, 0);
+	if(err < 0) goto error;
+
+	// configure HW params
+	snd_pcm_hw_params_alloca(&hwparams);
+
+	err = snd_pcm_hw_params_any(ad->pcm_handle, hwparams);
+	if(err < 0) goto error;
+
+	if(ad->mmap) {
+		err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
+				SND_PCM_ACCESS_MMAP_INTERLEAVED);
+		if(err < 0) {
+			ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
+					" %s\n", ad->device, 
+					snd_strerror(-err));
+			ERROR("Falling back to direct write mode\n");
+			ad->mmap = 0;
+		}
+		else ad->writei = snd_pcm_mmap_writei;
+	}
+
+	if(!ad->mmap) {
+		err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
+				SND_PCM_ACCESS_RW_INTERLEAVED);
+		if(err < 0) goto error;
+		ad->writei = snd_pcm_mmap_writei;
+	}
+
+	err = snd_pcm_hw_params_set_format(ad->pcm_handle, hwparams, bitformat);
+	if(err < 0) {
+		ERROR("Alsa device \"%s\" does not support %i bit audio: "
+				"%s\n", ad->device, (int)bitformat, 
+				snd_strerror(-err));
+		goto fail;
+	}
+
+	err = snd_pcm_hw_params_set_channels(ad->pcm_handle, hwparams, 
+			(unsigned int)audioFormat->channels);
+	if(err < 0) {
+		ERROR("Alsa device \"%s\" does not support %i channels: "
+				"%s\n", ad->device, (int)audioFormat->channels, 
+				snd_strerror(-err));
+		goto fail;
+	}
+
+	err = snd_pcm_hw_params_set_rate_near(ad->pcm_handle, hwparams, 
+			&sampleRate, 0);
+	if(err < 0 || sampleRate == 0) {
+		ERROR("Alsa device \"%s\" does not support %i Hz audio\n",
+				ad->device, (int)audioFormat->sampleRate);
+		goto fail;
+	}
+
+	err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm_handle, hwparams,
+			&alsa_buffer_time, 0);
+	if(err < 0) goto error;
+	
+	err = snd_pcm_hw_params_set_period_time_near(ad->pcm_handle, hwparams,
+			&alsa_period_time, 0);
+	if(err < 0) goto error;
+
+	err = snd_pcm_hw_params(ad->pcm_handle, hwparams);
+	if(err < 0) goto error;
+
+	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
+	if(err < 0) goto error;
+
+	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0);
+	if(err < 0) goto error;
+
+	// configure SW params
+	snd_pcm_sw_params_alloca(&swparams);
+	snd_pcm_sw_params_current(ad->pcm_handle, swparams);
+
+	err = snd_pcm_sw_params_set_start_threshold(ad->pcm_handle, swparams,
+			alsa_buffer_size - alsa_period_size);
+	if(err < 0) goto error;
+
+	err = snd_pcm_sw_params(ad->pcm_handle, swparams);
+	if(err < 0) goto error;
+	
+	audioOutput->open = 1;
+
+	return 0;
+
+error:
+	ERROR("Error opening alsa device \"%s\": %s\n", ad->device, 
+			snd_strerror(-err));
+fail:
+	if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle);
+	audioOutput->open = 0;
+	return -1;
+}
+
+static void alsa_closeDevice(AudioOutput * audioOutput) {
+	AlsaData * ad = audioOutput->data;
+
+	if(ad->pcm_handle) {
+		snd_pcm_drain(ad->pcm_handle);
+		ad->pcm_handle = NULL;
+	}
+
+	audioOutput->open = 0;
+}
+
+inline static int alsa_errorRecovery(AlsaData * ad, int err) {
+	if(err == -EPIPE) {
+		DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
+		err = snd_pcm_prepare(ad->pcm_handle);
+		if(err < 0) return -1;
+		return 0;
+	}
+
+	return err;
+}
+
+static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, 
+		int size) 
+{
+	AlsaData * ad = audioOutput->data;
+	int ret;
+
+	while (size > 0) {
+		ret = ad->writei(ad->pcm_handle, playChunk, size);
+
+		if(ret == -EAGAIN) continue;
+		
+		if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) {
+			ERROR("closing alsa device \"%s\" due to write error:"
+					" %s\n", ad->device, 
+					snd_strerror(-errno));
+			alsa_closeDevice(audioOutput);
+			return -1;
+		}
+		playChunk += ret;
+		size -= ret;
+	}
+
+	return 0;
+}
+
+AudioOutputPlugin alsaPlugin =
+{
+	"alsa",
+	alsa_initDriver,
+	alsa_finishDriver,
+	alsa_openDevice,
+	alsa_playAudio,
+	alsa_closeDevice,
+	NULL /* sendMetadataFunc */
+};
+
+#else /* HAVE ALSA */
+
+AudioOutputPlugin alsaPlugin =
+{
+	NULL,
+	NULL,
+	NULL,
+	NULL,
+	NULL,
+	NULL,
+	NULL /* sendMetadataFunc */
+};
+
+#endif /* HAVE_ALSA */
+
+