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-rw-r--r--src/audioOutputs/audioOutput_alsa.c293
1 files changed, 293 insertions, 0 deletions
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 */
+
+
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-09 15:50:36 +0000