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-rw-r--r--trunk/src/audioOutputs/audioOutput_alsa.c427
-rw-r--r--trunk/src/audioOutputs/audioOutput_ao.c246
-rw-r--r--trunk/src/audioOutputs/audioOutput_jack.c440
-rw-r--r--trunk/src/audioOutputs/audioOutput_mvp.c284
-rw-r--r--trunk/src/audioOutputs/audioOutput_oss.c575
-rw-r--r--trunk/src/audioOutputs/audioOutput_osx.c374
-rw-r--r--trunk/src/audioOutputs/audioOutput_pulse.c221
-rw-r--r--trunk/src/audioOutputs/audioOutput_shout.c636
8 files changed, 3203 insertions, 0 deletions
diff --git a/trunk/src/audioOutputs/audioOutput_alsa.c b/trunk/src/audioOutputs/audioOutput_alsa.c
new file mode 100644
index 000000000..3ade3df46
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_alsa.c
@@ -0,0 +1,427 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * 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_US 500000
+/* the default period time of xmms is 50 ms, so let's use that as well.
+ * a user can tweak this parameter via the "period_time" config parameter.
+ */
+#define MPD_ALSA_PERIOD_TIME_US 50000
+#define MPD_ALSA_RETRY_NR 5
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.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 *pcmHandle;
+ alsa_writei_t *writei;
+ unsigned int buffer_time;
+ unsigned int period_time;
+ int sampleSize;
+ int useMmap;
+ int canPause;
+ int canResume;
+} AlsaData;
+
+static AlsaData *newAlsaData(void)
+{
+ AlsaData *ret = xmalloc(sizeof(AlsaData));
+
+ ret->device = NULL;
+ ret->pcmHandle = NULL;
+ ret->writei = snd_pcm_writei;
+ ret->useMmap = 0;
+ ret->buffer_time = MPD_ALSA_BUFFER_TIME_US;
+ ret->period_time = MPD_ALSA_PERIOD_TIME_US;
+
+ return ret;
+}
+
+static void freeAlsaData(AlsaData * ad)
+{
+ if (ad->device)
+ free(ad->device);
+
+ free(ad);
+}
+
+static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+ AlsaData *ad = newAlsaData();
+
+ if (param) {
+ BlockParam *bp = getBlockParam(param, "device");
+ ad->device = bp ? xstrdup(bp->value) : xstrdup("default");
+
+ if ((bp = getBlockParam(param, "use_mmap")) &&
+ !strcasecmp(bp->value, "yes"))
+ 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);
+ } else
+ ad->device = xstrdup("default");
+ audioOutput->data = ad;
+
+ return 0;
+}
+
+static void alsa_finishDriver(AudioOutput * audioOutput)
+{
+ AlsaData *ad = audioOutput->data;
+
+ freeAlsaData(ad);
+}
+
+static int alsa_testDefault(void)
+{
+ snd_pcm_t *handle;
+
+ int ret = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK);
+ snd_config_update_free_global();
+
+ if (ret) {
+ WARNING("Error opening default alsa device: %s\n",
+ snd_strerror(-ret));
+ return -1;
+ } else
+ snd_pcm_close(handle);
+
+ return 0;
+}
+
+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;
+ unsigned int channels = audioFormat->channels;
+ snd_pcm_uframes_t alsa_buffer_size;
+ snd_pcm_uframes_t alsa_period_size;
+ int err;
+ const char *cmd = NULL;
+ int retry = MPD_ALSA_RETRY_NR;
+ unsigned int period_time, period_time_ro;
+ unsigned int buffer_time;
+
+ 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_S24;
+ break;
+ case 32:
+ bitformat = SND_PCM_FORMAT_S32;
+ break;
+ default:
+ ERROR("ALSA device \"%s\" doesn't support %i bit audio\n",
+ ad->device, audioFormat->bits);
+ return -1;
+ }
+
+ err = snd_pcm_open(&ad->pcmHandle, ad->device,
+ SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+ snd_config_update_free_global();
+ if (err < 0) {
+ ad->pcmHandle = NULL;
+ goto error;
+ }
+
+ cmd = "snd_pcm_nonblock";
+ err = snd_pcm_nonblock(ad->pcmHandle, 0);
+ if (err < 0)
+ goto error;
+
+ period_time_ro = period_time = ad->period_time;
+configure_hw:
+ /* configure HW params */
+ snd_pcm_hw_params_alloca(&hwparams);
+
+ cmd = "snd_pcm_hw_params_any";
+ err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
+ if (err < 0)
+ goto error;
+
+ if (ad->useMmap) {
+ err = snd_pcm_hw_params_set_access(ad->pcmHandle, 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->useMmap = 0;
+ } else
+ ad->writei = snd_pcm_mmap_writei;
+ }
+
+ if (!ad->useMmap) {
+ cmd = "snd_pcm_hw_params_set_access";
+ err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (err < 0)
+ goto error;
+ ad->writei = snd_pcm_writei;
+ }
+
+ err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
+ if (err < 0) {
+ ERROR("ALSA device \"%s\" does not support %i bit audio: "
+ "%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
+ goto fail;
+ }
+
+ err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
+ &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;
+ }
+ audioFormat->channels = channels;
+
+ err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
+ &sampleRate, NULL);
+ if (err < 0 || sampleRate == 0) {
+ ERROR("ALSA device \"%s\" does not support %i Hz audio\n",
+ ad->device, (int)audioFormat->sampleRate);
+ goto fail;
+ }
+ audioFormat->sampleRate = sampleRate;
+
+ buffer_time = ad->buffer_time;
+ cmd = "snd_pcm_hw_params_set_buffer_time_near";
+ err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
+ &buffer_time, NULL);
+ if (err < 0)
+ goto error;
+
+ period_time = period_time_ro;
+ cmd = "snd_pcm_hw_params_set_period_time_near";
+ err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
+ &period_time, NULL);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_hw_params";
+ err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
+ if (err == -EPIPE && --retry > 0) {
+ period_time_ro = period_time_ro >> 1;
+ goto configure_hw;
+ } else if (err < 0)
+ goto error;
+ if (retry != MPD_ALSA_RETRY_NR)
+ DEBUG("ALSA period_time set to %d\n", period_time);
+
+ cmd = "snd_pcm_hw_params_get_buffer_size";
+ err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_hw_params_get_period_size";
+ err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
+ NULL);
+ if (err < 0)
+ goto error;
+
+ ad->canPause = snd_pcm_hw_params_can_pause(hwparams);
+ ad->canResume = snd_pcm_hw_params_can_resume(hwparams);
+
+ /* configure SW params */
+ snd_pcm_sw_params_alloca(&swparams);
+
+ cmd = "snd_pcm_sw_params_current";
+ err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_start_threshold";
+ err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
+ alsa_buffer_size -
+ alsa_period_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_avail_min";
+ err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
+ alsa_period_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_xfer_align";
+ err = snd_pcm_sw_params_set_xfer_align(ad->pcmHandle, swparams, 1);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params";
+ err = snd_pcm_sw_params(ad->pcmHandle, swparams);
+ if (err < 0)
+ goto error;
+
+ ad->sampleSize = (audioFormat->bits / 8) * audioFormat->channels;
+
+ audioOutput->open = 1;
+
+ DEBUG("alsa device \"%s\" will be playing %i bit, %i channel audio at "
+ "%i Hz\n", ad->device, (int)audioFormat->bits,
+ channels, sampleRate);
+
+ return 0;
+
+error:
+ if (cmd) {
+ ERROR("Error opening alsa device \"%s\" (%s): %s\n",
+ ad->device, cmd, snd_strerror(-err));
+ } else {
+ ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
+ snd_strerror(-err));
+ }
+fail:
+ if (ad->pcmHandle)
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
+ audioOutput->open = 0;
+ return -1;
+}
+
+static int alsa_errorRecovery(AlsaData * ad, int err)
+{
+ if (err == -EPIPE) {
+ DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
+ } else if (err == -ESTRPIPE) {
+ DEBUG("alsa device \"%s\" was suspended\n", ad->device);
+ }
+
+ switch (snd_pcm_state(ad->pcmHandle)) {
+ case SND_PCM_STATE_PAUSED:
+ err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
+ break;
+ case SND_PCM_STATE_SUSPENDED:
+ err = ad->canResume ?
+ snd_pcm_resume(ad->pcmHandle) :
+ snd_pcm_prepare(ad->pcmHandle);
+ break;
+ case SND_PCM_STATE_SETUP:
+ case SND_PCM_STATE_XRUN:
+ err = snd_pcm_prepare(ad->pcmHandle);
+ break;
+ case SND_PCM_STATE_DISCONNECTED:
+ /* so alsa_closeDevice won't try to drain: */
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
+ break;
+ default:
+ /* unknown state, do nothing */
+ break;
+ }
+
+ return err;
+}
+
+static void alsa_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ AlsaData *ad = audioOutput->data;
+
+ alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
+}
+
+static void alsa_closeDevice(AudioOutput * audioOutput)
+{
+ AlsaData *ad = audioOutput->data;
+
+ if (ad->pcmHandle) {
+ snd_pcm_drain(ad->pcmHandle);
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
+ }
+
+ audioOutput->open = 0;
+}
+
+static int alsa_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+ AlsaData *ad = audioOutput->data;
+ int ret;
+
+ size /= ad->sampleSize;
+
+ while (size > 0) {
+ ret = ad->writei(ad->pcmHandle, playChunk, size);
+
+ if (ret == -EAGAIN || ret == -EINTR)
+ continue;
+
+ if (ret < 0) {
+ if (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;
+ }
+ continue;
+ }
+
+ playChunk += ret * ad->sampleSize;
+ size -= ret;
+ }
+
+ return 0;
+}
+
+AudioOutputPlugin alsaPlugin = {
+ "alsa",
+ alsa_testDefault,
+ alsa_initDriver,
+ alsa_finishDriver,
+ alsa_openDevice,
+ alsa_playAudio,
+ alsa_dropBufferedAudio,
+ alsa_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else /* HAVE ALSA */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
+#endif /* HAVE_ALSA */
diff --git a/trunk/src/audioOutputs/audioOutput_ao.c b/trunk/src/audioOutputs/audioOutput_ao.c
new file mode 100644
index 000000000..a7f437ef4
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_ao.c
@@ -0,0 +1,246 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * 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"
+
+#ifdef HAVE_AO
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <ao/ao.h>
+
+static int driverInitCount;
+
+typedef struct _AoData {
+ int writeSize;
+ int driverId;
+ ao_option *options;
+ ao_device *device;
+} AoData;
+
+static AoData *newAoData(void)
+{
+ AoData *ret = xmalloc(sizeof(AoData));
+ ret->device = NULL;
+ ret->options = NULL;
+
+ return ret;
+}
+
+static void audioOutputAo_error(void)
+{
+ if (errno == AO_ENOTLIVE) {
+ ERROR("not a live ao device\n");
+ } else if (errno == AO_EOPENDEVICE) {
+ ERROR("not able to open audio device\n");
+ } else if (errno == AO_EBADOPTION) {
+ ERROR("bad driver option\n");
+ }
+}
+
+static int audioOutputAo_initDriver(AudioOutput * audioOutput,
+ ConfigParam * param)
+{
+ ao_info *ai;
+ char *dup;
+ char *stk1;
+ char *stk2;
+ char *n1;
+ char *key;
+ char *value;
+ char *test;
+ AoData *ad = newAoData();
+ BlockParam *blockParam;
+
+ audioOutput->data = ad;
+
+ if ((blockParam = getBlockParam(param, "write_size"))) {
+ ad->writeSize = strtol(blockParam->value, &test, 10);
+ if (*test != '\0') {
+ FATAL("\"%s\" is not a valid write size at line %i\n",
+ blockParam->value, blockParam->line);
+ }
+ } else
+ ad->writeSize = 1024;
+
+ if (driverInitCount == 0) {
+ ao_initialize();
+ }
+ driverInitCount++;
+
+ blockParam = getBlockParam(param, "driver");
+
+ if (!blockParam || 0 == strcmp(blockParam->value, "default")) {
+ ad->driverId = ao_default_driver_id();
+ } else if ((ad->driverId = ao_driver_id(blockParam->value)) < 0) {
+ FATAL("\"%s\" is not a valid ao driver at line %i\n",
+ blockParam->value, blockParam->line);
+ }
+
+ if ((ai = ao_driver_info(ad->driverId)) == NULL) {
+ FATAL("problems getting driver info for device defined at line %i\n"
+ "you may not have permission to the audio device\n", param->line);
+ }
+
+ DEBUG("using ao driver \"%s\" for \"%s\"\n", ai->short_name,
+ audioOutput->name);
+
+ blockParam = getBlockParam(param, "options");
+
+ if (blockParam) {
+ dup = xstrdup(blockParam->value);
+ } else
+ dup = xstrdup("");
+
+ if (strlen(dup)) {
+ stk1 = NULL;
+ n1 = strtok_r(dup, ";", &stk1);
+ while (n1) {
+ stk2 = NULL;
+ key = strtok_r(n1, "=", &stk2);
+ if (!key)
+ FATAL("problems parsing options \"%s\"\n", n1);
+ /*found = 0;
+ for(i=0;i<ai->option_count;i++) {
+ if(strcmp(ai->options[i],key)==0) {
+ found = 1;
+ break;
+ }
+ }
+ if(!found) {
+ FATAL("\"%s\" is not an option for "
+ "\"%s\" ao driver\n",key,
+ ai->short_name);
+ } */
+ value = strtok_r(NULL, "", &stk2);
+ if (!value)
+ FATAL("problems parsing options \"%s\"\n", n1);
+ ao_append_option(&ad->options, key, value);
+ n1 = strtok_r(NULL, ";", &stk1);
+ }
+ }
+ free(dup);
+
+ return 0;
+}
+
+static void freeAoData(AoData * ad)
+{
+ ao_free_options(ad->options);
+ free(ad);
+}
+
+static void audioOutputAo_finishDriver(AudioOutput * audioOutput)
+{
+ AoData *ad = (AoData *) audioOutput->data;
+ freeAoData(ad);
+
+ driverInitCount--;
+
+ if (driverInitCount == 0)
+ ao_shutdown();
+}
+
+static void audioOutputAo_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ /* not supported by libao */
+}
+
+static void audioOutputAo_closeDevice(AudioOutput * audioOutput)
+{
+ AoData *ad = (AoData *) audioOutput->data;
+
+ if (ad->device) {
+ ao_close(ad->device);
+ ad->device = NULL;
+ }
+
+ audioOutput->open = 0;
+}
+
+static int audioOutputAo_openDevice(AudioOutput * audioOutput)
+{
+ ao_sample_format format;
+ AoData *ad = (AoData *) audioOutput->data;
+
+ if (ad->device) {
+ audioOutputAo_closeDevice(audioOutput);
+ }
+
+ format.bits = audioOutput->outAudioFormat.bits;
+ format.rate = audioOutput->outAudioFormat.sampleRate;
+ format.byte_format = AO_FMT_NATIVE;
+ format.channels = audioOutput->outAudioFormat.channels;
+
+ ad->device = ao_open_live(ad->driverId, &format, ad->options);
+
+ if (ad->device == NULL)
+ return -1;
+
+ audioOutput->open = 1;
+
+ return 0;
+}
+
+static int audioOutputAo_play(AudioOutput * audioOutput, char *playChunk,
+ int size)
+{
+ int send;
+ AoData *ad = (AoData *) audioOutput->data;
+
+ if (ad->device == NULL)
+ return -1;
+
+ while (size > 0) {
+ send = ad->writeSize > size ? size : ad->writeSize;
+
+ if (ao_play(ad->device, playChunk, send) == 0) {
+ audioOutputAo_error();
+ ERROR("closing audio device due to write error\n");
+ audioOutputAo_closeDevice(audioOutput);
+ return -1;
+ }
+
+ playChunk += send;
+ size -= send;
+ }
+
+ return 0;
+}
+
+AudioOutputPlugin aoPlugin = {
+ "ao",
+ NULL,
+ audioOutputAo_initDriver,
+ audioOutputAo_finishDriver,
+ audioOutputAo_openDevice,
+ audioOutputAo_play,
+ audioOutputAo_dropBufferedAudio,
+ audioOutputAo_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else
+
+#include <stdio.h>
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
+#endif
diff --git a/trunk/src/audioOutputs/audioOutput_jack.c b/trunk/src/audioOutputs/audioOutput_jack.c
new file mode 100644
index 000000000..1fdfaf4bb
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_jack.c
@@ -0,0 +1,440 @@
+/* jack plug in for the Music Player Daemon (MPD)
+ * (c)2006 by anarch(anarchsss@gmail.com)
+ *
+ * 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"
+
+#ifdef HAVE_JACK
+
+#include <stdlib.h>
+#include <errno.h>
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+#include <pthread.h>
+
+#include <jack/jack.h>
+#include <jack/types.h>
+#include <jack/ringbuffer.h>
+
+pthread_mutex_t play_audio_lock = PTHREAD_MUTEX_INITIALIZER;
+pthread_cond_t play_audio = PTHREAD_COND_INITIALIZER;
+
+/*#include "dmalloc.h"*/
+
+#define MIN(a, b) ((a) < (b) ? (a) : (b))
+/*#define SAMPLE_SIZE sizeof(jack_default_audio_sample_t);*/
+
+
+static char *name = "mpd";
+static char *output_ports[2];
+static int ringbuf_sz = 32768;
+size_t sample_size = sizeof(jack_default_audio_sample_t);
+
+typedef struct _JackData {
+ jack_port_t *ports[2];
+ jack_client_t *client;
+ jack_ringbuffer_t *ringbuffer[2];
+ int bps;
+ int shutdown;
+} JackData;
+
+/*JackData *jd = NULL;*/
+
+static JackData *newJackData(void)
+{
+ JackData *ret;
+ ret = xcalloc(sizeof(JackData), 1);
+
+ return ret;
+}
+
+static void freeJackData(AudioOutput *audioOutput)
+{
+ JackData *jd = audioOutput->data;
+ if (jd) {
+ if (jd->ringbuffer[0])
+ jack_ringbuffer_free(jd->ringbuffer[0]);
+ if (jd->ringbuffer[1])
+ jack_ringbuffer_free(jd->ringbuffer[1]);
+ free(jd);
+ audioOutput->data = NULL;
+ }
+}
+
+static void jack_finishDriver(AudioOutput *audioOutput)
+{
+ JackData *jd = audioOutput->data;
+ int i;
+
+ if ( jd && jd->client ) {
+ jack_deactivate(jd->client);
+ jack_client_close(jd->client);
+ }
+ DEBUG("disconnect_jack (pid=%d)\n", getpid ());
+
+ if ( strcmp(name, "mpd") ) {
+ free(name);
+ name = "mpd";
+ }
+
+ for ( i = ARRAY_SIZE(output_ports); --i >= 0; ) {
+ if (!output_ports[i])
+ continue;
+ free(output_ports[i]);
+ output_ports[i] = NULL;
+ }
+
+ freeJackData(audioOutput);
+}
+
+static int srate(jack_nframes_t rate, void *data)
+{
+ JackData *jd = (JackData *) ((AudioOutput*) data)->data;
+ AudioFormat *audioFormat = &(((AudioOutput*) data)->outAudioFormat);
+
+ audioFormat->sampleRate = (int)jack_get_sample_rate(jd->client);
+
+ return 0;
+}
+
+static int process(jack_nframes_t nframes, void *arg)
+{
+ size_t i;
+ JackData *jd = (JackData *) arg;
+ jack_default_audio_sample_t *out[2];
+ size_t avail_data, avail_frames;
+
+ if ( nframes <= 0 )
+ return 0;
+
+ out[0] = jack_port_get_buffer(jd->ports[0], nframes);
+ out[1] = jack_port_get_buffer(jd->ports[1], nframes);
+
+ while ( nframes ) {
+ avail_data = jack_ringbuffer_read_space(jd->ringbuffer[1]);
+
+ if ( avail_data > 0 ) {
+ avail_frames = avail_data / sample_size;
+
+ if (avail_frames > nframes) {
+ avail_frames = nframes;
+ avail_data = nframes*sample_size;
+ }
+
+ jack_ringbuffer_read(jd->ringbuffer[0], (char *)out[0],
+ avail_data);
+ jack_ringbuffer_read(jd->ringbuffer[1], (char *)out[1],
+ avail_data);
+
+ nframes -= avail_frames;
+ out[0] += avail_data;
+ out[1] += avail_data;
+ } else {
+ for (i = 0; i < nframes; i++)
+ out[0][i] = out[1][i] = 0.0;
+ nframes = 0;
+ }
+
+ if (pthread_mutex_trylock (&play_audio_lock) == 0) {
+ pthread_cond_signal (&play_audio);
+ pthread_mutex_unlock (&play_audio_lock);
+ }
+ }
+
+
+ /*DEBUG("process (pid=%d)\n", getpid());*/
+ return 0;
+}
+
+static void shutdown_callback(void *arg)
+{
+ JackData *jd = (JackData *) arg;
+ jd->shutdown = 1;
+}
+
+static void set_audioformat(AudioOutput *audioOutput)
+{
+ JackData *jd = audioOutput->data;
+ AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+
+ audioFormat->sampleRate = (int) jack_get_sample_rate(jd->client);
+ DEBUG("samplerate = %d\n", audioFormat->sampleRate);
+ audioFormat->channels = 2;
+ audioFormat->bits = 16;
+ jd->bps = audioFormat->channels
+ * sizeof(jack_default_audio_sample_t)
+ * audioFormat->sampleRate;
+}
+
+static void error_callback(const char *msg)
+{
+ ERROR("jack: %s\n", msg);
+}
+
+static int jack_initDriver(AudioOutput *audioOutput, ConfigParam *param)
+{
+ BlockParam *bp;
+ char *endptr;
+ int val;
+ char *cp = NULL;
+
+ DEBUG("jack_initDriver (pid=%d)\n", getpid());
+ if ( ! param ) return 0;
+
+ if ( (bp = getBlockParam(param, "ports")) ) {
+ DEBUG("output_ports=%s\n", bp->value);
+
+ if (!(cp = strchr(bp->value, ',')))
+ FATAL("expected comma and a second value for '%s' "
+ "at line %d: %s\n",
+ bp->name, bp->line, bp->value);
+
+ *cp = '\0';
+ output_ports[0] = xstrdup(bp->value);
+ *cp++ = ',';
+
+ if (!*cp)
+ FATAL("expected a second value for '%s' at line %d: "
+ "%s\n", bp->name, bp->line, bp->value);
+
+ output_ports[1] = xstrdup(cp);
+
+ if (strchr(cp,','))
+ FATAL("Only %d values are supported for '%s' "
+ "at line %d\n", (int)ARRAY_SIZE(output_ports),
+ bp->name, bp->line);
+ }
+
+ if ( (bp = getBlockParam(param, "ringbuffer_size")) ) {
+ errno = 0;
+ val = strtol(bp->value, &endptr, 10);
+
+ if ( errno == 0 && endptr != bp->value) {
+ ringbuf_sz = val < 32768 ? 32768 : val;
+ DEBUG("ringbuffer_size=%d\n", ringbuf_sz);
+ } else {
+ FATAL("%s is not a number; ringbuf_size=%d\n",
+ bp->value, ringbuf_sz);
+ }
+ }
+
+ if ( (bp = getBlockParam(param, "name"))
+ && (strcmp(bp->value, "mpd") != 0) ) {
+ name = xstrdup(bp->value);
+ DEBUG("name=%s\n", name);
+ }
+
+ return 0;
+}
+
+static int jack_testDefault(void)
+{
+ return 0;
+}
+
+static int connect_jack(AudioOutput *audioOutput)
+{
+ JackData *jd = audioOutput->data;
+ char **jports;
+ char *port_name;
+
+ if ( (jd->client = jack_client_new(name)) == NULL ) {
+ ERROR("jack server not running?\n");
+ freeJackData(audioOutput);
+ return -1;
+ }
+
+ jack_set_error_function(error_callback);
+ jack_set_process_callback(jd->client, process, (void *)jd);
+ jack_set_sample_rate_callback(jd->client, (JackProcessCallback)srate,
+ (void *)audioOutput);
+ jack_on_shutdown(jd->client, shutdown_callback, (void *)jd);
+
+ if ( jack_activate(jd->client) ) {
+ ERROR("cannot activate client");
+ freeJackData(audioOutput);
+ return -1;
+ }
+
+ jd->ports[0] = jack_port_register(jd->client, "left",
+ JACK_DEFAULT_AUDIO_TYPE,
+ JackPortIsOutput, 0);
+ if ( !jd->ports[0] ) {
+ ERROR("Cannot register left output port.\n");
+ freeJackData(audioOutput);
+ return -1;
+ }
+
+ jd->ports[1] = jack_port_register(jd->client, "right",
+ JACK_DEFAULT_AUDIO_TYPE,
+ JackPortIsOutput, 0);
+ if ( !jd->ports[1] ) {
+ ERROR("Cannot register right output port.\n");
+ freeJackData(audioOutput);
+ return -1;
+ }
+
+ /* hay que buscar que hay */
+ if ( !output_ports[1]
+ && (jports = (char **)jack_get_ports(jd->client, NULL, NULL,
+ JackPortIsPhysical|
+ JackPortIsInput)) ) {
+ output_ports[0] = jports[0];
+ output_ports[1] = jports[1] ? jports[1] : jports[0];
+ DEBUG("output_ports: %s %s\n", output_ports[0], output_ports[1]);
+ free(jports);
+ }
+
+ if ( output_ports[1] ) {
+ jd->ringbuffer[0] = jack_ringbuffer_create(ringbuf_sz);
+ jd->ringbuffer[1] = jack_ringbuffer_create(ringbuf_sz);
+ memset(jd->ringbuffer[0]->buf, 0, jd->ringbuffer[0]->size);
+ memset(jd->ringbuffer[1]->buf, 0, jd->ringbuffer[1]->size);
+
+ port_name = xmalloc(sizeof(char)*(7+strlen(name)));
+
+ sprintf(port_name, "%s:left", name);
+ if ( (jack_connect(jd->client, port_name,
+ output_ports[0])) != 0 ) {
+ ERROR("%s is not a valid Jack Client / Port ",
+ output_ports[0]);
+ freeJackData(audioOutput);
+ free(port_name);
+ return -1;
+ }
+ sprintf(port_name, "%s:right", name);
+ if ( (jack_connect(jd->client, port_name,
+ output_ports[1])) != 0 ) {
+ ERROR("%s is not a valid Jack Client / Port ",
+ output_ports[1]);
+ freeJackData(audioOutput);
+ free(port_name);
+ return -1;
+ }
+ free(port_name);
+ }
+
+ DEBUG("connect_jack (pid=%d)\n", getpid());
+ return 1;
+}
+
+static int jack_openDevice(AudioOutput *audioOutput)
+{
+ JackData *jd = audioOutput->data;
+
+ if ( !jd ) {
+ DEBUG("connect!\n");
+ jd = newJackData();
+ audioOutput->data = jd;
+
+ if (connect_jack(audioOutput) < 0) {
+ freeJackData(audioOutput);
+ audioOutput->open = 0;
+ return -1;
+ }
+ }
+
+ set_audioformat(audioOutput);
+ audioOutput->open = 1;
+
+ DEBUG("jack_openDevice (pid=%d)!\n", getpid ());
+ return 0;
+}
+
+
+static void jack_closeDevice(AudioOutput * audioOutput)
+{
+ /*jack_finishDriver(audioOutput);*/
+ audioOutput->open = 0;
+ DEBUG("jack_closeDevice (pid=%d)\n", getpid());
+}
+
+static void jack_dropBufferedAudio (AudioOutput * audioOutput)
+{
+}
+
+static int jack_playAudio(AudioOutput * audioOutput, char *buff, int size)
+{
+ JackData *jd = audioOutput->data;
+ size_t space;
+ int i;
+ short *buffer = (short *) buff;
+ jack_default_audio_sample_t sample;
+ size_t samples = size/4;
+
+ /*DEBUG("jack_playAudio: (pid=%d)!\n", getpid());*/
+
+ if ( jd->shutdown ) {
+ ERROR("Refusing to play, because there is no client thread.\n");
+ freeJackData(audioOutput);
+ audioOutput->open = 0;
+ return 0;
+ }
+
+ while ( samples && !jd->shutdown ) {
+
+ if ( (space = jack_ringbuffer_write_space(jd->ringbuffer[0]))
+ >= samples*sample_size ) {
+
+ /*space = MIN(space, samples*sample_size);*/
+ /*space = samples*sample_size;*/
+
+ /*for(i=0; i<space/sample_size; i++) {*/
+ for(i=0; i<samples; i++) {
+ sample = (jack_default_audio_sample_t) *(buffer++)/32768.0;
+
+ jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
+ sample_size);
+
+ sample = (jack_default_audio_sample_t) *(buffer++)/32768.0;
+
+ jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
+ sample_size);
+
+ /*samples--;*/
+ }
+ samples=0;
+
+ } else {
+ pthread_mutex_lock(&play_audio_lock);
+ pthread_cond_wait(&play_audio, &play_audio_lock);
+ pthread_mutex_unlock(&play_audio_lock);
+ }
+
+ }
+ return 0;
+}
+
+AudioOutputPlugin jackPlugin = {
+ "jack",
+ jack_testDefault,
+ jack_initDriver,
+ jack_finishDriver,
+ jack_openDevice,
+ jack_playAudio,
+ jack_dropBufferedAudio,
+ jack_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else /* HAVE JACK */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
+
+#endif /* HAVE_JACK */
diff --git a/trunk/src/audioOutputs/audioOutput_mvp.c b/trunk/src/audioOutputs/audioOutput_mvp.c
new file mode 100644
index 000000000..ea365c657
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_mvp.c
@@ -0,0 +1,284 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * Media MVP audio output based on code from MVPMC project:
+ * http://mvpmc.sourceforge.net/
+ *
+ * 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_MVP
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+
+typedef struct {
+ unsigned long dsp_status;
+ unsigned long stream_decode_type;
+ unsigned long sample_rate;
+ unsigned long bit_rate;
+ unsigned long raw[64 / sizeof(unsigned long)];
+} aud_status_t;
+
+#define MVP_SET_AUD_STOP _IOW('a',1,int)
+#define MVP_SET_AUD_PLAY _IOW('a',2,int)
+#define MVP_SET_AUD_PAUSE _IOW('a',3,int)
+#define MVP_SET_AUD_UNPAUSE _IOW('a',4,int)
+#define MVP_SET_AUD_SRC _IOW('a',5,int)
+#define MVP_SET_AUD_MUTE _IOW('a',6,int)
+#define MVP_SET_AUD_BYPASS _IOW('a',8,int)
+#define MVP_SET_AUD_CHANNEL _IOW('a',9,int)
+#define MVP_GET_AUD_STATUS _IOR('a',10,aud_status_t)
+#define MVP_SET_AUD_VOLUME _IOW('a',13,int)
+#define MVP_GET_AUD_VOLUME _IOR('a',14,int)
+#define MVP_SET_AUD_STREAMTYPE _IOW('a',15,int)
+#define MVP_SET_AUD_FORMAT _IOW('a',16,int)
+#define MVP_GET_AUD_SYNC _IOR('a',21,pts_sync_data_t*)
+#define MVP_SET_AUD_STC _IOW('a',22,long long int *)
+#define MVP_SET_AUD_SYNC _IOW('a',23,int)
+#define MVP_SET_AUD_END_STREAM _IOW('a',25,int)
+#define MVP_SET_AUD_RESET _IOW('a',26,int)
+#define MVP_SET_AUD_DAC_CLK _IOW('a',27,int)
+#define MVP_GET_AUD_REGS _IOW('a',28,aud_ctl_regs_t*)
+
+typedef struct _MvpData {
+ int fd;
+} MvpData;
+
+static int pcmfrequencies[][3] = {
+ {9, 8000, 32000},
+ {10, 11025, 44100},
+ {11, 12000, 48000},
+ {1, 16000, 32000},
+ {2, 22050, 44100},
+ {3, 24000, 48000},
+ {5, 32000, 32000},
+ {0, 44100, 44100},
+ {7, 48000, 48000},
+ {13, 64000, 32000},
+ {14, 88200, 44100},
+ {15, 96000, 48000}
+};
+
+static int numfrequencies = sizeof(pcmfrequencies) / 12;
+
+static int mvp_testDefault(void)
+{
+ int fd;
+
+ fd = open("/dev/adec_pcm", O_WRONLY);
+
+ if (fd) {
+ close(fd);
+ return 0;
+ }
+
+ WARNING("Error opening PCM device \"/dev/adec_pcm\": %s\n",
+ strerror(errno));
+
+ return -1;
+}
+
+static int mvp_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+ MvpData *md = xmalloc(sizeof(MvpData));
+ md->fd = -1;
+ audioOutput->data = md;
+
+ return 0;
+}
+
+static void mvp_finishDriver(AudioOutput * audioOutput)
+{
+ MvpData *md = audioOutput->data;
+ free(md);
+}
+
+static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
+ int big_endian, int bits)
+{
+ int iloop;
+ int mix[5];
+
+ if (channels == 1)
+ mix[0] = 1;
+ else if (channels == 2)
+ mix[0] = 0;
+ else
+ return -1;
+
+ /* 0,1=24bit(24) , 2,3=16bit */
+ if (bits == 16)
+ mix[1] = 2;
+ else if (bits == 24)
+ mix[1] = 0;
+ else
+ return -1;
+
+ mix[3] = 0; /* stream type? */
+
+ if (big_endian == 1)
+ mix[4] = 1;
+ else if (big_endian == 0)
+ mix[4] = 0;
+ else
+ return -1;
+
+ /*
+ * if there is an exact match for the frequency, use it.
+ */
+ for (iloop = 0; iloop < numfrequencies; iloop++) {
+ if (rate == pcmfrequencies[iloop][1]) {
+ mix[2] = pcmfrequencies[iloop][0];
+ break;
+ }
+ }
+
+ if (iloop >= numfrequencies) {
+ ERROR("Can not find suitable output frequency for %ld\n", rate);
+ return -1;
+ }
+
+ if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
+ ERROR("Can not set audio format\n");
+ return -1;
+ }
+
+ if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
+ ERROR("Can not set audio sync\n");
+ return -1;
+ }
+
+ if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
+ ERROR("Can not set audio play mode\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static int mvp_openDevice(AudioOutput * audioOutput)
+{
+ long long int stc = 0;
+ MvpData *md = audioOutput->data;
+ AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+ int mix[5] = { 0, 2, 7, 1, 0 };
+
+ if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
+ ERROR("Error opening /dev/adec_pcm: %s\n", strerror(errno));
+ return -1;
+ }
+ if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
+ ERROR("Error setting audio source: %s\n", strerror(errno));
+ return -1;
+ }
+ if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
+ ERROR("Error setting audio streamtype: %s\n", strerror(errno));
+ return -1;
+ }
+ if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
+ ERROR("Error setting audio format: %s\n", strerror(errno));
+ return -1;
+ }
+ ioctl(md->fd, MVP_SET_AUD_STC, &stc);
+ if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
+ ERROR("Error setting audio streamtype: %s\n", strerror(errno));
+ return -1;
+ }
+#ifdef WORDS_BIGENDIAN
+ mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 0,
+ audioFormat->bits);
+#else
+ mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 1,
+ audioFormat->bits);
+#endif
+ audioOutput->open = 1;
+ return 0;
+}
+
+static void mvp_closeDevice(AudioOutput * audioOutput)
+{
+ MvpData *md = audioOutput->data;
+ if (md->fd >= 0)
+ close(md->fd);
+ md->fd = -1;
+ audioOutput->open = 0;
+}
+
+static void mvp_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ MvpData *md = audioOutput->data;
+ if (md->fd >= 0) {
+ ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
+ close(md->fd);
+ md->fd = -1;
+ audioOutput->open = 0;
+ }
+}
+
+static int mvp_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+ MvpData *md = audioOutput->data;
+ int ret;
+
+ /* reopen the device since it was closed by dropBufferedAudio */
+ if (md->fd < 0)
+ mvp_openDevice(audioOutput);
+
+ while (size > 0) {
+ ret = write(md->fd, playChunk, size);
+ if (ret < 0) {
+ if (errno == EINTR)
+ continue;
+ ERROR("closing mvp PCM device due to write error: "
+ "%s\n", strerror(errno));
+ mvp_closeDevice(audioOutput);
+ return -1;
+ }
+ playChunk += ret;
+ size -= ret;
+ }
+ return 0;
+}
+
+AudioOutputPlugin mvpPlugin = {
+ "mvp",
+ mvp_testDefault,
+ mvp_initDriver,
+ mvp_finishDriver,
+ mvp_openDevice,
+ mvp_playAudio,
+ mvp_dropBufferedAudio,
+ mvp_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else /* HAVE_MVP */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
+#endif /* HAVE_MVP */
diff --git a/trunk/src/audioOutputs/audioOutput_oss.c b/trunk/src/audioOutputs/audioOutput_oss.c
new file mode 100644
index 000000000..01293cbd1
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_oss.c
@@ -0,0 +1,575 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * OSS audio output (c) 2004, 2005, 2006, 2007 by Eric Wong <eric@petta-tech.com>
+ * and Warren Dukes <warren.dukes@gmail.com>
+ *
+ * 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_OSS
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+
+#if defined(__OpenBSD__) || defined(__NetBSD__)
+# include <soundcard.h>
+#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+# include <sys/soundcard.h>
+#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+
+#ifdef WORDS_BIGENDIAN
+# define AFMT_S16_MPD AFMT_S16_BE
+#else
+# define AFMT_S16_MPD AFMT_S16_LE
+#endif /* WORDS_BIGENDIAN */
+
+typedef struct _OssData {
+ int fd;
+ const char *device;
+ int channels;
+ int sampleRate;
+ int bitFormat;
+ int bits;
+ int *supported[3];
+ int numSupported[3];
+ int *unsupported[3];
+ int numUnsupported[3];
+} OssData;
+
+#define OSS_SUPPORTED 1
+#define OSS_UNSUPPORTED 0
+#define OSS_UNKNOWN -1
+
+#define OSS_RATE 0
+#define OSS_CHANNELS 1
+#define OSS_BITS 2
+
+static int getIndexForParam(int param)
+{
+ int index = 0;
+
+ switch (param) {
+ case SNDCTL_DSP_SPEED:
+ index = OSS_RATE;
+ break;
+ case SNDCTL_DSP_CHANNELS:
+ index = OSS_CHANNELS;
+ break;
+ case SNDCTL_DSP_SAMPLESIZE:
+ index = OSS_BITS;
+ break;
+ }
+
+ return index;
+}
+
+static int findSupportedParam(OssData * od, int param, int val)
+{
+ int i;
+ int index = getIndexForParam(param);
+
+ for (i = 0; i < od->numSupported[index]; i++) {
+ if (od->supported[index][i] == val)
+ return 1;
+ }
+
+ return 0;
+}
+
+static int canConvert(int index, int val)
+{
+ switch (index) {
+ case OSS_BITS:
+ if (val != 16)
+ return 0;
+ break;
+ case OSS_CHANNELS:
+ if (val != 2)
+ return 0;
+ break;
+ }
+
+ return 1;
+}
+
+static int getSupportedParam(OssData * od, int param, int val)
+{
+ int i;
+ int index = getIndexForParam(param);
+ int ret = -1;
+ int least = val;
+ int diff;
+
+ for (i = 0; i < od->numSupported[index]; i++) {
+ diff = od->supported[index][i] - val;
+ if (diff < 0)
+ diff = -diff;
+ if (diff < least) {
+ if (!canConvert(index, od->supported[index][i])) {
+ continue;
+ }
+ least = diff;
+ ret = od->supported[index][i];
+ }
+ }
+
+ return ret;
+}
+
+static int findUnsupportedParam(OssData * od, int param, int val)
+{
+ int i;
+ int index = getIndexForParam(param);
+
+ for (i = 0; i < od->numUnsupported[index]; i++) {
+ if (od->unsupported[index][i] == val)
+ return 1;
+ }
+
+ return 0;
+}
+
+static void addSupportedParam(OssData * od, int param, int val)
+{
+ int index = getIndexForParam(param);
+
+ od->numSupported[index]++;
+ od->supported[index] = xrealloc(od->supported[index],
+ od->numSupported[index] * sizeof(int));
+ od->supported[index][od->numSupported[index] - 1] = val;
+}
+
+static void addUnsupportedParam(OssData * od, int param, int val)
+{
+ int index = getIndexForParam(param);
+
+ od->numUnsupported[index]++;
+ od->unsupported[index] = xrealloc(od->unsupported[index],
+ od->numUnsupported[index] *
+ sizeof(int));
+ od->unsupported[index][od->numUnsupported[index] - 1] = val;
+}
+
+static void removeSupportedParam(OssData * od, int param, int val)
+{
+ int i = 0;
+ int j = 0;
+ int index = getIndexForParam(param);
+
+ for (i = 0; i < od->numSupported[index] - 1; i++) {
+ if (od->supported[index][i] == val)
+ j = 1;
+ od->supported[index][i] = od->supported[index][i + j];
+ }
+
+ od->numSupported[index]--;
+ od->supported[index] = xrealloc(od->supported[index],
+ od->numSupported[index] * sizeof(int));
+}
+
+static void removeUnsupportedParam(OssData * od, int param, int val)
+{
+ int i = 0;
+ int j = 0;
+ int index = getIndexForParam(param);
+
+ for (i = 0; i < od->numUnsupported[index] - 1; i++) {
+ if (od->unsupported[index][i] == val)
+ j = 1;
+ od->unsupported[index][i] = od->unsupported[index][i + j];
+ }
+
+ od->numUnsupported[index]--;
+ od->unsupported[index] = xrealloc(od->unsupported[index],
+ od->numUnsupported[index] *
+ sizeof(int));
+}
+
+static int isSupportedParam(OssData * od, int param, int val)
+{
+ if (findSupportedParam(od, param, val))
+ return OSS_SUPPORTED;
+ if (findUnsupportedParam(od, param, val))
+ return OSS_UNSUPPORTED;
+ return OSS_UNKNOWN;
+}
+
+static void supportParam(OssData * od, int param, int val)
+{
+ int supported = isSupportedParam(od, param, val);
+
+ if (supported == OSS_SUPPORTED)
+ return;
+
+ if (supported == OSS_UNSUPPORTED) {
+ removeUnsupportedParam(od, param, val);
+ }
+
+ addSupportedParam(od, param, val);
+}
+
+static void unsupportParam(OssData * od, int param, int val)
+{
+ int supported = isSupportedParam(od, param, val);
+
+ if (supported == OSS_UNSUPPORTED)
+ return;
+
+ if (supported == OSS_SUPPORTED) {
+ removeSupportedParam(od, param, val);
+ }
+
+ addUnsupportedParam(od, param, val);
+}
+
+static OssData *newOssData(void)
+{
+ OssData *ret = xmalloc(sizeof(OssData));
+
+ ret->device = NULL;
+ ret->fd = -1;
+
+ ret->supported[OSS_RATE] = NULL;
+ ret->supported[OSS_CHANNELS] = NULL;
+ ret->supported[OSS_BITS] = NULL;
+ ret->unsupported[OSS_RATE] = NULL;
+ ret->unsupported[OSS_CHANNELS] = NULL;
+ ret->unsupported[OSS_BITS] = NULL;
+
+ ret->numSupported[OSS_RATE] = 0;
+ ret->numSupported[OSS_CHANNELS] = 0;
+ ret->numSupported[OSS_BITS] = 0;
+ ret->numUnsupported[OSS_RATE] = 0;
+ ret->numUnsupported[OSS_CHANNELS] = 0;
+ ret->numUnsupported[OSS_BITS] = 0;
+
+ supportParam(ret, SNDCTL_DSP_SPEED, 48000);
+ supportParam(ret, SNDCTL_DSP_SPEED, 44100);
+ supportParam(ret, SNDCTL_DSP_CHANNELS, 2);
+ supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16);
+
+ return ret;
+}
+
+static void freeOssData(OssData * od)
+{
+ if (od->supported[OSS_RATE])
+ free(od->supported[OSS_RATE]);
+ if (od->supported[OSS_CHANNELS])
+ free(od->supported[OSS_CHANNELS]);
+ if (od->supported[OSS_BITS])
+ free(od->supported[OSS_BITS]);
+ if (od->unsupported[OSS_RATE])
+ free(od->unsupported[OSS_RATE]);
+ if (od->unsupported[OSS_CHANNELS])
+ free(od->unsupported[OSS_CHANNELS]);
+ if (od->unsupported[OSS_BITS])
+ free(od->unsupported[OSS_BITS]);
+
+ free(od);
+}
+
+#define OSS_STAT_NO_ERROR 0
+#define OSS_STAT_NOT_CHAR_DEV -1
+#define OSS_STAT_NO_PERMS -2
+#define OSS_STAT_DOESN_T_EXIST -3
+#define OSS_STAT_OTHER -4
+
+static int oss_statDevice(const char *device, int *stErrno)
+{
+ struct stat st;
+
+ if (0 == stat(device, &st)) {
+ if (!S_ISCHR(st.st_mode)) {
+ return OSS_STAT_NOT_CHAR_DEV;
+ }
+ } else {
+ *stErrno = errno;
+
+ switch (errno) {
+ case ENOENT:
+ case ENOTDIR:
+ return OSS_STAT_DOESN_T_EXIST;
+ case EACCES:
+ return OSS_STAT_NO_PERMS;
+ default:
+ return OSS_STAT_OTHER;
+ }
+ }
+
+ return 0;
+}
+
+static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" };
+
+static int oss_testDefault(void)
+{
+ int fd, i;
+
+ for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+ if ((fd = open(default_devices[i], O_WRONLY)) >= 0) {
+ xclose(fd);
+ return 0;
+ }
+ WARNING("Error opening OSS device \"%s\": %s\n",
+ default_devices[i], strerror(errno));
+ }
+
+ return -1;
+}
+
+static int oss_open_default(AudioOutput *ao, ConfigParam *param, OssData *od)
+{
+ int i;
+ int err[ARRAY_SIZE(default_devices)];
+ int ret[ARRAY_SIZE(default_devices)];
+
+ for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+ ret[i] = oss_statDevice(default_devices[i], &err[i]);
+ if (ret[i] == 0) {
+ od->device = default_devices[i];
+ return 0;
+ }
+ }
+
+ if (param)
+ ERROR("error trying to open specified OSS device"
+ " at line %i\n", param->line);
+ else
+ ERROR("error trying to open default OSS device\n");
+
+ for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+ const char *dev = default_devices[i];
+ switch(ret[i]) {
+ case OSS_STAT_DOESN_T_EXIST:
+ ERROR("%s not found\n", dev);
+ break;
+ case OSS_STAT_NOT_CHAR_DEV:
+ ERROR("%s is not a character device\n", dev);
+ break;
+ case OSS_STAT_NO_PERMS:
+ ERROR("%s: permission denied\n", dev);
+ break;
+ default:
+ ERROR("Error accessing %s: %s", dev, strerror(err[i]));
+ }
+ }
+ exit(EXIT_FAILURE);
+ return 0; /* some compilers can be dumb... */
+}
+
+static int oss_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+ OssData *od = newOssData();
+ audioOutput->data = od;
+ if (param) {
+ BlockParam *bp = getBlockParam(param, "device");
+ if (bp) {
+ od->device = bp->value;
+ return 0;
+ }
+ }
+ return oss_open_default(audioOutput, param, od);
+}
+
+static void oss_finishDriver(AudioOutput * audioOutput)
+{
+ OssData *od = audioOutput->data;
+
+ freeOssData(od);
+}
+
+static int setParam(OssData * od, int param, int *value)
+{
+ int val = *value;
+ int copy;
+ int supported = isSupportedParam(od, param, val);
+
+ do {
+ if (supported == OSS_UNSUPPORTED) {
+ val = getSupportedParam(od, param, val);
+ if (copy < 0)
+ return -1;
+ }
+ copy = val;
+ if (ioctl(od->fd, param, &copy)) {
+ unsupportParam(od, param, val);
+ supported = OSS_UNSUPPORTED;
+ } else {
+ if (supported == OSS_UNKNOWN) {
+ supportParam(od, param, val);
+ supported = OSS_SUPPORTED;
+ }
+ val = copy;
+ }
+ } while (supported == OSS_UNSUPPORTED);
+
+ *value = val;
+
+ return 0;
+}
+
+static void oss_close(OssData * od)
+{
+ if (od->fd >= 0)
+ while (close(od->fd) && errno == EINTR) ;
+ od->fd = -1;
+}
+
+static int oss_open(AudioOutput * audioOutput)
+{
+ int tmp;
+ OssData *od = audioOutput->data;
+
+ if ((od->fd = open(od->device, O_WRONLY)) < 0) {
+ ERROR("Error opening OSS device \"%s\": %s\n", od->device,
+ strerror(errno));
+ goto fail;
+ }
+
+ if (setParam(od, SNDCTL_DSP_CHANNELS, &od->channels)) {
+ ERROR("OSS device \"%s\" does not support %i channels: %s\n",
+ od->device, od->channels, strerror(errno));
+ goto fail;
+ }
+
+ if (setParam(od, SNDCTL_DSP_SPEED, &od->sampleRate)) {
+ ERROR("OSS device \"%s\" does not support %i Hz audio: %s\n",
+ od->device, od->sampleRate, strerror(errno));
+ goto fail;
+ }
+
+ switch (od->bits) {
+ case 8:
+ tmp = AFMT_S8;
+ break;
+ case 16:
+ tmp = AFMT_S16_MPD;
+ }
+
+ if (setParam(od, SNDCTL_DSP_SAMPLESIZE, &tmp)) {
+ ERROR("OSS device \"%s\" does not support %i bit audio: %s\n",
+ od->device, tmp, strerror(errno));
+ goto fail;
+ }
+
+ audioOutput->open = 1;
+
+ return 0;
+
+fail:
+ oss_close(od);
+ audioOutput->open = 0;
+ return -1;
+}
+
+static int oss_openDevice(AudioOutput * audioOutput)
+{
+ int ret = -1;
+ OssData *od = audioOutput->data;
+ AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+
+ od->channels = audioFormat->channels;
+ od->sampleRate = audioFormat->sampleRate;
+ od->bits = audioFormat->bits;
+
+ if ((ret = oss_open(audioOutput)) < 0)
+ return ret;
+
+ audioFormat->channels = od->channels;
+ audioFormat->sampleRate = od->sampleRate;
+ audioFormat->bits = od->bits;
+
+ DEBUG("oss device \"%s\" will be playing %i bit %i channel audio at "
+ "%i Hz\n", od->device, od->bits, od->channels, od->sampleRate);
+
+ return ret;
+}
+
+static void oss_closeDevice(AudioOutput * audioOutput)
+{
+ OssData *od = audioOutput->data;
+
+ oss_close(od);
+
+ audioOutput->open = 0;
+}
+
+static void oss_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ OssData *od = audioOutput->data;
+
+ if (od->fd >= 0) {
+ ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+ oss_close(od);
+ }
+}
+
+static int oss_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+ OssData *od = audioOutput->data;
+ int ret;
+
+ /* reopen the device since it was closed by dropBufferedAudio */
+ if (od->fd < 0 && oss_open(audioOutput) < 0)
+ return -1;
+
+ while (size > 0) {
+ ret = write(od->fd, playChunk, size);
+ if (ret < 0) {
+ if (errno == EINTR)
+ continue;
+ ERROR("closing oss device \"%s\" due to write error: "
+ "%s\n", od->device, strerror(errno));
+ oss_closeDevice(audioOutput);
+ return -1;
+ }
+ playChunk += ret;
+ size -= ret;
+ }
+
+ return 0;
+}
+
+AudioOutputPlugin ossPlugin = {
+ "oss",
+ oss_testDefault,
+ oss_initDriver,
+ oss_finishDriver,
+ oss_openDevice,
+ oss_playAudio,
+ oss_dropBufferedAudio,
+ oss_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else /* HAVE OSS */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
+#endif /* HAVE_OSS */
diff --git a/trunk/src/audioOutputs/audioOutput_osx.c b/trunk/src/audioOutputs/audioOutput_osx.c
new file mode 100644
index 000000000..1caebade5
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_osx.c
@@ -0,0 +1,374 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * 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"
+
+#ifdef HAVE_OSX
+
+#include <AudioUnit/AudioUnit.h>
+#include <stdlib.h>
+#include <pthread.h>
+
+#include "../log.h"
+
+typedef struct _OsxData {
+ AudioUnit au;
+ pthread_mutex_t mutex;
+ pthread_cond_t condition;
+ char *buffer;
+ int bufferSize;
+ int pos;
+ int len;
+ int started;
+} OsxData;
+
+static OsxData *newOsxData()
+{
+ OsxData *ret = xmalloc(sizeof(OsxData));
+
+ pthread_mutex_init(&ret->mutex, NULL);
+ pthread_cond_init(&ret->condition, NULL);
+
+ ret->pos = 0;
+ ret->len = 0;
+ ret->started = 0;
+ ret->buffer = NULL;
+ ret->bufferSize = 0;
+
+ return ret;
+}
+
+static int osx_testDefault()
+{
+ /*AudioUnit au;
+ ComponentDescription desc;
+ Component comp;
+
+ desc.componentType = kAudioUnitType_Output;
+ desc.componentSubType = kAudioUnitSubType_Output;
+ desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+ desc.componentFlags = 0;
+ desc.componentFlagsMask = 0;
+
+ comp = FindNextComponent(NULL, &desc);
+ if(!comp) {
+ ERROR("Unable to open default OS X defice\n");
+ return -1;
+ }
+
+ if(OpenAComponent(comp, &au) != noErr) {
+ ERROR("Unable to open default OS X defice\n");
+ return -1;
+ }
+
+ CloseComponent(au); */
+
+ return 0;
+}
+
+static int osx_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+ OsxData *od = newOsxData();
+
+ audioOutput->data = od;
+
+ return 0;
+}
+
+static void freeOsxData(OsxData * od)
+{
+ if (od->buffer)
+ free(od->buffer);
+ pthread_mutex_destroy(&od->mutex);
+ pthread_cond_destroy(&od->condition);
+ free(od);
+}
+
+static void osx_finishDriver(AudioOutput * audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+ freeOsxData(od);
+}
+
+static void osx_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+
+ pthread_mutex_lock(&od->mutex);
+ od->len = 0;
+ pthread_mutex_unlock(&od->mutex);
+}
+
+static void osx_closeDevice(AudioOutput * audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+
+ pthread_mutex_lock(&od->mutex);
+ while (od->len) {
+ pthread_cond_wait(&od->condition, &od->mutex);
+ }
+ pthread_mutex_unlock(&od->mutex);
+
+ if (od->started) {
+ AudioOutputUnitStop(od->au);
+ od->started = 0;
+ }
+
+ CloseComponent(od->au);
+ AudioUnitUninitialize(od->au);
+
+ audioOutput->open = 0;
+}
+
+static OSStatus osx_render(void *vdata,
+ AudioUnitRenderActionFlags * ioActionFlags,
+ const AudioTimeStamp * inTimeStamp,
+ UInt32 inBusNumber, UInt32 inNumberFrames,
+ AudioBufferList * bufferList)
+{
+ OsxData *od = (OsxData *) vdata;
+ AudioBuffer *buffer = &bufferList->mBuffers[0];
+ int bufferSize = buffer->mDataByteSize;
+ int bytesToCopy;
+ int curpos = 0;
+
+ /*DEBUG("osx_render: enter : %i\n", (int)bufferList->mNumberBuffers);
+ DEBUG("osx_render: ioActionFlags: %p\n", ioActionFlags);
+ if(ioActionFlags) {
+ if(*ioActionFlags & kAudioUnitRenderAction_PreRender) {
+ DEBUG("prerender\n");
+ }
+ if(*ioActionFlags & kAudioUnitRenderAction_PostRender) {
+ DEBUG("post render\n");
+ }
+ if(*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence) {
+ DEBUG("post render\n");
+ }
+ if(*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight) {
+ DEBUG("prefilight\n");
+ }
+ if(*ioActionFlags & kAudioOfflineUnitRenderAction_Render) {
+ DEBUG("render\n");
+ }
+ if(*ioActionFlags & kAudioOfflineUnitRenderAction_Complete) {
+ DEBUG("complete\n");
+ }
+ } */
+
+ /* while(bufferSize) {
+ DEBUG("osx_render: lock\n"); */
+ pthread_mutex_lock(&od->mutex);
+ /*
+ DEBUG("%i:%i\n", bufferSize, od->len);
+ while(od->go && od->len < bufferSize &&
+ od->len < od->bufferSize)
+ {
+ DEBUG("osx_render: wait\n");
+ pthread_cond_wait(&od->condition, &od->mutex);
+ }
+ */
+
+ bytesToCopy = od->len < bufferSize ? od->len : bufferSize;
+ bufferSize = bytesToCopy;
+ od->len -= bytesToCopy;
+
+ if (od->pos + bytesToCopy > od->bufferSize) {
+ int bytes = od->bufferSize - od->pos;
+ memcpy(buffer->mData + curpos, od->buffer + od->pos, bytes);
+ od->pos = 0;
+ curpos += bytes;
+ bytesToCopy -= bytes;
+ }
+
+ memcpy(buffer->mData + curpos, od->buffer + od->pos, bytesToCopy);
+ od->pos += bytesToCopy;
+ curpos += bytesToCopy;
+
+ if (od->pos >= od->bufferSize)
+ od->pos = 0;
+ /* DEBUG("osx_render: unlock\n"); */
+ pthread_mutex_unlock(&od->mutex);
+ pthread_cond_signal(&od->condition);
+ /* } */
+
+ buffer->mDataByteSize = bufferSize;
+
+ if (!bufferSize) {
+ my_usleep(1000);
+ }
+
+ /* DEBUG("osx_render: leave\n"); */
+ return 0;
+}
+
+static int osx_openDevice(AudioOutput * audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+ ComponentDescription desc;
+ Component comp;
+ AURenderCallbackStruct callback;
+ AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+ AudioStreamBasicDescription streamDesc;
+
+ desc.componentType = kAudioUnitType_Output;
+ desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+ desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+ desc.componentFlags = 0;
+ desc.componentFlagsMask = 0;
+
+ comp = FindNextComponent(NULL, &desc);
+ if (comp == 0) {
+ ERROR("Error finding OS X component\n");
+ return -1;
+ }
+
+ if (OpenAComponent(comp, &od->au) != noErr) {
+ ERROR("Unable to open OS X component\n");
+ return -1;
+ }
+
+ if (AudioUnitInitialize(od->au) != 0) {
+ CloseComponent(od->au);
+ ERROR("Unable to initialize OS X audio unit\n");
+ return -1;
+ }
+
+ callback.inputProc = osx_render;
+ callback.inputProcRefCon = od;
+
+ if (AudioUnitSetProperty(od->au, kAudioUnitProperty_SetRenderCallback,
+ kAudioUnitScope_Input, 0,
+ &callback, sizeof(callback)) != 0) {
+ AudioUnitUninitialize(od->au);
+ CloseComponent(od->au);
+ ERROR("unable to set callback for OS X audio unit\n");
+ return -1;
+ }
+
+ streamDesc.mSampleRate = audioFormat->sampleRate;
+ streamDesc.mFormatID = kAudioFormatLinearPCM;
+ streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+#ifdef WORDS_BIGENDIAN
+ streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+
+ streamDesc.mBytesPerPacket =
+ audioFormat->channels * audioFormat->bits / 8;
+ streamDesc.mFramesPerPacket = 1;
+ streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket;
+ streamDesc.mChannelsPerFrame = audioFormat->channels;
+ streamDesc.mBitsPerChannel = audioFormat->bits;
+
+ if (AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
+ kAudioUnitScope_Input, 0,
+ &streamDesc, sizeof(streamDesc)) != 0) {
+ AudioUnitUninitialize(od->au);
+ CloseComponent(od->au);
+ ERROR("Unable to set format on OS X device\n");
+ return -1;
+ }
+
+ /* create a buffer of 1s */
+ od->bufferSize = (audioFormat->sampleRate) *
+ (audioFormat->bits >> 3) * (audioFormat->channels);
+ od->buffer = xrealloc(od->buffer, od->bufferSize);
+
+ od->pos = 0;
+ od->len = 0;
+
+ audioOutput->open = 1;
+
+ return 0;
+}
+
+static int osx_play(AudioOutput * audioOutput, char *playChunk, int size)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+ int bytesToCopy;
+ int curpos;
+
+ /* DEBUG("osx_play: enter\n"); */
+
+ if (!od->started) {
+ int err;
+ od->started = 1;
+ err = AudioOutputUnitStart(od->au);
+ if (err) {
+ ERROR("unable to start audio output: %i\n", err);
+ return -1;
+ }
+ }
+
+ pthread_mutex_lock(&od->mutex);
+
+ while (size) {
+ /* DEBUG("osx_play: lock\n"); */
+ curpos = od->pos + od->len;
+ if (curpos >= od->bufferSize)
+ curpos -= od->bufferSize;
+
+ bytesToCopy = od->bufferSize < size ? od->bufferSize : size;
+
+ while (od->len > od->bufferSize - bytesToCopy) {
+ /* DEBUG("osx_play: wait\n"); */
+ pthread_cond_wait(&od->condition, &od->mutex);
+ }
+
+ bytesToCopy = od->bufferSize - od->len;
+ bytesToCopy = bytesToCopy < size ? bytesToCopy : size;
+ size -= bytesToCopy;
+ od->len += bytesToCopy;
+
+ if (curpos + bytesToCopy > od->bufferSize) {
+ int bytes = od->bufferSize - curpos;
+ memcpy(od->buffer + curpos, playChunk, bytes);
+ curpos = 0;
+ playChunk += bytes;
+ bytesToCopy -= bytes;
+ }
+
+ memcpy(od->buffer + curpos, playChunk, bytesToCopy);
+ curpos += bytesToCopy;
+ playChunk += bytesToCopy;
+
+ }
+ /* DEBUG("osx_play: unlock\n"); */
+ pthread_mutex_unlock(&od->mutex);
+
+ /* DEBUG("osx_play: leave\n"); */
+ return 0;
+}
+
+AudioOutputPlugin osxPlugin = {
+ "osx",
+ osx_testDefault,
+ osx_initDriver,
+ osx_finishDriver,
+ osx_openDevice,
+ osx_play,
+ osx_dropBufferedAudio,
+ osx_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else
+
+#include <stdio.h>
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
+#endif
diff --git a/trunk/src/audioOutputs/audioOutput_pulse.c b/trunk/src/audioOutputs/audioOutput_pulse.c
new file mode 100644
index 000000000..8948e0263
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_pulse.c
@@ -0,0 +1,221 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * 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_PULSE
+
+#include "../conf.h"
+#include "../log.h"
+
+#include <string.h>
+#include <time.h>
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+
+#define MPD_PULSE_NAME "mpd"
+#define CONN_ATTEMPT_INTERVAL 60
+
+typedef struct _PulseData {
+ pa_simple *s;
+ char *server;
+ char *sink;
+ int connAttempts;
+ time_t lastAttempt;
+} PulseData;
+
+static PulseData *newPulseData(void)
+{
+ PulseData *ret;
+
+ ret = xmalloc(sizeof(PulseData));
+
+ ret->s = NULL;
+ ret->server = NULL;
+ ret->sink = NULL;
+ ret->connAttempts = 0;
+ ret->lastAttempt = 0;
+
+ return ret;
+}
+
+static void freePulseData(PulseData * pd)
+{
+ if (pd->server)
+ free(pd->server);
+ if (pd->sink)
+ free(pd->sink);
+ free(pd);
+}
+
+static int pulse_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+ BlockParam *server = NULL;
+ BlockParam *sink = NULL;
+ PulseData *pd;
+
+ if (param) {
+ server = getBlockParam(param, "server");
+ sink = getBlockParam(param, "sink");
+ }
+
+ pd = newPulseData();
+ pd->server = server ? xstrdup(server->value) : NULL;
+ pd->sink = sink ? xstrdup(sink->value) : NULL;
+ audioOutput->data = pd;
+
+ return 0;
+}
+
+static void pulse_finishDriver(AudioOutput * audioOutput)
+{
+ freePulseData((PulseData *) audioOutput->data);
+}
+
+static int pulse_testDefault(void)
+{
+ pa_simple *s;
+ pa_sample_spec ss;
+ int error;
+
+ ss.format = PA_SAMPLE_S16NE;
+ ss.rate = 44100;
+ ss.channels = 2;
+
+ s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
+ MPD_PULSE_NAME, &ss, NULL, NULL, &error);
+ if (!s) {
+ WARNING("Cannot connect to default PulseAudio server: %s\n",
+ pa_strerror(error));
+ return -1;
+ }
+
+ pa_simple_free(s);
+
+ return 0;
+}
+
+static int pulse_openDevice(AudioOutput * audioOutput)
+{
+ PulseData *pd;
+ AudioFormat *audioFormat;
+ pa_sample_spec ss;
+ time_t t;
+ int error;
+
+ t = time(NULL);
+ pd = audioOutput->data;
+ audioFormat = &audioOutput->outAudioFormat;
+
+ if (pd->connAttempts != 0 &&
+ (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
+ return -1;
+
+ pd->connAttempts++;
+ pd->lastAttempt = t;
+
+ if (audioFormat->bits != 16) {
+ ERROR("PulseAudio doesn't support %i bit audio\n",
+ audioFormat->bits);
+ return -1;
+ }
+
+ ss.format = PA_SAMPLE_S16NE;
+ ss.rate = audioFormat->sampleRate;
+ ss.channels = audioFormat->channels;
+
+ pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
+ pd->sink, audioOutput->name, &ss, NULL, NULL,
+ &error);
+ if (!pd->s) {
+ ERROR("Cannot connect to server in PulseAudio output "
+ "\"%s\" (attempt %i): %s\n", audioOutput->name,
+ pd->connAttempts, pa_strerror(error));
+ return -1;
+ }
+
+ pd->connAttempts = 0;
+ audioOutput->open = 1;
+
+ DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
+ "channel audio at %i Hz\n", audioOutput->name, audioFormat->bits,
+ audioFormat->channels, audioFormat->sampleRate);
+
+ return 0;
+}
+
+static void pulse_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ PulseData *pd;
+ int error;
+
+ pd = audioOutput->data;
+ if (pa_simple_flush(pd->s, &error) < 0)
+ WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
+ audioOutput->name, pa_strerror(error));
+}
+
+static void pulse_closeDevice(AudioOutput * audioOutput)
+{
+ PulseData *pd;
+
+ pd = audioOutput->data;
+ if (pd->s) {
+ pa_simple_drain(pd->s, NULL);
+ pa_simple_free(pd->s);
+ }
+
+ audioOutput->open = 0;
+}
+
+static int pulse_playAudio(AudioOutput * audioOutput, char *playChunk, int size)
+{
+ PulseData *pd;
+ int error;
+
+ pd = audioOutput->data;
+
+ if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
+ ERROR("PulseAudio output \"%s\" disconnecting due to write "
+ "error: %s\n", audioOutput->name, pa_strerror(error));
+ pulse_closeDevice(audioOutput);
+ return -1;
+ }
+
+ return 0;
+}
+
+AudioOutputPlugin pulsePlugin = {
+ "pulse",
+ pulse_testDefault,
+ pulse_initDriver,
+ pulse_finishDriver,
+ pulse_openDevice,
+ pulse_playAudio,
+ pulse_dropBufferedAudio,
+ pulse_closeDevice,
+ NULL, /* sendMetadataFunc */
+};
+
+#else /* HAVE_PULSE */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
+#endif /* HAVE_PULSE */
diff --git a/trunk/src/audioOutputs/audioOutput_shout.c b/trunk/src/audioOutputs/audioOutput_shout.c
new file mode 100644
index 000000000..7d93f8f85
--- /dev/null
+++ b/trunk/src/audioOutputs/audioOutput_shout.c
@@ -0,0 +1,636 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * 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_SHOUT
+
+#include "../conf.h"
+#include "../log.h"
+#include "../pcm_utils.h"
+
+#include <string.h>
+#include <time.h>
+
+#include <shout/shout.h>
+#include <vorbis/vorbisenc.h>
+
+#define CONN_ATTEMPT_INTERVAL 60
+
+static int shoutInitCount;
+
+/* lots of this code blatantly stolent from bossogg/bossao2 */
+
+typedef struct _ShoutData {
+ shout_t *shoutConn;
+ int shoutError;
+
+ ogg_stream_state os;
+ ogg_page og;
+ ogg_packet op;
+ ogg_packet header_main;
+ ogg_packet header_comments;
+ ogg_packet header_codebooks;
+
+ vorbis_dsp_state vd;
+ vorbis_block vb;
+ vorbis_info vi;
+ vorbis_comment vc;
+
+ float quality;
+ int bitrate;
+
+ int opened;
+
+ MpdTag *tag;
+ int tagToSend;
+
+ int connAttempts;
+ time_t lastAttempt;
+ int last_err;
+
+ /* just a pointer to audioOutput->outAudioFormat */
+ AudioFormat *audioFormat;
+} ShoutData;
+
+static ShoutData *newShoutData(void)
+{
+ ShoutData *ret = xmalloc(sizeof(ShoutData));
+
+ ret->shoutConn = shout_new();
+ ret->opened = 0;
+ ret->tag = NULL;
+ ret->tagToSend = 0;
+ ret->bitrate = -1;
+ ret->quality = -2.0;
+ ret->connAttempts = 0;
+ ret->lastAttempt = 0;
+ ret->audioFormat = NULL;
+ ret->last_err = SHOUTERR_UNCONNECTED;
+
+ return ret;
+}
+
+static void freeShoutData(ShoutData * sd)
+{
+ if (sd->shoutConn)
+ shout_free(sd->shoutConn);
+ if (sd->tag)
+ freeMpdTag(sd->tag);
+
+ free(sd);
+}
+
+#define checkBlockParam(name) { \
+ blockParam = getBlockParam(param, name); \
+ if (!blockParam) { \
+ FATAL("no \"%s\" defined for shout device defined at line " \
+ "%i\n", name, param->line); \
+ } \
+}
+
+static int myShout_initDriver(AudioOutput * audioOutput, ConfigParam * param)
+{
+ ShoutData *sd;
+ char *test;
+ int port;
+ char *host;
+ char *mount;
+ char *passwd;
+ char *user;
+ char *name;
+ BlockParam *blockParam;
+ unsigned int public = 0;
+
+ sd = newShoutData();
+
+ if (shoutInitCount == 0)
+ shout_init();
+
+ shoutInitCount++;
+
+ checkBlockParam("host");
+ host = blockParam->value;
+
+ checkBlockParam("mount");
+ mount = blockParam->value;
+
+ checkBlockParam("port");
+
+ port = strtol(blockParam->value, &test, 10);
+
+ if (*test != '\0' || port <= 0) {
+ FATAL("shout port \"%s\" is not a positive integer, line %i\n",
+ blockParam->value, blockParam->line);
+ }
+
+ checkBlockParam("password");
+ passwd = blockParam->value;
+
+ checkBlockParam("name");
+ name = blockParam->value;
+
+ blockParam = getBlockParam(param, "public");
+ if (blockParam) {
+ if (0 == strcmp(blockParam->value, "yes")) {
+ public = 1;
+ } else if (0 == strcmp(blockParam->value, "no")) {
+ public = 0;
+ } else {
+ FATAL("public \"%s\" is not \"yes\" or \"no\" at line "
+ "%i\n", param->value, param->line);
+ }
+ }
+
+ blockParam = getBlockParam(param, "user");
+ if (blockParam)
+ user = blockParam->value;
+ else
+ user = "source";
+
+ blockParam = getBlockParam(param, "quality");
+
+ if (blockParam) {
+ int line = blockParam->line;
+
+ sd->quality = strtod(blockParam->value, &test);
+
+ if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
+ FATAL("shout quality \"%s\" is not a number in the "
+ "range -1 to 10, line %i\n", blockParam->value,
+ blockParam->line);
+ }
+
+ blockParam = getBlockParam(param, "bitrate");
+
+ if (blockParam) {
+ FATAL("quality (line %i) and bitrate (line %i) are "
+ "both defined for shout output\n", line,
+ blockParam->line);
+ }
+ } else {
+ blockParam = getBlockParam(param, "bitrate");
+
+ if (!blockParam) {
+ FATAL("neither bitrate nor quality defined for shout "
+ "output at line %i\n", param->line);
+ }
+
+ sd->bitrate = strtol(blockParam->value, &test, 10);
+
+ if (*test != '\0' || sd->bitrate <= 0) {
+ FATAL("bitrate at line %i should be a positive integer "
+ "\n", blockParam->line);
+ }
+ }
+
+ checkBlockParam("format");
+ sd->audioFormat = &audioOutput->outAudioFormat;
+
+ if (shout_set_host(sd->shoutConn, host) != SHOUTERR_SUCCESS ||
+ shout_set_port(sd->shoutConn, port) != SHOUTERR_SUCCESS ||
+ shout_set_password(sd->shoutConn, passwd) != SHOUTERR_SUCCESS ||
+ shout_set_mount(sd->shoutConn, mount) != SHOUTERR_SUCCESS ||
+ shout_set_name(sd->shoutConn, name) != SHOUTERR_SUCCESS ||
+ shout_set_user(sd->shoutConn, user) != SHOUTERR_SUCCESS ||
+ shout_set_public(sd->shoutConn, public) != SHOUTERR_SUCCESS ||
+ shout_set_nonblocking(sd->shoutConn, 1) != SHOUTERR_SUCCESS ||
+ shout_set_format(sd->shoutConn, SHOUT_FORMAT_VORBIS)
+ != SHOUTERR_SUCCESS ||
+ shout_set_protocol(sd->shoutConn, SHOUT_PROTOCOL_HTTP)
+ != SHOUTERR_SUCCESS ||
+ shout_set_agent(sd->shoutConn, "MPD") != SHOUTERR_SUCCESS) {
+ FATAL("error configuring shout defined at line %i: %s\n",
+ param->line, shout_get_error(sd->shoutConn));
+ }
+
+ /* optional paramters */
+ blockParam = getBlockParam(param, "genre");
+ if (blockParam && shout_set_genre(sd->shoutConn, blockParam->value)) {
+ FATAL("error configuring shout defined at line %i: %s\n",
+ param->line, shout_get_error(sd->shoutConn));
+ }
+
+ blockParam = getBlockParam(param, "description");
+ if (blockParam && shout_set_description(sd->shoutConn,
+ blockParam->value)) {
+ FATAL("error configuring shout defined at line %i: %s\n",
+ param->line, shout_get_error(sd->shoutConn));
+ }
+
+ {
+ char temp[11];
+ memset(temp, 0, sizeof(temp));
+
+ snprintf(temp, sizeof(temp), "%d", sd->audioFormat->channels);
+ shout_set_audio_info(sd->shoutConn, SHOUT_AI_CHANNELS, temp);
+
+ snprintf(temp, sizeof(temp), "%d", sd->audioFormat->sampleRate);
+
+ shout_set_audio_info(sd->shoutConn, SHOUT_AI_SAMPLERATE, temp);
+
+ if (sd->quality >= -1.0) {
+ snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
+ shout_set_audio_info(sd->shoutConn, SHOUT_AI_QUALITY,
+ temp);
+ } else {
+ snprintf(temp, sizeof(temp), "%d", sd->bitrate);
+ shout_set_audio_info(sd->shoutConn, SHOUT_AI_BITRATE,
+ temp);
+ }
+ }
+
+ audioOutput->data = sd;
+
+ return 0;
+}
+
+static int myShout_handleError(ShoutData * sd, int err)
+{
+ switch (err) {
+ case SHOUTERR_SUCCESS:
+ break;
+ case SHOUTERR_UNCONNECTED:
+ case SHOUTERR_SOCKET:
+ ERROR("Lost shout connection to %s:%i : %s\n",
+ shout_get_host(sd->shoutConn),
+ shout_get_port(sd->shoutConn),
+ shout_get_error(sd->shoutConn));
+ sd->shoutError = 1;
+ return -1;
+ default:
+ ERROR("shout: connection to %s:%i error : %s\n",
+ shout_get_host(sd->shoutConn),
+ shout_get_port(sd->shoutConn),
+ shout_get_error(sd->shoutConn));
+ sd->shoutError = 1;
+ return -1;
+ }
+
+ return 0;
+}
+
+static int write_page(ShoutData * sd)
+{
+ int err = 0;
+
+ /*DEBUG("shout_delay: %i\n", shout_delay(sd->shoutConn)); */
+ shout_sync(sd->shoutConn);
+ err = shout_send(sd->shoutConn, sd->og.header, sd->og.header_len);
+ if (myShout_handleError(sd, err) < 0)
+ return -1;
+ err = shout_send(sd->shoutConn, sd->og.body, sd->og.body_len);
+ if (myShout_handleError(sd, err) < 0)
+ return -1;
+
+ return 0;
+}
+
+static void finishEncoder(ShoutData * sd)
+{
+ vorbis_analysis_wrote(&sd->vd, 0);
+
+ while (vorbis_analysis_blockout(&sd->vd, &sd->vb) == 1) {
+ vorbis_analysis(&sd->vb, NULL);
+ vorbis_bitrate_addblock(&sd->vb);
+ while (vorbis_bitrate_flushpacket(&sd->vd, &sd->op)) {
+ ogg_stream_packetin(&sd->os, &sd->op);
+ }
+ }
+}
+
+static int flushEncoder(ShoutData * sd)
+{
+ return (ogg_stream_pageout(&sd->os, &sd->og) > 0);
+}
+
+static void clearEncoder(ShoutData * sd)
+{
+ finishEncoder(sd);
+ while (1 == flushEncoder(sd)) {
+ if (!sd->shoutError)
+ write_page(sd);
+ }
+
+ vorbis_comment_clear(&sd->vc);
+ ogg_stream_clear(&sd->os);
+ vorbis_block_clear(&sd->vb);
+ vorbis_dsp_clear(&sd->vd);
+ vorbis_info_clear(&sd->vi);
+}
+
+static void myShout_closeShoutConn(ShoutData * sd)
+{
+ if (sd->opened) {
+ clearEncoder(sd);
+
+ if (shout_close(sd->shoutConn) != SHOUTERR_SUCCESS) {
+ ERROR("problem closing connection to shout server: "
+ "%s\n", shout_get_error(sd->shoutConn));
+ }
+ }
+
+ sd->last_err = SHOUTERR_UNCONNECTED;
+ sd->opened = 0;
+}
+
+static void myShout_finishDriver(AudioOutput * audioOutput)
+{
+ ShoutData *sd = (ShoutData *) audioOutput->data;
+
+ myShout_closeShoutConn(sd);
+
+ freeShoutData(sd);
+
+ shoutInitCount--;
+
+ if (shoutInitCount == 0)
+ shout_shutdown();
+}
+
+static void myShout_dropBufferedAudio(AudioOutput * audioOutput)
+{
+ /* needs to be implemented */
+}
+
+static void myShout_closeDevice(AudioOutput * audioOutput)
+{
+ ShoutData *sd = (ShoutData *) audioOutput->data;
+
+ myShout_closeShoutConn(sd);
+
+ audioOutput->open = 0;
+}
+
+#define addTag(name, value) { \
+ if(value) vorbis_comment_add_tag(&(sd->vc), name, value); \
+}
+
+static void copyTagToVorbisComment(ShoutData * sd)
+{
+ if (sd->tag) {
+ int i;
+
+ for (i = 0; i < sd->tag->numOfItems; i++) {
+ switch (sd->tag->items[i].type) {
+ case TAG_ITEM_ARTIST:
+ addTag("ARTIST", sd->tag->items[i].value);
+ break;
+ case TAG_ITEM_ALBUM:
+ addTag("ALBUM", sd->tag->items[i].value);
+ break;
+ case TAG_ITEM_TITLE:
+ addTag("TITLE", sd->tag->items[i].value);
+ break;
+ }
+ }
+ }
+}
+
+static int initEncoder(ShoutData * sd)
+{
+ vorbis_info_init(&(sd->vi));
+
+ if (sd->quality >= -1.0) {
+ if (0 != vorbis_encode_init_vbr(&(sd->vi),
+ sd->audioFormat->channels,
+ sd->audioFormat->sampleRate,
+ sd->quality * 0.1)) {
+ ERROR("problem setting up vorbis encoder for shout\n");
+ vorbis_info_clear(&(sd->vi));
+ return -1;
+ }
+ } else {
+ if (0 != vorbis_encode_init(&(sd->vi),
+ sd->audioFormat->channels,
+ sd->audioFormat->sampleRate, -1.0,
+ sd->bitrate * 1000, -1.0)) {
+ ERROR("problem setting up vorbis encoder for shout\n");
+ vorbis_info_clear(&(sd->vi));
+ return -1;
+ }
+ }
+
+ vorbis_analysis_init(&(sd->vd), &(sd->vi));
+ vorbis_block_init(&(sd->vd), &(sd->vb));
+
+ ogg_stream_init(&(sd->os), rand());
+
+ vorbis_comment_init(&(sd->vc));
+
+ return 0;
+}
+
+static int myShout_openShoutConn(AudioOutput * audioOutput)
+{
+ ShoutData *sd = (ShoutData *) audioOutput->data;
+ time_t t = time(NULL);
+
+ if (sd->connAttempts != 0 &&
+ (t - sd->lastAttempt) < CONN_ATTEMPT_INTERVAL) {
+ return -1;
+ }
+
+ sd->connAttempts++;
+
+ if (sd->last_err == SHOUTERR_UNCONNECTED)
+ sd->last_err = shout_open(sd->shoutConn);
+ switch (sd->last_err) {
+ case SHOUTERR_SUCCESS:
+ case SHOUTERR_CONNECTED:
+ break;
+ case SHOUTERR_BUSY:
+ sd->last_err = shout_get_connected(sd->shoutConn);
+ if (sd->last_err == SHOUTERR_CONNECTED)
+ break;
+ return -1;
+ default:
+ sd->lastAttempt = t;
+ ERROR("problem opening connection to shout server %s:%i "
+ "(attempt %i): %s\n",
+ shout_get_host(sd->shoutConn),
+ shout_get_port(sd->shoutConn),
+ sd->connAttempts, shout_get_error(sd->shoutConn));
+ return -1;
+ }
+
+ if (initEncoder(sd) < 0) {
+ shout_close(sd->shoutConn);
+ return -1;
+ }
+
+ sd->shoutError = 0;
+
+ copyTagToVorbisComment(sd);
+
+ vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
+ &(sd->header_comments),
+ &(sd->header_codebooks));
+
+ ogg_stream_packetin(&(sd->os), &(sd->header_main));
+ ogg_stream_packetin(&(sd->os), &(sd->header_comments));
+ ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
+
+ sd->opened = 1;
+ sd->tagToSend = 0;
+
+ while (ogg_stream_flush(&(sd->os), &(sd->og))) {
+ if (write_page(sd) < 0) {
+ myShout_closeShoutConn(sd);
+ return -1;
+ }
+ }
+
+ sd->connAttempts = 0;
+
+ return 0;
+}
+
+static int myShout_openDevice(AudioOutput * audioOutput)
+{
+ ShoutData *sd = (ShoutData *) audioOutput->data;
+
+ audioOutput->open = 1;
+
+ if (sd->opened)
+ return 0;
+
+ if (myShout_openShoutConn(audioOutput) < 0) {
+ audioOutput->open = 0;
+ return -1;
+ }
+
+ return 0;
+}
+
+static void myShout_sendMetadata(ShoutData * sd)
+{
+ if (!sd->opened || !sd->tag)
+ return;
+
+ clearEncoder(sd);
+ if (initEncoder(sd) < 0)
+ return;
+
+ copyTagToVorbisComment(sd);
+
+ vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
+ &(sd->header_comments),
+ &(sd->header_codebooks));
+
+ ogg_stream_packetin(&(sd->os), &(sd->header_main));
+ ogg_stream_packetin(&(sd->os), &(sd->header_comments));
+ ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
+
+ while (ogg_stream_flush(&(sd->os), &(sd->og))) {
+ if (write_page(sd) < 0) {
+ myShout_closeShoutConn(sd);
+ return;
+ }
+ }
+
+ /*if(sd->tag) freeMpdTag(sd->tag);
+ sd->tag = NULL; */
+ sd->tagToSend = 0;
+}
+
+static int myShout_play(AudioOutput * audioOutput, char *playChunk, int size)
+{
+ int i, j;
+ ShoutData *sd = (ShoutData *) audioOutput->data;
+ float **vorbbuf;
+ int samples;
+ int bytes = sd->audioFormat->bits / 8;
+
+ if (sd->opened && sd->tagToSend)
+ myShout_sendMetadata(sd);
+
+ if (!sd->opened) {
+ if (myShout_openShoutConn(audioOutput) < 0) {
+ return -1;
+ }
+ }
+
+ samples = size / (bytes * sd->audioFormat->channels);
+
+ /* this is for only 16-bit audio */
+
+ vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples);
+
+ for (i = 0; i < samples; i++) {
+ for (j = 0; j < sd->audioFormat->channels; j++) {
+ vorbbuf[j][i] = (*((mpd_sint16 *) playChunk)) / 32768.0;
+ playChunk += bytes;
+ }
+ }
+
+ vorbis_analysis_wrote(&(sd->vd), samples);
+
+ while (1 == vorbis_analysis_blockout(&(sd->vd), &(sd->vb))) {
+ vorbis_analysis(&(sd->vb), NULL);
+ vorbis_bitrate_addblock(&(sd->vb));
+
+ while (vorbis_bitrate_flushpacket(&(sd->vd), &(sd->op))) {
+ ogg_stream_packetin(&(sd->os), &(sd->op));
+ }
+ }
+
+ while (ogg_stream_pageout(&(sd->os), &(sd->og)) != 0) {
+ if (write_page(sd) < 0) {
+ myShout_closeShoutConn(sd);
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+static void myShout_setTag(AudioOutput * audioOutput, MpdTag * tag)
+{
+ ShoutData *sd = (ShoutData *) audioOutput->data;
+
+ if (sd->tag)
+ freeMpdTag(sd->tag);
+ sd->tag = NULL;
+ sd->tagToSend = 0;
+
+ if (!tag)
+ return;
+
+ sd->tag = mpdTagDup(tag);
+ sd->tagToSend = 1;
+}
+
+AudioOutputPlugin shoutPlugin = {
+ "shout",
+ NULL,
+ myShout_initDriver,
+ myShout_finishDriver,
+ myShout_openDevice,
+ myShout_play,
+ myShout_dropBufferedAudio,
+ myShout_closeDevice,
+ myShout_setTag,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
+#endif
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-23 04:33:47 +0000