/* GStreamer * Copyright (C) <2005> Arwed v. Merkatz * * Roughly based on the gstreamer 0.8 esdsink plugin: * Copyright (C) <2001> Richard Boulton * * esdsink.c: an EsounD audio sink * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-esdsink * @see_also: #GstAlsaSink, #GstAutoAudioSink * * * * This element outputs sound to an already-running Enlightened Sound Daemon * (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned * through this element (regardless of the system configuration), since this * is actively prevented by the element. If you must use esd, you need to * make sure it is started automatically with your session or otherwise. * * * TODO: insert some comments about how sucky esd is and that all the cool * kids use pulseaudio or whatever these days. * * * Simple example pipeline that plays an Ogg/Vorbis file via esd: * * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! esdsink * * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "esdsink.h" #include #include #include #include /* wtay: from my esd.h (debian unstable libesd0-dev 0.2.36-3) */ #ifndef ESD_MAX_WRITE_SIZE #define ESD_MAX_WRITE_SIZE (21 * 4096) #endif GST_DEBUG_CATEGORY_EXTERN (esd_debug); #define GST_CAT_DEFAULT esd_debug /* elementfactory information */ static const GstElementDetails esdsink_details = GST_ELEMENT_DETAILS ("Esound audio sink", "Sink/Audio", "Plays audio to an esound server", "Arwed von Merkatz "); enum { PROP_0, PROP_HOST }; static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) BYTE_ORDER, " "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; " "audio/x-raw-int, " "signed = (boolean) { true, false }, " "width = (int) 8, " "depth = (int) 8, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]") ); static void gst_esdsink_finalize (GObject * object); static GstCaps *gst_esdsink_getcaps (GstBaseSink * bsink); static gboolean gst_esdsink_open (GstAudioSink * asink); static gboolean gst_esdsink_close (GstAudioSink * asink); static gboolean gst_esdsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec); static gboolean gst_esdsink_unprepare (GstAudioSink * asink); static guint gst_esdsink_write (GstAudioSink * asink, gpointer data, guint length); static guint gst_esdsink_delay (GstAudioSink * asink); static void gst_esdsink_reset (GstAudioSink * asink); static void gst_esdsink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_esdsink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); GST_BOILERPLATE (GstEsdSink, gst_esdsink, GstAudioSink, GST_TYPE_AUDIO_SINK); static void gst_esdsink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_factory)); gst_element_class_set_details (element_class, &esdsink_details); } static void gst_esdsink_class_init (GstEsdSinkClass * klass) { GObjectClass *gobject_class; GstBaseSinkClass *gstbasesink_class; GstBaseAudioSinkClass *gstbaseaudiosink_class; GstAudioSinkClass *gstaudiosink_class; gobject_class = (GObjectClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; gstaudiosink_class = (GstAudioSinkClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_esdsink_finalize; gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_esdsink_getcaps); gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_esdsink_open); gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_esdsink_close); gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_esdsink_prepare); gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_esdsink_unprepare); gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_esdsink_write); gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_esdsink_delay); gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_esdsink_reset); gobject_class->set_property = gst_esdsink_set_property; gobject_class->get_property = gst_esdsink_get_property; /* default value is filled in the _init method */ g_object_class_install_property (gobject_class, PROP_HOST, g_param_spec_string ("host", "Host", "The host running the esound daemon", NULL, G_PARAM_READWRITE)); } static void gst_esdsink_init (GstEsdSink * esdsink, GstEsdSinkClass * klass) { esdsink->fd = -1; esdsink->ctrl_fd = -1; esdsink->host = g_strdup (g_getenv ("ESPEAKER")); } static void gst_esdsink_finalize (GObject * object) { GstEsdSink *esdsink = GST_ESDSINK (object); gst_caps_replace (&esdsink->cur_caps, NULL); g_free (esdsink->host); G_OBJECT_CLASS (parent_class)->finalize (object); } static GstCaps * gst_esdsink_getcaps (GstBaseSink * bsink) { GstEsdSink *esdsink; esdsink = GST_ESDSINK (bsink); /* no fd, we're done with the template caps */ if (esdsink->ctrl_fd < 0 || esdsink->cur_caps == NULL) { GST_LOG_OBJECT (esdsink, "getcaps called, returning template caps"); return NULL; } GST_LOG_OBJECT (esdsink, "returning %" GST_PTR_FORMAT, esdsink->cur_caps); return gst_caps_ref (esdsink->cur_caps); } static gboolean gst_esdsink_open (GstAudioSink * asink) { esd_server_info_t *server_info; GstPadTemplate *pad_template; GstEsdSink *esdsink; gchar *saved_env; gint i; esdsink = GST_ESDSINK (asink); GST_DEBUG_OBJECT (esdsink, "open"); /* ensure libesd doesn't auto-spawn a sound daemon if none is running yet */ saved_env = g_strdup (g_getenv ("ESD_NO_SPAWN")); g_setenv ("ESD_NO_SPAWN", "1", TRUE); /* now try to connect to any existing/running sound daemons */ esdsink->ctrl_fd = esd_open_sound (esdsink->host); /* and restore the previous state */ if (saved_env != NULL) { g_setenv ("ESD_NO_SPAWN", saved_env, TRUE); } else { g_unsetenv ("ESD_NO_SPAWN"); } g_free (saved_env); if (esdsink->ctrl_fd < 0) goto couldnt_connect; /* get server info */ server_info = esd_get_server_info (esdsink->ctrl_fd); if (!server_info) goto no_server_info; GST_INFO_OBJECT (esdsink, "got server info rate: %i", server_info->rate); pad_template = gst_static_pad_template_get (&sink_factory); esdsink->cur_caps = gst_caps_copy (gst_pad_template_get_caps (pad_template)); for (i = 0; i < esdsink->cur_caps->structs->len; i++) { GstStructure *s; s = gst_caps_get_structure (esdsink->cur_caps, i); gst_structure_set (s, "rate", G_TYPE_INT, server_info->rate, NULL); } esd_free_server_info (server_info); GST_INFO_OBJECT (esdsink, "server caps: %" GST_PTR_FORMAT, esdsink->cur_caps); return TRUE; /* ERRORS */ couldnt_connect: { GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE, (_("Could not establish connection to sound server")), ("can't open connection to esound server")); return FALSE; } no_server_info: { GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE, (_("Failed to query sound server capabilities")), ("couldn't get server info!")); return FALSE; } } static gboolean gst_esdsink_close (GstAudioSink * asink) { GstEsdSink *esdsink = GST_ESDSINK (asink); GST_DEBUG_OBJECT (esdsink, "close"); gst_caps_replace (&esdsink->cur_caps, NULL); esd_close (esdsink->ctrl_fd); esdsink->ctrl_fd = -1; return TRUE; } static gboolean gst_esdsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) { GstEsdSink *esdsink = GST_ESDSINK (asink); esd_format_t esdformat; /* Name used by esound for this connection. */ const char connname[] = "GStreamer"; GST_DEBUG_OBJECT (esdsink, "prepare"); /* Bitmap describing audio format. */ esdformat = ESD_STREAM | ESD_PLAY; switch (spec->depth) { case 8: esdformat |= ESD_BITS8; break; case 16: esdformat |= ESD_BITS16; break; default: goto unsupported_depth; } switch (spec->channels) { case 1: esdformat |= ESD_MONO; break; case 2: esdformat |= ESD_STEREO; break; default: goto unsupported_channels; } GST_INFO_OBJECT (esdsink, "attempting to open data connection to esound server"); esdsink->fd = esd_play_stream (esdformat, spec->rate, esdsink->host, connname); if ((esdsink->fd < 0) || (esdsink->ctrl_fd < 0)) goto cannot_open; esdsink->rate = spec->rate; spec->segsize = ESD_BUF_SIZE; spec->segtotal = (ESD_MAX_WRITE_SIZE / spec->segsize); /* FIXME: this is wrong for signed ints (and the * audioringbuffers should do it for us anyway) */ spec->silence_sample[0] = 0; spec->silence_sample[1] = 0; spec->silence_sample[2] = 0; spec->silence_sample[3] = 0; GST_INFO_OBJECT (esdsink, "successfully opened connection to esound server"); return TRUE; /* ERRORS */ unsupported_depth: { GST_ELEMENT_ERROR (esdsink, STREAM, WRONG_TYPE, (NULL), ("can't handle sample depth of %d, only 8 or 16 supported", spec->depth)); return FALSE; } unsupported_channels: { GST_ELEMENT_ERROR (esdsink, STREAM, WRONG_TYPE, (NULL), ("can't handle %d channels, only 1 or 2 supported", spec->channels)); return FALSE; } cannot_open: { GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE, (_("Could not establish connection to sound server")), ("can't open connection to esound server")); return FALSE; } } static gboolean gst_esdsink_unprepare (GstAudioSink * asink) { GstEsdSink *esdsink = GST_ESDSINK (asink); if ((esdsink->fd < 0) && (esdsink->ctrl_fd < 0)) return TRUE; close (esdsink->fd); esdsink->fd = -1; GST_INFO_OBJECT (esdsink, "closed sound device"); return TRUE; } static guint gst_esdsink_write (GstAudioSink * asink, gpointer data, guint length) { GstEsdSink *esdsink = GST_ESDSINK (asink); gint to_write = 0; to_write = length; while (to_write > 0) { int done; done = write (esdsink->fd, data, to_write); if (done < 0) goto write_error; to_write -= done; data = (char *) data + done; } return length; /* ERRORS */ write_error: { GST_ELEMENT_ERROR (esdsink, RESOURCE, WRITE, ("Failed to write data to the esound daemon"), GST_ERROR_SYSTEM); return 0; } } static guint gst_esdsink_delay (GstAudioSink * asink) { GstEsdSink *esdsink = GST_ESDSINK (asink); guint latency; latency = esd_get_latency (esdsink->ctrl_fd); if (latency == (guint) - 1) { GST_WARNING_OBJECT (asink, "couldn't get latency"); return 0; } /* latency is measured in samples at a rate of 44100, this * cannot overflow. */ latency = latency * G_GINT64_CONSTANT (44100) / esdsink->rate; GST_DEBUG_OBJECT (asink, "got latency: %u", latency); return latency; } static void gst_esdsink_reset (GstAudioSink * asink) { GST_DEBUG_OBJECT (asink, "reset called"); } static void gst_esdsink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstEsdSink *esdsink = GST_ESDSINK (object); switch (prop_id) { case PROP_HOST: g_free (esdsink->host); esdsink->host = g_value_dup_string (value); break; default: break; } } static void gst_esdsink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstEsdSink *esdsink = GST_ESDSINK (object); switch (prop_id) { case PROP_HOST: g_value_set_string (value, esdsink->host); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } }