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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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468 lines
12 KiB
C
468 lines
12 KiB
C
/* GStreamer
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* Copyright (C) <2005> Arwed v. Merkatz <v.merkatz@gmx.net>
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*
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* Roughly based on the gstreamer 0.8 esdsink plugin:
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* Copyright (C) <2001> Richard Boulton <richard-gst@tartarus.org>
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*
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* esdsink.c: an EsounD audio sink
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-esdsink
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* @see_also: #GstAlsaSink, #GstAutoAudioSink
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*
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* This element outputs sound to an already-running Enlightened Sound Daemon
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* (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned
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* through this element (regardless of the system configuration), since this
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* is actively prevented by the element. If you must use esd, you need to
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* make sure it is started automatically with your session or otherwise.
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*
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* TODO: insert some comments about how sucky esd is and that all the cool
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* kids use pulseaudio or whatever these days.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! esdsink
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* ]| play an Ogg/Vorbis audio file via esd
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "esdsink.h"
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#include <esd.h>
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#include <unistd.h>
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#include <errno.h>
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#include <gst/gst-i18n-plugin.h>
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/* wtay: from my esd.h (debian unstable libesd0-dev 0.2.36-3) */
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#ifndef ESD_MAX_WRITE_SIZE
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#define ESD_MAX_WRITE_SIZE (21 * 4096)
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#endif
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GST_DEBUG_CATEGORY_EXTERN (esd_debug);
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#define GST_CAT_DEFAULT esd_debug
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enum
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{
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PROP_0,
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PROP_HOST
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};
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) TRUE, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-raw-int, "
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"signed = (boolean) { true, false }, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
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);
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static void gst_esdsink_finalize (GObject * object);
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static GstCaps *gst_esdsink_getcaps (GstBaseSink * bsink);
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static gboolean gst_esdsink_open (GstAudioSink * asink);
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static gboolean gst_esdsink_close (GstAudioSink * asink);
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static gboolean gst_esdsink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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static gboolean gst_esdsink_unprepare (GstAudioSink * asink);
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static guint gst_esdsink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_esdsink_delay (GstAudioSink * asink);
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static void gst_esdsink_reset (GstAudioSink * asink);
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static void gst_esdsink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_esdsink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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GST_BOILERPLATE (GstEsdSink, gst_esdsink, GstAudioSink, GST_TYPE_AUDIO_SINK);
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static void
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gst_esdsink_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_static_pad_template (element_class, &sink_factory);
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gst_element_class_set_details_simple (element_class, "Esound audio sink",
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"Sink/Audio",
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"Plays audio to an esound server",
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"Arwed von Merkatz <v.merkatz@gmx.net>");
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}
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static void
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gst_esdsink_class_init (GstEsdSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseSinkClass *gstbasesink_class;
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GstAudioSinkClass *gstaudiosink_class;
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gobject_class = (GObjectClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gstaudiosink_class = (GstAudioSinkClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_esdsink_finalize;
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_esdsink_getcaps);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_esdsink_open);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_esdsink_close);
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_esdsink_prepare);
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gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_esdsink_unprepare);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_esdsink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_esdsink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_esdsink_reset);
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gobject_class->set_property = gst_esdsink_set_property;
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gobject_class->get_property = gst_esdsink_get_property;
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/* default value is filled in the _init method */
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g_object_class_install_property (gobject_class, PROP_HOST,
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g_param_spec_string ("host", "Host",
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"The host running the esound daemon", NULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_esdsink_init (GstEsdSink * esdsink, GstEsdSinkClass * klass)
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{
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esdsink->fd = -1;
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esdsink->ctrl_fd = -1;
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esdsink->host = g_strdup (g_getenv ("ESPEAKER"));
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}
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static void
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gst_esdsink_finalize (GObject * object)
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{
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GstEsdSink *esdsink = GST_ESDSINK (object);
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gst_caps_replace (&esdsink->cur_caps, NULL);
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g_free (esdsink->host);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstCaps *
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gst_esdsink_getcaps (GstBaseSink * bsink)
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{
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GstEsdSink *esdsink;
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esdsink = GST_ESDSINK (bsink);
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/* no fd, we're done with the template caps */
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if (esdsink->ctrl_fd < 0 || esdsink->cur_caps == NULL) {
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GST_LOG_OBJECT (esdsink, "getcaps called, returning template caps");
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return NULL;
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}
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GST_LOG_OBJECT (esdsink, "returning %" GST_PTR_FORMAT, esdsink->cur_caps);
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return gst_caps_ref (esdsink->cur_caps);
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}
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static gboolean
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gst_esdsink_open (GstAudioSink * asink)
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{
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esd_server_info_t *server_info;
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GstPadTemplate *pad_template;
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GstEsdSink *esdsink;
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gchar *saved_env;
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gint i;
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esdsink = GST_ESDSINK (asink);
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GST_DEBUG_OBJECT (esdsink, "open");
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/* ensure libesd doesn't auto-spawn a sound daemon if none is running yet */
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saved_env = g_strdup (g_getenv ("ESD_NO_SPAWN"));
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g_setenv ("ESD_NO_SPAWN", "1", TRUE);
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/* now try to connect to any existing/running sound daemons */
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esdsink->ctrl_fd = esd_open_sound (esdsink->host);
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/* and restore the previous state */
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if (saved_env != NULL) {
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g_setenv ("ESD_NO_SPAWN", saved_env, TRUE);
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} else {
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g_unsetenv ("ESD_NO_SPAWN");
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}
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g_free (saved_env);
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if (esdsink->ctrl_fd < 0)
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goto couldnt_connect;
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/* get server info */
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server_info = esd_get_server_info (esdsink->ctrl_fd);
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if (!server_info)
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goto no_server_info;
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GST_INFO_OBJECT (esdsink, "got server info rate: %i", server_info->rate);
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pad_template = gst_static_pad_template_get (&sink_factory);
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esdsink->cur_caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
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gst_object_unref (pad_template);
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for (i = 0; i < esdsink->cur_caps->structs->len; i++) {
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GstStructure *s;
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s = gst_caps_get_structure (esdsink->cur_caps, i);
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gst_structure_set (s, "rate", G_TYPE_INT, server_info->rate, NULL);
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}
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esd_free_server_info (server_info);
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GST_INFO_OBJECT (esdsink, "server caps: %" GST_PTR_FORMAT, esdsink->cur_caps);
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return TRUE;
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/* ERRORS */
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couldnt_connect:
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{
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GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE,
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(_("Could not establish connection to sound server")),
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("can't open connection to esound server"));
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return FALSE;
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}
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no_server_info:
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{
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GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE,
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(_("Failed to query sound server capabilities")),
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("couldn't get server info!"));
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return FALSE;
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}
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}
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static gboolean
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gst_esdsink_close (GstAudioSink * asink)
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{
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GstEsdSink *esdsink = GST_ESDSINK (asink);
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GST_DEBUG_OBJECT (esdsink, "close");
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gst_caps_replace (&esdsink->cur_caps, NULL);
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esd_close (esdsink->ctrl_fd);
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esdsink->ctrl_fd = -1;
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return TRUE;
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}
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static gboolean
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gst_esdsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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{
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GstEsdSink *esdsink = GST_ESDSINK (asink);
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esd_format_t esdformat;
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/* Name used by esound for this connection. */
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const char connname[] = "GStreamer";
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GST_DEBUG_OBJECT (esdsink, "prepare");
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/* Bitmap describing audio format. */
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esdformat = ESD_STREAM | ESD_PLAY;
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switch (spec->depth) {
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case 8:
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esdformat |= ESD_BITS8;
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break;
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case 16:
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esdformat |= ESD_BITS16;
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break;
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default:
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goto unsupported_depth;
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}
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switch (spec->channels) {
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case 1:
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esdformat |= ESD_MONO;
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break;
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case 2:
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esdformat |= ESD_STEREO;
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break;
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default:
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goto unsupported_channels;
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}
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GST_INFO_OBJECT (esdsink,
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"attempting to open data connection to esound server");
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esdsink->fd =
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esd_play_stream (esdformat, spec->rate, esdsink->host, connname);
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if ((esdsink->fd < 0) || (esdsink->ctrl_fd < 0))
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goto cannot_open;
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esdsink->rate = spec->rate;
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spec->segsize = ESD_BUF_SIZE;
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spec->segtotal = (ESD_MAX_WRITE_SIZE / spec->segsize);
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/* FIXME: this is wrong for signed ints (and the
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* audioringbuffers should do it for us anyway) */
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spec->silence_sample[0] = 0;
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spec->silence_sample[1] = 0;
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spec->silence_sample[2] = 0;
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spec->silence_sample[3] = 0;
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GST_INFO_OBJECT (esdsink, "successfully opened connection to esound server");
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return TRUE;
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/* ERRORS */
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unsupported_depth:
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{
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GST_ELEMENT_ERROR (esdsink, STREAM, WRONG_TYPE, (NULL),
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("can't handle sample depth of %d, only 8 or 16 supported",
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spec->depth));
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return FALSE;
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}
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unsupported_channels:
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{
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GST_ELEMENT_ERROR (esdsink, STREAM, WRONG_TYPE, (NULL),
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("can't handle %d channels, only 1 or 2 supported", spec->channels));
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return FALSE;
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}
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cannot_open:
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{
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GST_ELEMENT_ERROR (esdsink, RESOURCE, OPEN_WRITE,
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(_("Could not establish connection to sound server")),
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("can't open connection to esound server"));
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return FALSE;
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}
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}
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static gboolean
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gst_esdsink_unprepare (GstAudioSink * asink)
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{
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GstEsdSink *esdsink = GST_ESDSINK (asink);
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if ((esdsink->fd < 0) && (esdsink->ctrl_fd < 0))
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return TRUE;
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close (esdsink->fd);
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esdsink->fd = -1;
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GST_INFO_OBJECT (esdsink, "closed sound device");
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return TRUE;
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}
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static guint
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gst_esdsink_write (GstAudioSink * asink, gpointer data, guint length)
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{
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GstEsdSink *esdsink = GST_ESDSINK (asink);
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gint to_write = 0;
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to_write = length;
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while (to_write > 0) {
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int done;
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done = write (esdsink->fd, data, to_write);
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if (done < 0)
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goto write_error;
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to_write -= done;
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data = (char *) data + done;
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}
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return length;
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/* ERRORS */
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write_error:
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{
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GST_ELEMENT_ERROR (esdsink, RESOURCE, WRITE,
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("Failed to write data to the esound daemon"), GST_ERROR_SYSTEM);
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return -1;
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}
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}
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static guint
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gst_esdsink_delay (GstAudioSink * asink)
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{
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GstEsdSink *esdsink = GST_ESDSINK (asink);
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guint latency;
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latency = esd_get_latency (esdsink->ctrl_fd);
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if (latency == (guint) - 1) {
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GST_WARNING_OBJECT (asink, "couldn't get latency");
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return 0;
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}
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/* latency is measured in samples at a rate of 44100, this
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* cannot overflow. */
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latency = latency * G_GINT64_CONSTANT (44100) / esdsink->rate;
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GST_DEBUG_OBJECT (asink, "got latency: %u", latency);
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return latency;
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}
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static void
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gst_esdsink_reset (GstAudioSink * asink)
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{
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GST_DEBUG_OBJECT (asink, "reset called");
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}
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static void
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gst_esdsink_set_property (GObject * object, guint prop_id, const GValue * value,
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GParamSpec * pspec)
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{
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GstEsdSink *esdsink = GST_ESDSINK (object);
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switch (prop_id) {
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case PROP_HOST:
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g_free (esdsink->host);
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esdsink->host = g_value_dup_string (value);
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break;
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default:
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break;
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}
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}
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static void
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gst_esdsink_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstEsdSink *esdsink = GST_ESDSINK (object);
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switch (prop_id) {
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case PROP_HOST:
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g_value_set_string (value, esdsink->host);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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