--- /dev/null
+/* GStreamer
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a
+ * copy of this software and associated documentation files (the "Software"),
+ * to deal in the Software without restriction, including without limitation
+ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
+ * and/or sell copies of the Software, and to permit persons to whom the
+ * Software is furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+ * DEALINGS IN THE SOFTWARE.
+ *
+ * Alternatively, the contents of this file may be used under the
+ * GNU Lesser General Public License Version 2.1 (the "LGPL"), in
+ * which case the following provisions apply instead of the ones
+ * mentioned above:
+ *
+ * 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-jackaudiosrc
+ * @see_also: #GstBaseAudioSrc, #GstRingBuffer
+ *
+ * A Src that inputs data from Jack ports.
+ *
+ * It will create N Jack ports named in_<name>_<num> where
+ * <name> is the element name and <num> is starting from 1.
+ * Each port corresponds to a gstreamer channel.
+ *
+ * The samplerate as exposed on the caps is always the same as the samplerate of
+ * the jack server.
+ *
+ * When the #GstJackAudioSrc:connect property is set to auto, this element
+ * will try to connect each input port to a random physical jack output pin.
+ *
+ * When the #GstJackAudioSrc:connect property is set to none, the element will
+ * accept any number of output channels and will create (but not connect) an
+ * input port for each channel.
+ *
+ * The element will generate an error when the Jack server is shut down when it
+ * was PAUSED or PLAYING. This element does not support dynamic rate and buffer
+ * size changes at runtime.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0
+ * ]| Get audio input into gstreamer from jack.
+ * </refsect2>
+ *
+ * Last reviewed on 2008-07-22 (0.10.4)
+ */
+
+#include <gst/gst.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "gstjackaudiosrc.h"
+#include "gstjackringbuffer.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_jackaudiosrc_debug);
+#define GST_CAT_DEFAULT gst_jackaudiosrc_debug
+
+static gboolean
+gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
+{
+ jack_client_t *client;
+
+ client = gst_jack_audio_client_get_client (src->client);
+
+ /* remove ports we don't need */
+ while (src->port_count > channels)
+ jack_port_unregister (client, src->ports[--src->port_count]);
+
+ /* alloc enough input ports */
+ src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
+
+ /* create an input port for each channel */
+ while (src->port_count < channels) {
+ gchar *name;
+
+ /* port names start from 1 and are local to the element */
+ name =
+ g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
+ src->port_count + 1);
+ src->ports[src->port_count] =
+ jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
+ JackPortIsInput, 0);
+ if (src->ports[src->port_count] == NULL)
+ return FALSE;
+
+ src->port_count++;
+
+ g_free (name);
+ }
+ return TRUE;
+}
+
+static void
+gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
+{
+ gint res, i = 0;
+ jack_client_t *client;
+
+ client = gst_jack_audio_client_get_client (src->client);
+
+ /* get rid of all ports */
+ while (src->port_count) {
+ GST_LOG_OBJECT (src, "unregister port %d", i);
+ if ((res = jack_port_unregister (client, src->ports[i++])))
+ GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
+
+ src->port_count--;
+ }
+ g_free (src->ports);
+ src->ports = NULL;
+}
+
+/* ringbuffer abstract base class */
+static GType
+gst_jack_ring_buffer_get_type ()
+{
+ static GType ringbuffer_type = 0;
+
+ if (!ringbuffer_type) {
+ static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_jack_ring_buffer_class_init,
+ NULL,
+ NULL,
+ sizeof (GstJackRingBuffer),
+ 0,
+ (GInstanceInitFunc) gst_jack_ring_buffer_init,
+ NULL
+ };
+
+ ringbuffer_type =
+ g_type_register_static (GST_TYPE_RING_BUFFER,
+ "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
+ }
+ return ringbuffer_type;
+}
+
+static void
+gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstObjectClass *gstobject_class;
+ GstRingBufferClass *gstringbuffer_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstobject_class = (GstObjectClass *) klass;
+ gstringbuffer_class = (GstRingBufferClass *) klass;
+
+ ring_parent_class = g_type_class_peek_parent (klass);
+
+ gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_dispose);
+ gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_finalize);
+
+ gstringbuffer_class->open_device =
+ GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
+ gstringbuffer_class->close_device =
+ GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
+ gstringbuffer_class->acquire =
+ GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
+ gstringbuffer_class->release =
+ GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
+ gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
+ gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
+ gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
+ gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
+
+ gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
+}
+
+/* this is the callback of jack. This should be RT-safe.
+ * Writes samples from the jack input port's buffer to the gst ring buffer.
+ */
+static int
+jack_process_cb (jack_nframes_t nframes, void *arg)
+{
+ GstJackAudioSrc *src;
+ GstRingBuffer *buf;
+ GstJackRingBuffer *abuf;
+ gint len, givenLen;
+ guint8 *writeptr, *dataStart;
+ gint writeseg;
+ gint channels, i, j;
+ sample_t **buffers, *data;
+
+ buf = GST_RING_BUFFER_CAST (arg);
+ abuf = GST_JACK_RING_BUFFER_CAST (arg);
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ channels = buf->spec.channels;
+ len = sizeof (sample_t) * nframes * channels;
+
+ /* alloc pointers to samples */
+ buffers = g_alloca (sizeof (sample_t *) * channels);
+ data = g_alloca (len);
+
+ /* get input buffers */
+ for (i = 0; i < channels; i++)
+ buffers[i] = (sample_t *) jack_port_get_buffer (src->ports[i], nframes);
+
+ //writeptr = data;
+ dataStart = (guint8 *) data;
+
+ /* the samples in the jack input buffers have to be interleaved into the
+ * ringbuffer
+ */
+
+ for (i = 0; i < nframes; ++i)
+ for (j = 0; j < channels; ++j)
+ *data++ = buffers[j][i];
+
+ if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &givenLen)) {
+ memcpy (writeptr, (char *) dataStart, givenLen);
+
+ GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
+ len / channels, channels);
+
+ /* clear written samples in the ringbuffer */
+ // gst_ring_buffer_clear(buf, 0);
+
+ /* we wrote one segment */
+ gst_ring_buffer_advance (buf, 1);
+ }
+ return 0;
+}
+
+/* we error out */
+static int
+jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
+{
+ GstJackAudioSrc *src;
+ GstJackRingBuffer *abuf;
+
+ abuf = GST_JACK_RING_BUFFER_CAST (arg);
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
+
+ if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
+ goto not_supported;
+
+ return 0;
+
+ /* ERRORS */
+not_supported:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
+ (NULL), ("Jack changed the sample rate, which is not supported"));
+ return 1;
+ }
+}
+
+/* we error out */
+static int
+jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
+{
+ GstJackAudioSrc *src;
+ GstJackRingBuffer *abuf;
+
+ abuf = GST_JACK_RING_BUFFER_CAST (arg);
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
+
+ if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
+ goto not_supported;
+
+ return 0;
+
+ /* ERRORS */
+not_supported:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
+ (NULL), ("Jack changed the buffer size, which is not supported"));
+ return 1;
+ }
+}
+
+static void
+jack_shutdown_cb (void *arg)
+{
+ GstJackAudioSrc *src;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
+
+ GST_DEBUG_OBJECT (src, "shutdown");
+
+ GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
+ (NULL), ("Jack server shutdown"));
+}
+
+static void
+gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
+ GstJackRingBufferClass * g_class)
+{
+ buf->channels = -1;
+ buf->buffer_size = -1;
+ buf->sample_rate = -1;
+}
+
+static void
+gst_jack_ring_buffer_dispose (GObject * object)
+{
+ G_OBJECT_CLASS (ring_parent_class)->dispose (object);
+}
+
+static void
+gst_jack_ring_buffer_finalize (GObject * object)
+{
+ GstJackRingBuffer *ringbuffer;
+ ringbuffer = GST_JACK_RING_BUFFER_CAST (object);
+ G_OBJECT_CLASS (ring_parent_class)->finalize (object);
+}
+
+/* the _open_device method should make a connection with the server
+*/
+static gboolean
+gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+ jack_status_t status = 0;
+ const gchar *name;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "open");
+
+ name = g_get_application_name ();
+ if (!name)
+ name = "GStreamer";
+
+ src->client = gst_jack_audio_client_new (name, src->server,
+ GST_JACK_CLIENT_SOURCE,
+ jack_shutdown_cb,
+ jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
+ if (src->client == NULL)
+ goto could_not_open;
+
+ GST_DEBUG_OBJECT (src, "opened");
+
+ return TRUE;
+
+ /* ERRORS */
+could_not_open:
+ {
+ if (status & JackServerFailed) {
+ GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
+ (NULL), ("Cannot connect to the Jack server (status %d)", status));
+ } else {
+ GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE,
+ (NULL), ("Jack client open error (status %d)", status));
+ }
+ return FALSE;
+ }
+}
+
+/* close the connection with the server
+*/
+static gboolean
+gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "close");
+
+ gst_jack_audio_src_free_channels (src);
+ gst_jack_audio_client_free (src->client);
+ src->client = NULL;
+
+ return TRUE;
+}
+
+
+/* allocate a buffer and setup resources to process the audio samples of
+ * the format as specified in @spec.
+ *
+ * We allocate N jack ports, one for each channel. If we are asked to
+ * automatically make a connection with physical ports, we connect as many
+ * ports as there are physical ports, leaving leftover ports unconnected.
+ *
+ * It is assumed that samplerate and number of channels are acceptable since our
+ * getcaps method will always provide correct values. If unacceptable caps are
+ * received for some reason, we fail here.
+ */
+static gboolean
+gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
+{
+ GstJackAudioSrc *src;
+ GstJackRingBuffer *abuf;
+ const char **ports;
+ gint sample_rate, buffer_size;
+ gint i, channels, res;
+ jack_client_t *client;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+ abuf = GST_JACK_RING_BUFFER_CAST (buf);
+
+ GST_DEBUG_OBJECT (src, "acquire");
+
+ client = gst_jack_audio_client_get_client (src->client);
+
+ /* sample rate must be that of the server */
+ sample_rate = jack_get_sample_rate (client);
+ if (sample_rate != spec->rate)
+ goto wrong_samplerate;
+
+ channels = spec->channels;
+
+ if (!gst_jack_audio_src_allocate_channels (src, channels))
+ goto out_of_ports;
+
+ buffer_size = jack_get_buffer_size (client);
+
+ /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
+ * for all channels */
+ spec->segsize = buffer_size * sizeof (gfloat) * channels;
+ spec->latency_time = gst_util_uint64_scale (spec->segsize,
+ (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
+ /* segtotal based on buffer-time latency */
+ spec->segtotal = spec->buffer_time / spec->latency_time;
+
+ GST_DEBUG_OBJECT (src, "segsize %d, segtotal %d", spec->segsize,
+ spec->segtotal);
+
+ /* allocate the ringbuffer memory now */
+ buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
+ memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
+
+ if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
+ goto could_not_activate;
+
+ /* if we need to automatically connect the ports, do so now. We must do this
+ * after activating the client. */
+ if (src->connect == GST_JACK_CONNECT_AUTO) {
+ /* find all the physical output ports. A physical output port is a port
+ * associated with a hardware device. Someone needs connect to a physical
+ * port in order to capture something. */
+ ports =
+ jack_get_ports (client, NULL, NULL,
+ JackPortIsPhysical | JackPortIsOutput);
+ if (ports == NULL) {
+ /* no ports? fine then we don't do anything except for posting a warning
+ * message. */
+ GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
+ ("No physical output ports found, leaving ports unconnected"));
+ goto done;
+ }
+
+ for (i = 0; i < channels; i++) {
+ /* stop when all output ports are exhausted */
+ if (ports[i] == NULL) {
+ /* post a warning that we could not connect all ports */
+ GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
+ ("No more physical ports, leaving some ports unconnected"));
+ break;
+ }
+ GST_DEBUG_OBJECT (src, "try connecting to %s",
+ jack_port_name (src->ports[i]));
+ /* connect the physical port to a port */
+
+ res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
+ g_print ("connecting to %s\n", jack_port_name (src->ports[i]));
+ if (res != 0 && res != EEXIST)
+ goto cannot_connect;
+ }
+ free (ports);
+ }
+done:
+
+ abuf->sample_rate = sample_rate;
+ abuf->buffer_size = buffer_size;
+ abuf->channels = spec->channels;
+
+ return TRUE;
+
+ /* ERRORS */
+wrong_samplerate:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+ ("Wrong samplerate, server is running at %d and we received %d",
+ sample_rate, spec->rate));
+ return FALSE;
+ }
+out_of_ports:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+ ("Cannot allocate more Jack ports"));
+ return FALSE;
+ }
+could_not_activate:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+ ("Could not activate client (%d:%s)", res, g_strerror (res)));
+ return FALSE;
+ }
+cannot_connect:
+ {
+ GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+ ("Could not connect input ports to physical ports (%d:%s)",
+ res, g_strerror (res)));
+ free (ports);
+ return FALSE;
+ }
+}
+
+/* function is called with LOCK */
+static gboolean
+gst_jack_ring_buffer_release (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+ GstJackRingBuffer *abuf;
+ gint res;
+
+ abuf = GST_JACK_RING_BUFFER_CAST (buf);
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "release");
+
+ if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
+ /* we only warn, this means the server is probably shut down and the client
+ * is gone anyway. */
+ GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
+ ("Could not deactivate Jack client (%d)", res));
+ }
+
+ abuf->channels = -1;
+ abuf->buffer_size = -1;
+ abuf->sample_rate = -1;
+
+ /* free the buffer */
+ gst_buffer_unref (buf->data);
+ buf->data = NULL;
+
+ return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_start (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "start");
+
+ return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_pause (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "pause");
+
+ return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_stop (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "stop");
+
+ return TRUE;
+}
+
+static guint
+gst_jack_ring_buffer_delay (GstRingBuffer * buf)
+{
+ GstJackAudioSrc *src;
+ guint res = 0;
+
+ src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+ GST_DEBUG_OBJECT (src, "delay %u", res);
+
+ return res;
+}
+
+/* Audiosrc signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER NULL
+
+enum
+{
+ PROP_0,
+ PROP_CONNECT,
+ PROP_SERVER,
+ PROP_LAST
+};
+
+
+/* the capabilities of the inputs and outputs.
+ *
+ * describe the real formats here.
+ */
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+ "width = (int) 32, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+ );
+
+#define _do_init(bla) \
+ GST_DEBUG_CATEGORY_INIT(gst_jackaudiosrc_debug, "jacksrc", 0, "jacksrc element");
+
+GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc,
+ GST_TYPE_BASE_AUDIO_SRC, _do_init);
+
+static void gst_jackaudiosrc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_jackaudiosrc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc);
+static GstRingBuffer *gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc *
+ src);
+
+/* GObject vmethod implementations */
+
+static void
+gst_jackaudiosrc_base_init (gpointer gclass)
+{
+ static GstElementDetails gst_jackaudiosrc_details = {
+ "Audio Source (Jack)",
+ "Source/Audio",
+ "Input from Jack",
+ "Tristan Matthews <tristan@sat.qc.ca>"
+ };
+ GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_factory));
+ gst_element_class_set_details (element_class, &gst_jackaudiosrc_details);
+}
+
+/* initialize the jackaudiosrc's class */
+static void
+gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseSrcClass *gstbasesrc_class;
+ GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ gstbasesrc_class = (GstBaseSrcClass *) klass;
+ gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+
+ gobject_class->set_property =
+ GST_DEBUG_FUNCPTR (gst_jackaudiosrc_set_property);
+ gobject_class->get_property =
+ GST_DEBUG_FUNCPTR (gst_jackaudiosrc_get_property);
+
+ g_object_class_install_property (gobject_class, PROP_CONNECT,
+ g_param_spec_enum ("connect", "Connect",
+ "Specify how the input ports will be connected",
+ GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SERVER,
+ g_param_spec_string ("server", "Server",
+ "The Jack server to connect to (NULL = default)",
+ DEFAULT_PROP_SERVER, G_PARAM_READWRITE));
+
+ gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jackaudiosrc_getcaps);
+ gstbaseaudiosrc_class->create_ringbuffer =
+ GST_DEBUG_FUNCPTR (gst_jackaudiosrc_create_ringbuffer);
+
+ /* ref class from a thread-safe context to work around missing bit of
+ * thread-safety in GObject */
+ g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
+
+ gst_jack_audio_client_init ();
+}
+
+/* initialize the new element
+ * instantiate pads and add them to element
+ * set pad calback functions
+ * initialize instance structure
+ */
+static void
+gst_jackaudiosrc_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
+{
+ //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
+ src->connect = DEFAULT_PROP_CONNECT;
+ src->server = g_strdup (DEFAULT_PROP_SERVER);
+ src->ports = NULL;
+ src->port_count = 0;
+}
+
+static void
+gst_jackaudiosrc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
+
+ switch (prop_id) {
+ case PROP_CONNECT:
+ src->connect = g_value_get_enum (value);
+ break;
+ case PROP_SERVER:
+ g_free (src->server);
+ src->server = g_value_dup_string (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_jackaudiosrc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
+
+ switch (prop_id) {
+ case PROP_CONNECT:
+ g_value_set_enum (value, src->connect);
+ break;
+ case PROP_SERVER:
+ g_value_set_string (value, src->server);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstCaps *
+gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc)
+{
+ GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
+ const char **ports;
+ gint min, max;
+ gint rate;
+ jack_client_t *client;
+
+ if (src->client == NULL)
+ goto no_client;
+
+ client = gst_jack_audio_client_get_client (src->client);
+
+ if (src->connect == GST_JACK_CONNECT_AUTO) {
+ /* get a port count, this is the number of channels we can automatically
+ * connect. */
+ ports = jack_get_ports (client, NULL, NULL,
+ JackPortIsPhysical | JackPortIsOutput);
+ max = 0;
+ if (ports != NULL) {
+ for (; ports[max]; max++);
+
+ free (ports);
+ } else
+ max = 0;
+ } else {
+ /* we allow any number of pads, something else is going to connect the
+ * pads. */
+ max = G_MAXINT;
+ }
+ min = MIN (1, max);
+
+ rate = jack_get_sample_rate (client);
+
+ GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
+
+ if (!src->caps) {
+ src->caps = gst_caps_new_simple ("audio/x-raw-float",
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "width", G_TYPE_INT, 32,
+ "rate", G_TYPE_INT, rate,
+ "channels", GST_TYPE_INT_RANGE, min, max, NULL);
+ }
+ GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
+
+ return gst_caps_ref (src->caps);
+
+ /* ERRORS */
+no_client:
+ {
+ GST_DEBUG_OBJECT (src, "device not open, using template caps");
+ /* base class will get template caps for us when we return NULL */
+ return NULL;
+ }
+}
+
+static GstRingBuffer *
+gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * src)
+{
+ GstRingBuffer *buffer;
+
+ buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
+ GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
+
+ return buffer;
+}
projects / platform / upstream / gstreamer.git / commitdiff