/* GStreamer * Copyright (C) 2008 Tristan Matthews * * 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. * * * Example launch line * |[ * gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0 * ]| Get audio input into gstreamer from jack. * * * Last reviewed on 2008-07-22 (0.10.4) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include "gstjackaudiosrc.h" #include "gstjackringbuffer.h" #include "gstjackutil.h" GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug); #define GST_CAT_DEFAULT gst_jack_audio_src_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); src->buffers = g_realloc (src->buffers, sizeof (sample_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; g_free (src->buffers); src->buffers = NULL; } /* ringbuffer abstract base class */ static GType gst_jack_ring_buffer_get_type (void) { 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); 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; gint len; guint8 *writeptr; gint writeseg; gint channels, i, j, flen; sample_t *data; buf = GST_RING_BUFFER_CAST (arg); src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); channels = buf->spec.channels; /* get input buffers */ for (i = 0; i < channels; i++) src->buffers[i] = (sample_t *) jack_port_get_buffer (src->ports[i], nframes); if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) { flen = len / channels; /* the number of samples must be exactly the segment size */ if (nframes * sizeof (sample_t) != flen) goto wrong_size; /* the samples in the jack input buffers have to be interleaved into the * ringbuffer */ data = (sample_t *) writeptr; for (i = 0; i < nframes; ++i) for (j = 0; j < channels; ++j) *data++ = src->buffers[j][i]; GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr, len / channels, channels); /* we wrote one segment */ gst_ring_buffer_advance (buf, 1); } return 0; /* ERRORS */ wrong_size: { GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)", (gint) (nframes * sizeof (sample_t)), flen); return 1; } } /* 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; } /* 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, src->jclient, 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, (_("Jack server not found")), ("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; gst_jack_set_layout_on_caps (&spec->caps, channels); 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; if (spec->segtotal < 2) { spec->segtotal = 2; spec->buffer_time = spec->latency_time * spec->segtotal; } GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec", spec->buffer_time); GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec", spec->latency_time); GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d", buffer_size, 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 || src->connect == GST_JACK_CONNECT_AUTO_FORCED) { /* 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])); 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 i, res = 0, latency; jack_client_t *client; src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); client = gst_jack_audio_client_get_client (src->client); for (i = 0; i < src->port_count; i++) { latency = jack_port_get_total_latency (client, src->ports[i]); if (latency > res) res = latency; } 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_CLIENT, 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_jack_audio_src_debug, "jacksrc", 0, "jacksrc element"); GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc, GST_TYPE_BASE_AUDIO_SRC, _do_init); static void gst_jack_audio_src_dispose (GObject * object); static void gst_jack_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_jack_audio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc); static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src); /* GObject vmethod implementations */ static void gst_jack_audio_src_base_init (gpointer gclass) { 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_simple (element_class, "Audio Source (Jack)", "Source/Audio", "Input from Jack", "Tristan Matthews "); } /* initialize the jack_audio_src's class */ static void gst_jack_audio_src_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->dispose = gst_jack_audio_src_dispose; gobject_class->set_property = gst_jack_audio_src_set_property; gobject_class->get_property = gst_jack_audio_src_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_PARAM_STATIC_STRINGS)); 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 | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CLIENT, g_param_spec_boxed ("client", "JackClient", "Handle for jack client", GST_TYPE_JACK_CLIENT, GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps); gstbaseaudiosrc_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_jack_audio_src_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_jack_audio_src_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->jclient = NULL; src->ports = NULL; src->port_count = 0; src->buffers = NULL; } static void gst_jack_audio_src_dispose (GObject * object) { GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); gst_caps_replace (&src->caps, NULL); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_jack_audio_src_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; case PROP_CLIENT: if (GST_STATE (src) == GST_STATE_NULL || GST_STATE (src) == GST_STATE_READY) { src->jclient = g_value_get_boxed (value); } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_jack_audio_src_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; case PROP_CLIENT: g_value_set_boxed (value, src->jclient); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstCaps * gst_jack_audio_src_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_jack_audio_src_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; }