diff options
Diffstat (limited to 'ext/jack')
| -rw-r--r-- | ext/jack/Makefile.am | 4 | ||||
| -rw-r--r-- | ext/jack/gstjack.c | 23 | ||||
| -rw-r--r-- | ext/jack/gstjackaudioclient.c | 3 | ||||
| -rw-r--r-- | ext/jack/gstjackaudiosink.c | 72 | ||||
| -rw-r--r-- | ext/jack/gstjackaudiosink.h | 17 | ||||
| -rw-r--r-- | ext/jack/gstjackaudiosrc.c | 840 | ||||
| -rw-r--r-- | ext/jack/gstjackaudiosrc.h | 94 | ||||
| -rw-r--r-- | ext/jack/gstjackringbuffer.h | 90 |
8 files changed, 1054 insertions, 89 deletions
diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 17efdfa9..abcc39a5 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,11 +1,11 @@ plugin_LTLIBRARIES = libgstjack.la -libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c gstjackaudioclient.c +libgstjack_la_SOURCES = gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -noinst_HEADERS = gstjackaudiosink.h gstjackaudioclient.h +noinst_HEADERS = gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h EXTRA_DIST = README diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 72f501d0..96afd06e 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -21,11 +21,34 @@ #include "config.h" #endif +#include "gstjackaudiosrc.h" #include "gstjackaudiosink.h" +GType +gst_jack_connect_get_type (void) +{ + static GType jack_connect_type = 0; + static const GEnumValue jack_connect[] = { + {GST_JACK_CONNECT_NONE, + "Don't automatically connect ports to physical ports", "none"}, + {GST_JACK_CONNECT_AUTO, + "Automatically connect ports to physical ports", "auto"}, + {0, NULL, NULL}, + }; + + if (!jack_connect_type) { + jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); + } + return jack_connect_type; +} + + static gboolean plugin_init (GstPlugin * plugin) { + if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY, + GST_TYPE_JACK_AUDIO_SRC)) + return FALSE; if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY, GST_TYPE_JACK_AUDIO_SINK)) return FALSE; diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index 9777cd97..1aa1baf8 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -127,6 +127,9 @@ jack_shutdown_cb (void *arg) GstJackAudioConnection *conn = (GstJackAudioConnection *) arg; GList *walk; + GST_DEBUG ("disconnect client %s from server %s", conn->id, + GST_STR_NULL (conn->server)); + g_mutex_lock (conn->lock); for (walk = conn->src_clients; walk; walk = g_list_next (walk)) { GstJackAudioClient *client = (GstJackAudioClient *) walk->data; diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 05571b2c..ec257deb 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -59,62 +59,11 @@ #include <string.h> #include "gstjackaudiosink.h" +#include "gstjackringbuffer.h" GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug); #define GST_CAT_DEFAULT gst_jack_audio_sink_debug -typedef jack_default_audio_sample_t sample_t; - -#define GST_TYPE_JACK_RING_BUFFER \ - (gst_jack_ring_buffer_get_type()) -#define GST_JACK_RING_BUFFER(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer)) -#define GST_JACK_RING_BUFFER_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) -#define GST_JACK_RING_BUFFER_GET_CLASS(obj) \ - (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER, GstJackRingBufferClass)) -#define GST_JACK_RING_BUFFER_CAST(obj) \ - ((GstJackRingBuffer *)obj) -#define GST_IS_JACK_RING_BUFFER(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER)) -#define GST_IS_JACK_RING_BUFFER_CLASS(klass)\ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER)) - -typedef struct _GstJackRingBuffer GstJackRingBuffer; -typedef struct _GstJackRingBufferClass GstJackRingBufferClass; - -struct _GstJackRingBuffer -{ - GstRingBuffer object; - - gint sample_rate; - gint buffer_size; - gint channels; -}; - -struct _GstJackRingBufferClass -{ - GstRingBufferClass parent_class; -}; - -static void gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass); -static void gst_jack_ring_buffer_init (GstJackRingBuffer * ringbuffer, - GstJackRingBufferClass * klass); -static void gst_jack_ring_buffer_dispose (GObject * object); -static void gst_jack_ring_buffer_finalize (GObject * object); - -static GstRingBufferClass *ring_parent_class = NULL; - -static gboolean gst_jack_ring_buffer_open_device (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_close_device (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_acquire (GstRingBuffer * buf, - GstRingBufferSpec * spec); -static gboolean gst_jack_ring_buffer_release (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_start (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_pause (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_stop (GstRingBuffer * buf); -static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf); - static gboolean gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels) { @@ -689,25 +638,6 @@ enum PROP_LAST }; -#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) -static GType -gst_jack_connect_get_type (void) -{ - static GType jack_connect_type = 0; - static const GEnumValue jack_connect[] = { - {GST_JACK_CONNECT_NONE, - "Don't automatically connect ports to physical ports", "none"}, - {GST_JACK_CONNECT_AUTO, - "Automatically connect ports to physical ports", "auto"}, - {0, NULL, NULL}, - }; - - if (!jack_connect_type) { - jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); - } - return jack_connect_type; -} - #define _do_init(bla) \ GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element"); diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 12c82a83..b4a77033 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -27,6 +27,7 @@ #include <gst/gst.h> #include <gst/audio/gstbaseaudiosink.h> +#include "gstjack.h" #include "gstjackaudioclient.h" G_BEGIN_DECLS @@ -42,22 +43,6 @@ typedef struct _GstJackAudioSink GstJackAudioSink; typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass; /** - * GstJackConnect: - * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports. - * In this mode, the element will accept any number of input channels and will - * create (but not connect) an output port for each channel. - * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each - * output port to a random physical jack input pin. The sink will - * expose the number of physical channels on its pad caps. - * - * Specify how the output ports will be connected. - */ -typedef enum { - GST_JACK_CONNECT_NONE, - GST_JACK_CONNECT_AUTO -} GstJackConnect; - -/** * GstJackAudioSink: * * Opaque #GstJackAudioSink. diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c new file mode 100644 index 00000000..d41b62ff --- /dev/null +++ b/ext/jack/gstjackaudiosrc.c @@ -0,0 +1,840 @@ +/* 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; +} diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h new file mode 100644 index 00000000..69230977 --- /dev/null +++ b/ext/jack/gstjackaudiosrc.h @@ -0,0 +1,94 @@ +/* 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. + */ + +#ifndef __GST_JACK_AUDIO_SRC_H__ +#define __GST_JACK_AUDIO_SRC_H__ + +#include <jack/jack.h> + +#include <gst/gst.h> +#include <gst/audio/gstaudiosrc.h> + +#include "gstjackaudioclient.h" +#include "gstjack.h" + +G_BEGIN_DECLS + +#define GST_TYPE_JACK_AUDIO_SRC (gst_jackaudiosrc_get_type()) +#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc)) +#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass)) +#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC)) +#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC)) + +typedef struct _GstJackAudioSrc GstJackAudioSrc; +typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass; + +struct _GstJackAudioSrc +{ + GstBaseAudioSrc src; + + /*< private >*/ + /* cached caps */ + GstCaps *caps; + + /* properties */ + GstJackConnect connect; + gchar *server; + + /* our client */ + GstJackAudioClient *client; + + /* our ports */ + jack_port_t **ports; + int port_count; +}; + +struct _GstJackAudioSrcClass +{ + GstBaseAudioSrcClass parent_class; +}; + +GType gst_jackaudiosrc_get_type (void); + +G_END_DECLS + +#endif /* __GST_JACK_AUDIO_SRC_H__ */ diff --git a/ext/jack/gstjackringbuffer.h b/ext/jack/gstjackringbuffer.h new file mode 100644 index 00000000..dfe21b1a --- /dev/null +++ b/ext/jack/gstjackringbuffer.h @@ -0,0 +1,90 @@ +/* + * GStreamer + * Copyright (C) 2006 Wim Taymans <wim@fluendo.com> + * 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. + */ + +#ifndef __GST_JACK_RING_BUFFER_H__ +#define __GST_JACK_RING_BUFFER_H__ + +#define GST_TYPE_JACK_RING_BUFFER (gst_jack_ring_buffer_get_type()) +#define GST_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer)) +#define GST_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) +#define GST_JACK_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) +#define GST_JACK_RING_BUFFER_CAST(obj) ((GstJackRingBuffer *)obj) +#define GST_IS_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER)) +#define GST_IS_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER)) + +typedef struct _GstJackRingBuffer GstJackRingBuffer; +typedef struct _GstJackRingBufferClass GstJackRingBufferClass; + +struct _GstJackRingBuffer +{ + GstRingBuffer object; + + gint sample_rate; + gint buffer_size; + gint channels; +}; + +struct _GstJackRingBufferClass +{ + GstRingBufferClass parent_class; +}; + +static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass); +static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer, + GstJackRingBufferClass * klass); +static void gst_jack_ring_buffer_dispose(GObject * object); +static void gst_jack_ring_buffer_finalize(GObject * object); + +static GstRingBufferClass *ring_parent_class = NULL; + +static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec); +static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf); +static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf); + +#endif |