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ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse.

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Original commit message from CVS:
patch by: Tristan Matthews <tristan@sat.qc.ca>
* ext/jack/Makefile.am:
* ext/jack/gstjack.c:
* ext/jack/gstjackaudioclient.c:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosink.h:
* ext/jack/gstjackaudiosrc.c:
* ext/jack/gstjackaudiosrc.h:
* ext/jack/gstjackringbuffer.h:
Add a jackaudiosrc. Refactor sink slightly for better code reuse.
Fixes #545197.
This commit is contained in:
Tristan Matthews 2008-08-07 08:47:40 +00:00 committed by Stefan Kost
parent 020d3ca531
commit 3fcdc01db8
9 changed files with 1069 additions and 89 deletions
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15
ChangeLog
View file

@ -1,3 +1,18 @@
2008-08-07 Stefan Kost <ensonic@users.sf.net>
patch by: Tristan Matthews <tristan@sat.qc.ca>
* ext/jack/Makefile.am:
* ext/jack/gstjack.c:
* ext/jack/gstjackaudioclient.c:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosink.h:
* ext/jack/gstjackaudiosrc.c:
* ext/jack/gstjackaudiosrc.h:
* ext/jack/gstjackringbuffer.h:
Add a jackaudiosrc. Refactor sink slightly for better code reuse.
Fixes #545197.
2008-08-06 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* docs/plugins/Makefile.am:

4
ext/jack/Makefile.am
View file

@ -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

23
ext/jack/gstjack.c
View file

@ -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;

3
ext/jack/gstjackaudioclient.c
View file

@ -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;

72
ext/jack/gstjackaudiosink.c
View file

@ -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");

17
ext/jack/gstjackaudiosink.h
View file

@ -27,6 +27,7 @@
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosink.h>
#include "gstjack.h"
#include "gstjackaudioclient.h"
G_BEGIN_DECLS
@ -41,22 +42,6 @@ G_BEGIN_DECLS
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:
*

840
ext/jack/gstjackaudiosrc.c Normal file
View file

@ -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_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; 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;
}

94
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/* 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__ */

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/*
* 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