gstreamer/ext/jack/gstjackaudiosrc.c
Matthew Waters 612102fdbc gst: don't use volatile to mean atomic
volatile is not sufficient to provide atomic guarantees and real atomics
should be used instead.  GCC 11 has started warning about using volatile
with atomic operations.

https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719

Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/911>
2021-03-18 19:52:53 +11:00

990 lines
28 KiB
C

/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-jackaudiosrc
* @title: jackaudiosrc
* @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer
*
* 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.
*
* ## Example launch line
* |[
* gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
* ]| Get audio input into gstreamer from jack.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include <stdlib.h>
#include <string.h>
#include <gst/audio/audio.h>
#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 gsize ringbuffer_type = 0;
if (g_once_init_enter (&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
};
GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
"GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
g_once_init_leave (&ringbuffer_type, tmp);
}
return (GType) ringbuffer_type;
}
static void
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
{
GstAudioRingBufferClass *gstringbuffer_class;
gstringbuffer_class = (GstAudioRingBufferClass *) 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;
GstAudioRingBuffer *buf;
gint len;
guint8 *writeptr;
gint writeseg;
gint channels, i, j, flen;
sample_t *data;
buf = GST_AUDIO_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* get input buffers */
for (i = 0; i < channels; i++)
src->buffers[i] =
(sample_t *) jack_port_get_buffer (src->ports[i], nframes);
if (gst_audio_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_audio_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 (GstAudioRingBuffer * 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");
if (src->client_name) {
name = src->client_name;
} else {
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 | JackFailure)) {
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_READ,
(NULL), ("Jack client open error (status %d)", status));
}
return FALSE;
}
}
/* close the connection with the server
*/
static gboolean
gst_jack_ring_buffer_close_device (GstAudioRingBuffer * 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 (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, bpf, rate, 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);
rate = GST_AUDIO_INFO_RATE (&spec->info);
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
if (sample_rate != rate)
goto wrong_samplerate;
bpf = GST_AUDIO_INFO_BPF (&spec->info);
channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (!gst_jack_audio_src_allocate_channels (src, channels))
goto out_of_ports;
gst_jack_set_layout (buf, spec);
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), rate * bpf);
/* 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->size = spec->segtotal * spec->segsize;
buf->memory = g_malloc0 (buf->size);
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. */
if (src->port_pattern == NULL) {
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput);
} else {
ports = jack_get_ports (client, src->port_pattern, NULL,
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;
}
jack_free (ports);
}
done:
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
abuf->channels = 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, 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)));
jack_free (ports);
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstAudioRingBuffer * 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 */
g_free (buf->memory);
buf->memory = NULL;
return TRUE;
}
static gboolean
gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "start");
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
jack_transport_start (client);
}
return TRUE;
}
static gboolean
gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "pause");
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
jack_transport_stop (client);
}
return TRUE;
}
static gboolean
gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "stop");
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
jack_transport_stop (client);
}
return TRUE;
}
#if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
static guint
gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstJackAudioSrc *src;
guint i, res = 0;
jack_latency_range_t range;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
for (i = 0; i < src->port_count; i++) {
jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
if (range.max > res)
res = range.max;
}
GST_DEBUG_OBJECT (src, "delay %u", res);
return res;
}
#else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
static guint
gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstJackAudioSrc *src;
guint i, res = 0;
guint 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;
}
#endif
/* Audiosrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
#define DEFAULT_PROP_CLIENT_NAME NULL
#define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS
#define DEFAULT_PROP_PORT_PATTERN NULL
enum
{
PROP_0,
PROP_CONNECT,
PROP_SERVER,
PROP_CLIENT,
PROP_CLIENT_NAME,
PROP_PORT_PATTERN,
PROP_TRANSPORT,
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, "
"format = (string) " GST_JACK_FORMAT_STR ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
#define gst_jack_audio_src_parent_class parent_class
G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_AUDIO_BASE_SRC);
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,
GstCaps * filter);
static GstAudioRingBuffer *gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc
* src);
/* GObject vmethod implementations */
/* 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;
GstAudioBaseSrcClass *gstaudiobasesrc_class;
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_src_debug, "jacksrc", 0,
"jacksrc element");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) 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));
/**
* GstJackAudioSrc:client-name:
*
* The client name to use.
*/
g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
g_param_spec_string ("client-name", "Client name",
"The client name of the Jack instance (NULL = default)",
DEFAULT_PROP_CLIENT_NAME,
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));
/**
* GstJackAudioSrc:port-pattern
*
* autoconnect to ports matching pattern, when NULL connect to physical ports
*
* Since: 1.6
*/
g_object_class_install_property (gobject_class, PROP_PORT_PATTERN,
g_param_spec_string ("port-pattern", "port pattern",
"A pattern to select which ports to connect to (NULL = first physical ports)",
DEFAULT_PROP_PORT_PATTERN,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstJackAudioSink:transport:
*
* The jack transport behaviour for the client.
*/
g_object_class_install_property (gobject_class, PROP_TRANSPORT,
g_param_spec_flags ("transport", "Transport mode",
"Jack transport behaviour of the client",
GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
gst_element_class_set_static_metadata (gstelement_class,
"Audio Source (Jack)", "Source/Audio",
"Captures audio from a JACK server",
"Tristan Matthews <tristan@sat.qc.ca>");
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
gstaudiobasesrc_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 ();
}
static void
gst_jack_audio_src_init (GstJackAudioSrc * src)
{
//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;
src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
src->transport = DEFAULT_PROP_TRANSPORT;
}
static void
gst_jack_audio_src_dispose (GObject * object)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
gst_caps_replace (&src->caps, NULL);
if (src->client_name != NULL) {
g_free (src->client_name);
src->client_name = NULL;
}
if (src->port_pattern != NULL) {
g_free (src->port_pattern);
src->port_pattern = 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_CLIENT_NAME:
g_free (src->client_name);
src->client_name = g_value_dup_string (value);
break;
case PROP_PORT_PATTERN:
g_free (src->port_pattern);
src->port_pattern = g_value_dup_string (value);
break;
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;
case PROP_TRANSPORT:
src->transport = g_value_get_flags (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_CLIENT_NAME:
g_value_set_string (value, src->client_name);
break;
case PROP_PORT_PATTERN:
g_value_set_string (value, src->port_pattern);
break;
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;
case PROP_TRANSPORT:
g_value_set_flags (value, src->transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_jack_audio_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
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",
"format", G_TYPE_STRING, GST_JACK_FORMAT_STR,
"layout", G_TYPE_STRING, "interleaved",
"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 GstAudioRingBuffer *
gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
{
GstAudioRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
return buffer;
}