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gstreamer/ext/jack/gstjackaudiosink.c

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/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudiosink.c: jack audio sink implementation
*
* 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-jackaudiosink
* @see_also: #GstAudioBaseSink, #GstAudioRingBuffer
*
* A Sink that outputs data to Jack ports.
*
* It will create N Jack ports named out_&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 #GstJackAudioSink:connect property is set to auto, this element
* will try to connect each output port to a random physical jack input pin. In
* this mode, the sink will expose the number of physical channels on its pad
* caps.
*
* When the #GstJackAudioSink:connect property is set to none, the element will
* accept any number of input channels and will create (but not connect) an
* output 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-1.0 audiotestsrc ! jackaudiosink
* ]| Play a sine wave to using jack.
* </refsect2>
*/
#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 "gstjackaudiosink.h"
#include "gstjackringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
#define GST_CAT_DEFAULT gst_jack_audio_sink_debug
static gboolean
gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
{
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
/* remove ports we don't need */
while (sink->port_count > channels) {
jack_port_unregister (client, sink->ports[--sink->port_count]);
}
/* alloc enough output ports */
sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels);
sink->buffers = g_realloc (sink->buffers, sizeof (sample_t *) * channels);
/* create an output port for each channel */
while (sink->port_count < channels) {
gchar *name;
/* port names start from 1 and are local to the element */
name =
g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
sink->port_count + 1);
sink->ports[sink->port_count] =
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsOutput, 0);
if (sink->ports[sink->port_count] == NULL)
return FALSE;
sink->port_count++;
g_free (name);
}
return TRUE;
}
static void
gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
{
gint res, i = 0;
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
/* get rid of all ports */
while (sink->port_count) {
GST_LOG_OBJECT (sink, "unregister port %d", i);
if ((res = jack_port_unregister (client, sink->ports[i++]))) {
GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
}
sink->port_count--;
}
g_free (sink->ports);
sink->ports = NULL;
g_free (sink->buffers);
sink->buffers = NULL;
}
/* ringbuffer abstract base class */
static GType
gst_jack_ring_buffer_get_type (void)
{
static volatile 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,
"GstJackAudioSinkRingBuffer", &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 RT-safe.
*/
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstAudioRingBuffer *buf;
gint readseg, len;
guint8 *readptr;
gint i, j, flen, channels;
sample_t *data;
buf = GST_AUDIO_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* get target buffers */
for (i = 0; i < channels; i++) {
sink->buffers[i] =
(sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
}
if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
nframes, readptr, flen, channels);
data = (sample_t *) readptr;
/* the samples in the ringbuffer have the channels interleaved, we need to
* deinterleave into the jack target buffers */
for (i = 0; i < nframes; i++) {
for (j = 0; j < channels; j++) {
sink->buffers[j][i] = *data++;
}
}
/* clear written samples in the ringbuffer */
gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
/* We are not allowed to read from the ringbuffer, write silence to all
* jack output buffers */
for (i = 0; i < channels; i++) {
memset (sink->buffers[i], 0, nframes * sizeof (sample_t));
}
}
return 0;
/* ERRORS */
wrong_size:
{
GST_ERROR_OBJECT (sink, "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)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (sink, 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)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
(NULL), ("Jack changed the buffer size, which is not supported"));
return 1;
}
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
GST_DEBUG_OBJECT (sink, "shutdown");
GST_ELEMENT_ERROR (sink, 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)
{
GstJackAudioSink *sink;
jack_status_t status = 0;
const gchar *name;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "open");
if (sink->client_name) {
name = sink->client_name;
} else {
name = g_get_application_name ();
}
if (!name)
name = "GStreamer";
sink->client = gst_jack_audio_client_new (name, sink->server,
sink->jclient,
GST_JACK_CLIENT_SINK,
jack_shutdown_cb,
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
if (sink->client == NULL)
goto could_not_open;
GST_DEBUG_OBJECT (sink, "opened");
return TRUE;
/* ERRORS */
could_not_open:
{
if (status & JackServerFailed) {
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
(_("Jack server not found")),
("Cannot connect to the Jack server (status %d)", status));
} else {
GST_ELEMENT_ERROR (sink, 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 (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "close");
gst_jack_audio_sink_free_channels (sink);
gst_jack_audio_client_free (sink->client);
sink->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)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, rate, bpf, channels, res;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (sink, "acquire");
client = gst_jack_audio_client_get_client (sink->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;
channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
bpf = GST_AUDIO_INFO_BPF (&spec->info);
if (!gst_jack_audio_sink_allocate_channels (sink, 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), 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 (sink, "buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG_OBJECT (sink, "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 (sink->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 (sink->connect == GST_JACK_CONNECT_AUTO
|| sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
/* find all the physical input ports. A physical input port is a port
* associated with a hardware device. Someone needs connect to a physical
* port in order to hear something. */
if (sink->port_pattern == NULL) {
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsInput);
} else {
ports = jack_get_ports (client, sink->port_pattern, NULL,
JackPortIsInput);
}
if (ports == NULL) {
/* no ports? fine then we don't do anything except for posting a warning
* message. */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No physical input ports found, leaving ports unconnected"));
goto done;
}
for (i = 0; i < channels; i++) {
/* stop when all input ports are exhausted */
if (ports[i] == NULL) {
/* post a warning that we could not connect all ports */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No more physical ports, leaving some ports unconnected"));
break;
}
GST_DEBUG_OBJECT (sink, "try connecting to %s",
jack_port_name (sink->ports[i]));
/* connect the port to a physical port */
res = jack_connect (client, jack_port_name (sink->ports[i]), 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 = channels;
return TRUE;
/* ERRORS */
wrong_samplerate:
{
GST_ELEMENT_ERROR (sink, 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 (sink, RESOURCE, SETTINGS, (NULL),
("Cannot allocate more Jack ports"));
return FALSE;
}
could_not_activate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not activate client (%d:%s)", res, g_strerror (res)));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not connect output 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 (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
gint res;
abuf = GST_JACK_RING_BUFFER_CAST (buf);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "release");
if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_ELEMENT_WARNING (sink, 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)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "start");
if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
jack_transport_start (client);
}
return TRUE;
}
static gboolean
gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "pause");
if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
jack_transport_stop (client);
}
return TRUE;
}
static gboolean
gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "stop");
if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->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)
{
GstJackAudioSink *sink;
guint i, res = 0;
jack_latency_range_t range;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
for (i = 0; i < sink->port_count; i++) {
jack_port_get_latency_range (sink->ports[i], JackPlaybackLatency, &range);
if (range.max > res)
res = range.max;
}
GST_LOG_OBJECT (sink, "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)
{
GstJackAudioSink *sink;
guint i, res = 0;
guint latency;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (sink->client);
for (i = 0; i < sink->port_count; i++) {
latency = jack_port_get_total_latency (client, sink->ports[i]);
if (latency > res)
res = latency;
}
GST_LOG_OBJECT (sink, "delay %u", res);
return res;
}
#endif
static GstStaticPadTemplate jackaudiosink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
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 ]")
);
/* AudioSink signals and args */
enum
{
/* FILL ME */
SIGNAL_LAST
};
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
#define DEFAULT_PROP_CLIENT_NAME NULL
#define DEFAULT_PROP_PORT_PATTERN NULL
#define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS
enum
{
PROP_0,
PROP_CONNECT,
PROP_SERVER,
PROP_CLIENT,
PROP_CLIENT_NAME,
PROP_PORT_PATTERN,
PROP_TRANSPORT,
PROP_LAST
};
#define gst_jack_audio_sink_parent_class parent_class
G_DEFINE_TYPE (GstJackAudioSink, gst_jack_audio_sink, GST_TYPE_AUDIO_BASE_SINK);
static void gst_jack_audio_sink_dispose (GObject * object);
static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink,
GstCaps * filter);
static GstAudioRingBuffer
* gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink);
static void
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstAudioBaseSinkClass *gstaudiobasesink_class;
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0,
"jacksink element");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
gobject_class->dispose = gst_jack_audio_sink_dispose;
gobject_class->get_property = gst_jack_audio_sink_get_property;
gobject_class->set_property = gst_jack_audio_sink_set_property;
g_object_class_install_property (gobject_class, PROP_CONNECT,
g_param_spec_enum ("connect", "Connect",
"Specify how the output 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));
/**
* GstJackAudioSink: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));
/**
* GstJackAudioSink: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_set_static_metadata (gstelement_class, "Audio Sink (Jack)",
"Sink/Audio", "Output audio to a JACK server",
"Wim Taymans <wim.taymans@gmail.com>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&jackaudiosink_sink_factory));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
gstaudiobasesink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_sink_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_sink_init (GstJackAudioSink * sink)
{
sink->connect = DEFAULT_PROP_CONNECT;
sink->server = g_strdup (DEFAULT_PROP_SERVER);
sink->jclient = NULL;
sink->ports = NULL;
sink->port_count = 0;
sink->buffers = NULL;
sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
sink->transport = DEFAULT_PROP_TRANSPORT;
}
static void
gst_jack_audio_sink_dispose (GObject * object)
{
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
gst_caps_replace (&sink->caps, NULL);
if (sink->client_name != NULL) {
g_free (sink->client_name);
sink->client_name = NULL;
}
if (sink->port_pattern != NULL) {
g_free (sink->port_pattern);
sink->port_pattern = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (object);
switch (prop_id) {
case PROP_CLIENT_NAME:
g_free (sink->client_name);
sink->client_name = g_value_dup_string (value);
break;
case PROP_PORT_PATTERN:
g_free (sink->port_pattern);
sink->port_pattern = g_value_dup_string (value);
break;
case PROP_CONNECT:
sink->connect = g_value_get_enum (value);
break;
case PROP_SERVER:
g_free (sink->server);
sink->server = g_value_dup_string (value);
break;
case PROP_CLIENT:
if (GST_STATE (sink) == GST_STATE_NULL ||
GST_STATE (sink) == GST_STATE_READY) {
sink->jclient = g_value_get_boxed (value);
}
break;
case PROP_TRANSPORT:
sink->transport = g_value_get_flags (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (object);
switch (prop_id) {
case PROP_CLIENT_NAME:
g_value_set_string (value, sink->client_name);
break;
case PROP_PORT_PATTERN:
g_value_set_string (value, sink->port_pattern);
break;
case PROP_CONNECT:
g_value_set_enum (value, sink->connect);
break;
case PROP_SERVER:
g_value_set_string (value, sink->server);
break;
case PROP_CLIENT:
g_value_set_boxed (value, sink->jclient);
break;
case PROP_TRANSPORT:
g_value_set_flags (value, sink->transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_jack_audio_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
const char **ports;
gint min, max;
gint rate;
jack_client_t *client;
if (sink->client == NULL)
goto no_client;
client = gst_jack_audio_client_get_client (sink->client);
if (sink->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 | JackPortIsInput);
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 (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!sink->caps) {
sink->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 (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
return gst_caps_ref (sink->caps);
/* ERRORS */
no_client:
{
GST_DEBUG_OBJECT (sink, "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_sink_create_ringbuffer (GstAudioBaseSink * sink)
{
GstAudioRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
return buffer;
}
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