static guint
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
guint i, res = 0;
#ifdef HAVE_JACK_0_120_2
jack_latency_range_t range;
#else
guint latency;
#endif
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++) {
#ifdef HAVE_JACK_0_120_2
jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
if (range.max > res)
res = range.max;
#else
latency = jack_port_get_total_latency (client, src->ports[i]);
if (latency > res)
res = latency;
#endif
}
GST_DEBUG_OBJECT (src, "delay %u", res);
return res;
}
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 GstCaps *
gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
const char **ports;
gint min, max;
gint rate;
jack_client_t *client;
if (src->client == NULL)
goto no_client;
client = gst_jack_audio_client_get_client (src->client);
if (src->connect == GST_JACK_CONNECT_AUTO) {
/* get a port count, this is the number of channels we can automatically
* connect. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput);
max = 0;
if (ports != NULL) {
for (; ports[max]; max++);
free (ports);
} else
max = 0;
} else {
/* we allow any number of pads, something else is going to connect the
* pads. */
max = G_MAXINT;
}
min = MIN (1, max);
rate = jack_get_sample_rate (client);
GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!src->caps) {
src->caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 32,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
return gst_caps_ref (src->caps);
/* ERRORS */
no_client:
{
GST_DEBUG_OBJECT (src, "device not open, using template caps");
/* base class will get template caps for us when we return NULL */
return NULL;
}
}
static gboolean
gst_jack_ring_buffer_stop (GstRingBuffer * 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;
}
static gboolean
gst_jack_ring_buffer_pause (GstRingBuffer * 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 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;
}
static guint
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
guint i, res = 0, latency;
jack_client_t *client;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (src->client);
for (i = 0; i < src->port_count; i++) {
latency = jack_port_get_total_latency (client, src->ports[i]);
if (latency > res)
res = latency;
}
GST_DEBUG_OBJECT (src, "delay %u", res);
return res;
}
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
return TRUE;
}
static gboolean
gst_jack_ring_buffer_start (GstRingBuffer * 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 (GstRingBuffer * 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 (GstRingBuffer * 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 (GstRingBuffer * 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 (GstRingBuffer * 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;
}
/* 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;
}
}
