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_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 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_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;
}
}
/* 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 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 void
gst_jack_audio_src_dispose (GObject * object)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
gst_caps_replace (&src->caps, NULL);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
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");
return TRUE;
}
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"));
}
/* 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;
}
/* 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;
gint len;
guint8 *writeptr;
gint writeseg;
gint channels, i, j, flen;
sample_t *data;
buf = GST_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
channels = buf->spec.channels;
/* get input buffers */
for (i = 0; i < channels; i++)
src->buffers[i] =
(sample_t *) jack_port_get_buffer (src->ports[i], nframes);
if (gst_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_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;
}
}
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;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
/* 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;
}
/* 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");
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;
}
}
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 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_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 guint
gst_jack_ring_buffer_delay (GstRingBuffer * 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;
}
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;
}
/* 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;
}
}
/* 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;
}
}
