mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-04 23:46:43 +00:00
688 lines
19 KiB
C
688 lines
19 KiB
C
/* GStreamer
|
|
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
*
|
|
* gstjackaudioclient.c: jack audio client 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.
|
|
*/
|
|
|
|
#include <string.h>
|
|
|
|
#include "gstjackaudioclient.h"
|
|
#include "gstjack.h"
|
|
|
|
#include <gst/glib-compat-private.h>
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug);
|
|
#define GST_CAT_DEFAULT gst_jack_audio_client_debug
|
|
|
|
static void
|
|
jack_log_error (const gchar * msg)
|
|
{
|
|
GST_ERROR ("%s", msg);
|
|
}
|
|
|
|
static void
|
|
jack_info_error (const gchar * msg)
|
|
{
|
|
GST_INFO ("%s", msg);
|
|
}
|
|
|
|
void
|
|
gst_jack_audio_client_init (void)
|
|
{
|
|
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0,
|
|
"jackclient helpers");
|
|
|
|
jack_set_error_function (jack_log_error);
|
|
jack_set_info_function (jack_info_error);
|
|
}
|
|
|
|
/* a list of global connections indexed by id and server. */
|
|
G_LOCK_DEFINE_STATIC (connections_lock);
|
|
static GList *connections;
|
|
|
|
/* the connection to a server */
|
|
typedef struct
|
|
{
|
|
gint refcount;
|
|
GMutex lock;
|
|
GCond flush_cond;
|
|
|
|
/* id/server pair and the connection */
|
|
gchar *id;
|
|
gchar *server;
|
|
jack_client_t *client;
|
|
|
|
/* lists of GstJackAudioClients */
|
|
gint n_clients;
|
|
GList *src_clients;
|
|
GList *sink_clients;
|
|
|
|
/* transport state handling */
|
|
gint cur_ts;
|
|
GstState transport_state;
|
|
} GstJackAudioConnection;
|
|
|
|
/* an object sharing a jack_client_t connection. */
|
|
struct _GstJackAudioClient
|
|
{
|
|
GstJackAudioConnection *conn;
|
|
|
|
GstJackClientType type;
|
|
gboolean active;
|
|
gboolean deactivate;
|
|
gboolean server_down;
|
|
|
|
JackShutdownCallback shutdown;
|
|
JackProcessCallback process;
|
|
JackBufferSizeCallback buffer_size;
|
|
JackSampleRateCallback sample_rate;
|
|
gpointer user_data;
|
|
};
|
|
|
|
typedef struct
|
|
{
|
|
jack_nframes_t nframes;
|
|
gpointer user_data;
|
|
} JackCB;
|
|
|
|
static gboolean
|
|
jack_handle_transport_change (GstJackAudioClient * client, GstState state)
|
|
{
|
|
GstObject *obj = GST_OBJECT_PARENT (client->user_data);
|
|
guint mode;
|
|
|
|
g_object_get (obj, "transport", &mode, NULL);
|
|
if ((mode & GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) {
|
|
GST_INFO_OBJECT (obj, "requesting state change: %s",
|
|
gst_element_state_get_name (state));
|
|
gst_element_post_message (GST_ELEMENT (obj),
|
|
gst_message_new_request_state (obj, state));
|
|
return TRUE;
|
|
}
|
|
return FALSE;
|
|
}
|
|
|
|
static int
|
|
jack_process_cb (jack_nframes_t nframes, void *arg)
|
|
{
|
|
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
|
|
GList *walk;
|
|
int res = 0;
|
|
jack_transport_state_t ts = jack_transport_query (conn->client, NULL);
|
|
|
|
if (ts != conn->cur_ts) {
|
|
conn->cur_ts = ts;
|
|
switch (ts) {
|
|
case JackTransportStopped:
|
|
GST_DEBUG ("transport state is 'stopped'");
|
|
conn->transport_state = GST_STATE_PAUSED;
|
|
break;
|
|
case JackTransportStarting:
|
|
GST_DEBUG ("transport state is 'starting'");
|
|
conn->transport_state = GST_STATE_READY;
|
|
break;
|
|
case JackTransportRolling:
|
|
GST_DEBUG ("transport state is 'rolling'");
|
|
conn->transport_state = GST_STATE_PLAYING;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
GST_DEBUG ("num of clients: src=%d, sink=%d",
|
|
g_list_length (conn->src_clients), g_list_length (conn->sink_clients));
|
|
}
|
|
|
|
g_mutex_lock (&conn->lock);
|
|
/* call sources first, then sinks. Sources will either push data into the
|
|
* ringbuffer of the sinks, which will then pull the data out of it, or
|
|
* sinks will pull the data from the sources. */
|
|
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
|
|
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
|
|
/* only call active clients */
|
|
if ((client->active || client->deactivate) && client->process) {
|
|
res = client->process (nframes, client->user_data);
|
|
if (client->deactivate) {
|
|
client->deactivate = FALSE;
|
|
g_cond_signal (&conn->flush_cond);
|
|
}
|
|
}
|
|
}
|
|
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
|
|
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
|
|
/* only call active clients */
|
|
if ((client->active || client->deactivate) && client->process) {
|
|
res = client->process (nframes, client->user_data);
|
|
if (client->deactivate) {
|
|
client->deactivate = FALSE;
|
|
g_cond_signal (&conn->flush_cond);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* handle transport state requisition, do sinks first, stop after the first
|
|
* element that handled it */
|
|
if (conn->transport_state != GST_STATE_VOID_PENDING) {
|
|
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
|
|
if (jack_handle_transport_change ((GstJackAudioClient *) walk->data,
|
|
conn->transport_state)) {
|
|
conn->transport_state = GST_STATE_VOID_PENDING;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
if (conn->transport_state != GST_STATE_VOID_PENDING) {
|
|
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
|
|
if (jack_handle_transport_change ((GstJackAudioClient *) walk->data,
|
|
conn->transport_state)) {
|
|
conn->transport_state = GST_STATE_VOID_PENDING;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
g_mutex_unlock (&conn->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
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;
|
|
|
|
client->server_down = TRUE;
|
|
g_cond_signal (&conn->flush_cond);
|
|
if (client->shutdown)
|
|
client->shutdown (client->user_data);
|
|
}
|
|
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
|
|
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
|
|
client->server_down = TRUE;
|
|
g_cond_signal (&conn->flush_cond);
|
|
if (client->shutdown)
|
|
client->shutdown (client->user_data);
|
|
}
|
|
g_mutex_unlock (&conn->lock);
|
|
}
|
|
|
|
/* we error out */
|
|
static int
|
|
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
|
|
{
|
|
jack_shutdown_cb (arg);
|
|
return 0;
|
|
}
|
|
|
|
/* we error out */
|
|
static int
|
|
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
|
|
{
|
|
jack_shutdown_cb (arg);
|
|
return 0;
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
const gchar *id;
|
|
const gchar *server;
|
|
} FindData;
|
|
|
|
static gint
|
|
connection_find (GstJackAudioConnection * conn, FindData * data)
|
|
{
|
|
/* id's must match */
|
|
if (strcmp (conn->id, data->id))
|
|
return 1;
|
|
|
|
/* both the same or NULL */
|
|
if (conn->server == data->server)
|
|
return 0;
|
|
|
|
/* we cannot compare NULL */
|
|
if (conn->server == NULL || data->server == NULL)
|
|
return 1;
|
|
|
|
if (strcmp (conn->server, data->server))
|
|
return 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* make a connection with @id and @server. Returns NULL on failure with the
|
|
* status set. */
|
|
static GstJackAudioConnection *
|
|
gst_jack_audio_make_connection (const gchar * id, const gchar * server,
|
|
jack_client_t * jclient, jack_status_t * status)
|
|
{
|
|
GstJackAudioConnection *conn;
|
|
jack_options_t options;
|
|
gint res;
|
|
|
|
*status = 0;
|
|
|
|
GST_DEBUG ("new client %s, connecting to server %s", id,
|
|
GST_STR_NULL (server));
|
|
|
|
/* never start a server */
|
|
options = JackNoStartServer;
|
|
/* if we have a servername, use it */
|
|
if (server != NULL)
|
|
options |= JackServerName;
|
|
/* open the client */
|
|
if (jclient == NULL)
|
|
jclient = jack_client_open (id, options, status, server);
|
|
if (jclient == NULL)
|
|
goto could_not_open;
|
|
|
|
/* now create object */
|
|
conn = g_new (GstJackAudioConnection, 1);
|
|
conn->refcount = 1;
|
|
g_mutex_init (&conn->lock);
|
|
g_cond_init (&conn->flush_cond);
|
|
conn->id = g_strdup (id);
|
|
conn->server = g_strdup (server);
|
|
conn->client = jclient;
|
|
conn->n_clients = 0;
|
|
conn->src_clients = NULL;
|
|
conn->sink_clients = NULL;
|
|
conn->cur_ts = -1;
|
|
conn->transport_state = GST_STATE_VOID_PENDING;
|
|
|
|
/* set our callbacks */
|
|
jack_set_process_callback (jclient, jack_process_cb, conn);
|
|
/* these callbacks cause us to error */
|
|
jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn);
|
|
jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn);
|
|
jack_on_shutdown (jclient, jack_shutdown_cb, conn);
|
|
|
|
/* all callbacks are set, activate the client */
|
|
GST_INFO ("activate jack_client %p", jclient);
|
|
if ((res = jack_activate (jclient)))
|
|
goto could_not_activate;
|
|
|
|
GST_DEBUG ("opened connection %p", conn);
|
|
|
|
return conn;
|
|
|
|
/* ERRORS */
|
|
could_not_open:
|
|
{
|
|
GST_DEBUG ("failed to open jack client, %d", *status);
|
|
return NULL;
|
|
}
|
|
could_not_activate:
|
|
{
|
|
GST_ERROR ("Could not activate client (%d)", res);
|
|
*status = JackFailure;
|
|
g_mutex_clear (&conn->lock);
|
|
g_free (conn->id);
|
|
g_free (conn->server);
|
|
g_free (conn);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstJackAudioConnection *
|
|
gst_jack_audio_get_connection (const gchar * id, const gchar * server,
|
|
jack_client_t * jclient, jack_status_t * status)
|
|
{
|
|
GstJackAudioConnection *conn;
|
|
GList *found;
|
|
FindData data;
|
|
|
|
GST_DEBUG ("getting connection for id %s, server %s", id,
|
|
GST_STR_NULL (server));
|
|
|
|
data.id = id;
|
|
data.server = server;
|
|
|
|
G_LOCK (connections_lock);
|
|
found =
|
|
g_list_find_custom (connections, &data, (GCompareFunc) connection_find);
|
|
if (found != NULL && jclient != NULL) {
|
|
/* we found it, increase refcount and return it */
|
|
conn = (GstJackAudioConnection *) found->data;
|
|
conn->refcount++;
|
|
|
|
GST_DEBUG ("found connection %p", conn);
|
|
} else {
|
|
/* make new connection */
|
|
conn = gst_jack_audio_make_connection (id, server, jclient, status);
|
|
if (conn != NULL) {
|
|
GST_DEBUG ("created connection %p", conn);
|
|
/* add to list on success */
|
|
connections = g_list_prepend (connections, conn);
|
|
} else {
|
|
GST_WARNING ("could not create connection");
|
|
}
|
|
}
|
|
G_UNLOCK (connections_lock);
|
|
|
|
return conn;
|
|
}
|
|
|
|
static void
|
|
gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
|
|
{
|
|
gint res;
|
|
gboolean zero;
|
|
|
|
GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount);
|
|
|
|
G_LOCK (connections_lock);
|
|
conn->refcount--;
|
|
if ((zero = (conn->refcount == 0))) {
|
|
GST_DEBUG ("closing connection %p", conn);
|
|
/* remove from list, we can release the mutex after removing the connection
|
|
* from the list because after that, nobody can access the connection anymore. */
|
|
connections = g_list_remove (connections, conn);
|
|
}
|
|
G_UNLOCK (connections_lock);
|
|
|
|
/* if we are zero, close and cleanup the connection */
|
|
if (zero) {
|
|
/* don't use conn->lock here. two reasons:
|
|
*
|
|
* 1) its not necessary: jack_deactivate() will not return until the JACK thread
|
|
* associated with this connection is cleaned up by a thread join, hence
|
|
* no more callbacks can occur or be in progress.
|
|
*
|
|
* 2) it would deadlock anyway, because jack_deactivate() will sleep
|
|
* waiting for the JACK thread, and can thus cause deadlock in
|
|
* jack_process_cb()
|
|
*/
|
|
GST_INFO ("deactivate jack_client %p", conn->client);
|
|
if ((res = jack_deactivate (conn->client))) {
|
|
/* we only warn, this means the server is probably shut down and the client
|
|
* is gone anyway. */
|
|
GST_WARNING ("Could not deactivate Jack client (%d)", res);
|
|
}
|
|
/* close connection */
|
|
if ((res = jack_client_close (conn->client))) {
|
|
/* we assume the client is gone. */
|
|
GST_WARNING ("close failed (%d)", res);
|
|
}
|
|
|
|
/* free resources */
|
|
g_mutex_clear (&conn->lock);
|
|
g_cond_clear (&conn->flush_cond);
|
|
g_free (conn->id);
|
|
g_free (conn->server);
|
|
g_free (conn);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_jack_audio_connection_add_client (GstJackAudioConnection * conn,
|
|
GstJackAudioClient * client)
|
|
{
|
|
g_mutex_lock (&conn->lock);
|
|
switch (client->type) {
|
|
case GST_JACK_CLIENT_SOURCE:
|
|
conn->src_clients = g_list_append (conn->src_clients, client);
|
|
conn->n_clients++;
|
|
break;
|
|
case GST_JACK_CLIENT_SINK:
|
|
conn->sink_clients = g_list_append (conn->sink_clients, client);
|
|
conn->n_clients++;
|
|
break;
|
|
default:
|
|
g_warning ("trying to add unknown client type");
|
|
break;
|
|
}
|
|
g_mutex_unlock (&conn->lock);
|
|
}
|
|
|
|
static void
|
|
gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn,
|
|
GstJackAudioClient * client)
|
|
{
|
|
g_mutex_lock (&conn->lock);
|
|
switch (client->type) {
|
|
case GST_JACK_CLIENT_SOURCE:
|
|
conn->src_clients = g_list_remove (conn->src_clients, client);
|
|
conn->n_clients--;
|
|
break;
|
|
case GST_JACK_CLIENT_SINK:
|
|
conn->sink_clients = g_list_remove (conn->sink_clients, client);
|
|
conn->n_clients--;
|
|
break;
|
|
default:
|
|
g_warning ("trying to remove unknown client type");
|
|
break;
|
|
}
|
|
g_mutex_unlock (&conn->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_jack_audio_client_get:
|
|
* @id: the client id
|
|
* @server: the server to connect to or NULL for the default server
|
|
* @type: the client type
|
|
* @shutdown: a callback when the jack server shuts down
|
|
* @process: a callback when samples are available
|
|
* @buffer_size: a callback when the buffer_size changes
|
|
* @sample_rate: a callback when the sample_rate changes
|
|
* @user_data: user data passed to the callbacks
|
|
* @status: pointer to hold the jack status code in case of errors
|
|
*
|
|
* Get the jack client connection for @id and @server. Connections to the same
|
|
* @id and @server will receive the same physical Jack client connection and
|
|
* will therefore be scheduled in the same process callback.
|
|
*
|
|
* Returns: a #GstJackAudioClient.
|
|
*/
|
|
GstJackAudioClient *
|
|
gst_jack_audio_client_new (const gchar * id, const gchar * server,
|
|
jack_client_t * jclient, GstJackClientType type,
|
|
void (*shutdown) (void *arg), JackProcessCallback process,
|
|
JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate,
|
|
gpointer user_data, jack_status_t * status)
|
|
{
|
|
GstJackAudioClient *client;
|
|
GstJackAudioConnection *conn;
|
|
|
|
g_return_val_if_fail (id != NULL, NULL);
|
|
g_return_val_if_fail (status != NULL, NULL);
|
|
|
|
/* first get a connection for the id/server pair */
|
|
conn = gst_jack_audio_get_connection (id, server, jclient, status);
|
|
if (conn == NULL)
|
|
goto no_connection;
|
|
|
|
GST_INFO ("new client %s", id);
|
|
|
|
/* make new client using the connection */
|
|
client = g_new (GstJackAudioClient, 1);
|
|
client->active = client->deactivate = FALSE;
|
|
client->conn = conn;
|
|
client->type = type;
|
|
client->shutdown = shutdown;
|
|
client->process = process;
|
|
client->buffer_size = buffer_size;
|
|
client->sample_rate = sample_rate;
|
|
client->user_data = user_data;
|
|
client->server_down = FALSE;
|
|
|
|
/* add the client to the connection */
|
|
gst_jack_audio_connection_add_client (conn, client);
|
|
|
|
return client;
|
|
|
|
/* ERRORS */
|
|
no_connection:
|
|
{
|
|
GST_DEBUG ("Could not get server connection (%d)", *status);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_jack_audio_client_free:
|
|
* @client: a #GstJackAudioClient
|
|
*
|
|
* Free the resources used by @client.
|
|
*/
|
|
void
|
|
gst_jack_audio_client_free (GstJackAudioClient * client)
|
|
{
|
|
GstJackAudioConnection *conn;
|
|
|
|
g_return_if_fail (client != NULL);
|
|
|
|
GST_INFO ("free client");
|
|
|
|
conn = client->conn;
|
|
|
|
/* remove from connection first so that it's not scheduled anymore after this
|
|
* call */
|
|
gst_jack_audio_connection_remove_client (conn, client);
|
|
gst_jack_audio_unref_connection (conn);
|
|
|
|
g_free (client);
|
|
}
|
|
|
|
/**
|
|
* gst_jack_audio_client_get_client:
|
|
* @client: a #GstJackAudioClient
|
|
*
|
|
* Get the jack audio client for @client. This function is used to perform
|
|
* operations on the jack server from this client.
|
|
*
|
|
* Returns: The jack audio client.
|
|
*/
|
|
jack_client_t *
|
|
gst_jack_audio_client_get_client (GstJackAudioClient * client)
|
|
{
|
|
g_return_val_if_fail (client != NULL, NULL);
|
|
|
|
/* no lock needed, the connection and the client does not change
|
|
* once the client is created. */
|
|
return client->conn->client;
|
|
}
|
|
|
|
/**
|
|
* gst_jack_audio_client_set_active:
|
|
* @client: a #GstJackAudioClient
|
|
* @active: new mode for the client
|
|
*
|
|
* Activate or deactivate @client. When a client is activated it will receive
|
|
* callbacks when data should be processed.
|
|
*
|
|
* Returns: 0 if all ok.
|
|
*/
|
|
gint
|
|
gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
|
|
{
|
|
g_return_val_if_fail (client != NULL, -1);
|
|
|
|
/* make sure that we are not dispatching the client */
|
|
g_mutex_lock (&client->conn->lock);
|
|
if (client->active && !active) {
|
|
/* we need to process once more to flush the port */
|
|
client->deactivate = TRUE;
|
|
|
|
/* need to wait for process_cb run once more */
|
|
while (client->deactivate && !client->server_down)
|
|
g_cond_wait (&client->conn->flush_cond, &client->conn->lock);
|
|
}
|
|
client->active = active;
|
|
g_mutex_unlock (&client->conn->lock);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* gst_jack_audio_client_get_transport_state:
|
|
* @client: a #GstJackAudioClient
|
|
*
|
|
* Check the current transport state. The client can use this to request a state
|
|
* change from the application.
|
|
*
|
|
* Returns: the state, %GST_STATE_VOID_PENDING for no change in the transport
|
|
* state
|
|
*/
|
|
GstState
|
|
gst_jack_audio_client_get_transport_state (GstJackAudioClient * client)
|
|
{
|
|
GstState state = client->conn->transport_state;
|
|
|
|
client->conn->transport_state = GST_STATE_VOID_PENDING;
|
|
return state;
|
|
}
|
|
|
|
/**
|
|
* gst_jack_audio_client_get_port_names_from_string:
|
|
* @jclient: a jack_client_t handle
|
|
* @port_names: comma-separated jack port name(s)
|
|
* @port_flags: JackPortFlags
|
|
*
|
|
* Returns: a newly-allocated %NULL-terminated array of strings or %NULL
|
|
* if @port_names contains invalid port name. Use g_strfreev() to free it.
|
|
*/
|
|
gchar **
|
|
gst_jack_audio_client_get_port_names_from_string (jack_client_t * jclient,
|
|
const gchar * port_names, gint port_flags)
|
|
{
|
|
gchar **p = NULL;
|
|
guint i, len;
|
|
|
|
g_return_val_if_fail (jclient != NULL, NULL);
|
|
|
|
if (!port_names)
|
|
return NULL;
|
|
|
|
p = g_strsplit (port_names, ",", 0);
|
|
len = g_strv_length (p);
|
|
|
|
if (len < 1)
|
|
goto invalid;
|
|
|
|
for (i = 0; i < len; i++) {
|
|
jack_port_t *port = jack_port_by_name (jclient, p[i]);
|
|
int flags;
|
|
|
|
if (!port) {
|
|
GST_WARNING ("Couldn't get jack port by name %s", p[i]);
|
|
goto invalid;
|
|
}
|
|
|
|
flags = jack_port_flags (port);
|
|
if ((flags & port_flags) != port_flags) {
|
|
GST_WARNING ("Port flags 0x%x doesn't match expected flags 0x%x",
|
|
flags, port_flags);
|
|
goto invalid;
|
|
}
|
|
}
|
|
|
|
return p;
|
|
|
|
invalid:
|
|
g_strfreev (p);
|
|
return NULL;
|
|
}
|