/* 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 deactive @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;
}