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
print_plugin (const gchar * marker, GstRegistry * registry, GstPlugin * plugin)
{
const gchar *name;
GList *features, *f;
name = gst_plugin_get_name (plugin);
GST_DEBUG ("%s: plugin %p %d %s file: %s", marker, plugin,
GST_OBJECT_REFCOUNT (plugin), name,
GST_STR_NULL (gst_plugin_get_filename (plugin)));
features = gst_registry_get_feature_list_by_plugin (registry, name);
for (f = features; f != NULL; f = f->next) {
GstPluginFeature *feature;
feature = GST_PLUGIN_FEATURE (f->data);
GST_LOG ("%s: feature: %p %s", marker, feature,
GST_OBJECT_NAME (feature));
}
gst_plugin_feature_list_free (features);
}
/**
* gst_element_factory_make:
* @factoryname: a named factory to instantiate
* @name: (allow-none): name of new element, or NULL to automatically create
* a unique name
*
* Create a new element of the type defined by the given element factory.
* If name is NULL, then the element will receive a guaranteed unique name,
* consisting of the element factory name and a number.
* If name is given, it will be given the name supplied.
*
* Returns: (transfer full): new #GstElement or NULL if unable to create element
*/
GstElement *
gst_element_factory_make (const gchar * factoryname, const gchar * name)
{
GstElementFactory *factory;
GstElement *element;
g_return_val_if_fail (factoryname != NULL, NULL);
g_return_val_if_fail (gst_is_initialized (), NULL);
GST_LOG ("gstelementfactory: make \"%s\" \"%s\"",
factoryname, GST_STR_NULL (name));
factory = gst_element_factory_find (factoryname);
if (factory == NULL)
goto no_factory;
GST_LOG_OBJECT (factory, "found factory %p", factory);
element = gst_element_factory_create (factory, name);
if (element == NULL)
goto create_failed;
gst_object_unref (factory);
return element;
/* ERRORS */
no_factory:
{
GST_INFO ("no such element factory \"%s\"!", factoryname);
return NULL;
}
create_failed:
{
GST_INFO_OBJECT (factory, "couldn't create instance!");
gst_object_unref (factory);
return NULL;
}
}
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;
if (client->shutdown)
client->shutdown (client->user_data);
}
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
if (client->shutdown)
client->shutdown (client->user_data);
}
g_mutex_unlock (conn->lock);
}
static gboolean
gst_mms_do_seek (GstBaseSrc * src, GstSegment * segment)
{
mms_off_t start;
GstMMS *mmssrc = GST_MMS (src);
if (segment->format == GST_FORMAT_TIME) {
if (!mmsx_time_seek (NULL, mmssrc->connection,
(double) segment->start / GST_SECOND)) {
GST_LOG_OBJECT (mmssrc, "mmsx_time_seek() failed");
return FALSE;
}
start = mmsx_get_current_pos (mmssrc->connection);
GST_INFO_OBJECT (mmssrc, "sought to %" GST_TIME_FORMAT ", offset after "
"seek: %" G_GINT64_FORMAT, GST_TIME_ARGS (segment->start), start);
} else if (segment->format == GST_FORMAT_BYTES) {
start = mmsx_seek (NULL, mmssrc->connection, segment->start, SEEK_SET);
/* mmsx_seek will close and reopen the connection when seeking with the
mmsh protocol, if the reopening fails this is indicated with -1 */
if (start == -1) {
GST_DEBUG_OBJECT (mmssrc, "connection broken during seek");
return FALSE;
}
GST_INFO_OBJECT (mmssrc, "sought to: %" G_GINT64_FORMAT " bytes, "
"result: %" G_GINT64_FORMAT, segment->start, start);
} else {
GST_DEBUG_OBJECT (mmssrc, "unsupported seek segment format: %s",
GST_STR_NULL (gst_format_get_name (segment->format)));
return FALSE;
}
gst_segment_init (segment, GST_FORMAT_BYTES);
gst_segment_set_seek (segment, segment->rate, GST_FORMAT_BYTES,
segment->flags, GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE,
segment->stop, NULL);
return TRUE;
}
void
totem_gst_message_print (GstMessage *msg,
GstElement *play,
const char *filename)
{
GError *err = NULL;
char *dbg = NULL;
g_return_if_fail (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR);
if (play != NULL) {
g_return_if_fail (filename != NULL);
GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN_CAST (play),
GST_DEBUG_GRAPH_SHOW_ALL ^ GST_DEBUG_GRAPH_SHOW_NON_DEFAULT_PARAMS,
filename);
}
gst_message_parse_error (msg, &err, &dbg);
if (err) {
char *uri;
g_object_get (play, "uri", &uri, NULL);
GST_ERROR ("message = %s", GST_STR_NULL (err->message));
GST_ERROR ("domain = %d (%s)", err->domain,
GST_STR_NULL (g_quark_to_string (err->domain)));
GST_ERROR ("code = %d", err->code);
GST_ERROR ("debug = %s", GST_STR_NULL (dbg));
GST_ERROR ("source = %" GST_PTR_FORMAT, msg->src);
GST_ERROR ("uri = %s", GST_STR_NULL (uri));
g_free (uri);
g_message ("Error: %s\n%s\n", GST_STR_NULL (err->message),
GST_STR_NULL (dbg));
g_error_free (err);
}
g_free (dbg);
}
static gboolean
gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpG722Depay *rtpg722depay;
gint clock_rate, payload, samplerate;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_G722:
channels = 1;
clock_rate = 8000;
samplerate = 16000;
break;
default:
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
samplerate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
if (clock_rate == 8000)
samplerate = 16000;
if (samplerate == 0)
samplerate = clock_rate;
channels =
gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
rtpg722depay->rate = samplerate;
rtpg722depay->channels = channels;
srccaps = gst_caps_new_simple ("audio/G722",
"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
if (order) {
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
order->pos);
} else {
GstAudioChannelPosition *pos;
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
pos = gst_rtp_channels_create_default (channels);
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
g_free (pos);
}
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static gboolean
gst_gsettings_audio_sink_change_child (GstGSettingsAudioSink * sink)
{
const gchar *key = NULL;
gchar *new_string;
GError *err = NULL;
GstElement *new_kid;
GST_OBJECT_LOCK (sink);
switch (sink->profile) {
case GST_GSETTINGS_AUDIOSINK_PROFILE_SOUNDS:
key = GST_GSETTINGS_KEY_SOUNDS_AUDIOSINK;
break;
case GST_GSETTINGS_AUDIOSINK_PROFILE_MUSIC:
key = GST_GSETTINGS_KEY_MUSIC_AUDIOSINK;
break;
case GST_GSETTINGS_AUDIOSINK_PROFILE_CHAT:
key = GST_GSETTINGS_KEY_CHAT_AUDIOSINK;
break;
default:
break;
}
new_string = g_settings_get_string (sink->settings, key);
if (new_string != NULL && sink->gsettings_str != NULL &&
(strlen (new_string) == 0 ||
strcmp (sink->gsettings_str, new_string) == 0)) {
g_free (new_string);
GST_DEBUG_OBJECT (sink,
"GSettings key was updated, but it didn't change. Ignoring");
GST_OBJECT_UNLOCK (sink);
return TRUE;
}
GST_OBJECT_UNLOCK (sink);
GST_DEBUG_OBJECT (sink, "GSettings key changed from '%s' to '%s'",
GST_STR_NULL (sink->gsettings_str), GST_STR_NULL (new_string));
if (new_string) {
new_kid = gst_parse_bin_from_description (new_string, TRUE, &err);
if (err) {
GST_ERROR_OBJECT (sink, "error creating bin '%s': %s", new_string,
err->message);
g_error_free (err);
}
} else {
new_kid = NULL;
}
if (new_kid == NULL) {
GST_ELEMENT_ERROR (sink, LIBRARY, SETTINGS, (NULL),
("Failed to render audio sink from GSettings"));
goto fail;
}
if (!gst_switch_sink_set_child (GST_SWITCH_SINK (sink), new_kid)) {
GST_WARNING_OBJECT (sink, "Failed to update child element");
goto fail;
}
g_free (sink->gsettings_str);
sink->gsettings_str = new_string;
return TRUE;
fail:
g_free (new_string);
return FALSE;
}
static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
gchar *name = gst_pulse_client_name ();
pa_threaded_mainloop_lock (pulsesrc->mainloop);
g_assert (!pulsesrc->context);
g_assert (!pulsesrc->stream);
GST_DEBUG_OBJECT (pulsesrc, "opening device");
if (!(pulsesrc->context =
pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop),
name))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"),
(NULL));
goto unlock_and_fail;
}
pa_context_set_state_callback (pulsesrc->context,
gst_pulsesrc_context_state_cb, pulsesrc);
GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
GST_STR_NULL (pulsesrc->server));
if (pa_context_connect (pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
for (;;) {
pa_context_state_t state;
state = pa_context_get_state (pulsesrc->context);
if (!PA_CONTEXT_IS_GOOD (state)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (state == PA_CONTEXT_READY)
break;
/* Wait until the context is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
GST_DEBUG_OBJECT (pulsesrc, "connected");
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
g_free (name);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
gst_pulsesrc_destroy_context (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
g_free (name);
return FALSE;
}
}
/* 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;
}
}
int
main (int argc, char **argv)
{
GstBus *bus;
GOptionContext *ctx;
GIOChannel *io_stdin;
GError *err = NULL;
gboolean res;
GOptionEntry options[] = {
{NULL}
};
GThread *rthread;
/* Clear application state */
memset (state, 0, sizeof (*state));
state->animate = TRUE;
/* must initialise the threading system before using any other GLib funtion */
if (!g_thread_supported ())
g_thread_init (NULL);
ctx = g_option_context_new ("[ADDITIONAL ARGUMENTS]");
g_option_context_add_main_entries (ctx, options, NULL);
g_option_context_add_group (ctx, gst_init_get_option_group ());
if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
exit (1);
}
g_option_context_free (ctx);
if (argc != 2) {
g_print ("Usage: %s <URI> or <PIPELINE-DESCRIPTION>\n", argv[0]);
exit (1);
}
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* initialize inter thread comunnication */
init_intercom (state);
TRACE_VC_MEMORY ("state 0");
if (!(rthread = g_thread_new ("render", (GThreadFunc) render_func, NULL))) {
g_print ("Render thread create failed\n");
exit (1);
}
/* Initialize player */
if (gst_uri_is_valid (argv[1])) {
res = init_playbin_player (state, argv[1]);
} else {
res = init_parse_launch_player (state, argv[1]);
}
if (!res)
goto done;
/* Create a GLib Main Loop and set it to run */
state->main_loop = g_main_loop_new (NULL, FALSE);
/* Add a keyboard watch so we get notified of keystrokes */
io_stdin = g_io_channel_unix_new (fileno (stdin));
g_io_add_watch (io_stdin, G_IO_IN, (GIOFunc) handle_keyboard, state);
g_io_channel_unref (io_stdin);
/* *INDENT-OFF* */
g_print ("Available commands: \n"
" a - Toggle animation \n"
" p - Pause playback \n"
" r - Resume playback \n"
" l - Query position/duration\n"
" f - Seek 30 seconds forward \n"
" b - Seek 30 seconds backward \n"
" q - Quit \n");
/* *INDENT-ON* */
/* Connect the bus handlers */
bus = gst_element_get_bus (state->pipeline);
gst_bus_set_sync_handler (bus, (GstBusSyncHandler) bus_sync_handler, state,
NULL);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
gst_bus_enable_sync_message_emission (bus);
g_signal_connect (G_OBJECT (bus), "message::error", (GCallback) error_cb,
state);
g_signal_connect (G_OBJECT (bus), "message::buffering",
(GCallback) buffering_cb, state);
g_signal_connect (G_OBJECT (bus), "message::eos", (GCallback) eos_cb, state);
g_signal_connect (G_OBJECT (bus), "message::qos", (GCallback) qos_cb, state);
g_signal_connect (G_OBJECT (bus), "message::state-changed",
(GCallback) state_changed_cb, state);
gst_object_unref (bus);
/* Make player start playing */
gst_element_set_state (state->pipeline, GST_STATE_PLAYING);
/* Start the mainloop */
state->main_loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (state->main_loop);
done:
/* Release pipeline */
if (state->pipeline) {
gst_element_set_state (state->pipeline, GST_STATE_NULL);
if (state->vsink) {
gst_object_unref (state->vsink);
state->vsink = NULL;
}
gst_object_unref (state->pipeline);
}
/* Unref the mainloop */
if (state->main_loop) {
g_main_loop_unref (state->main_loop);
}
/* Stop rendering thread */
state->running = FALSE;
g_thread_join (rthread);
terminate_intercom (state);
TRACE_VC_MEMORY ("at exit");
return 0;
}
static gboolean
gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpL16Depay *rtpL16depay;
gint clock_rate, payload;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_L16_STEREO:
channels = 2;
clock_rate = 44100;
break;
case GST_RTP_PAYLOAD_L16_MONO:
channels = 1;
clock_rate = 44100;
break;
default:
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
channels =
gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
info = &rtpL16depay->info;
gst_audio_info_init (info);
info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
info->rate = clock_rate;
info->channels = channels;
info->bpf = (info->finfo->width / 8) * channels;
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
rtpL16depay->order = order;
if (order) {
memcpy (info->position, order->pos,
sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (info->position, info->channels);
} else {
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
gst_rtp_channels_create_default (channels, info->position);
}
srccaps = gst_audio_info_to_caps (info);
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static gboolean
gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstAmcAudioDec *self;
GstStructure *s;
GstAmcFormat *format;
const gchar *mime;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
gchar *format_string;
gint rate, channels;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
is_format_change |= (!self->input_caps
|| !gst_caps_is_equal (self->input_caps, caps));
needs_disable = self->started;
/* If the component is not started and a real format change happens
* we have to restart the component. If no real format change
* happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
/* Framerate or something minor changed */
self->input_caps_changed = TRUE;
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
gst_amc_audio_dec_drain (self);
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_audio_dec_close (GST_AUDIO_DECODER (self));
if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to open codec again");
return FALSE;
}
if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to start codec again");
}
}
/* srcpad task is not running at this point */
mime = caps_to_mime (caps);
if (!mime) {
GST_ERROR_OBJECT (self, "Failed to convert caps to mime");
return FALSE;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Failed to get rate/channels");
return FALSE;
}
format = gst_amc_format_new_audio (mime, rate, channels, &err);
if (!format) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
/* FIXME: These buffers needs to be valid until the codec is stopped again */
g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
if (gst_structure_has_field (s, "codec_data")) {
const GValue *h = gst_structure_get_value (s, "codec_data");
GstBuffer *codec_data = gst_value_get_buffer (h);
GstMapInfo minfo;
guint8 *data;
gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (codec_data, &minfo);
} else if (gst_structure_has_field (s, "streamheader")) {
const GValue *sh = gst_structure_get_value (s, "streamheader");
gint nsheaders = gst_value_array_get_size (sh);
GstBuffer *buf;
const GValue *h;
gint i, j;
gchar *fname;
GstMapInfo minfo;
guint8 *data;
for (i = 0, j = 0; i < nsheaders; i++) {
h = gst_value_array_get_value (sh, i);
buf = gst_value_get_buffer (h);
if (strcmp (mime, "audio/vorbis") == 0) {
guint8 header_type;
gst_buffer_extract (buf, 0, &header_type, 1);
/* Only use the identification and setup packets */
if (header_type != 0x01 && header_type != 0x05)
continue;
}
fname = g_strdup_printf ("csd-%d", j);
gst_buffer_map (buf, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (buf, &minfo);
g_free (fname);
j++;
}
}
format_string = gst_amc_format_to_string (format, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
GST_STR_NULL (format_string));
g_free (format_string);
if (!gst_amc_codec_configure (self->codec, format, 0, &err)) {
GST_ERROR_OBJECT (self, "Failed to configure codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
gst_amc_format_free (format);
if (!gst_amc_codec_start (self->codec, &err)) {
GST_ERROR_OBJECT (self, "Failed to start codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->spf = -1;
/* TODO: Implement for other codecs too */
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int (s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
gint layer = -1, mpegaudioversion = -1;
gst_structure_get_int (s, "layer", &layer);
gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
if (layer == 1)
self->spf = 384;
else if (layer == 2)
self->spf = 1152;
else if (layer == 3 && mpegaudioversion != -1)
self->spf = (mpegaudioversion == 1 ? 1152 : 576);
}
}
self->started = TRUE;
self->input_caps_changed = TRUE;
/* Start the srcpad loop again */
self->flushing = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
return TRUE;
}
static gboolean
do_toggle_element (GstGConfVideoSink * sink)
{
GstPad *targetpad;
gchar *new_gconf_str;
GstState cur, next;
new_gconf_str = gst_gconf_get_string (GST_GCONF_VIDEOSINK_KEY);
if (new_gconf_str != NULL && sink->gconf_str != NULL &&
(strlen (new_gconf_str) == 0 ||
strcmp (sink->gconf_str, new_gconf_str) == 0)) {
g_free (new_gconf_str);
GST_DEBUG_OBJECT (sink, "GConf key was updated, but it didn't change");
return TRUE;
}
/* Sometime, it would be lovely to allow sink changes even when
* already running, but this involves sending an appropriate new-segment
* and possibly prerolling etc */
GST_OBJECT_LOCK (sink);
cur = GST_STATE (sink);
next = GST_STATE_PENDING (sink);
GST_OBJECT_UNLOCK (sink);
if (cur > GST_STATE_READY || next == GST_STATE_PAUSED) {
GST_DEBUG_OBJECT (sink,
"Auto-sink is already running. Ignoring GConf change");
g_free (new_gconf_str);
return TRUE;
}
GST_DEBUG_OBJECT (sink, "GConf key changed: '%s' to '%s'",
GST_STR_NULL (sink->gconf_str), GST_STR_NULL (new_gconf_str));
g_free (sink->gconf_str);
sink->gconf_str = new_gconf_str;
/* kill old element */
if (sink->kid) {
GST_DEBUG_OBJECT (sink, "Removing old kid");
gst_element_set_state (sink->kid, GST_STATE_NULL);
gst_bin_remove (GST_BIN (sink), sink->kid);
sink->kid = NULL;
}
GST_DEBUG_OBJECT (sink, "Creating new kid");
if (!(sink->kid = gst_gconf_get_default_video_sink ())) {
GST_ELEMENT_ERROR (sink, LIBRARY, SETTINGS, (NULL),
("Failed to render video sink from GConf"));
return FALSE;
}
gst_element_set_state (sink->kid, GST_STATE (sink));
gst_bin_add (GST_BIN (sink), sink->kid);
/* re-attach ghostpad */
GST_DEBUG_OBJECT (sink, "Creating new ghostpad");
targetpad = gst_element_get_static_pad (sink->kid, "sink");
gst_ghost_pad_set_target (GST_GHOST_PAD (sink->pad), targetpad);
gst_object_unref (targetpad);
GST_DEBUG_OBJECT (sink, "done changing gconf video sink");
return TRUE;
}
GstMixerOptions *
gst_sunaudiomixer_options_new (GstSunAudioMixerCtrl * mixer, gint track_num)
{
GstMixerOptions *opts;
GstSunAudioMixerOptions *sun_opts;
GstMixerTrack *track;
const gchar *label;
gint i;
struct audio_info audioinfo;
if ((mixer == NULL) || (mixer->mixer_fd == -1)) {
g_warning ("mixer not initialized");
return NULL;
}
if (track_num != GST_SUNAUDIO_TRACK_RECSRC) {
g_warning ("invalid options track");
return (NULL);
}
label = N_("Record Source");
opts = g_object_new (GST_TYPE_SUNAUDIO_MIXER_OPTIONS,
"untranslated-label", label, NULL);
sun_opts = GST_SUNAUDIO_MIXER_OPTIONS (opts);
track = GST_MIXER_TRACK (opts);
GST_DEBUG_OBJECT (opts, "New mixer options, track %d: %s",
track_num, GST_STR_NULL (label));
/* save off names for the record sources */
sun_opts->names[0] = g_quark_from_string (_("Microphone"));
sun_opts->names[1] = g_quark_from_string (_("Line In"));
sun_opts->names[2] = g_quark_from_string (_("Internal CD"));
sun_opts->names[3] = g_quark_from_string (_("SPDIF In"));
sun_opts->names[4] = g_quark_from_string (_("AUX 1 In"));
sun_opts->names[5] = g_quark_from_string (_("AUX 2 In"));
sun_opts->names[6] = g_quark_from_string (_("Codec Loopback"));
sun_opts->names[7] = g_quark_from_string (_("SunVTS Loopback"));
/* set basic information */
track->label = g_strdup (_(label));
track->num_channels = 0;
track->min_volume = 0;
track->max_volume = 0;
track->flags =
GST_MIXER_TRACK_INPUT | GST_MIXER_TRACK_WHITELIST |
GST_MIXER_TRACK_NO_RECORD;
if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
g_warning ("Error getting audio device settings");
g_object_unref (G_OBJECT (sun_opts));
return NULL;
}
sun_opts->avail = audioinfo.record.avail_ports;
sun_opts->track_num = track_num;
for (i = 0; i < 8; i++) {
if ((1 << i) & audioinfo.record.avail_ports) {
const char *s = g_quark_to_string (sun_opts->names[i]);
opts->values = g_list_append (opts->values, g_strdup (s));
GST_DEBUG_OBJECT (opts, "option for track %d: %s",
track_num, GST_STR_NULL (s));
}
}
return opts;
}
/* returns TRUE if there was an error or we caught a keyboard interrupt. */
static gboolean
event_loop (GstElement * pipeline, gboolean blocking, GstState target_state)
{
GstBus *bus;
GstMessage *message = NULL;
gboolean res = FALSE;
gboolean buffering = FALSE;
bus = gst_element_get_bus (GST_ELEMENT (pipeline));
#ifndef DISABLE_FAULT_HANDLER
g_timeout_add (50, (GSourceFunc) check_intr, pipeline);
#endif
while (TRUE) {
message = gst_bus_poll (bus, GST_MESSAGE_ANY, blocking ? -1 : 0);
/* if the poll timed out, only when !blocking */
if (message == NULL)
goto exit;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_NEW_CLOCK:
{
GstClock *clock;
gst_message_parse_new_clock (message, &clock);
g_print ("New clock: %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
break;
}
case GST_MESSAGE_EOS:
g_print ("Got EOS from element \"%s\".\n",
GST_STR_NULL (GST_ELEMENT_NAME (GST_MESSAGE_SRC (message))));
goto exit;
case GST_MESSAGE_TAG:
if (tags) {
GstTagList *tags;
gst_message_parse_tag (message, &tags);
g_print ("FOUND TAG : found by element \"%s\".\n",
GST_STR_NULL (GST_ELEMENT_NAME (GST_MESSAGE_SRC (message))));
gst_tag_list_foreach (tags, print_tag, NULL);
gst_tag_list_free (tags);
}
break;
case GST_MESSAGE_INFO:{
GError *gerror;
gchar *debug;
gchar *name = gst_object_get_path_string (GST_MESSAGE_SRC (message));
gst_message_parse_info (message, &gerror, &debug);
if (debug) {
g_print ("INFO:\n%s\n", debug);
}
g_error_free (gerror);
g_free (debug);
g_free (name);
break;
}
case GST_MESSAGE_WARNING:{
GError *gerror;
gchar *debug;
gchar *name = gst_object_get_path_string (GST_MESSAGE_SRC (message));
/* dump graph on warning */
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "gst-launch.warning");
gst_message_parse_warning (message, &gerror, &debug);
g_print ("WARNING: from element %s: %s\n", name, gerror->message);
if (debug) {
g_print ("Additional debug info:\n%s\n", debug);
}
g_error_free (gerror);
g_free (debug);
g_free (name);
break;
}
case GST_MESSAGE_ERROR:{
GError *gerror;
gchar *debug;
/* dump graph on error */
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "gst-launch.error");
gst_message_parse_error (message, &gerror, &debug);
gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
g_error_free (gerror);
g_free (debug);
/* we have an error */
res = TRUE;
goto exit;
}
case GST_MESSAGE_STATE_CHANGED:{
GstState old, newX, pending;
gst_message_parse_state_changed (message, &old, &newX, &pending);
/* we only care about pipeline state change messages */
if (GST_MESSAGE_SRC (message) != GST_OBJECT_CAST (pipeline))
break;
/* dump graph for pipeline state changes */
{
gchar *dump_name = g_strdup_printf ("gst-launch.%s_%s",
gst_element_state_get_name (old),
gst_element_state_get_name (newX));
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
}
/* ignore when we are buffering since then we mess with the states
* ourselves. */
if (buffering) {
fprintf (stderr,
"Prerolled, waiting for buffering to finish...\n");
break;
}
/* if we reached the final target state, exit */
if (target_state == GST_STATE_PAUSED && newX == target_state)
goto exit;
/* else not an interesting message */
break;
}
case GST_MESSAGE_BUFFERING:{
gint percent;
gst_message_parse_buffering (message, &percent);
fprintf (stderr, "%s %d%% \r", "buffering...", percent);
/* no state management needed for live pipelines */
if (is_live)
break;
if (percent == 100) {
/* a 100% message means buffering is done */
buffering = FALSE;
/* if the desired state is playing, go back */
if (target_state == GST_STATE_PLAYING) {
fprintf (stderr,
"Done buffering, setting pipeline to PLAYING ...\n");
gst_element_set_state (pipeline, GST_STATE_PLAYING);
} else
goto exit;
} else {
/* buffering busy */
if (buffering == FALSE && target_state == GST_STATE_PLAYING) {
/* we were not buffering but PLAYING, PAUSE the pipeline. */
fprintf (stderr, "Buffering, setting pipeline to PAUSED ...\n");
gst_element_set_state (pipeline, GST_STATE_PAUSED);
}
buffering = TRUE;
}
break;
}
case GST_MESSAGE_LATENCY:
{
fprintf (stderr, "Redistribute latency...\n");
gst_bin_recalculate_latency (GST_BIN (pipeline));
break;
}
case GST_MESSAGE_APPLICATION:{
const GstStructure *s;
s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstLaunchInterrupt")) {
/* this application message is posted when we caught an interrupt and
* we need to stop the pipeline. */
fprintf (stderr, "Interrupt: Stopping pipeline ...\n");
/* return TRUE when we caught an interrupt */
res = TRUE;
goto exit;
}
}
default:
/* just be quiet by default */
break;
}
if (message)
gst_message_unref (message);
}
g_assert_not_reached ();
exit:
{
if (message)
gst_message_unref (message);
gst_object_unref (bus);
return res;
}
}
static gboolean
gst_cd_paranoia_src_open (GstAudioCdSrc * audiocdsrc, const gchar * device)
{
GstCdParanoiaSrc *src = GST_CD_PARANOIA_SRC (audiocdsrc);
gint i, cache_size;
GST_DEBUG_OBJECT (src, "trying to open device %s (generic-device=%s) ...",
device, GST_STR_NULL (src->generic_device));
/* find the device */
if (src->generic_device != NULL) {
src->d = cdda_identify_scsi (src->generic_device, device, FALSE, NULL);
} else {
if (device != NULL) {
src->d = cdda_identify (device, FALSE, NULL);
} else {
src->d = cdda_identify ("/dev/cdrom", FALSE, NULL);
}
}
/* fail if the device couldn't be found */
if (src->d == NULL)
goto no_device;
/* set verbosity mode */
cdda_verbose_set (src->d, CDDA_MESSAGE_FORGETIT, CDDA_MESSAGE_FORGETIT);
/* open the disc */
if (cdda_open (src->d))
goto open_failed;
GST_INFO_OBJECT (src, "set read speed to %d", src->read_speed);
cdda_speed_set (src->d, src->read_speed);
for (i = 1; i < src->d->tracks + 1; i++) {
GstAudioCdSrcTrack track = { 0, };
track.num = i;
track.is_audio = IS_AUDIO (src->d, i - 1);
track.start = cdda_track_firstsector (src->d, i);
track.end = cdda_track_lastsector (src->d, i);
track.tags = NULL;
gst_audio_cd_src_add_track (GST_AUDIO_CD_SRC (src), &track);
}
/* create the paranoia struct and set it up */
src->p = paranoia_init (src->d);
if (src->p == NULL)
goto init_failed;
paranoia_modeset (src->p, src->paranoia_mode);
GST_INFO_OBJECT (src, "set paranoia mode to 0x%02x", src->paranoia_mode);
if (src->search_overlap != -1) {
paranoia_overlapset (src->p, src->search_overlap);
GST_INFO_OBJECT (src, "search overlap set to %u", src->search_overlap);
}
cache_size = src->cache_size;
if (cache_size == -1) {
/* if paranoia mode is low (the default), assume we're doing playback */
if (src->paranoia_mode <= PARANOIA_MODE_FRAGMENT)
cache_size = 150;
else
cache_size = paranoia_cachemodel_size (src->p, -1);
}
paranoia_cachemodel_size (src->p, cache_size);
GST_INFO_OBJECT (src, "set cachemodel size to %u", cache_size);
src->next_sector = -1;
return TRUE;
/* ERRORS */
no_device:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
(_("Could not open CD device for reading.")), ("cdda_identify failed"));
return FALSE;
}
open_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
(_("Could not open CD device for reading.")), ("cdda_open failed"));
cdda_close (src->d);
src->d = NULL;
return FALSE;
}
init_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT,
("failed to initialize paranoia"), ("failed to initialize paranoia"));
return FALSE;
}
}
int main (int argc, char *argv[])
{
HTTPMgmt *httpmgmt;
HTTPStreaming *httpstreaming;
GMainLoop *loop;
GOptionContext *ctx;
GError *err = NULL;
gboolean foreground;
struct rlimit rlim;
GDateTime *datetime;
gchar exe_path[512], *date;
ctx = g_option_context_new (NULL);
g_option_context_add_main_entries (ctx, options, NULL);
g_option_context_add_group (ctx, gst_init_get_option_group ());
if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
exit (1);
}
g_option_context_free (ctx);
GST_DEBUG_CATEGORY_INIT (GSTREAMILL, "gstreamill", 0, "gstreamill log");
if (version) {
print_version_info ();
exit (0);
}
/* stop gstreamill. */
if (stop) {
gchar *pid_str;
gint pid;
g_file_get_contents (PID_FILE, &pid_str, NULL, NULL);
if (pid_str == NULL) {
g_print ("File %s not found, check if gstreamill is running.\n", PID_FILE);
exit (1);
}
pid = atoi (pid_str);
g_free (pid_str);
g_print ("stoping gstreamill with pid %d ...\n", pid);
kill (pid, SIGTERM);
exit (0);
}
/* readlink exe path before setuid, on CentOS, readlink exe path after setgid/setuid failure on permission denied */
memset (exe_path, '\0', sizeof (exe_path));
if (readlink ("/proc/self/exe", exe_path, sizeof (exe_path)) == -1) {
g_print ("Read /proc/self/exe error: %s", g_strerror (errno));
exit (2);
}
if (prepare_gstreamill_run_dir () != 0) {
g_print ("Can't create gstreamill run directory\n");
exit (3);
}
/*
if (set_user_and_group () != 0) {
g_print ("set user and group failure\n");
exit (4);
}
*/
if (job_file != NULL) {
/* gstreamill command with job, run in foreground */
foreground = TRUE;
} else {
/* gstreamill command without job, run in background */
foreground = FALSE;
}
if (gst_debug_get_default_threshold () < GST_LEVEL_WARNING) {
gst_debug_set_default_threshold (GST_LEVEL_WARNING);
}
/* initialize ts segment static plugin */
if (!gst_plugin_register_static (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"tssegment",
"ts segment plugin",
ts_segment_plugin_init,
"0.1.0",
"GPL",
"GStreamer",
"GStreamer",
"http://gstreamer.net/")) {
GST_ERROR ("registe tssegment error");
exit (17);
}
/* subprocess, create_job_process */
if (shm_name != NULL) {
gint fd;
gchar *job_desc, *p;
Job *job;
gchar *log_path, *name;
gint ret;
/* set subprocess maximum of core file */
rlim.rlim_cur = 0;
rlim.rlim_max = 0;
if (setrlimit (RLIMIT_CORE, &rlim) == -1) {
GST_ERROR ("setrlimit error: %s", g_strerror (errno));
}
/* read job description from share memory */
job_desc = NULL;
fd = shm_open (shm_name, O_CREAT | O_RDWR, S_IRUSR | S_IWUSR);
if (ftruncate (fd, job_length) == -1) {
exit (5);
}
p = mmap (NULL, job_length, PROT_READ|PROT_WRITE, MAP_SHARED, fd, 0);
job_desc = g_strdup (p);
if ((job_desc != NULL) && (!jobdesc_is_valid (job_desc))) {
exit (6);
}
/* initialize log */
name = (gchar *)jobdesc_get_name (job_desc);
if (!jobdesc_is_live (job_desc)) {
gchar *p;
p = jobdesc_get_log_path (job_desc);
log_path = g_build_filename (p, "gstreamill.log", NULL);
g_free (p);
} else {
log_path = g_build_filename (log_dir, name, "gstreamill.log", NULL);
}
ret = init_log (log_path);
g_free (log_path);
if (ret != 0) {
exit (7);
}
/* launch a job. */
datetime = g_date_time_new_now_local ();
date = g_date_time_format (datetime, "%b %d %H:%M:%S");
fprintf (_log->log_hd, "\n*** %s : job %s starting ***\n\n", date, name);
g_date_time_unref (datetime);
g_free (date);
job = job_new ("name", name, "job", job_desc, NULL);
job->is_live = jobdesc_is_live (job_desc);
job->eos = FALSE;
loop = g_main_loop_new (NULL, FALSE);
GST_INFO ("Initializing job ...");
if (job_initialize (job, TRUE) != 0) {
GST_ERROR ("initialize job failure, exit");
exit (8);
}
GST_INFO ("Initializing job done");
GST_INFO ("Initializing job's encoders output ...");
if (job_encoders_output_initialize (job) != 0) {
GST_ERROR ("initialize job encoders' output failure, exit");
exit (8);
}
GST_INFO ("Initializing job's encoders output done");
GST_INFO ("Starting job ...");
if (job_start (job) != 0) {
GST_ERROR ("start livejob failure, exit");
exit (9);
}
datetime = g_date_time_new_now_local ();
date = g_date_time_format (datetime, "%b %d %H:%M:%S");
fprintf (_log->log_hd, "\n*** %s : job %s started ***\n\n", date, name);
g_date_time_unref (datetime);
g_free (date);
g_free (name);
g_free (job_desc);
signal (SIGPIPE, SIG_IGN);
signal (SIGUSR1, sighandler);
signal (SIGTERM, stop_job);
g_main_loop_run (loop);
} else {
/* set parent process maximum of core file */
rlim.rlim_cur = RLIM_INFINITY;
rlim.rlim_max = RLIM_INFINITY;
if (setrlimit (RLIMIT_CORE, &rlim) == -1) {
GST_ERROR ("setrlimit error: %s", g_strerror (errno));
}
}
/* run in background? */
if (!foreground) {
gchar *path;
gint ret;
/* pid file exist? */
if (g_file_test (PID_FILE, G_FILE_TEST_EXISTS)) {
g_print ("file %s found, gstreamill already running !!!\n", PID_FILE);
exit (10);
}
/* media directory */
path = g_strdup_printf ("%s/dvr", MEDIA_LOCATION);
if (!g_file_test (path, G_FILE_TEST_EXISTS)) {
g_printf ("Create DVR directory: %s", path);
if (g_mkdir_with_parents (path, 0755) != 0) {
g_printf ("Create DVR directory failure: %s", path);
}
}
g_free (path);
path = g_strdup_printf ("%s/transcode/in", MEDIA_LOCATION);
if (!g_file_test (path, G_FILE_TEST_EXISTS)) {
g_printf ("Create transcode directory: %s", path);
if (g_mkdir_with_parents (path, 0755) != 0) {
g_printf ("Create transcode directory failure: %s", path);
}
}
g_free (path);
path = g_strdup_printf ("%s/transcode/out", MEDIA_LOCATION);
if (!g_file_test (path, G_FILE_TEST_EXISTS)) {
g_printf ("Create transcode directory: %s", path);
if (g_mkdir_with_parents (path, 0755) != 0) {
g_printf ("Create transcode directory failure: %s", path);
}
}
g_free (path);
/* log to file */
path = g_build_filename (log_dir, "gstreamill.log", NULL);
ret = init_log (path);
g_free (path);
if (ret != 0) {
g_print ("Init log error, ret %d.\n", ret);
exit (11);
}
/* daemonize */
if (daemon (0, 0) != 0) {
fprintf (_log->log_hd, "Failed to daemonize");
remove_pid_file ();
exit (1);
}
/* create pid file */
if (create_pid_file () != 0) {
exit (1);
}
/* customize signal */
signal (SIGUSR1, sighandler);
signal (SIGTERM, stop_gstreamill);
datetime = g_date_time_new_now_local ();
date = g_date_time_format (datetime, "%b %d %H:%M:%S");
fprintf (_log->log_hd, "\n*** %s : gstreamill started ***\n\n", date);
g_free (date);
g_date_time_unref (datetime);
}
/* ignore SIGPIPE */
signal (SIGPIPE, SIG_IGN);
loop = g_main_loop_new (NULL, FALSE);
/* gstreamill */
gstreamill = gstreamill_new ("daemon", !foreground, "log_dir", log_dir, "exe_path", exe_path, NULL);
if (gstreamill_start (gstreamill) != 0) {
GST_ERROR ("start gstreamill error, exit.");
remove_pid_file ();
exit (12);
}
/* httpstreaming, pull */
httpstreaming = httpstreaming_new ("gstreamill", gstreamill, "address", http_streaming, NULL);
if (httpstreaming_start (httpstreaming, 10) != 0) {
GST_ERROR ("start httpstreaming error, exit.");
remove_pid_file ();
exit (13);
}
if (!foreground) {
/* run in background, management via http */
httpmgmt = httpmgmt_new ("gstreamill", gstreamill, "address", http_mgmt, NULL);
if (httpmgmt_start (httpmgmt) != 0) {
GST_ERROR ("start http mangment error, exit.");
remove_pid_file ();
exit (14);
}
} else {
/* run in foreground, start job */
gchar *job, *p, *result;
JSON_Value *val;
JSON_Object *obj;
/* ctrl-c, stop gstreamill */
signal (SIGINT, stop_gstreamill);
/* ctrl-\, stop gstreamill */
signal (SIGQUIT, stop_gstreamill);
if (!g_file_get_contents (job_file, &job, NULL, NULL)) {
GST_ERROR ("Read job file %s error.", job_file);
exit (15);
}
p = gstreamill_job_start (gstreamill, job);
val = json_parse_string (p);
obj = json_value_get_object (val);
result = (gchar *)json_object_get_string (obj, "result");
GST_INFO ("start job result: %s.", result);
if (g_strcmp0 (result, "success") != 0) {
exit (16);
}
json_value_free (val);
g_free (p);
}
g_main_loop_run (loop);
return 0;
}
int
main (int argc, char **argv)
{
GPtrArray *files;
gchar **args = NULL;
guint num, i;
GError *err = NULL;
GOptionContext *ctx;
GOptionEntry options[] = {
{G_OPTION_REMAINING, 0, 0, G_OPTION_ARG_FILENAME_ARRAY, &args, NULL},
{NULL}
};
GTimer *timer;
if (!g_thread_supported ())
g_thread_init (NULL);
ctx = g_option_context_new ("FILES OR DIRECTORIES WITH AUDIO FILES");
g_option_context_add_main_entries (ctx, options, NULL);
g_option_context_add_group (ctx, gst_init_get_option_group ());
if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
exit (1);
}
g_option_context_free (ctx);
if (args == NULL || *args == NULL) {
g_print ("Please provide one or more directories with audio files\n\n");
return 1;
}
files = g_ptr_array_new ();
num = g_strv_length (args);
for (i = 0; i < num; ++i) {
if (g_path_is_absolute (args[i])) {
check_arg (files, args[i]);
} else {
g_warning ("Argument '%s' is not an absolute file path", args[i]);
}
}
if (files->len == 0) {
g_print ("Did not find any files\n\n");
return 1;
}
timer = g_timer_new ();
while (g_timer_elapsed (timer, NULL) < TEST_RUNTIME) {
gint32 idx;
idx = g_random_int_range (0, files->len);
play_file ((const gchar *) g_ptr_array_index (files, idx));
}
g_strfreev (args);
g_timer_destroy (timer);
return 0;
}
static GstFlowReturn
gst_test_http_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** retbuf)
{
GstTestHTTPSrc *src = GST_TEST_HTTP_SRC (basesrc);
guint bytes_read;
GstFlowReturn ret = GST_FLOW_OK;
guint blocksize;
fail_unless (gst_test_http_src_callbacks != NULL);
fail_unless (gst_test_http_src_callbacks->src_create != NULL);
GST_OBJECT_LOCK (src);
blocksize = basesrc->blocksize;
GST_OBJECT_UNLOCK (src);
g_mutex_lock (&src->mutex);
GST_DEBUG ("gst_test_http_src_create feeding from %" G_GUINT64_FORMAT,
src->position);
if (src->uri == NULL) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
g_mutex_unlock (&src->mutex);
return GST_FLOW_ERROR;
}
if (src->input.status_code < 200 || src->input.status_code >= 300) {
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, ("%s",
"Generated requested error"), ("%s (%d), URL: %s, Redirect to: %s",
"Generated requested error", src->input.status_code, src->uri,
GST_STR_NULL (NULL)));
g_mutex_unlock (&src->mutex);
return GST_FLOW_ERROR;
}
if (src->http_method == METHOD_INVALID) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, ("%s",
"Invalid HTTP method"), ("%s (%s), URL: %s",
"Invalid HTTP method", src->http_method_name, src->uri));
g_mutex_unlock (&src->mutex);
return GST_FLOW_ERROR;
} else if (src->http_method == METHOD_HEAD) {
ret = GST_FLOW_EOS;
goto http_events;
}
fail_unless_equals_uint64 (offset, src->position);
bytes_read = MIN ((src->segment_end - src->position), blocksize);
if (bytes_read == 0) {
ret = GST_FLOW_EOS;
goto http_events;
}
ret = gst_test_http_src_callbacks->src_create (src,
offset, bytes_read, retbuf,
src->input.context, gst_test_http_src_callback_user_data);
if (ret != GST_FLOW_OK) {
goto http_events;
}
GST_BUFFER_OFFSET (*retbuf) = src->position;
GST_BUFFER_OFFSET_END (*retbuf) = src->position + bytes_read;
src->position += bytes_read;
http_events:
if (src->http_headers_event) {
gst_pad_push_event (GST_BASE_SRC_PAD (src), src->http_headers_event);
src->http_headers_event = NULL;
}
if (src->duration_changed) {
src->duration_changed = FALSE;
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_duration_changed (GST_OBJECT (src)));
}
g_mutex_unlock (&src->mutex);
return ret;
}
gint
main (gint argc, gchar * argv[])
{
GstStateChangeReturn ret;
GstElement *pipeline;
GstElement *filter, *sink;
GstElement *sourcebin;
GError *error = NULL;
GtkWidget *window;
GtkWidget *screen;
GtkWidget *vbox, *combo;
GtkWidget *hbox;
GtkWidget *play, *pause, *null, *ready;
gchar **source_desc_array = NULL;
gchar *source_desc = NULL;
GOptionContext *context;
GOptionEntry options[] = {
{"source-bin", 's', 0, G_OPTION_ARG_STRING_ARRAY, &source_desc_array,
"Use a custom source bin description (gst-launch style)", NULL}
,
{NULL}
};
context = g_option_context_new (NULL);
g_option_context_add_main_entries (context, options, NULL);
g_option_context_add_group (context, gst_init_get_option_group ());
g_option_context_add_group (context, gtk_get_option_group (TRUE));
if (!g_option_context_parse (context, &argc, &argv, &error)) {
g_print ("Inizialization error: %s\n", GST_STR_NULL (error->message));
return -1;
}
g_option_context_free (context);
if (source_desc_array != NULL) {
source_desc = g_strjoinv (" ", source_desc_array);
g_strfreev (source_desc_array);
}
if (source_desc == NULL) {
source_desc =
g_strdup
("videotestsrc ! video/x-raw, width=352, height=288 ! identity");
}
sourcebin =
gst_parse_bin_from_description (g_strdup (source_desc), TRUE, &error);
g_free (source_desc);
if (error) {
g_print ("Error while parsing source bin description: %s\n",
GST_STR_NULL (error->message));
return -1;
}
g_set_application_name ("gst-gl-effects test app");
window = gtk_window_new (GTK_WINDOW_TOPLEVEL);
gtk_container_set_border_width (GTK_CONTAINER (window), 3);
pipeline = gst_pipeline_new ("pipeline");
filter = gst_element_factory_make ("gleffects", "flt");
sink = gst_element_factory_make ("glimagesink", "glsink");
gst_bin_add_many (GST_BIN (pipeline), sourcebin, filter, sink, NULL);
if (!gst_element_link_many (sourcebin, filter, sink, NULL)) {
g_print ("Failed to link one or more elements!\n");
return -1;
}
g_signal_connect (G_OBJECT (window), "delete-event",
G_CALLBACK (destroy_cb), pipeline);
g_signal_connect (G_OBJECT (window), "destroy-event",
G_CALLBACK (destroy_cb), pipeline);
screen = gtk_drawing_area_new ();
gtk_widget_set_size_request (screen, 640, 480); // 500 x 376
vbox = gtk_box_new (GTK_ORIENTATION_VERTICAL, 2);
gtk_box_pack_start (GTK_BOX (vbox), screen, TRUE, TRUE, 0);
combo = gtk_combo_box_text_new ();
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "identity");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "mirror");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "squeeze");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "stretch");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "fisheye");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "twirl");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "bulge");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "tunnel");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "square");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "heat");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "xpro");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "lumaxpro");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "sepia");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "xray");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "sin");
gtk_combo_box_text_append_text (GTK_COMBO_BOX_TEXT (combo), "glow");
g_signal_connect (G_OBJECT (combo), "changed", G_CALLBACK (apply_fx), filter);
gtk_box_pack_start (GTK_BOX (vbox), combo, FALSE, FALSE, 0);
hbox = gtk_box_new (GTK_ORIENTATION_HORIZONTAL, 0);
play = gtk_button_new_with_label ("PLAY");
g_signal_connect (G_OBJECT (play), "clicked", G_CALLBACK (play_cb), pipeline);
pause = gtk_button_new_with_label ("PAUSE");
g_signal_connect (G_OBJECT (pause), "clicked",
G_CALLBACK (pause_cb), pipeline);
null = gtk_button_new_with_label ("NULL");
g_signal_connect (G_OBJECT (null), "clicked", G_CALLBACK (null_cb), pipeline);
ready = gtk_button_new_with_label ("READY");
g_signal_connect (G_OBJECT (ready), "clicked",
G_CALLBACK (ready_cb), pipeline);
gtk_box_pack_start (GTK_BOX (hbox), null, TRUE, TRUE, 0);
gtk_box_pack_start (GTK_BOX (hbox), ready, TRUE, TRUE, 0);
gtk_box_pack_start (GTK_BOX (hbox), play, TRUE, TRUE, 0);
gtk_box_pack_start (GTK_BOX (hbox), pause, TRUE, TRUE, 0);
gtk_box_pack_start (GTK_BOX (vbox), hbox, FALSE, FALSE, 0);
gtk_container_add (GTK_CONTAINER (window), vbox);
g_signal_connect (screen, "realize", G_CALLBACK (expose_cb), pipeline);
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_print ("Failed to start up pipeline!\n");
return -1;
}
gtk_widget_show_all (GTK_WIDGET (window));
gtk_main ();
return 0;
}
/* open the file, necessary to go to RUNNING state */
static gboolean
gst_vcdsrc_start (GstBaseSrc * bsrc)
{
int i;
GstVCDSrc *src = GST_VCDSRC (bsrc);
struct stat buf;
/* open the device */
src->fd = open (src->device, O_RDONLY);
if (src->fd < 0)
goto open_failed;
if (fstat (src->fd, &buf) < 0)
goto toc_failed;
/* If it's not a block device, then we need to try and
* parse the cue file if there is one
* FIXME implement */
if (!S_ISBLK (buf.st_mode)) {
GST_DEBUG ("Reading CUE files not handled yet, cannot process %s",
GST_STR_NULL (src->device));
goto toc_failed;
}
/* read the table of contents */
if (ioctl (src->fd, CDROMREADTOCHDR, &src->tochdr))
goto toc_failed;
/* allocate enough track structs for disk */
src->numtracks = (src->tochdr.cdth_trk1 - src->tochdr.cdth_trk0) + 1;
src->tracks = g_new (struct cdrom_tocentry, src->numtracks + 1);
/* read each track entry */
for (i = 0; i <= src->numtracks; i++) {
src->tracks[i].cdte_track = i == src->numtracks ? CDROM_LEADOUT : i + 1;
src->tracks[i].cdte_format = CDROM_MSF;
if (ioctl (src->fd, CDROMREADTOCENTRY, &src->tracks[i]))
goto toc_entry_failed;
GST_DEBUG ("track %d begins at %d:%02d.%02d", i,
src->tracks[i].cdte_addr.msf.minute,
src->tracks[i].cdte_addr.msf.second,
src->tracks[i].cdte_addr.msf.frame);
}
src->curoffset = 0;
gst_vcdsrc_recalculate (src);
return TRUE;
/* ERRORS */
open_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), GST_ERROR_SYSTEM);
return FALSE;
}
toc_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), GST_ERROR_SYSTEM);
close (src->fd);
return FALSE;
}
toc_entry_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), GST_ERROR_SYSTEM);
g_free (src->tracks);
close (src->fd);
return FALSE;
}
}
GstElement *mmsGstLaunch(const char *pipeline_description) {
GError *error = NULL;
gint res = 0;
#ifdef ENABLE_NLS
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
textdomain (GETTEXT_PACKAGE);
#endif
if (!g_thread_supported ())
g_thread_init (NULL);
gst_alloc_trace_set_flags_all (GST_ALLOC_TRACE_LIVE);
#ifndef DISABLE_FAULT_HANDLER
fault_setup ();
sigint_setup ();
play_signal_setup ();
#endif
// parse the pipeline description
pipeline = gst_parse_launch(pipeline_description, &error);
if (!pipeline) {
if (error) {
fprintf (stderr, "ERROR: pipeline could not be constructed: %s.\n",
GST_STR_NULL (error->message));
g_error_free (error);
} else {
fprintf (stderr, "ERROR: pipeline could not be constructed.\n");
}
return NULL;
} else if (error) {
fprintf (stderr, "WARNING: erroneous pipeline: %s\n",
GST_STR_NULL (error->message));
g_error_free (error);
return NULL;
}
GstState state;
GstStateChangeReturn ret;
/* If the top-level object is not a pipeline, place it in a pipeline. */
if (!GST_IS_PIPELINE (pipeline)) {
GstElement *real_pipeline = gst_element_factory_make ("pipeline", NULL);
if (real_pipeline == NULL) {
fprintf (stderr, "ERROR: the 'pipeline' element wasn't found.\n");
return NULL;
}
gst_bin_add (GST_BIN (real_pipeline), pipeline);
pipeline = real_pipeline;
}
fprintf (stderr, "Setting pipeline to PAUSED ...\n");
ret = gst_element_set_state (pipeline, GST_STATE_PAUSED);
switch (ret) {
case GST_STATE_CHANGE_FAILURE:
fprintf (stderr, "ERROR: Pipeline doesn't want to pause.\n");
res = -1;
event_loop (pipeline, FALSE, GST_STATE_VOID_PENDING);
goto end;
case GST_STATE_CHANGE_NO_PREROLL:
fprintf (stderr, "Pipeline is live and does not need PREROLL ...\n");
is_live = TRUE;
break;
case GST_STATE_CHANGE_ASYNC:
fprintf (stderr, "Pipeline is PREROLLING ...\n");
caught_error = event_loop (pipeline, TRUE, GST_STATE_PAUSED);
if (caught_error) {
fprintf (stderr, "ERROR: pipeline doesn't want to preroll.\n");
goto end;
}
state = GST_STATE_PAUSED;
// fallthrough
case GST_STATE_CHANGE_SUCCESS:
fprintf (stderr, "Pipeline is PREROLLED ...\n");
break;
}
// pipe successfully created
return pipeline;
end:
mmsGstFree();
return NULL;
}
static gboolean
wildmidi_open_config (void)
{
gchar *path = g_strdup (g_getenv ("WILDMIDI_CFG"));
gint ret;
GST_DEBUG ("trying %s", GST_STR_NULL (path));
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
if (path == NULL) {
path =
g_build_path (G_DIR_SEPARATOR_S, g_get_home_dir (), ".wildmidirc",
NULL);
GST_DEBUG ("trying %s", path);
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
}
if (path == NULL) {
path =
g_build_path (G_DIR_SEPARATOR_S, G_DIR_SEPARATOR_S "etc",
"wildmidi.cfg", NULL);
GST_DEBUG ("trying %s", path);
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
}
if (path == NULL) {
path =
g_build_path (G_DIR_SEPARATOR_S, G_DIR_SEPARATOR_S "etc", "wildmidi",
"wildmidi.cfg", NULL);
GST_DEBUG ("trying %s", path);
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
}
if (path == NULL) {
path = g_strdup (WILDMIDI_CFG);
GST_DEBUG ("trying %s", path);
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
}
if (path == NULL) {
path =
g_build_path (G_DIR_SEPARATOR_S, G_DIR_SEPARATOR_S "etc",
"timidity.cfg", NULL);
GST_DEBUG ("trying %s", path);
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
}
if (path == NULL) {
path =
g_build_path (G_DIR_SEPARATOR_S, G_DIR_SEPARATOR_S "etc", "timidity",
"timidity.cfg", NULL);
GST_DEBUG ("trying %s", path);
if (path && (g_access (path, R_OK) == -1)) {
g_free (path);
path = NULL;
}
}
if (path == NULL) {
/* I've created a symlink to get it playing
* ln -s /usr/share/timidity/timidity.cfg /etc/wildmidi.cfg
* we could make it use : WILDMIDI_CFG
* but unfortunately it fails to create a proper filename if the config
* has a redirect
* http://sourceforge.net/tracker/index.php?func=detail&aid=1657358&group_id=42635&atid=433744
*/
GST_WARNING ("no config file, can't initialise");
return FALSE;
}
/* this also initializes a some filter and stuff and thus is slow */
ret = WildMidi_Init (path, WILDMIDI_RATE, 0);
g_free (path);
return (ret == 0);
}
static GstFlowReturn
gst_rdt_manager_chain_rtcp (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstRDTManager *src;
#ifdef HAVE_RTCP
gboolean valid;
GstRTCPPacket packet;
gboolean more;
#endif
src = GST_RDT_MANAGER (parent);
GST_DEBUG_OBJECT (src, "got rtcp packet");
#ifdef HAVE_RTCP
valid = gst_rtcp_buffer_validate (buffer);
if (!valid)
goto bad_packet;
/* position on first packet */
more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
while (more) {
switch (gst_rtcp_packet_get_type (&packet)) {
case GST_RTCP_TYPE_SR:
{
guint32 ssrc, rtptime, packet_count, octet_count;
guint64 ntptime;
guint count, i;
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
&packet_count, &octet_count);
GST_DEBUG_OBJECT (src,
"got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
", RTP %u, PC %u, OC %u", ssrc, ntptime, rtptime, packet_count,
octet_count);
count = gst_rtcp_packet_get_rb_count (&packet);
for (i = 0; i < count; i++) {
guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
guint8 fractionlost;
gint32 packetslost;
gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
packetslost, exthighestseq, jitter, lsr, dlsr);
}
break;
}
case GST_RTCP_TYPE_RR:
{
guint32 ssrc;
guint count, i;
ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
GST_DEBUG_OBJECT (src, "got RR packet: SSRC %08x", ssrc);
count = gst_rtcp_packet_get_rb_count (&packet);
for (i = 0; i < count; i++) {
guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
guint8 fractionlost;
gint32 packetslost;
gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
packetslost, exthighestseq, jitter, lsr, dlsr);
}
break;
}
case GST_RTCP_TYPE_SDES:
{
guint chunks, i, j;
gboolean more_chunks, more_items;
chunks = gst_rtcp_packet_sdes_get_chunk_count (&packet);
GST_DEBUG_OBJECT (src, "got SDES packet with %d chunks", chunks);
more_chunks = gst_rtcp_packet_sdes_first_chunk (&packet);
i = 0;
while (more_chunks) {
guint32 ssrc;
ssrc = gst_rtcp_packet_sdes_get_ssrc (&packet);
GST_DEBUG_OBJECT (src, "chunk %d, SSRC %08x", i, ssrc);
more_items = gst_rtcp_packet_sdes_first_item (&packet);
j = 0;
while (more_items) {
GstRTCPSDESType type;
guint8 len;
gchar *data;
gst_rtcp_packet_sdes_get_item (&packet, &type, &len, &data);
GST_DEBUG_OBJECT (src, "item %d, type %d, len %d, data %s", j,
type, len, data);
more_items = gst_rtcp_packet_sdes_next_item (&packet);
j++;
}
more_chunks = gst_rtcp_packet_sdes_next_chunk (&packet);
i++;
}
break;
}
case GST_RTCP_TYPE_BYE:
{
guint count, i;
gchar *reason;
reason = gst_rtcp_packet_bye_get_reason (&packet);
GST_DEBUG_OBJECT (src, "got BYE packet (reason: %s)",
GST_STR_NULL (reason));
g_free (reason);
count = gst_rtcp_packet_bye_get_ssrc_count (&packet);
for (i = 0; i < count; i++) {
guint32 ssrc;
ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, i);
GST_DEBUG_OBJECT (src, "SSRC: %08x", ssrc);
}
break;
}
case GST_RTCP_TYPE_APP:
GST_DEBUG_OBJECT (src, "got APP packet");
break;
default:
GST_WARNING_OBJECT (src, "got unknown RTCP packet");
break;
}
more = gst_rtcp_packet_move_to_next (&packet);
}
gst_buffer_unref (buffer);
return GST_FLOW_OK;
bad_packet:
{
GST_WARNING_OBJECT (src, "got invalid RTCP packet");
return GST_FLOW_OK;
}
#else
return GST_FLOW_OK;
#endif
}
static void
test_taglib_id3mux_with_tags (GstTagList * tags, guint32 mask)
{
GstMessage *msg;
GstTagList *tags_read = NULL;
GstElement *pipeline, *id3mux, *id3demux, *fakesrc, *identity, *fakesink;
GstBus *bus;
guint64 offset;
GstBuffer *outbuf = NULL;
GstBuffer *tagbuf = NULL;
GstStateChangeReturn state_result;
pipeline = gst_pipeline_new ("pipeline");
g_assert (pipeline != NULL);
fakesrc = gst_element_factory_make ("fakesrc", "fakesrc");
g_assert (fakesrc != NULL);
id3mux = gst_element_factory_make ("id3v2mux", "id3v2mux");
g_assert (id3mux != NULL);
identity = gst_element_factory_make ("identity", "identity");
g_assert (identity != NULL);
id3demux = gst_element_factory_make ("id3demux", "id3demux");
g_assert (id3demux != NULL);
fakesink = gst_element_factory_make ("fakesink", "fakesink");
g_assert (fakesink != NULL);
/* set up sink */
outbuf = NULL;
g_object_set (fakesink, "signal-handoffs", TRUE, NULL);
g_signal_connect (fakesink, "handoff", G_CALLBACK (got_buffer), &outbuf);
gst_bin_add (GST_BIN (pipeline), fakesrc);
gst_bin_add (GST_BIN (pipeline), id3mux);
gst_bin_add (GST_BIN (pipeline), identity);
gst_bin_add (GST_BIN (pipeline), id3demux);
gst_bin_add (GST_BIN (pipeline), fakesink);
gst_tag_setter_merge_tags (GST_TAG_SETTER (id3mux), tags,
GST_TAG_MERGE_APPEND);
gst_element_link_many (fakesrc, id3mux, identity, id3demux, fakesink, NULL);
/* set up source */
g_object_set (fakesrc, "signal-handoffs", TRUE, "can-activate-pull", FALSE,
"filltype", 2, "sizetype", 2, "sizemax", MP3_FRAME_SIZE,
"num-buffers", 16, NULL);
offset = 0;
g_signal_connect (fakesrc, "handoff", G_CALLBACK (fill_mp3_buffer), &offset);
/* set up identity to catch tag buffer */
g_signal_connect (identity, "handoff", G_CALLBACK (identity_cb), &tagbuf);
GST_LOG ("setting and getting state ...");
gst_element_set_state (pipeline, GST_STATE_PLAYING);
state_result = gst_element_get_state (pipeline, NULL, NULL, -1);
fail_unless (state_result == GST_STATE_CHANGE_SUCCESS,
"Unexpected result from get_state(). Expected success, got %d",
state_result);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
GST_LOG ("Waiting for tag ...");
msg =
gst_bus_poll (bus, GST_MESSAGE_TAG | GST_MESSAGE_EOS | GST_MESSAGE_ERROR,
-1);
if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
GError *err;
gchar *dbg;
gst_message_parse_error (msg, &err, &dbg);
g_printerr ("ERROR from element %s: %s\n%s\n",
GST_OBJECT_NAME (msg->src), err->message, GST_STR_NULL (dbg));
g_error_free (err);
g_free (dbg);
} else if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS) {
g_printerr ("EOS message, but were waiting for TAGS!\n");
}
fail_unless (msg->type == GST_MESSAGE_TAG);
gst_message_parse_tag (msg, &tags_read);
gst_message_unref (msg);
GST_LOG ("Got tags: %" GST_PTR_FORMAT, tags_read);
test_taglib_id3mux_check_tags (tags_read, mask);
gst_tag_list_unref (tags_read);
fail_unless (tagbuf != NULL);
test_taglib_id3mux_check_tag_buffer (tagbuf, mask);
gst_buffer_unref (tagbuf);
GST_LOG ("Waiting for EOS ...");
msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
GError *err;
gchar *dbg;
gst_message_parse_error (msg, &err, &dbg);
g_printerr ("ERROR from element %s: %s\n%s\n",
GST_OBJECT_NAME (msg->src), err->message, GST_STR_NULL (dbg));
g_error_free (err);
g_free (dbg);
}
fail_unless (msg->type == GST_MESSAGE_EOS);
gst_message_unref (msg);
gst_object_unref (bus);
GST_LOG ("Got EOS, shutting down ...");
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
test_taglib_id3mux_check_output_buffer (outbuf);
gst_buffer_unref (outbuf);
GST_LOG ("Done");
}
static gboolean
gst_split_file_src_start (GstBaseSrc * basesrc)
{
GstSplitFileSrc *src = GST_SPLIT_FILE_SRC (basesrc);
GCancellable *cancel;
gboolean ret = FALSE;
guint64 offset;
GError *err = NULL;
gchar *basename = NULL;
gchar *dirname = NULL;
gchar **files;
guint i;
GST_OBJECT_LOCK (src);
if (src->location != NULL && src->location[0] != '\0') {
basename = g_path_get_basename (src->location);
dirname = g_path_get_dirname (src->location);
}
GST_OBJECT_UNLOCK (src);
files = gst_split_file_src_find_files (src, dirname, basename, &err);
if (files == NULL || *files == NULL)
goto no_files;
src->num_parts = g_strv_length (files);
src->parts = g_new0 (GstFilePart, src->num_parts);
cancel = src->cancellable;
offset = 0;
for (i = 0; i < src->num_parts; ++i) {
GFileInputStream *stream;
GFileInfo *info;
goffset size;
GFile *file;
file = g_file_new_for_path (files[i]);
stream = g_file_read (file, cancel, &err);
g_object_unref (file);
if (err != NULL)
goto open_read_error;
info = g_file_input_stream_query_info (stream, "standard::*", NULL, &err);
if (err != NULL) {
g_object_unref (stream);
goto query_info_error;
}
size = g_file_info_get_size (info);
g_object_unref (info);
src->parts[i].stream = stream;
src->parts[i].path = g_strdup (files[i]);
src->parts[i].start = offset;
src->parts[i].stop = offset + size - 1;
GST_DEBUG ("[%010" G_GUINT64_FORMAT "-%010" G_GUINT64_FORMAT "] %s",
src->parts[i].start, src->parts[i].stop, src->parts[i].path);
offset += size;
}
GST_INFO ("Successfully opened %u file parts for reading", src->num_parts);
src->cur_part = 0;
src->cancellable = g_cancellable_new ();
ret = TRUE;
done:
if (err != NULL)
g_error_free (err);
g_strfreev (files);
g_free (basename);
g_free (dirname);
return ret;
/* ERRORS */
no_files:
{
if (err->code == G_IO_ERROR_CANCELLED)
goto cancelled;
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, ("%s", err->message),
("Failed to find files in '%s' for pattern '%s'",
GST_STR_NULL (dirname), GST_STR_NULL (basename)));
goto done;
}
open_read_error:
{
if (err->code == G_IO_ERROR_CANCELLED)
goto cancelled;
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, ("%s", err->message),
("Failed to open file '%s' for reading", files[i]));
goto done;
}
query_info_error:
{
if (err->code == G_IO_ERROR_CANCELLED)
goto cancelled;
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, ("%s", err->message),
("Failed to query info for file '%s'", files[i]));
goto done;
}
cancelled:
{
GST_DEBUG_OBJECT (src, "I/O operation cancelled from another thread");
goto done;
}
}
/* First some utils, then the mixer implementation */
static gboolean
gst_alsa_mixer_open (GstAlsaMixer * mixer)
{
gint err;
snd_ctl_t *ctl;
snd_ctl_card_info_t *card_info;
g_return_val_if_fail (mixer->handle == NULL, FALSE);
/* open and initialize the mixer device */
err = snd_mixer_open (&mixer->handle, 0);
if (err < 0 || mixer->handle == NULL)
goto open_failed;
if ((err = snd_mixer_attach (mixer->handle, mixer->device)) < 0) {
GST_WARNING ("Cannot open mixer for sound device '%s': %s", mixer->device,
snd_strerror (err));
goto error;
}
if ((err = snd_mixer_selem_register (mixer->handle, NULL, NULL)) < 0) {
GST_WARNING ("Cannot register mixer elements: %s", snd_strerror (err));
goto error;
}
if ((err = snd_mixer_load (mixer->handle)) < 0) {
GST_WARNING ("Cannot load mixer settings: %s", snd_strerror (err));
goto error;
}
snd_mixer_set_callback_private (mixer->handle, mixer);
snd_mixer_set_callback (mixer->handle, gst_alsa_mixer_handle_callback);
/* now get the device name, any of this is not fatal */
g_free (mixer->cardname);
if ((err = snd_ctl_open (&ctl, mixer->device, 0)) < 0) {
GST_WARNING ("Cannot open CTL: %s", snd_strerror (err));
goto no_card_name;
}
snd_ctl_card_info_malloc (&card_info);
if ((err = snd_ctl_card_info (ctl, card_info)) < 0) {
GST_WARNING ("Cannot get card info: %s", snd_strerror (err));
snd_ctl_close (ctl);
goto no_card_name;
}
mixer->cardname = g_strdup (snd_ctl_card_info_get_name (card_info));
GST_DEBUG ("Card name = %s", GST_STR_NULL (mixer->cardname));
snd_ctl_card_info_free (card_info);
snd_ctl_close (ctl);
no_card_name:
if (mixer->cardname == NULL) {
mixer->cardname = g_strdup ("Unknown");
GST_DEBUG ("Cannot find card name");
}
GST_INFO ("Successfully opened mixer for device '%s'.", mixer->device);
return TRUE;
/* ERROR */
open_failed:
{
GST_WARNING ("Cannot open mixer: %s", snd_strerror (err));
mixer->handle = NULL;
return FALSE;
}
error:
{
snd_mixer_close (mixer->handle);
mixer->handle = NULL;
return FALSE;
}
}
int
main (int argc, char **argv)
{
GstPlay *play;
GPtrArray *playlist;
gboolean verbose = FALSE;
gboolean print_version = FALSE;
gboolean interactive = TRUE;
gboolean gapless = FALSE;
gboolean shuffle = FALSE;
gdouble volume = -1;
gchar **filenames = NULL;
gchar *audio_sink = NULL;
gchar *video_sink = NULL;
gchar **uris;
gchar *flags = NULL;
guint num, i;
GError *err = NULL;
GOptionContext *ctx;
gchar *playlist_file = NULL;
GOptionEntry options[] = {
{"verbose", 'v', 0, G_OPTION_ARG_NONE, &verbose,
N_("Output status information and property notifications"), NULL},
{"flags", 0, 0, G_OPTION_ARG_STRING, &flags,
N_("Control playback behaviour setting playbin 'flags' property"),
NULL},
{"version", 0, 0, G_OPTION_ARG_NONE, &print_version,
N_("Print version information and exit"), NULL},
{"videosink", 0, 0, G_OPTION_ARG_STRING, &video_sink,
N_("Video sink to use (default is autovideosink)"), NULL},
{"audiosink", 0, 0, G_OPTION_ARG_STRING, &audio_sink,
N_("Audio sink to use (default is autoaudiosink)"), NULL},
{"gapless", 0, 0, G_OPTION_ARG_NONE, &gapless,
N_("Enable gapless playback"), NULL},
{"shuffle", 0, 0, G_OPTION_ARG_NONE, &shuffle,
N_("Shuffle playlist"), NULL},
{"no-interactive", 0, G_OPTION_FLAG_REVERSE, G_OPTION_ARG_NONE,
&interactive,
N_("Disable interactive control via the keyboard"), NULL},
{"volume", 0, 0, G_OPTION_ARG_DOUBLE, &volume,
N_("Volume"), NULL},
{"playlist", 0, 0, G_OPTION_ARG_FILENAME, &playlist_file,
N_("Playlist file containing input media files"), NULL},
{"quiet", 'q', 0, G_OPTION_ARG_NONE, &quiet,
N_("Do not print any output (apart from errors)"), NULL},
{G_OPTION_REMAINING, 0, 0, G_OPTION_ARG_FILENAME_ARRAY, &filenames, NULL},
{NULL}
};
setlocale (LC_ALL, "");
#ifdef ENABLE_NLS
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
textdomain (GETTEXT_PACKAGE);
#endif
g_set_prgname ("gst-play-" GST_API_VERSION);
ctx = g_option_context_new ("FILE1|URI1 [FILE2|URI2] [FILE3|URI3] ...");
g_option_context_add_main_entries (ctx, options, GETTEXT_PACKAGE);
g_option_context_add_group (ctx, gst_init_get_option_group ());
if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
g_option_context_free (ctx);
g_clear_error (&err);
return 1;
}
g_option_context_free (ctx);
GST_DEBUG_CATEGORY_INIT (play_debug, "play", 0, "gst-play");
if (print_version) {
gchar *version_str;
version_str = gst_version_string ();
g_print ("%s version %s\n", g_get_prgname (), PACKAGE_VERSION);
g_print ("%s\n", version_str);
g_print ("%s\n", GST_PACKAGE_ORIGIN);
g_free (version_str);
g_free (audio_sink);
g_free (video_sink);
g_free (playlist_file);
return 0;
}
playlist = g_ptr_array_new ();
if (playlist_file != NULL) {
gchar *playlist_contents = NULL;
gchar **lines = NULL;
if (g_file_get_contents (playlist_file, &playlist_contents, NULL, &err)) {
lines = g_strsplit (playlist_contents, "\n", 0);
num = g_strv_length (lines);
for (i = 0; i < num; i++) {
if (lines[i][0] != '\0') {
GST_LOG ("Playlist[%d]: %s", i + 1, lines[i]);
add_to_playlist (playlist, lines[i]);
}
}
g_strfreev (lines);
g_free (playlist_contents);
} else {
g_printerr ("Could not read playlist: %s\n", err->message);
g_clear_error (&err);
}
g_free (playlist_file);
playlist_file = NULL;
}
if (playlist->len == 0 && (filenames == NULL || *filenames == NULL)) {
g_printerr (_("Usage: %s FILE1|URI1 [FILE2|URI2] [FILE3|URI3] ..."),
"gst-play-" GST_API_VERSION);
g_printerr ("\n\n"),
g_printerr ("%s\n\n",
_("You must provide at least one filename or URI to play."));
/* No input provided. Free array */
g_ptr_array_free (playlist, TRUE);
g_free (audio_sink);
g_free (video_sink);
return 1;
}
/* fill playlist */
if (filenames != NULL && *filenames != NULL) {
num = g_strv_length (filenames);
for (i = 0; i < num; ++i) {
GST_LOG ("command line argument: %s", filenames[i]);
add_to_playlist (playlist, filenames[i]);
}
g_strfreev (filenames);
}
num = playlist->len;
g_ptr_array_add (playlist, NULL);
uris = (gchar **) g_ptr_array_free (playlist, FALSE);
if (shuffle)
shuffle_uris (uris, num);
/* prepare */
play =
play_new (uris, audio_sink, video_sink, gapless, volume, verbose, flags);
if (play == NULL) {
g_printerr
("Failed to create 'playbin' element. Check your GStreamer installation.\n");
return EXIT_FAILURE;
}
if (interactive) {
if (gst_play_kb_set_key_handler (keyboard_cb, play)) {
g_print (_("Press 'k' to see a list of keyboard shortcuts.\n"));
atexit (restore_terminal);
} else {
g_print ("Interactive keyboard handling in terminal not available.\n");
}
}
/* play */
do_play (play);
/* clean up */
play_free (play);
g_free (audio_sink);
g_free (video_sink);
g_print ("\n");
gst_deinit ();
return 0;
}
int
main (int argc, char **argv)
{
GstPlay *play;
GPtrArray *playlist;
gboolean print_version = FALSE;
gboolean interactive = FALSE; /* FIXME: maybe enable by default? */
gboolean shuffle = FALSE;
gdouble volume = 1.0;
gchar **filenames = NULL;
gchar **uris;
guint num, i;
GError *err = NULL;
GOptionContext *ctx;
gchar *playlist_file = NULL;
GOptionEntry options[] = {
{"version", 0, 0, G_OPTION_ARG_NONE, &print_version,
"Print version information and exit", NULL},
{"shuffle", 0, 0, G_OPTION_ARG_NONE, &shuffle,
"Shuffle playlist", NULL},
{"interactive", 0, 0, G_OPTION_ARG_NONE, &interactive,
"Interactive control via keyboard", NULL},
{"volume", 0, 0, G_OPTION_ARG_DOUBLE, &volume,
"Volume", NULL},
{"playlist", 0, 0, G_OPTION_ARG_FILENAME, &playlist_file,
"Playlist file containing input media files", NULL},
{G_OPTION_REMAINING, 0, 0, G_OPTION_ARG_FILENAME_ARRAY, &filenames, NULL},
{NULL}
};
g_set_prgname ("gst-play");
ctx = g_option_context_new ("FILE1|URI1 [FILE2|URI2] [FILE3|URI3] ...");
g_option_context_add_main_entries (ctx, options, NULL);
g_option_context_add_group (ctx, gst_init_get_option_group ());
if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
return 1;
}
g_option_context_free (ctx);
GST_DEBUG_CATEGORY_INIT (play_debug, "play", 0, "gst-play");
if (print_version) {
gchar *version_str;
version_str = gst_version_string ();
g_print ("%s version %s\n", g_get_prgname (), "1.0");
g_print ("%s\n", version_str);
g_free (version_str);
g_free (playlist_file);
return 0;
}
playlist = g_ptr_array_new ();
if (playlist_file != NULL) {
gchar *playlist_contents = NULL;
gchar **lines = NULL;
if (g_file_get_contents (playlist_file, &playlist_contents, NULL, &err)) {
lines = g_strsplit (playlist_contents, "\n", 0);
num = g_strv_length (lines);
for (i = 0; i < num; i++) {
if (lines[i][0] != '\0') {
GST_LOG ("Playlist[%d]: %s", i + 1, lines[i]);
add_to_playlist (playlist, lines[i]);
}
}
g_strfreev (lines);
g_free (playlist_contents);
} else {
g_printerr ("Could not read playlist: %s\n", err->message);
g_clear_error (&err);
}
g_free (playlist_file);
playlist_file = NULL;
}
if (playlist->len == 0 && (filenames == NULL || *filenames == NULL)) {
g_printerr ("Usage: %s FILE1|URI1 [FILE2|URI2] [FILE3|URI3] ...",
"gst-play");
g_printerr ("\n\n"),
g_printerr ("%s\n\n",
"You must provide at least one filename or URI to play.");
/* No input provided. Free array */
g_ptr_array_free (playlist, TRUE);
return 1;
}
/* fill playlist */
if (filenames != NULL && *filenames != NULL) {
num = g_strv_length (filenames);
for (i = 0; i < num; ++i) {
GST_LOG ("command line argument: %s", filenames[i]);
add_to_playlist (playlist, filenames[i]);
}
g_strfreev (filenames);
}
num = playlist->len;
g_ptr_array_add (playlist, NULL);
uris = (gchar **) g_ptr_array_free (playlist, FALSE);
if (shuffle)
shuffle_uris (uris, num);
/* prepare */
play = play_new (uris, volume);
if (interactive) {
if (gst_play_kb_set_key_handler (keyboard_cb, play)) {
atexit (restore_terminal);
} else {
g_print ("Interactive keyboard handling in terminal not available.\n");
}
}
/* play */
do_play (play);
/* clean up */
play_free (play);
g_print ("\n");
return 0;
}