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examples/gstplay/player.c: Don't iterate.
Original commit message from CVS: * examples/gstplay/player.c: (main): Don't iterate. * examples/seeking/seek.c: (fixate), (make_playerbin_pipeline): Add visualizations. * ext/a52dec/gsta52dec.c: (gst_a52dec_push), (gst_a52dec_handle_frame): Set duration. * ext/dvdnav/gst-dvd: Add audioconvert. Fixes #161325. * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get): Explicitely case to gint64. Possible valgrind error. * gst-libs/gst/play/play.c: (caps_set), (setup_size), (gst_play_tick_callback), (gst_play_change_state), (gst_play_dispose), (gst_play_init), (gst_play_class_init), (gst_play_set_location), (gst_play_get_location), (gst_play_seek_to_time), (gst_play_set_data_src), (gst_play_set_video_sink), (gst_play_set_audio_sink), (gst_play_set_visualization), (gst_play_connect_visualization), (gst_play_get_framerate), (gst_play_get_all_by_interface), (gst_play_new): Use playbin. Fixes #139749 and #147744. * gst/apetag/apedemux.c: (gst_ape_demux_parse_tags): Add genre tag. * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type), (audioscale_get_type), (gst_audioscale_base_init), (gst_audioscale_class_init), (gst_audioscale_expand_caps), (gst_audioscale_getcaps), (gst_audioscale_fixate), (gst_audioscale_link), (gst_audioscale_get_buffer), (gst_audioscale_decrease_rate), (gst_audioscale_increase_rate), (gst_audioscale_init), (gst_audioscale_dispose), (gst_audioscale_chain), (gst_audioscale_set_property), (gst_audioscale_get_property), (plugin_init): Indent properly. * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private): Fix LPCM. * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num), (qtdemux_tag_add_gnre), (qtdemux_video_caps): Add more metadata (fixes #162656).
This commit is contained in:
Ronald S. Bultje
parent
795f383c21
commit
6e1c77c85e
7 changed files with 473 additions and 1063 deletions
42
ChangeLog
42
ChangeLog
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@ -1,3 +1,45 @@
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2005-01-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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* examples/gstplay/player.c: (main):
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Don't iterate.
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* examples/seeking/seek.c: (fixate), (make_playerbin_pipeline):
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Add visualizations.
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* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
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(gst_a52dec_handle_frame):
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Set duration.
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* ext/dvdnav/gst-dvd:
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Add audioconvert. Fixes #161325.
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* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get):
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Explicitely case to gint64. Possible valgrind error.
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* gst-libs/gst/play/play.c: (caps_set), (setup_size),
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(gst_play_tick_callback), (gst_play_change_state),
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(gst_play_dispose), (gst_play_init), (gst_play_class_init),
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(gst_play_set_location), (gst_play_get_location),
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(gst_play_seek_to_time), (gst_play_set_data_src),
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(gst_play_set_video_sink), (gst_play_set_audio_sink),
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(gst_play_set_visualization), (gst_play_connect_visualization),
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(gst_play_get_framerate), (gst_play_get_all_by_interface),
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(gst_play_new):
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Use playbin. Fixes #139749 and #147744.
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* gst/apetag/apedemux.c: (gst_ape_demux_parse_tags):
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Add genre tag.
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* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
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(audioscale_get_type), (gst_audioscale_base_init),
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(gst_audioscale_class_init), (gst_audioscale_expand_caps),
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(gst_audioscale_getcaps), (gst_audioscale_fixate),
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(gst_audioscale_link), (gst_audioscale_get_buffer),
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(gst_audioscale_decrease_rate), (gst_audioscale_increase_rate),
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(gst_audioscale_init), (gst_audioscale_dispose),
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(gst_audioscale_chain), (gst_audioscale_set_property),
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(gst_audioscale_get_property), (plugin_init):
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Indent properly.
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* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private):
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Fix LPCM.
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* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta),
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(qtdemux_tag_add_str), (qtdemux_tag_add_num),
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(qtdemux_tag_add_gnre), (qtdemux_video_caps):
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Add more metadata (fixes #162656).
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2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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@ -90,13 +90,6 @@ seek_timer (GstPlay * play)
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return FALSE;
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}
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static gboolean
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idle_iterate (GstPlay * play)
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{
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gst_bin_iterate (GST_BIN (play));
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return (GST_STATE (GST_ELEMENT (play)) == GST_STATE_PLAYING);
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}
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int
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main (int argc, char *argv[])
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{
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@ -168,7 +161,6 @@ main (int argc, char *argv[])
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GST_STATE_PLAYING) == GST_STATE_FAILURE)
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g_error ("Could not set state to PLAYING");
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g_idle_add ((GSourceFunc) idle_iterate, play);
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g_timeout_add (20000, (GSourceFunc) seek_timer, play);
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g_main_loop_run (loop);
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@ -551,15 +551,45 @@ make_mpegnt_pipeline (const gchar * location)
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return pipeline;
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}
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static GstCaps *
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fixate (GstPad * pad, const GstCaps * in_caps, gpointer data)
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{
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GstCaps *caps;
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GstStructure *s;
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if (gst_caps_get_size (in_caps) > 1)
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return NULL;
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/* nothing if fixed already */
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s = gst_caps_get_structure (in_caps, 0);
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if (gst_structure_has_field_typed (s, "width", G_TYPE_INT) &&
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gst_structure_has_field_typed (s, "height", G_TYPE_INT) &&
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gst_structure_has_field_typed (s, "framerate", G_TYPE_DOUBLE))
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return NULL;
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/* fixate */
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caps = gst_caps_copy (in_caps);
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s = gst_caps_get_structure (caps, 0);
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gst_caps_structure_fixate_field_nearest_int (s, "width", 200);
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gst_caps_structure_fixate_field_nearest_int (s, "height", 150);
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gst_caps_structure_fixate_field_nearest_double (s, "framerate", 10.0);
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return caps;
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}
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static GstElement *
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make_playerbin_pipeline (const gchar * location)
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{
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GstElement *player;
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GstElement *player, *vis;
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player = gst_element_factory_make ("playbin", "player");
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vis = gst_element_factory_make ("synaesthesia", "vis");
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g_assert (player);
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g_assert (vis);
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g_object_set (G_OBJECT (player), "uri", location, NULL);
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g_signal_connect (gst_element_get_pad (vis, "src"), "fixate",
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G_CALLBACK (fixate), NULL);
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g_object_set (G_OBJECT (player), "uri", location, "vis-plugin", vis, NULL);
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seekable_elements = g_list_prepend (seekable_elements, player);
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File diff suppressed because it is too large
Load diff
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@ -66,42 +66,43 @@ enum
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"signed = (boolean) true")
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#if 0
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/* disabled because it segfaults */
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"audio/x-raw-float, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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#define NOTHING "audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " "width = (int) 32")
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#endif
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static GstStaticPadTemplate gst_audioscale_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static GstStaticPadTemplate gst_audioscale_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static GstStaticPadTemplate gst_audioscale_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static GstStaticPadTemplate gst_audioscale_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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#define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type())
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static GType gst_audioscale_method_get_type (void)
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{
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static GType audioscale_method_type = 0;
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static GEnumValue audioscale_methods[] =
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static GType
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gst_audioscale_method_get_type (void)
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{
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{
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GST_RESAMPLE_NEAREST, "0", "Nearest"}
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,
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{
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GST_RESAMPLE_BILINEAR, "1", "Bilinear"}
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, {
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GST_RESAMPLE_SINC, "2", "Sinc"}
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, {
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0, NULL, NULL}
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,};
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static GType audioscale_method_type = 0;
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static GEnumValue audioscale_methods[] = {
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{
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GST_RESAMPLE_NEAREST, "0", "Nearest"}
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,
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{
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GST_RESAMPLE_BILINEAR, "1", "Bilinear"}
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, {
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GST_RESAMPLE_SINC, "2", "Sinc"}
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, {
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0, NULL, NULL}
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,
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};
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if (!audioscale_method_type) {
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audioscale_method_type = g_enum_register_static ("GstAudioscaleMethod",
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audioscale_methods);
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}
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return audioscale_method_type;
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}
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if (!audioscale_method_type) {
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audioscale_method_type = g_enum_register_static ("GstAudioscaleMethod",
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audioscale_methods);
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}
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return audioscale_method_type;
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}
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static void gst_audioscale_base_init (gpointer g_class);
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static void gst_audioscale_class_init (AudioscaleClass * klass);
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@ -122,20 +123,21 @@ static GstElementClass *parent_class = NULL;
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/*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */
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GType audioscale_get_type (void)
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GType
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audioscale_get_type (void)
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{
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static GType audioscale_type = 0;
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if (!audioscale_type)
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{
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if (!audioscale_type) {
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static const GTypeInfo audioscale_info = {
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sizeof (AudioscaleClass),
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gst_audioscale_base_init,
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NULL,
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(GClassInitFunc) gst_audioscale_class_init,
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NULL,
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NULL,
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sizeof (Audioscale), 0, (GInstanceInitFunc) gst_audioscale_init,};
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sizeof (AudioscaleClass),
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gst_audioscale_base_init,
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NULL,
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(GClassInitFunc) gst_audioscale_class_init,
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NULL,
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NULL,
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sizeof (Audioscale), 0, (GInstanceInitFunc) gst_audioscale_init,
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};
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audioscale_type =
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g_type_register_static (GST_TYPE_ELEMENT, "Audioscale",
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@ -144,7 +146,8 @@ GType audioscale_get_type (void)
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return audioscale_type;
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}
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static void gst_audioscale_base_init (gpointer g_class)
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static void
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gst_audioscale_base_init (gpointer g_class)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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@ -156,7 +159,8 @@ static void gst_audioscale_base_init (gpointer g_class)
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gst_element_class_set_details (gstelement_class, &gst_audioscale_details);
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}
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static void gst_audioscale_class_init (AudioscaleClass * klass)
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static void
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gst_audioscale_class_init (AudioscaleClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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@ -185,25 +189,25 @@ static void gst_audioscale_class_init (AudioscaleClass * klass)
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}
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static GstStaticCaps gst_audioscale_passthru_caps =
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GST_STATIC_CAPS ("audio/x-raw-int, channels = [ 3, MAX ]");
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GST_STATIC_CAPS ("audio/x-raw-int, channels = [ 3, MAX ]");
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static GstStaticCaps gst_audioscale_convert_caps =
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GST_STATIC_CAPS ("audio/x-raw-int, channels = [ 1, 2 ]");
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GST_STATIC_CAPS ("audio/x-raw-int, channels = [ 1, 2 ]");
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static GstCaps *gst_audioscale_expand_caps (const GstCaps * caps)
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static GstCaps *
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gst_audioscale_expand_caps (const GstCaps * caps)
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{
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GstCaps *caps1, *caps2;
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int i;
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caps1 = gst_caps_intersect (caps,
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caps1 = gst_caps_intersect (caps,
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gst_static_caps_get (&gst_audioscale_passthru_caps));
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caps2 = gst_caps_intersect (caps,
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caps2 = gst_caps_intersect (caps,
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gst_static_caps_get (&gst_audioscale_convert_caps));
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for (i = 0; i < gst_caps_get_size (caps2); i++)
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{
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for (i = 0; i < gst_caps_get_size (caps2); i++) {
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GstStructure *structure = gst_caps_get_structure (caps2, i);
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gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
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gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
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NULL);
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}
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@ -212,7 +216,8 @@ static GstCaps *gst_audioscale_expand_caps (const GstCaps * caps)
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return caps1;
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}
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static GstCaps *gst_audioscale_getcaps (GstPad * pad)
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static GstCaps *
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gst_audioscale_getcaps (GstPad * pad)
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{
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Audioscale *audioscale;
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GstPad *otherpad;
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@ -230,7 +235,8 @@ static GstCaps *gst_audioscale_getcaps (GstPad * pad)
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return caps;
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}
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static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
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static GstCaps *
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gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
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{
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Audioscale *audioscale;
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gst_resample_t *r;
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@ -239,13 +245,12 @@ static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
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GstCaps *copy;
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GstStructure *structure;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = &(audioscale->gst_resample_template);
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = &(audioscale->gst_resample_template);
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if (pad == audioscale->srcpad) {
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otherpad = audioscale->sinkpad;
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rate = r->i_rate;
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} else
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{
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} else {
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otherpad = audioscale->srcpad;
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rate = r->o_rate;
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}
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@ -262,7 +267,8 @@ static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
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return NULL;
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}
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static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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static GstPadLinkReturn
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gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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{
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Audioscale *audioscale;
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gst_resample_t *r;
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@ -276,15 +282,14 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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GstPad *otherpad;
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GstCaps *copy;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = &(audioscale->gst_resample_template);
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = &(audioscale->gst_resample_template);
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if (pad == audioscale->srcpad) {
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otherpad = audioscale->sinkpad;
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rate = &r->o_rate;
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otherrate = &r->i_rate;
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} else
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{
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} else {
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otherpad = audioscale->srcpad;
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rate = &r->i_rate;
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otherrate = &r->o_rate;
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@ -380,26 +385,28 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
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return link_ret;
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}
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static void *gst_audioscale_get_buffer (void *priv, unsigned int size)
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static void *
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gst_audioscale_get_buffer (void *priv, unsigned int size)
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{
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Audioscale *audioscale = priv;
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GST_DEBUG ("size requested: %u irate: %f orate: %f", size,
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GST_DEBUG ("size requested: %u irate: %f orate: %f", size,
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audioscale->gst_resample->i_rate, audioscale->gst_resample->o_rate);
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audioscale->outbuf = gst_buffer_new ();
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GST_BUFFER_SIZE (audioscale->outbuf) = size;
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GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size);
|
||||
GST_BUFFER_TIMESTAMP (audioscale->outbuf) =
|
||||
audioscale->outbuf = gst_buffer_new ();
|
||||
GST_BUFFER_SIZE (audioscale->outbuf) = size;
|
||||
GST_BUFFER_DATA (audioscale->outbuf) = g_malloc (size);
|
||||
GST_BUFFER_TIMESTAMP (audioscale->outbuf) =
|
||||
audioscale->gst_resample_offset * GST_SECOND /
|
||||
audioscale->gst_resample->o_rate;
|
||||
audioscale->gst_resample_offset +=
|
||||
audioscale->gst_resample_offset +=
|
||||
size / sizeof (gint16) / audioscale->gst_resample->channels;
|
||||
|
||||
return GST_BUFFER_DATA (audioscale->outbuf);
|
||||
return GST_BUFFER_DATA (audioscale->outbuf);
|
||||
}
|
||||
|
||||
/* reduces rate by factor of 2 */
|
||||
GstBuffer *gst_audioscale_decrease_rate (Audioscale * audioscale,
|
||||
GstBuffer *
|
||||
gst_audioscale_decrease_rate (Audioscale * audioscale,
|
||||
GstBuffer * buf, double outrate, int cur_iteration)
|
||||
{
|
||||
gint i, j, curoffset;
|
||||
|
|
@ -436,7 +443,8 @@ GstBuffer *gst_audioscale_decrease_rate (Audioscale * audioscale,
|
|||
}
|
||||
|
||||
/* increases rate by factor of 2 */
|
||||
GstBuffer *gst_audioscale_increase_rate (Audioscale * audioscale,
|
||||
GstBuffer *
|
||||
gst_audioscale_increase_rate (Audioscale * audioscale,
|
||||
GstBuffer * buf, double outrate, int cur_iteration)
|
||||
{
|
||||
gint i, j, curoffset;
|
||||
|
|
@ -473,7 +481,8 @@ GstBuffer *gst_audioscale_increase_rate (Audioscale * audioscale,
|
|||
return outbuf;
|
||||
}
|
||||
|
||||
static void gst_audioscale_init (Audioscale * audioscale)
|
||||
static void
|
||||
gst_audioscale_init (Audioscale * audioscale)
|
||||
{
|
||||
gst_resample_t *r;
|
||||
|
||||
|
|
@ -516,7 +525,8 @@ static void gst_audioscale_init (Audioscale * audioscale)
|
|||
audioscale->increase = FALSE;
|
||||
}
|
||||
|
||||
static void gst_audioscale_dispose (GObject * object)
|
||||
static void
|
||||
gst_audioscale_dispose (GObject * object)
|
||||
{
|
||||
Audioscale *audioscale = GST_AUDIOSCALE (object);
|
||||
|
||||
|
|
@ -532,7 +542,8 @@ static void gst_audioscale_dispose (GObject * object)
|
|||
G_OBJECT_CLASS (parent_class)->dispose (object);
|
||||
}
|
||||
|
||||
static void gst_audioscale_chain (GstPad * pad, GstData * _data)
|
||||
static void
|
||||
gst_audioscale_chain (GstPad * pad, GstData * _data)
|
||||
{
|
||||
GstBuffer *buf = GST_BUFFER (_data);
|
||||
GstBuffer *tempbuf, *tempbuf2;
|
||||
|
|
@ -596,7 +607,8 @@ static void gst_audioscale_chain (GstPad * pad, GstData * _data)
|
|||
|
||||
}
|
||||
|
||||
static GstElementStateReturn gst_audioscale_change_state (GstElement * element)
|
||||
static GstElementStateReturn
|
||||
gst_audioscale_change_state (GstElement * element)
|
||||
{
|
||||
Audioscale *audioscale = GST_AUDIOSCALE (element);
|
||||
|
||||
|
|
@ -612,16 +624,16 @@ static GstElementStateReturn gst_audioscale_change_state (GstElement * element)
|
|||
}
|
||||
|
||||
static void
|
||||
gst_audioscale_set_property (GObject * object, guint prop_id,
|
||||
gst_audioscale_set_property (GObject * object, guint prop_id,
|
||||
const GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
Audioscale *src;
|
||||
gst_resample_t *r;
|
||||
|
||||
/* it's not null if we got it, but it might not be ours */
|
||||
g_return_if_fail (GST_IS_AUDIOSCALE (object));
|
||||
src = GST_AUDIOSCALE (object);
|
||||
r = &(src->gst_resample_template);
|
||||
g_return_if_fail (GST_IS_AUDIOSCALE (object));
|
||||
src = GST_AUDIOSCALE (object);
|
||||
r = &(src->gst_resample_template);
|
||||
|
||||
switch (prop_id) {
|
||||
case ARG_FILTERLEN:
|
||||
|
|
@ -629,9 +641,11 @@ static void
|
|||
GST_DEBUG_OBJECT (GST_ELEMENT (src), "new filter length %d\n",
|
||||
r->filter_length);
|
||||
break;
|
||||
case ARG_METHOD:r->method = g_value_get_enum (value);
|
||||
case ARG_METHOD:
|
||||
r->method = g_value_get_enum (value);
|
||||
break;
|
||||
default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
|
||||
|
|
@ -639,7 +653,7 @@ static void
|
|||
}
|
||||
|
||||
static void
|
||||
gst_audioscale_get_property (GObject * object, guint prop_id,
|
||||
gst_audioscale_get_property (GObject * object, guint prop_id,
|
||||
GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
Audioscale *src;
|
||||
|
|
@ -662,7 +676,8 @@ static void
|
|||
}
|
||||
|
||||
|
||||
static gboolean plugin_init (GstPlugin * plugin)
|
||||
static gboolean
|
||||
plugin_init (GstPlugin * plugin)
|
||||
{
|
||||
/* load support library */
|
||||
if (!gst_library_load ("gstresample"))
|
||||
|
|
|
|||
|
|
@ -551,15 +551,45 @@ make_mpegnt_pipeline (const gchar * location)
|
|||
return pipeline;
|
||||
}
|
||||
|
||||
static GstCaps *
|
||||
fixate (GstPad * pad, const GstCaps * in_caps, gpointer data)
|
||||
{
|
||||
GstCaps *caps;
|
||||
GstStructure *s;
|
||||
|
||||
if (gst_caps_get_size (in_caps) > 1)
|
||||
return NULL;
|
||||
|
||||
/* nothing if fixed already */
|
||||
s = gst_caps_get_structure (in_caps, 0);
|
||||
if (gst_structure_has_field_typed (s, "width", G_TYPE_INT) &&
|
||||
gst_structure_has_field_typed (s, "height", G_TYPE_INT) &&
|
||||
gst_structure_has_field_typed (s, "framerate", G_TYPE_DOUBLE))
|
||||
return NULL;
|
||||
|
||||
/* fixate */
|
||||
caps = gst_caps_copy (in_caps);
|
||||
s = gst_caps_get_structure (caps, 0);
|
||||
gst_caps_structure_fixate_field_nearest_int (s, "width", 200);
|
||||
gst_caps_structure_fixate_field_nearest_int (s, "height", 150);
|
||||
gst_caps_structure_fixate_field_nearest_double (s, "framerate", 10.0);
|
||||
|
||||
return caps;
|
||||
}
|
||||
|
||||
static GstElement *
|
||||
make_playerbin_pipeline (const gchar * location)
|
||||
{
|
||||
GstElement *player;
|
||||
GstElement *player, *vis;
|
||||
|
||||
player = gst_element_factory_make ("playbin", "player");
|
||||
vis = gst_element_factory_make ("synaesthesia", "vis");
|
||||
g_assert (player);
|
||||
g_assert (vis);
|
||||
|
||||
g_object_set (G_OBJECT (player), "uri", location, NULL);
|
||||
g_signal_connect (gst_element_get_pad (vis, "src"), "fixate",
|
||||
G_CALLBACK (fixate), NULL);
|
||||
g_object_set (G_OBJECT (player), "uri", location, "vis-plugin", vis, NULL);
|
||||
|
||||
seekable_elements = g_list_prepend (seekable_elements, player);
|
||||
|
||||
|
|
|
|||
|
|
@ -90,13 +90,6 @@ seek_timer (GstPlay * play)
|
|||
return FALSE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
idle_iterate (GstPlay * play)
|
||||
{
|
||||
gst_bin_iterate (GST_BIN (play));
|
||||
return (GST_STATE (GST_ELEMENT (play)) == GST_STATE_PLAYING);
|
||||
}
|
||||
|
||||
int
|
||||
main (int argc, char *argv[])
|
||||
{
|
||||
|
|
@ -168,7 +161,6 @@ main (int argc, char *argv[])
|
|||
GST_STATE_PLAYING) == GST_STATE_FAILURE)
|
||||
g_error ("Could not set state to PLAYING");
|
||||
|
||||
g_idle_add ((GSourceFunc) idle_iterate, play);
|
||||
g_timeout_add (20000, (GSourceFunc) seek_timer, play);
|
||||
|
||||
g_main_loop_run (loop);
|
||||
|
|
|
|||
Loading…
Reference in a new issue