mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-25 03:01:03 +00:00
Code Issues 1 Projects Releases Wiki Activity

Better error recovery in teh MPEG1 decoder.

Browse source 
Original commit message from CVS:
Better error recovery in teh MPEG1 decoder.
Removed MPEG2 stuff from the MPEG1 decoderand make it ignore MPEG2
streams.
Some extra framerates in the MPEG2 decoder (from libmpeg2)
Round image size up to the nearest multiple of 16 (MPEG2 decoder)
This commit is contained in:
Wim Taymans 2000-07-13 19:49:50 +00:00
parent 9920b88c3f
commit 2ab64206bb
6 changed files with 47 additions and 29 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

9
gst/elements/gstaudiosink.c
View file

@ -224,15 +224,16 @@ void gst_audiosink_chain(GstPad *pad,GstBuffer *buf) {
//g_print("audiosink: writing to soundcard\n");
if (audiosink->fd > 2) {
if (!audiosink->mute) {
if (gst_clock_current_diff(audiosink->clock, GST_BUFFER_TIMESTAMP(buf)) > 500000) {
}
else {
//if (gst_clock_current_diff(audiosink->clock, GST_BUFFER_TIMESTAMP(buf)) > 500000) {
//}
//else {
gst_clock_wait(audiosink->clock, GST_BUFFER_TIMESTAMP(buf), GST_OBJECT(audiosink));
ioctl(audiosink->fd,SNDCTL_DSP_GETOSPACE,&ospace);
DEBUG("audiosink: (%d bytes buffer)\n", ospace.bytes);
write(audiosink->fd,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
//}
//gst_clock_set(audiosink->clock, GST_BUFFER_TIMESTAMP(buf));
}
//}
}
}
}

3
gst/gstclock.c
View file

@ -126,7 +126,8 @@ void gst_clock_wait(GstClock *clock, GstClockTime time, GstObject *obj) {
if (!tfnow.tv_sec) {
select(0, NULL, NULL, NULL, &tfnow);
}
else fprintf(stderr, "gst_clock: waiting %u %llu %llu %llu seconds\n", (int)tfnow.tv_sec, now, diff, time);
else printf("gst_clock: %s waiting %u %llu %llu %llu seconds\n", gst_element_get_name(GST_ELEMENT(obj)),
(int)tfnow.tv_sec, now, diff, time);
//DEBUG("gst_clock: 0x%p waiting for time %llu %llu %lld %llu\n", obj, time, target, diff, now);
//DEBUG("waiting %d.%08d\n",tfnow.tv_sec, tfnow.tv_usec);
//DEBUG("gst_clock: 0x%p waiting done time %llu \n", obj, time);

6
gst/gstthread.c
View file

@ -253,9 +253,10 @@ static gboolean gst_thread_change_state(GstElement *element,
GST_FLAG_UNSET(thread,GST_THREAD_STATE_SPINNING);
GST_FLAG_SET(thread,GST_THREAD_STATE_REAPING);
gst_thread_signal_thread(thread);
pthread_join(thread->thread_id,0);
//pthread_join(thread->thread_id,0);
/* tear down the internal state */
gst_info("gstthread: tearing down thread's iteration state\n");
gst_info("gstthread: tearing down thread's \"%s\" iteration state\n",
gst_element_get_name(GST_ELEMENT(element)));
/* FIXME do stuff */
break;
case GST_STATE_PLAYING:
@ -303,6 +304,7 @@ void *gst_thread_main_loop(void *arg) {
}
GST_FLAG_UNSET(thread,GST_THREAD_STATE_REAPING);
//pthread_join(thread->thread_id,0);
gst_info("gstthread: thread \"%s\" is stopped\n",
gst_element_get_name(GST_ELEMENT(thread)));

24
gstplay/gstplay.c
View file

@ -95,6 +95,23 @@ void mute_audio(gboolean mute) {
gtk_object_set(GTK_OBJECT(audio_play),"mute",mute,NULL);
}
gint delete_event(GtkWidget *widget, GdkEvent *event, gpointer data)
{
gdk_threads_leave();
g_print("setting to ~PLAYING state\n");
gst_element_set_state(GST_ELEMENT(pipeline),~GST_STATE_PLAYING);
g_print("setting to ~RUNNING state\n");
gst_element_set_state(GST_ELEMENT(pipeline),~GST_STATE_RUNNING);
gdk_threads_enter();
return FALSE;
}
void destroy(GtkWidget *widget, gpointer data)
{
gtk_main_quit();
}
void gstplay_parse_state_changed(GstElement *element, gint state, gpointer data)
{
printf("gstplay: element \"%s\" state changed %d\n", gst_element_get_name(element), state);
@ -221,9 +238,14 @@ main (int argc, char *argv[])
g_return_val_if_fail(video_render_thread != NULL, -1);
show = gst_elementfactory_make("videosink","show");
g_return_val_if_fail(show != NULL, -1);
gtk_object_set(GTK_OBJECT(show),"xv_enabled",FALSE,NULL);
//gtk_object_set(GTK_OBJECT(show),"xv_enabled",FALSE,NULL);
window1 = create_window1 (gst_util_get_widget_arg(GTK_OBJECT(show),"widget"));
gtk_widget_show (window1);
gtk_signal_connect(GTK_OBJECT(window1),"delete_event",
GTK_SIGNAL_FUNC(delete_event),NULL);
gtk_signal_connect(GTK_OBJECT(window1),"destroy",
GTK_SIGNAL_FUNC(destroy),pipeline);
gtk_signal_connect(GTK_OBJECT(show),"frame_displayed",
GTK_SIGNAL_FUNC(frame_displayed),NULL);
gst_bin_add(GST_BIN(video_render_thread),GST_ELEMENT(show));

25
gstplay/mpeg2.c
View file

@ -1,5 +1,5 @@
#define BUFFER 15
#define BUFFER 20
#define VIDEO_DECODER "mpeg2play"
#ifdef HAVE_CONFIG_H
@ -13,7 +13,7 @@
extern gboolean _gst_plugin_spew;
extern GstElement *show, *audio_play;
extern GstElement *video_render_queue, *audio_render_queue;
void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
{
@ -25,7 +25,7 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
// connect to audio pad
//if (0) {
if (strncmp(gst_pad_get_name(pad), "private_stream_1.0", 18) == 0 && audio_play) {
if (strncmp(gst_pad_get_name(pad), "private_stream_1.0", 18) == 0) {
gst_plugin_load("ac3parse");
gst_plugin_load("ac3dec");
// construct internal pipeline elements
@ -39,7 +39,6 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
g_return_if_fail(audio_thread != NULL);
gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(parse_audio));
gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(decode));
gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(audio_play));
// set up pad connections
gst_element_add_ghost_pad(GST_ELEMENT(audio_thread),
@ -47,7 +46,7 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
gst_pad_connect(gst_element_get_pad(parse_audio,"src"),
gst_element_get_pad(decode,"sink"));
gst_pad_connect(gst_element_get_pad(decode,"src"),
gst_element_get_pad(audio_play,"sink"));
gst_element_get_pad(audio_render_queue,"sink"));
// construct queue and connect everything in the main pipelie
audio_queue = gst_elementfactory_make("queue","audio_queue");
@ -63,12 +62,10 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
gtk_object_set(GTK_OBJECT(audio_thread),"create_thread",TRUE,NULL);
g_print("setting to RUNNING state\n");
gst_element_set_state(GST_ELEMENT(audio_thread),GST_STATE_RUNNING);
g_print("setting to PLAYING state\n");
gst_element_set_state(GST_ELEMENT(audio_thread),GST_STATE_PLAYING);
}
// connect to audio pad
//if (0) {
if (strncmp(gst_pad_get_name(pad), "audio_", 6) == 0 && audio_play) {
if (strncmp(gst_pad_get_name(pad), "audio_", 6) == 0) {
gst_plugin_load("mp3parse");
gst_plugin_load("mpg123");
// construct internal pipeline elements
@ -82,7 +79,6 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
g_return_if_fail(audio_thread != NULL);
gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(parse_audio));
gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(decode));
gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(audio_play));
// set up pad connections
gst_element_add_ghost_pad(GST_ELEMENT(audio_thread),
@ -90,7 +86,7 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
gst_pad_connect(gst_element_get_pad(parse_audio,"src"),
gst_element_get_pad(decode,"sink"));
gst_pad_connect(gst_element_get_pad(decode,"src"),
gst_element_get_pad(audio_play,"sink"));
gst_element_get_pad(audio_render_queue,"sink"));
// construct queue and connect everything in the main pipelie
audio_queue = gst_elementfactory_make("queue","audio_queue");
@ -106,11 +102,9 @@ void mpeg2_new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline)
gtk_object_set(GTK_OBJECT(audio_thread),"create_thread",TRUE,NULL);
g_print("setting to RUNNING state\n");
gst_element_set_state(GST_ELEMENT(audio_thread),GST_STATE_RUNNING);
g_print("setting to PLAYING state\n");
gst_element_set_state(GST_ELEMENT(audio_thread),GST_STATE_PLAYING);
} else if (strncmp(gst_pad_get_name(pad), "video_", 6) == 0) {
//} else if (0) {
mpeg2_setup_video_thread(pad, show, pipeline);
mpeg2_setup_video_thread(pad, video_render_queue, pipeline);
}
}
@ -133,7 +127,6 @@ void mpeg2_setup_video_thread(GstPad *pad, GstElement *show, GstElement *pipelin
g_return_if_fail(video_thread != NULL);
gst_bin_add(GST_BIN(video_thread),GST_ELEMENT(parse_video));
gst_bin_add(GST_BIN(video_thread),GST_ELEMENT(decode_video));
gst_bin_add(GST_BIN(video_thread),GST_ELEMENT(show));
// set up pad connections
gst_element_add_ghost_pad(GST_ELEMENT(video_thread),
@ -141,7 +134,7 @@ void mpeg2_setup_video_thread(GstPad *pad, GstElement *show, GstElement *pipelin
gst_pad_connect(gst_element_get_pad(parse_video,"src"),
gst_element_get_pad(decode_video,"sink"));
gst_pad_connect(gst_element_get_pad(decode_video,"src"),
gst_element_get_pad(show,"sink"));
gst_element_get_pad(video_render_queue,"sink"));
// construct queue and connect everything in the main pipeline
video_queue = gst_elementfactory_make("queue","video_queue");
@ -157,8 +150,6 @@ void mpeg2_setup_video_thread(GstPad *pad, GstElement *show, GstElement *pipelin
gtk_object_set(GTK_OBJECT(video_thread),"create_thread",TRUE,NULL);
g_print("setting to RUNNING state\n");
gst_element_set_state(GST_ELEMENT(video_thread),GST_STATE_RUNNING);
g_print("setting to PLAYING state\n");
gst_element_set_state(GST_ELEMENT(video_thread),GST_STATE_PLAYING);
g_print("\n");
}

9
plugins/elements/gstaudiosink.c
View file

@ -224,15 +224,16 @@ void gst_audiosink_chain(GstPad *pad,GstBuffer *buf) {
//g_print("audiosink: writing to soundcard\n");
if (audiosink->fd > 2) {
if (!audiosink->mute) {
if (gst_clock_current_diff(audiosink->clock, GST_BUFFER_TIMESTAMP(buf)) > 500000) {
}
else {
//if (gst_clock_current_diff(audiosink->clock, GST_BUFFER_TIMESTAMP(buf)) > 500000) {
//}
//else {
gst_clock_wait(audiosink->clock, GST_BUFFER_TIMESTAMP(buf), GST_OBJECT(audiosink));
ioctl(audiosink->fd,SNDCTL_DSP_GETOSPACE,&ospace);
DEBUG("audiosink: (%d bytes buffer)\n", ospace.bytes);
write(audiosink->fd,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
//}
//gst_clock_set(audiosink->clock, GST_BUFFER_TIMESTAMP(buf));
}
//}
}
}
}