Rewrote the mpeg1 system stream parser.

Original commit message from CVS:
Rewrote the mpeg1 system stream parser.
Fixed a motion compensation bug (was a bug on Intels site)
Faster/Better/more accurate motion detection in the mpeg encoder
Added an audio resampler (convert 48000 kHz to 44100 for example) does not work
Fixed rate control in the mpeg2 encoder.

diff --git a/configure.in b/configure.in
index 542948b..b3349b3 100644 (file)
--- a/configure.in
+++ b/configure.in
@@ -361,6 +361,7 @@ plugins/visualization/synaesthesia/Makefile
 plugins/visualization/smoothwave/Makefile
 plugins/videosink/Makefile
 plugins/videoscale/Makefile
+plugins/audioscale/Makefile
 plugins/dvdsrc/Makefile
 plugins/vcdsrc/Makefile
 plugins/cobin/Makefile

diff --git a/gst/elements/Makefile.am b/gst/elements/Makefile.am
index 49cc8dc..f07e48c 100644 (file)
--- a/gst/elements/Makefile.am
+++ b/gst/elements/Makefile.am
@@ -21,6 +21,7 @@ libgstelements_la_SOURCES =   \
        gstaudiosink.c          \
        gstaudiosrc.c           \
        gstfdsink.c             \
+       gstpipefilter.c         \
        gstqueue.c              \
        gstsinesrc.c
 
@@ -35,6 +36,7 @@ noinst_HEADERS =              \
        gstaudiosink.h          \
        gstaudiosrc.h           \
        gstfdsink.h             \
+       gstpipefilter.h         \
        gstqueue.h              \
        gstsinesrc.h
 

diff --git a/gst/elements/gstelements.c b/gst/elements/gstelements.c
index 030ea57..bf4ba0e 100644 (file)
--- a/gst/elements/gstelements.c
+++ b/gst/elements/gstelements.c
@@ -32,6 +32,7 @@
 #if HAVE_LIBGHTTP
 #include <gsthttpsrc.h>
 #endif /* HAVE_LIBGHTTP */
+#include <gstpipefilter.h>
 #include <gstqueue.h>
 #include <gstsinesrc.h>
 
@@ -55,6 +56,7 @@ struct _elements_entry _elements[] = {
 #if HAVE_LIBGHTTP
   { "httpsrc", gst_httpsrc_get_type, &gst_httpsrc_details },
 #endif /* HAVE_LIBGHTTP */
+  { "pipefilter", gst_pipefilter_get_type, &gst_pipefilter_details },
   { "queue", gst_queue_get_type, &gst_queue_details },
   { "sinesrc", gst_sinesrc_get_type, &gst_sinesrc_details },
   { NULL, 0 },

diff --git a/plugins/elements/Makefile.am b/plugins/elements/Makefile.am
index 49cc8dc..f07e48c 100644 (file)
--- a/plugins/elements/Makefile.am
+++ b/plugins/elements/Makefile.am
@@ -21,6 +21,7 @@ libgstelements_la_SOURCES =   \
        gstaudiosink.c          \
        gstaudiosrc.c           \
        gstfdsink.c             \
+       gstpipefilter.c         \
        gstqueue.c              \
        gstsinesrc.c
 
@@ -35,6 +36,7 @@ noinst_HEADERS =              \
        gstaudiosink.h          \
        gstaudiosrc.h           \
        gstfdsink.h             \
+       gstpipefilter.h         \
        gstqueue.h              \
        gstsinesrc.h
 

diff --git a/plugins/elements/gstelements.c b/plugins/elements/gstelements.c
index 030ea57..bf4ba0e 100644 (file)
--- a/plugins/elements/gstelements.c
+++ b/plugins/elements/gstelements.c
@@ -32,6 +32,7 @@
 #if HAVE_LIBGHTTP
 #include <gsthttpsrc.h>
 #endif /* HAVE_LIBGHTTP */
+#include <gstpipefilter.h>
 #include <gstqueue.h>
 #include <gstsinesrc.h>
 
@@ -55,6 +56,7 @@ struct _elements_entry _elements[] = {
 #if HAVE_LIBGHTTP
   { "httpsrc", gst_httpsrc_get_type, &gst_httpsrc_details },
 #endif /* HAVE_LIBGHTTP */
+  { "pipefilter", gst_pipefilter_get_type, &gst_pipefilter_details },
   { "queue", gst_queue_get_type, &gst_queue_details },
   { "sinesrc", gst_sinesrc_get_type, &gst_sinesrc_details },
   { NULL, 0 },

diff --git a/test/Makefile.am b/test/Makefile.am
index 25dce54..223e7bb 100644 (file)
--- a/test/Makefile.am
+++ b/test/Makefile.am
@@ -1,7 +1,7 @@
 #noinst_PROGRAMS = basic m types a r plugin w s args mpg123 mcut push qtest
 noinst_PROGRAMS = qtest spectrum record wave mp3 teardown buffer mp3parse \
                  mpeg2parse mp1parse mp3play ac3parse ac3play dvdcat fake cobin videotest \
-                       aviparse vidcapture avi2mpg mp2tomp1 mp1tomp1
+                       aviparse vidcapture avi2mpg mp2tomp1 mp1tomp1 pipetest
 
 SUBDIRS = xml cothreads bindings
 

diff --git a/test/avi2mpg.c b/test/avi2mpg.c
index 23203e6..784f5c3 100644 (file)
--- a/test/avi2mpg.c
+++ b/test/avi2mpg.c
@@ -67,7 +67,8 @@ void new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline) {
     smooth = gst_elementfactory_make("smooth","smooth");
     //smooth = gst_elementfactory_make("median","median");
     g_return_if_fail(smooth != NULL);
-    //gtk_object_set(GTK_OBJECT(smooth),"filtersize",9,NULL);
+    gtk_object_set(GTK_OBJECT(smooth),"filtersize",2,NULL);
+    gtk_object_set(GTK_OBJECT(smooth),"tolerance",4,NULL);
     encode = gst_elementfactory_make("mpeg2enc","encode");
     g_return_if_fail(encode != NULL);
     fd = open(outfile,O_CREAT|O_RDWR|O_TRUNC);
@@ -92,6 +93,7 @@ void new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline) {
     // set up pad connections
     gst_element_add_ghost_pad(GST_ELEMENT(video_thread),
                               gst_element_get_pad(smooth,"sink"));
+                              //gst_element_get_pad(encode,"sink"));
 
     // construct queue and connect everything in the main pipeline
     video_queue = gst_elementfactory_make("queue","video_queue");

diff --git a/test/mp2tomp1.c b/test/mp2tomp1.c
index 7d7b3ef..7ee5249 100644 (file)
--- a/test/mp2tomp1.c
+++ b/test/mp2tomp1.c
@@ -6,6 +6,7 @@
 #include <gst/gst.h>
 
 int fd;
+int audiofd;
 char *outfile;
 
 void eof(GstSrc *src) {
@@ -17,15 +18,15 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
   GstElement *parse_audio, *parse_video, *decode, *decode_video, *play, *encode;
   GstElement *audio_queue, *video_queue;
   GstElement *audio_thread, *video_thread;
-  GstElement *videoscale;
-  GstElement *fdsink;
-  GstElementFactory *fdsinkfactory;
+  GstElement *videoscale, *audio_encode;
+  GstElement *fdsink, *audiofdsink;
+  GstElementFactory *fdsinkfactory, *audiofdsinkfactory;
 
   g_print("***** a new pad %s was created\n", gst_pad_get_name(pad));
 
   // connect to audio pad
-  if (0) {
-  //if (strncmp(gst_pad_get_name(pad), "private_stream_1.0", 18) == 0) {
+  //if (0) {
+  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
@@ -33,15 +34,23 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
     g_return_if_fail(parse_audio != NULL);
     decode = gst_elementfactory_make("ac3dec","decode_audio");
     g_return_if_fail(decode != NULL);
-    play = gst_elementfactory_make("audiosink","play_audio");
-    g_return_if_fail(play != NULL);
+    audio_encode = gst_elementfactory_make("pipefilter","audio_encode");
+    g_return_if_fail(audio_encode != NULL);
 
     // create the thread and pack stuff into it
     audio_thread = gst_thread_new("audio_thread");
     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(play));
+    gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(audio_encode));
+
+    audiofd = open("/opt2/audio.mp3",O_CREAT|O_RDWR|O_TRUNC);
+    audiofdsinkfactory = gst_elementfactory_find("fdsink");
+    g_return_if_fail(audiofdsinkfactory != NULL);
+    audiofdsink = gst_elementfactory_create(audiofdsinkfactory,"fdsink");
+    g_return_if_fail(audiofdsink != NULL);
+    gtk_object_set(GTK_OBJECT(audiofdsink),"fd",audiofd,NULL);
+    gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(audiofdsink));
 
     // set up pad connections
     gst_element_add_ghost_pad(GST_ELEMENT(audio_thread),
@@ -49,7 +58,9 @@ void mp2tomp1(GstElement *parser,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(play,"sink"));
+                    gst_element_get_pad(audio_encode,"sink"));
+    gst_pad_connect(gst_element_get_pad(audio_encode,"src"),
+                    gst_element_get_pad(audiofdsink,"sink"));
 
     // construct queue and connect everything in the main pipelie
     audio_queue = gst_elementfactory_make("queue","audio_queue");
@@ -124,6 +135,7 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
     g_return_if_fail(decode_video != NULL);
     videoscale = gst_elementfactory_make("videoscale","videoscale");
     g_return_if_fail(videoscale != NULL);
+    gtk_object_set(GTK_OBJECT(videoscale),"width",352, "height", 224,NULL);
     encode = gst_elementfactory_make("mpeg2enc","encode");
     //encode = gst_elementfactory_make("mpeg1encoder","encode");
     g_return_if_fail(encode != NULL);