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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.
This commit is contained in:
Wim Taymans
parent
49899cf0d8
commit
cbb196fd31
8 changed files with 34 additions and 11 deletions
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@ -361,6 +361,7 @@ plugins/visualization/synaesthesia/Makefile
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plugins/visualization/smoothwave/Makefile
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plugins/videosink/Makefile
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plugins/videoscale/Makefile
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plugins/audioscale/Makefile
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plugins/dvdsrc/Makefile
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plugins/vcdsrc/Makefile
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plugins/cobin/Makefile
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@ -21,6 +21,7 @@ libgstelements_la_SOURCES = \
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gstaudiosink.c \
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gstaudiosrc.c \
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gstfdsink.c \
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gstpipefilter.c \
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gstqueue.c \
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gstsinesrc.c
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@ -35,6 +36,7 @@ noinst_HEADERS = \
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gstaudiosink.h \
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gstaudiosrc.h \
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gstfdsink.h \
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gstpipefilter.h \
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gstqueue.h \
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gstsinesrc.h
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@ -32,6 +32,7 @@
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#if HAVE_LIBGHTTP
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#include <gsthttpsrc.h>
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#endif /* HAVE_LIBGHTTP */
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#include <gstpipefilter.h>
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#include <gstqueue.h>
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#include <gstsinesrc.h>
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@ -55,6 +56,7 @@ struct _elements_entry _elements[] = {
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#if HAVE_LIBGHTTP
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{ "httpsrc", gst_httpsrc_get_type, &gst_httpsrc_details },
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#endif /* HAVE_LIBGHTTP */
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{ "pipefilter", gst_pipefilter_get_type, &gst_pipefilter_details },
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{ "queue", gst_queue_get_type, &gst_queue_details },
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{ "sinesrc", gst_sinesrc_get_type, &gst_sinesrc_details },
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{ NULL, 0 },
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@ -21,6 +21,7 @@ libgstelements_la_SOURCES = \
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gstaudiosink.c \
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gstaudiosrc.c \
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gstfdsink.c \
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gstpipefilter.c \
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gstqueue.c \
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gstsinesrc.c
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@ -35,6 +36,7 @@ noinst_HEADERS = \
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gstaudiosink.h \
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gstaudiosrc.h \
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gstfdsink.h \
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gstpipefilter.h \
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gstqueue.h \
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gstsinesrc.h
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@ -32,6 +32,7 @@
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#if HAVE_LIBGHTTP
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#include <gsthttpsrc.h>
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#endif /* HAVE_LIBGHTTP */
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#include <gstpipefilter.h>
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#include <gstqueue.h>
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#include <gstsinesrc.h>
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@ -55,6 +56,7 @@ struct _elements_entry _elements[] = {
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#if HAVE_LIBGHTTP
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{ "httpsrc", gst_httpsrc_get_type, &gst_httpsrc_details },
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#endif /* HAVE_LIBGHTTP */
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{ "pipefilter", gst_pipefilter_get_type, &gst_pipefilter_details },
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{ "queue", gst_queue_get_type, &gst_queue_details },
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{ "sinesrc", gst_sinesrc_get_type, &gst_sinesrc_details },
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{ NULL, 0 },
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@ -1,7 +1,7 @@
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#noinst_PROGRAMS = basic m types a r plugin w s args mpg123 mcut push qtest
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noinst_PROGRAMS = qtest spectrum record wave mp3 teardown buffer mp3parse \
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mpeg2parse mp1parse mp3play ac3parse ac3play dvdcat fake cobin videotest \
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aviparse vidcapture avi2mpg mp2tomp1 mp1tomp1
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aviparse vidcapture avi2mpg mp2tomp1 mp1tomp1 pipetest
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SUBDIRS = xml cothreads bindings
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@ -67,7 +67,8 @@ void new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline) {
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smooth = gst_elementfactory_make("smooth","smooth");
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//smooth = gst_elementfactory_make("median","median");
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g_return_if_fail(smooth != NULL);
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//gtk_object_set(GTK_OBJECT(smooth),"filtersize",9,NULL);
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gtk_object_set(GTK_OBJECT(smooth),"filtersize",2,NULL);
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gtk_object_set(GTK_OBJECT(smooth),"tolerance",4,NULL);
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encode = gst_elementfactory_make("mpeg2enc","encode");
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g_return_if_fail(encode != NULL);
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fd = open(outfile,O_CREAT|O_RDWR|O_TRUNC);
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@ -92,6 +93,7 @@ void new_pad_created(GstElement *parse,GstPad *pad,GstElement *pipeline) {
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// set up pad connections
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gst_element_add_ghost_pad(GST_ELEMENT(video_thread),
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gst_element_get_pad(smooth,"sink"));
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//gst_element_get_pad(encode,"sink"));
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// construct queue and connect everything in the main pipeline
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video_queue = gst_elementfactory_make("queue","video_queue");
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@ -6,6 +6,7 @@
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#include <gst/gst.h>
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int fd;
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int audiofd;
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char *outfile;
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void eof(GstSrc *src) {
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@ -17,15 +18,15 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
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GstElement *parse_audio, *parse_video, *decode, *decode_video, *play, *encode;
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GstElement *audio_queue, *video_queue;
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GstElement *audio_thread, *video_thread;
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GstElement *videoscale;
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GstElement *fdsink;
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GstElementFactory *fdsinkfactory;
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GstElement *videoscale, *audio_encode;
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GstElement *fdsink, *audiofdsink;
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GstElementFactory *fdsinkfactory, *audiofdsinkfactory;
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g_print("***** a new pad %s was created\n", gst_pad_get_name(pad));
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// connect to audio pad
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if (0) {
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//if (strncmp(gst_pad_get_name(pad), "private_stream_1.0", 18) == 0) {
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//if (0) {
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if (strncmp(gst_pad_get_name(pad), "private_stream_1.0", 18) == 0) {
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gst_plugin_load("ac3parse");
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gst_plugin_load("ac3dec");
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// construct internal pipeline elements
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@ -33,15 +34,23 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
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g_return_if_fail(parse_audio != NULL);
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decode = gst_elementfactory_make("ac3dec","decode_audio");
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g_return_if_fail(decode != NULL);
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play = gst_elementfactory_make("audiosink","play_audio");
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g_return_if_fail(play != NULL);
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audio_encode = gst_elementfactory_make("pipefilter","audio_encode");
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g_return_if_fail(audio_encode != NULL);
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// create the thread and pack stuff into it
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audio_thread = gst_thread_new("audio_thread");
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g_return_if_fail(audio_thread != NULL);
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gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(parse_audio));
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gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(decode));
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gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(play));
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gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(audio_encode));
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audiofd = open("/opt2/audio.mp3",O_CREAT|O_RDWR|O_TRUNC);
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audiofdsinkfactory = gst_elementfactory_find("fdsink");
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g_return_if_fail(audiofdsinkfactory != NULL);
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audiofdsink = gst_elementfactory_create(audiofdsinkfactory,"fdsink");
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g_return_if_fail(audiofdsink != NULL);
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gtk_object_set(GTK_OBJECT(audiofdsink),"fd",audiofd,NULL);
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gst_bin_add(GST_BIN(audio_thread),GST_ELEMENT(audiofdsink));
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// set up pad connections
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gst_element_add_ghost_pad(GST_ELEMENT(audio_thread),
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@ -49,7 +58,9 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
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gst_pad_connect(gst_element_get_pad(parse_audio,"src"),
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gst_element_get_pad(decode,"sink"));
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gst_pad_connect(gst_element_get_pad(decode,"src"),
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gst_element_get_pad(play,"sink"));
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gst_element_get_pad(audio_encode,"sink"));
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gst_pad_connect(gst_element_get_pad(audio_encode,"src"),
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gst_element_get_pad(audiofdsink,"sink"));
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// construct queue and connect everything in the main pipelie
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audio_queue = gst_elementfactory_make("queue","audio_queue");
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@ -124,6 +135,7 @@ void mp2tomp1(GstElement *parser,GstPad *pad, GstElement *pipeline) {
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g_return_if_fail(decode_video != NULL);
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videoscale = gst_elementfactory_make("videoscale","videoscale");
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g_return_if_fail(videoscale != NULL);
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gtk_object_set(GTK_OBJECT(videoscale),"width",352, "height", 224,NULL);
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encode = gst_elementfactory_make("mpeg2enc","encode");
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//encode = gst_elementfactory_make("mpeg1encoder","encode");
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g_return_if_fail(encode != NULL);
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