我想在Windows 7中使用GStreamer播放.AVI文件。按照此link中的说明安装了GStreamer SDK。然后创建了一个GStreamer SDK项目,并将以下代码添加到C文件中,如此link给出了建议的更正。项目属性 - >配置属性 - >调试 - >工作目录更改为“$(GSTREAMER_SDK_ROOT_X86)\ bin”,并按照上面给出的安装链接中的建议将其添加到Windows PATH变量中。运行代码时,它只是退出而不播放视频,输出的最后几行如下所示。请注意,我在64位Windows 7上安装了32位Gstreamer SDK。
代码:
#include<stdio.h>
#include<gst/gst.h>
#include<glib.h>
//Function to process message on bus of pipeline
gboolean process_message(GstBus *bus, GstMessage *msg,gpointer data);
//Function to add pad dynamically for ogg demux
void dynamic_addpad(GstElement *element, GstPad *pad, gpointer data);
void dynamic_decodepad (GstElement* object, GstPad* arg0, gboolean arg1,gpointer user_data);
GstElement *source, *demuxer, *audio_decoder, *video_decoder, *audio_convertor,*video_convertor, *audio_sink,*video_sink,*audioqueue,*videoqueue;//*audio_demuxer, *video_demuxer,
int main(int argc,char* argv[]){
GstPipeline *pipeline;
GstBin *Bin;
GstBus *bus;
GMainLoop *Mainloop;
gst_init (&argc,&argv);
Mainloop = g_main_loop_new(NULL,FALSE);//NULL to use the current context and False to tell its not in running state
GstElement *pipeline = gst_pipeline_new("PIPELINE");
Bin = GST_BIN(pipeline);
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
source = gst_element_factory_make("filesrc","file-source");
g_object_set(G_OBJECT(source), "location", "file:///C:/Video.avi", NULL);
demuxer = gst_element_factory_make("avidemux","avi-demuxer");
audioqueue = gst_element_factory_make("queue","Queue for audio");
videoqueue = gst_element_factory_make("queue","Queue for video");
audio_decoder = gst_element_factory_make("decodebin","a_decodebin");
video_decoder = gst_element_factory_make("decodebin","decoderbin");//"Vorbis audio decoder","vorbis");
audio_convertor = gst_element_factory_make("audioconvert","audio convertor");//"Audio converter","audioconvert");
video_convertor = gst_element_factory_make("videoscale","video convertor");//"Audio converter","audioconvert");
audio_sink = gst_element_factory_make("autoaudiosink","Auto audio sink");
video_sink = gst_element_factory_make("xvimagesink","XV video sink ");
if(!source || !demuxer || !audioqueue || !videoqueue || !video_decoder ||!audio_convertor || ! video_convertor || !audio_sink || !video_sink ){
g_print("Could not not create element\n");
return 0;
}
gst_bin_add(Bin,source);
gst_bin_add_many(
Bin,
demuxer,
audioqueue,videoqueue,
audio_decoder,audio_convertor,
video_decoder,video_convertor,
audio_sink,video_sink,
NULL);
gst_element_link(source,demuxer);
gst_element_link_many(audioqueue,audio_decoder,NULL);
gst_element_link_many(audio_convertor,audio_sink,NULL);
gst_element_link_many(videoqueue,video_decoder,NULL);
gst_element_link_many(video_convertor,video_sink,NULL);
g_signal_connect(demuxer,"pad-added",G_CALLBACK(dynamic_addpad),NULL);//demuxer and decoder are passed as instance and data as pads of both the elements are linked in dynamic_addpad
g_signal_connect(audio_decoder,"new-decoded-pad",G_CALLBACK(dynamic_decodepad),NULL);
g_signal_connect(video_decoder,"new-decoded-pad",G_CALLBACK(dynamic_decodepad),NULL);//demuxer and decoder are passed as instance and data as pads of both the elements are linked in dynamic_addpad
gst_bus_add_watch(bus,process_message,Mainloop); //Mainloop is passed as user data as in the process_message actions are taken on the loop
g_object_unref(bus);
g_print("In playing state\n");
gst_element_set_state(pipeline, GST_STATE_PLAYING);//Pipeline is also a bin and bin is also an element at abstract level and hence gst_element_set_state call is used to set state of pipeline.
g_main_loop_run(Mainloop);
g_print("In playing state2\n");
gst_element_set_state(pipeline, GST_STATE_NULL);
g_object_unref(G_OBJECT(pipeline));
}
//Function to process message on bus of pipeline
gboolean process_message(GstBus *bus, GstMessage *msg,gpointer data){
GError *error;
gchar *debug;
GMainLoop *loop = (GMainLoop *)data;
g_print(" In process message msg->type : %d\n",GST_MESSAGE_TYPE(msg));
switch(GST_MESSAGE_TYPE(msg)){
case GST_MESSAGE_UNKNOWN :
g_print("GST_MESSAGE_UNKNOWN \n");
break;
case GST_MESSAGE_EOS :
g_print("GST_MESSAGE_EOS \n");
g_main_loop_quit(loop);
break;
case GST_MESSAGE_ERROR :
g_print("GST_MESSAGE_ERROR \n");
gst_message_parse_error (msg, &error, &debug);
g_free(debug);
//if(!error)
{
g_print("GST_MESSAGE_ERROR message : %s \n",error->message);
}
g_main_loop_quit(loop);
break;
case GST_MESSAGE_WARNING :
g_print("GST_MESSAGE_WARNING \n");
break;
case GST_MESSAGE_INFO :
g_print("GST_MESSAGE_INFO \n");
break;
case GST_MESSAGE_TAG :
g_print("GST_MESSAGE_TAG \n");
break;
case GST_MESSAGE_BUFFERING:
g_print("GST_MESSAGE_BUFFERING \n");
break;
case GST_MESSAGE_STATE_CHANGED:
g_print("GST_MESSAGE_STATE_CHANGED \n");
break;
default :
g_print("default \n");
break;
}
return TRUE; //returns true always as it has to be always registered returning false will deregister the function
}
void dynamic_addpad(GstElement *element, GstPad *pad, gpointer data) {
char* pad_name = gst_pad_get_name(pad);
g_print(" In dynamic ADDING PAD %s\n", pad_name);
if (g_str_has_prefix(pad_name,"audio")) {
GstPad *audiodemuxsink = gst_element_get_static_pad(audioqueue,"sink");
gst_pad_link(pad,audiodemuxsink );
}
else if (g_str_has_prefix(pad_name,"video")) {
GstPad *videodemuxsink = gst_element_get_static_pad(videoqueue,"sink");
gst_pad_link(pad,videodemuxsink );
}
g_free (pad_name);
}
void dynamic_decodepad (GstElement* object, GstPad* pad, gboolean arg1,gpointer user_data) {
GstPad* videoconvertsink = gst_element_get_static_pad(video_convertor,"sink");
if (gst_pad_can_link(pad,videoconvertsink)) {
gst_pad_link(pad,videoconvertsink);
}
GstPad* audioconvertsink = gst_element_get_static_pad(audio_convertor,"sink");
if (gst_pad_can_link(pad,audioconvertsink)) {
gst_pad_link(pad,audioconvertsink);
}
}
输出:
The thread 'Win32 Thread' (0x19c4) has exited with code 0 (0x0).
The thread 'Win32 Thread' (0x2370) has exited with code 0 (0x0).
The thread 'Win32 Thread' (0x2040) has exited with code 0 (0x0).
The program '[5368] GstProject2.exe: Native' has exited with code 0 (0x0).
答案 0 :(得分:1)
最后,我能够使用以下代码播放AVI文件,该代码基于Gstreamer SDK网站中的this示例。
在命令提示符下: 选项1:
gst-launch-0.10 filesrc location=C:\\Video.avi ! decodebin2 name=dec ! queue ! ffmpegcolorspace ! autovideosink dec. ! queue ! audioconvert ! audioresample ! autoaudiosink
选项2:
gst-launch-0.10 filesrc location=C:\\Video.avi ! decodebin2 name=dec ! ffmpegcolorspace ! autovideosink dec. ! audioconvert ! audioresample ! autoaudiosink
选项3:
gst-launch-0.10 uridecodebin uri=file:///C:/Video.avi name=dec ! ffmpegcolorspace ! autovideosink dec. ! audioconvert ! autoaudiosink
在Visual Studio中:
#include <gst/gst.h>
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *audio_sink;
GstElement *colorspace;
GstElement *video_sink;
} CustomData;
/* Handler for the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.source = gst_element_factory_make ("uridecodebin", "source");
data.convert = gst_element_factory_make ("audioconvert", "convert");
data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
data.colorspace = gst_element_factory_make ("ffmpegcolorspace", "colorspace");
data.video_sink = gst_element_factory_make ("autovideosink", "video_sink");
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.convert || !data.audio_sink || !data.colorspace || !data.video_sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Build the pipeline. Note that we are NOT linking the source at this
* point. We will do it later. */
gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert , data.audio_sink, data.colorspace, data.video_sink, NULL);
if (!(gst_element_link (data.convert, data.audio_sink) && gst_element_link (data.colorspace, data.video_sink))) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Set the URI to play */
g_object_set (data.source, "uri", "file:///C:/Video.avi", NULL);
/* Connect to the pad-added signal */
g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Listen to the bus */
bus = gst_element_get_bus (data.pipeline);
do {
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
g_clear_error (&err);
g_free (debug_info);
terminate = TRUE;
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
terminate = TRUE;
break;
case GST_MESSAGE_STATE_CHANGED:
/* We are only interested in state-changed messages from the pipeline */
if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
GstState old_state, new_state, pending_state;
gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state);
g_print ("Pipeline state changed from %s to %s:\n",
gst_element_state_get_name (old_state), gst_element_state_get_name (new_state));
}
break;
default:
/* We should not reach here */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
} while (!terminate);
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
/* This function will be called by the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
GstPad *sink_pad_audio = gst_element_get_static_pad (data->convert, "sink");
GstPad *sink_pad_video = gst_element_get_static_pad (data->colorspace, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
///* If our converter is already linked, we have nothing to do here */
//if (gst_pad_is_linked (sink_pad)) {
// g_print (" We are already linked. Ignoring.\n");
// goto exit;
//}
/* Check the new pad's type */
new_pad_caps = gst_pad_get_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) {
g_print (" It has type '%s' which is raw video. Connecting.\n", new_pad_type);
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad_video);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
goto exit;
}
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad_audio);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad_audio);
gst_object_unref (sink_pad_video);
}