Gstreamer教程 --動態pipeline
阿新 • • 發佈：2018-11-07
demuxer在沒有看到容器檔案之前無法確定需要做的工作，不能生成對應的內容。也就是說，demuxer開始時是沒有source pad給其他element連線用的。
解決方法是隻管建立pipeline，讓source和demuxer連線起來，然後開始執行。當demuxer接收到資料之後它就有了足夠的資訊生成source pad。這時我們就可以繼續把其他部分和demuxer新生成的pad連線起來，生成一個完整的pipeline。
g_signal_connect()
#define
Connects a GCallback functionto a signal for a particular object.
The handlerwill be called before the default handler of the signal.
instance : |
the instance to connect to. |
detailed_signal : |
a string of the form "signal-name::detail". |
c_handler : |
the GCallback to connect. |
data : |
data to pass to c_handler calls. |
Returns : |
the handler id |
只能相鄰狀態改變，
絕大多數應用都是在PLAYING狀態開始播放，然後跳轉到PAUSE狀態來提供暫停功能，最後在退出時退到NULL狀態。
來自 <http://blog.csdn.net/sakulafly/article/details/20936067>
#include <gst/gst.h>
/* Structure to contain all our information, so we can passit to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *sink;
} CustomData;
/* Handler for the pad-added signal */
staticvoid 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;
/* InitializeGStreamer */
gst_init (&argc,&argv);
/* Create the elements */
data.source = gst_element_factory_make ("uridecodebin","source");
data.convert = gst_element_factory_make ("audioconvert","convert");
data.sink = gst_element_factory_make ("autoaudiosink","sink");
/*
uridecodebin自己會在內部初始化必要的element，然後把一個URI變成一個原始音視訊流輸出，它差不多做了playbin2的一半工作。因為它自己帶著demuxer，所以它的source pad沒有初始化，我們等會會用到。
audioconvert在不同的音訊格式轉換時很有用。這裡用這個element是為了確保應用的平臺無關性。
autoaudiosink,這個element的輸出就是直接送往音效卡的音訊流。
*/
/* Create the emptypipeline */
data.pipeline =gst_pipeline_new ("test-pipeline");
if (!data.pipeline ||!data.source || !data.convert || !data.sink) {
g_printerr ("Notall elements could be created.\n");
return -1;
}
/* Build thepipeline. Note that we are NOT linking the source at this
* point. We will doit later. */
gst_bin_add_many(GST_BIN (data.pipeline), data.source, data.convert , data.sink, NULL);
/* 連線convert，sink，由於目前沒有source pad，之後再連線 */
if (!gst_element_link (data.convert, data.sink)) {
g_printerr("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Set the URI toplay */
g_object_set(data.source, "uri", "http://docs.gstreamer.com/media/sintel_trailer-480p.webm",NULL);
/* Connect to the pad-added signal */
/*source element最後獲得足夠的資料時，它就會自動生成source pad，並且觸發“pad-added”訊號*/
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)) {
caseGST_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;
caseGST_MESSAGE_EOS:
g_print ("End-Of-Streamreached.\n");
terminate =TRUE;
break;
case GST_MESSAGE_STATE_CHANGED:
/* We are onlyinterested in state-changed messages from the pipeline */
if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
GstStateold_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 shouldnot 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) {
/*獲取convert的sinkpad */
GstPad *sink_pad =gst_element_get_static_pad (data->convert, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps =NULL;
GstStructure*new_pad_struct = NULL;
const gchar*new_pad_type = NULL;
g_print ("Receivednew pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME(src));
//uridecodebin會自動建立許多的pad，對於每一個pad，這個回撥函式都會被呼叫。
/* If our converter isalready linked, we have nothing to do here */
if (gst_pad_is_linked(sink_pad)) {
g_print (" We are already linked. Ignoring.\n");
goto exit;
}
//gst_pad_get_caps()方法會獲得pad的capability（也就是pad支援的資料型別），是被封裝起來的GstCaps結構。一個pad可以有多個capability，GstCaps可以包含多個GstStructure，每個都描述了一個不同的capability。
/* Check the newpad'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);
/*判斷是否以audio/x-raw開頭，是否為音訊*/
if (!g_str_has_prefix(new_pad_type, "audio/x-raw")) {
g_print (" It has type '%s' which is not raw audio.Ignoring.\n", new_pad_type);
goto exit;
}
/* Attempt thelink src新產生的pad和convert的sink pad*/
ret = gst_pad_link(new_pad, sink_pad);
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 newpad's caps, if we got them */
if (new_pad_caps !=NULL)
gst_caps_unref(new_pad_caps);
/* Unreference the sinkpad */
gst_object_unref(sink_pad);
}