c++ gstreamer使用2
1，播放教程playbin
#include <gst/gst.h> #include <stdio.h> /* Structure to contain all our information, so we can pass it around */ typedef struct _CustomData { GstElement *playbin; /* Our one and only element */ gint n_video; /* Number of embedded video streams */ gint n_audio; /* Number of embedded audio streams */ gint n_text; /* Number of embedded subtitle streams */ gint current_video; /* Currently playing video stream */ gint current_audio; /* Currently playing audio stream */ gint current_text; /* Currently playing subtitle stream */ GMainLoop *main_loop; /* GLib's Main Loop */ } CustomData; /* playbin flags */ typedef enum { GST_PLAY_FLAG_VIDEO = (1 << 0), /* We want video output */ GST_PLAY_FLAG_AUDIO = (1 << 1), /* We want audio output */ GST_PLAY_FLAG_TEXT = (1 << 2) /* We want subtitle output */ } GstPlayFlags; /* Forward definition for the message and keyboard processing functions */ static gboolean handle_message(GstBus *bus, GstMessage *msg, CustomData *data); static gboolean handle_keyboard(GIOChannel *source, GIOCondition cond, CustomData *data); int main(int argc, char *argv[]) { CustomData data; GstBus *bus; GstStateChangeReturn ret; gint flags; GIOChannel *io_stdin; /* Initialize GStreamer */ gst_init(&argc, &argv); /* Create the elements */ data.playbin = gst_element_factory_make("playbin", "playbin"); if (!data.playbin) { g_printerr("Not all elements could be created.\n"); return -1; } /* Set the URI to play */ g_object_set(data.playbin, "uri", "file:///D:/gstreamer/1.mp4", NULL); //rtsp://xxx:xxx@xxx/h264/ch1/main/av_stream /* Set flags to show Audio and Video but ignore Subtitles */ g_object_get(data.playbin, "flags", &flags, NULL); flags |= GST_PLAY_FLAG_VIDEO | GST_PLAY_FLAG_AUDIO; flags &= ~GST_PLAY_FLAG_TEXT; g_object_set(data.playbin, "flags", flags, NULL); /* Set connection speed. This will affect some internal decisions of playbin */ g_object_set(data.playbin, "connection-speed", 56, NULL); /* Add a bus watch, so we get notified when a message arrives */ bus = gst_element_get_bus(data.playbin); gst_bus_add_watch(bus, (GstBusFunc)handle_message, &data); /* Add a keyboard watch so we get notified of keystrokes */ #ifdef G_OS_WIN32 io_stdin = g_io_channel_win32_new_fd(_fileno(stdin)); #else io_stdin = g_io_channel_unix_new(fileno(stdin)); #endif g_io_add_watch(io_stdin, G_IO_IN, (GIOFunc)handle_keyboard, &data); /* Start playing */ ret = gst_element_set_state(data.playbin, 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.playbin); return -1; } /* Create a GLib Main Loop and set it to run */ data.main_loop = g_main_loop_new(NULL, FALSE); g_main_loop_run(data.main_loop); /* Free resources */ g_main_loop_unref(data.main_loop); g_io_channel_unref(io_stdin); gst_object_unref(bus); gst_element_set_state(data.playbin, GST_STATE_NULL); gst_object_unref(data.playbin); return 0; } /* Extract some metadata from the streams and print it on the screen */ static void analyze_streams(CustomData *data) { gint i; GstTagList *tags; gchar *str; guint rate; /* Read some properties */ g_object_get(data->playbin, "n-video", &data->n_video, NULL); g_object_get(data->playbin, "n-audio", &data->n_audio, NULL); g_object_get(data->playbin, "n-text", &data->n_text, NULL); g_print("%d video stream(s), %d audio stream(s), %d text stream(s)\n", data->n_video, data->n_audio, data->n_text); g_print("\n"); for (i = 0; i < data->n_video; i++) { tags = NULL; /* Retrieve the stream's video tags */ g_signal_emit_by_name(data->playbin, "get-video-tags", i, &tags); if (tags) { g_print("video stream %d:\n", i); gst_tag_list_get_string(tags, GST_TAG_VIDEO_CODEC, &str); g_print(" codec: %s\n", str ? str : "unknown"); g_free(str); gst_tag_list_free(tags); } } g_print("\n"); for (i = 0; i < data->n_audio; i++) { tags = NULL; /* Retrieve the stream's audio tags */ g_signal_emit_by_name(data->playbin, "get-audio-tags", i, &tags); if (tags) { g_print("audio stream %d:\n", i); if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &str)) { g_print(" codec: %s\n", str); g_free(str); } if (gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &str)) { g_print(" language: %s\n", str); g_free(str); } if (gst_tag_list_get_uint(tags, GST_TAG_BITRATE, &rate)) { g_print(" bitrate: %d\n", rate); } gst_tag_list_free(tags); } } g_print("\n"); for (i = 0; i < data->n_text; i++) { tags = NULL; /* Retrieve the stream's subtitle tags */ g_signal_emit_by_name(data->playbin, "get-text-tags", i, &tags); if (tags) { g_print("subtitle stream %d:\n", i); if (gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &str)) { g_print(" language: %s\n", str); g_free(str); } gst_tag_list_free(tags); } } g_object_get(data->playbin, "current-video", &data->current_video, NULL); g_object_get(data->playbin, "current-audio", &data->current_audio, NULL); g_object_get(data->playbin, "current-text", &data->current_text, NULL); g_print("\n"); g_print("Currently playing video stream %d, audio stream %d and text stream %d\n", data->current_video, data->current_audio, data->current_text); g_print("Type any number and hit ENTER to select a different audio stream\n"); } /* Process messages from GStreamer */ static gboolean handle_message(GstBus *bus, GstMessage *msg, CustomData *data) { 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); g_main_loop_quit(data->main_loop); break; case GST_MESSAGE_EOS: g_print("End-Of-Stream reached.\n"); g_main_loop_quit(data->main_loop); break; case GST_MESSAGE_STATE_CHANGED: { GstState old_state, new_state, pending_state; gst_message_parse_state_changed(msg, &old_state, &new_state, &pending_state); if (GST_MESSAGE_SRC(msg) == GST_OBJECT(data->playbin)) { if (new_state == GST_STATE_PLAYING) { /* Once we are in the playing state, analyze the streams */ analyze_streams(data); } } } break; } /* We want to keep receiving messages */ return TRUE; } /* Process keyboard input */ static gboolean handle_keyboard(GIOChannel *source, GIOCondition cond, CustomData *data) { gchar *str = NULL; if (g_io_channel_read_line(source, &str, NULL, NULL, NULL) == G_IO_STATUS_NORMAL) { int index = g_ascii_strtoull(str, NULL, 0); if (index < 0 || index >= data->n_audio) { g_printerr("Index out of bounds\n"); } else { /* If the input was a valid audio stream index, set the current audio stream */ g_print("Setting current audio stream to %d\n", index); g_object_set(data->playbin, "current-audio", index, NULL); } } g_free(str); return TRUE; }
此代码应该是和命令行里面的playbin一样的，啥都不需要你做，就能播放，但是这同样代表着什么你都无法优化，直接一个playbin管道就结束了。实测rtsp延时挺严重的。
2，自定义衬垫链接：
#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 *decode; GstElement *convert; GstElement *sink; } CustomData; //先建立一个结构，里面放了一个pipeline指针和四个元件指针 /* Handler for the pad-added signal */ static void pad_added_handler(GstElement *src, GstPad *pad, CustomData *data); static void pad_added_handler2(GstElement *src, GstPad *pad, CustomData *data); //声明一个两个回调函数，一个负责链接source和decode，另一个负责链接decode和convert int main(int argc, char *argv[]) { CustomData data; GstBus *bus; GstMessage *msg; GstStateChangeReturn ret; gboolean terminate = FALSE; /* Initialize GStreamer */ gst_init(&argc, &argv); //同样需要先初始化 /* Create the elements */ data.source = gst_element_factory_make("rtspsrc", "source"); data.decode = gst_element_factory_make("decodebin", "decode"); data.convert = gst_element_factory_make("videoconvert", "convert"); data.sink = gst_element_factory_make("autovideosink", "sink"); /* Create the empty pipeline */ data.pipeline = gst_pipeline_new("test-pipeline"); //先把data里的信息创建出来，创建了一个pipeline和四个元件 if (!data.pipeline || !data.source || !data.decode || !data.convert || !data.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.decode, data.convert, data.sink, NULL); //把元件都添加到管道里 if (!gst_element_link_many( data.convert, data.sink, NULL)) { //把元件都链接起来，为什么不连接source和decode？因为这俩需要回调函数进行特殊的链接，一般的链接是要报错的 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, "location", "rtsp://admin:abc12345@192.168.3.198/h264/ch1/main/av_stream", NULL); //大约是将source元件的数据源给怼进去 /* Connect to the pad-added signal */ g_signal_connect(data.source, "pad-added", G_CALLBACK(pad_added_handler), &data); g_signal_connect(data.decode, "pad-added", G_CALLBACK(pad_added_handler2), &data); //给source和decode添加衬垫，衬垫关联的是回调函数，我理解的回调函数的参数:源元件（给谁添加衬垫，就是谁），新添加的衬垫，用来传递数据的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 */ //获取一个总线，总线可以监视pipeline的运行状态，是否播放完毕等，然后进行相应的处理。 bus = gst_element_get_bus(data.pipeline); do { msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ANY); //等待执行结束并且返回 //顺带说一句，以前的老语法是GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS这样的，所以下文中的case用的是这几个错误信息，但是现在这个语法不被支持了。嗯嗯，所以改用GST_MESSAGE_ANY /* Parse message */ if (msg != NULL) { GError *err; gchar *debug_info; g_print("error msg:%d\n", GST_MESSAGE_TYPE(msg)); 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); //只要不中止，就一直监视执行结束的状态 /* 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 = gst_element_get_static_pad(data->decode, "sink"); //pipeline的链接顺序是：source-decode-convert-sink，我们为source添加了回调，然后此处在回调内部获取了convert的对应的衬垫 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; } //此处应该是检查新为source添加的衬垫是不是已经链接到了convert衬垫 /* Check the new pad's type */ new_pad_caps = gst_pad_get_current_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, "application/x-rtp")) { g_print("It has type '%s' which is not raw rtsp. Ignoring.\n", new_pad_type); goto exit; } //检查这个衬垫当前输出的数据类型，经过一番解析，如果发现里面没有"audio/x-raw"，那说明这不是解码音频的 /* Attempt the link */ 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); } //如果两个衬垫没链接，那就人为地链接起来 exit: //这个语法就厉害了，首先定义了一个exit标号，如果前文中goto exit;那转到的就将会是此处 /* 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); } static void pad_added_handler2(GstElement *src, GstPad *new_pad, CustomData *data) { GstPad *sink_pad = gst_element_get_static_pad(data->convert, "sink"); //pipeline的链接顺序是：source-decode-convert-sink，我们为source添加了回调，然后此处在回调内部获取了convert的对应的衬垫 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; } //此处应该是检查新为source添加的衬垫是不是已经链接到了convert衬垫 /* Check the new pad's type */ new_pad_caps = gst_pad_get_current_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, "video/x-raw")) { g_print("It has type '%s' which is not raw rtsp. Ignoring.\n", new_pad_type); goto exit; } //检查这个衬垫当前输出的数据类型，经过一番解析，如果发现里面没有"audio/x-raw"，那说明这不是解码音频的 /* Attempt the link */ 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("Link222 succeeded (type '%s').\n", new_pad_type); } //如果两个衬垫没链接，那就人为地链接起来 exit: //这个语法就厉害了，首先定义了一个exit标号，如果前文中goto exit;那转到的就将会是此处 /* 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); }
3，从rtsp解码视频，转码为jpg并且写出到本地文件，注意，文件会变得很大
#include <gst/gst.h> #include <iostream> using namespace std; /* Structure to contain all our information, so we can pass it to callbacks */ typedef struct _CustomData { GstElement *pipeline; GstElement *source; GstElement *decode; GstElement *convert; GstElement *sink; } CustomData; //先建立一个结构，里面放了一个pipeline指针和四个元件指针 /* Handler for the pad-added signal */ static void pad_added_handler(GstElement *src, GstPad *pad, CustomData *data); static void pad_added_handler2(GstElement *src, GstPad *pad, CustomData *data); static void daqing_function(GstElement* object, GstBuffer* arg0, GstPad* arg1, gpointer user_data); //声明一个两个回调函数，一个负责链接source和decode，另一个负责链接decode和convert int main(int argc, char *argv[]) { CustomData data; GstBus *bus; GstMessage *msg; GstStateChangeReturn ret; gboolean terminate = FALSE; /* Initialize GStreamer */ gst_init(&argc, &argv); //同样需要先初始化 /* Create the elements */ data.source = gst_element_factory_make("rtspsrc", "source"); data.decode = gst_element_factory_make("decodebin", "decode"); data.convert = gst_element_factory_make("jpegenc", "convert");//jpegenc avenc_bmp data.sink = gst_element_factory_make("filesink", "sink"); /* Create the empty pipeline */ data.pipeline = gst_pipeline_new("test-pipeline"); //先把data里的信息创建出来，创建了一个pipeline和四个元件 if (!data.pipeline || !data.source || !data.decode || !data.convert || !data.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.decode, data.convert, data.sink, NULL);// //把元件都添加到管道里 if (!gst_element_link_many(data.convert, data.sink, NULL)) { //data.convert, //把元件都链接起来，为什么不连接source和decode？因为这俩需要回调函数进行特殊的链接，一般的链接是要报错的 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, "location", "rtsp://admin:abc12345@192.168.3.198/h264/ch1/main/av_stream", NULL); g_object_set(data.sink, "location", "D:\\tmp\\test.jpg", NULL); //g_object_set(data.sink, "max-lateness", 1000000000, NULL); //g_object_set(data.sink, "blocksize", 900000, NULL); //大约是将source元件的数据源给怼进去 /* Connect to the pad-added signal */ g_signal_connect(data.source, "pad-added", G_CALLBACK(pad_added_handler), &data); g_signal_connect(data.decode, "pad-added", G_CALLBACK(pad_added_handler2), &data); //g_signal_connect(data.sink, "convert-sample", G_CALLBACK(daqing_function), &data); ////GstBuffer buffer; //GstSample *sample; //g_signal_emit_by_name(data.sink, "convert-sample", &sample, NULL); //给source和decode添加衬垫，衬垫关联的是回调函数，我理解的回调函数的参数:源元件（给谁添加衬垫，就是谁），新添加的衬垫，用来传递数据的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 */ //获取一个总线，总线可以监视pipeline的运行状态，是否播放完毕等，然后进行相应的处理。 bus = gst_element_get_bus(data.pipeline); do { msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ANY); //等待执行结束并且返回 //顺带说一句，以前的老语法是GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS这样的，所以下文中的case用的是这几个错误信息，但是现在这个语法不被支持了。嗯嗯，所以改用GST_MESSAGE_ANY /* 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)); } case GST_MESSAGE_LATENCY: g_print("bus: error msg:%d\n", GST_MESSAGE_TYPE(msg)); //GstMessage ftmsg; GstObject * src; src = msg->src; cout <<"message->src:"<< src->name << endl; break; default: /* We should not reach here */ //g_printerr("Unexpected message received.\n"); break; } gst_message_unref(msg); } } while (!terminate); //只要不中止，就一直监视执行结束的状态 /* 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 = gst_element_get_static_pad(data->decode, "sink"); //pipeline的链接顺序是：source-decode-convert-sink，我们为source添加了回调，然后此处在回调内部获取了convert的对应的衬垫 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; } //此处应该是检查新为source添加的衬垫是不是已经链接到了convert衬垫 /* Check the new pad's type */ new_pad_caps = gst_pad_get_current_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, "application/x-rtp")) { g_print("It has type '%s' which is not raw rtsp. Ignoring.\n", new_pad_type); goto exit; } //检查这个衬垫当前输出的数据类型，经过一番解析，如果发现里面没有"audio/x-raw"，那说明这不是解码音频的 /* Attempt the link */ 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); } //如果两个衬垫没链接，那就人为地链接起来 exit: //这个语法就厉害了，首先定义了一个exit标号，如果前文中goto exit;那转到的就将会是此处 /* 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); } static void pad_added_handler2(GstElement *src, GstPad *new_pad, CustomData *data) { GstPad *sink_pad = gst_element_get_static_pad(data->convert, "sink"); //pipeline的链接顺序是：source-decode-convert-sink，我们为source添加了回调，然后此处在回调内部获取了convert的对应的衬垫 GstPadLinkReturn ret; GstCaps *new_pad_caps = NULL; GstStructure *new_pad_struct = NULL; const gchar *new_pad_type = NULL; g_print("22Received 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("22We are already linked. Ignoring.\n"); goto exit; } //此处应该是检查新为source添加的衬垫是不是已经链接到了convert衬垫 /* Check the new pad's type */ new_pad_caps = gst_pad_get_current_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, "video/x-raw")) { g_print("22It has type '%s' which is not raw rtsp. Ignoring.\n", new_pad_type); goto exit; } //检查这个衬垫当前输出的数据类型，经过一番解析，如果发现里面没有"audio/x-raw"，那说明这不是解码音频的 /* Attempt the link */ ret = gst_pad_link(new_pad, sink_pad); if (GST_PAD_LINK_FAILED(ret)) { g_print("22Type is '%s' but link failed.\n", new_pad_type); } else { g_print("22Link succeeded (type '%s').\n", new_pad_type); } //如果两个衬垫没链接，那就人为地链接起来 exit: //这个语法就厉害了，首先定义了一个exit标号，如果前文中goto exit;那转到的就将会是此处 /* 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); } void daqing_function(GstElement* object, GstBuffer* arg0, GstPad* arg1, gpointer user_data) { g_print("hello callback=============="); //cout << "test buffer:" << sizeof(arg0) << endl; //cout << "test pad:"<< sizeof(arg1) << endl; GstBufferPool *test = arg0->pool; //guint test = gst_buffer_n_memory(arg0); cout << test << endl; printf("%p ppp\n", test); int a = 57; int *p = &a; for (int i = 0; i < 4; i++) { printf("%c cc\n", *p); //cout << *(p ++ )<< endl; } }
