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:[email protected]/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:[email protected]/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;
}
}
标签：gst,gstreamer,c++,pad,sink,使用,new,data,GST
From： https://www.cnblogs.com/kn-zheng/p/17098315.html