C++ gstreamer函数使用总结
目录
1、GSteamer的基本API的使用
这个播放mp4报错，
这个创建play_bin，返回0，不能运行，
这个值得看：
2、创建元件并且链接起来
3、添加衬垫，添加回调，手动链接衬垫
4、打印gstreamer的版本信息
5、gstreamer封装的argparse
6、创建gst元件对象
元件的四种状态：
7、查看插件
8、链接元件
9、箱柜（箱柜本身是一个元件，但是它内部还可以是一串链接起来的元件）
10、bus总线
1、GSteamer的基本API的使用
gst_init（）初始化GStreamer 。
gst_parse_launch（）从文本描述快速构建管道 。
playbin创建自动播放管道。
gst_element_set_state（）通知GStreamer开始播放 。
gst_element_get_bus（）和gst_bus_timed_pop_filtered（）来释放资源
#include
#include
#include
int main(int argc, char *argv[]) {
GstElement *pipeline;
GstBus *bus;
GstMessage *msg;
/* Initialize GStreamer */
gst_init(&argc, &argv);
//初始化gstream
/* Build the pipeline */
pipeline =gst_parse_launch("playbin uri=file:///D:/gstream/1.mp4",NULL);
//gst_parse_launch使用系统预设的管道来处理流媒体。gst_parse_launch创建的是一个由playbin单元素组成的管道
/* Start playing */
gst_element_set_state(pipeline, GST_STATE_PLAYING);
//将我们的元素设置为playing状态才能开始播放
/* Wait until error or EOS */
bus = gst_element_get_bus(pipeline);
msg =gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,GST_MESSAGE_ERROR ); //| GST_MESSAGE_EOS
//遇到错误或者播放完毕以后gst_bus_timed_pop_filtered（）会返回一条消息
/* Free resources */
if (msg != NULL) {
gst_message_unref(msg);
//需要使用gst_message_unref（）将msg释放,此函数专门清除gst_bus_timed_pop_filtered
gst_object_unref(bus);
gst_element_set_state(pipeline, GST_STATE_NULL);
gst_object_unref(pipeline);
}
printf("finish");
return 0;
} 这个播放mp4报错，
报错：
(Project1.exe:10556): GStreamer-CRITICAL **: 02:09:36.630: gst_element_set_state: assertion 'GST_IS_ELEMENT (element)' failed
(Project1.exe:10556): GStreamer-CRITICAL **: 02:09:36.630: gst_element_get_bus: assertion 'GST_IS_ELEMENT (element)' failed
(Project1.exe:10556): GStreamer-CRITICAL **: 02:09:36.630: gst_bus_timed_pop_filtered: assertion 'GST_IS_BUS (bus)' failed
c++ gstreamer使用2 - 走看看
这个创建play_bin，返回0，不能运行，
#include
#include
#include
#include
/* 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.");
return -1;
}
/* Set the URI to play */
g_object_set(data.playbin, "uri", "file:///D:/data/person/skivideo.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.");
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)",data->n_video, data->n_audio, data->n_text);
g_print(" ");
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:", i);
gst_tag_list_get_string(tags, GST_TAG_VIDEO_CODEC, &str);
g_print(" codec: %s", str ? str : "unknown");
g_free(str);
gst_tag_list_free(tags);
}
}
g_print(" ");
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:", i);
if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &str)) {
g_print(" codec: %s", str);
g_free(str);
}
if (gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &str)) {
g_print(" language: %s", str);
g_free(str);
}
if (gst_tag_list_get_uint(tags, GST_TAG_BITRATE, &rate)) {
g_print(" bitrate: %d", rate);
}
gst_tag_list_free(tags);
}
}
g_print(" ");
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:", i);
if (gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &str)) {
g_print(" language: %s", 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(" ");
g_print("Currently playing video stream %d, audio stream %d and text stream %d",data->current_video, data->current_audio, data->current_text);
g_print("Type any number and hit ENTER to select a different audio stream");
}
/* 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", GST_OBJECT_NAME(msg->src), err->message);
g_printerr("Debugging information: %s", 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.");
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");
}
else {
/* If the input was a valid audio stream index, set the current audio stream */
g_print("Setting current audio stream to %d ", index);
g_object_set(data->playbin, "current-audio", index, NULL);
}
}
g_free(str);
return TRUE;
} 这个值得看：
gstreamer播放教程一：playbin——获取媒体的流信息、切换流。_胖子呀的博客-CSDN博客_playbin
2、创建元件并且链接起来
#include
int main(int argc, char *argv[]) {
GstElement *pipeline, *source, *sink;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init(&argc, &argv);
/* Create the elements */
source = gst_element_factory_make("videotestsrc", "source");
sink = gst_element_factory_make("autovideosink", "sink");
//创建元件，参数：元件的类型，元件名称
//videotestsrc是一个源元素（它产生数据），它创建一个测试视频模式。
//autovideosink是一个接收器元素（它消耗数据），它在窗口上显示它接收到的图像。
/* Create the empty pipeline */
pipeline = gst_pipeline_new("test-pipeline");
//创建管道，管道是一种特殊类型的bin，（估计就是箱柜）
if (!pipeline || !source || !sink) {
g_printerr("Not all elements could be created.
");
return -1;
}
/* Build the pipeline */
gst_bin_add_many(GST_BIN(pipeline), source, sink, NULL);
//向管道中添加元件，以null结尾，添加单个和可以，函数是：gst_bin_add()
if (gst_element_link(source, sink) != TRUE) {
g_printerr("Elements could not be linked.
");
gst_object_unref(pipeline);
return -1;
}
/* Modify the source's properties */
g_object_set(source, "pattern", 0, NULL);
//修改元件的属性
/* Start playing */
ret = gst_element_set_state(pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr("Unable to set the pipeline to the playing state.
");
gst_object_unref(pipeline);
return -1;
}
//设置管道开始工作
//调用gst_element_set_state()，并且检查其返回值是否有错误。
/* Wait until error or EOS */
bus = gst_element_get_bus(pipeline);
//获取pipeline的总线
msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR );
//gst_bus_timed_pop_filtered()等待执行结束并返回GstMessage
/* Parse message：
GstMessage是一种非常通用的结构，
通过使用 GST_MESSAGE_TYPE()宏可以获得其中的消息
*/
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
", GST_OBJECT_NAME(msg->src), err->message);
g_printerr("Debugging information: %s
", debug_info ? debug_info : "none");
g_clear_error(&err);
g_free(debug_info);
break;
case GST_MESSAGE_EOS:
g_print("End-Of-Stream reached.
");
break;
default:
/* We should not reach here because we only asked for ERRORs and EOS */
g_printerr("Unexpected message received.
");
break;
}
gst_message_unref(msg);
}
/* Free resources */
gst_object_unref(bus);
gst_element_set_state(pipeline, GST_STATE_NULL);
gst_object_unref(pipeline);
return 0;
} 3、添加衬垫，添加回调，手动链接衬垫
#include
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *resample;
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);
//声明一个函数，叫添加衬垫的函数pad_added_handler。
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("uridecodebin", "source");
data.convert = gst_element_factory_make("audioconvert", "convert");
data.resample = gst_element_factory_make("audioresample", "resample");
data.sink = gst_element_factory_make("autoaudiosink", "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.convert || !data.resample || !data.sink) {
g_printerr("Not all elements could be created.
");
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.resample, data.sink, NULL);
if (!gst_element_link_many(data.convert, data.resample, data.sink, NULL)) {
g_printerr("Elements could not be linked.
");
gst_object_unref(data.pipeline);
return -1;
}
/* Set the URI to play */
g_object_set(data.source, "uri", "https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm", NULL);
//大约是将source元件的衬垫链接到某个网址上
/* Connect to the pad-added signal */
g_signal_connect(data.source, "pad-added", G_CALLBACK(pad_added_handler), &data);
//GSignals是GStreamer中的关键点。它们使您可以在发生事情时（通过回调）得到通知，所以我们为source元件添加了一个回调
//这个回调好像没有传递参数啊喂，好吧，官方是真么说的：src是GstElement触发信号的。在此示例中，它只能是uridecodebin。newpad是刚刚添加到src元素中的，我理解为哦我们为source添加回调这件事就是增加了一个衬垫，data是当作指针来传递信号的
/* 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.
");
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_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
/* 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
", GST_OBJECT_NAME(msg->src), err->message);
g_printerr("Debugging information: %s
", 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.
");
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:
",
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.
");
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->convert, "sink");
//pipeline的链接顺序是：source-convert-resample-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':
", 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.
");
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, "audio/x-raw")) {
g_print("It has type '%s' which is not raw audio. Ignoring.
", 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.
", new_pad_type);
}
else {
g_print("Link succeeded (type '%s').
", 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);
}
4、打印gstreamer的版本信息
#include
#include
#include
//#include
int main(int argc,char *argv[])
{
const gchar *nano_str;
guint major, minor, micro, nano;
gst_init(&argc, &argv);
gst_version(&major, &minor, µ, &nano);
if (nano == 1)
nano_str = "(CVS)";
else if (nano == 2)
nano_str = "(Prerelease)";
else
nano_str = "";
printf("This program is linked against GStreamer %d.%d.%d %s
",major, minor, micro, nano_str);
return 0;
}
5、gstreamer封装的argparse
#include
#include
#include
#include
int main(int argc,char *argv[])
{
gboolean silent = FALSE;
gchar *savefile = NULL;
GOptionContext *ctx;
GError *err = NULL;
GOptionEntry entries[] = {
{ "silent", 's', 0, G_OPTION_ARG_NONE, &silent,"do not output status information", NULL },
{ "output", 'o', 0, G_OPTION_ARG_STRING, &savefile,"save xml representation of pipeline to FILE and exit", "FILE" },
{ NULL }
};
ctx = g_option_context_new("- Your application");
g_option_context_add_main_entries(ctx, entries, NULL);
g_option_context_add_group(ctx, gst_init_get_option_group());
if (!g_option_context_parse(ctx, &argc, &argv, &err)) {
g_print("Failed to initialize: %s
", err->message);
g_error_free(err);
return 1;
}
printf("Run me with --help to see the Application options appended.
");
return 0;
}
6、创建gst元件对象
#include
#include
#include
int main(int argc, char *argv[])
{
GstElement *element;
gchar *name;
/* init GStreamer */
gst_init(&argc, &argv);
/* create element */
element = gst_element_factory_make("fakesrc", "source");
//创建一个jst元件
if (!element) {
g_print("Failed to create element of type 'fakesrc'
");
return -1;
}
else {
g_print("gstelement ok!
");
}
element = gst_element_factory_make("fakesrc", "source");
/* get name */
g_object_get(G_OBJECT(element), "name", &name, NULL);
//g_object_get获取gobject对象的名字属性
g_print("The name of the element is '%s'.
", name);
g_free(name);
gst_object_unref(GST_OBJECT(element));
//释放jst元件，必须手动释放
return 0;
}
元件的四种状态：
GST_STATE_NULL: 默认状态
该状态将会回收所有被该元件占用的资源。
GST_STATE_READY: 准备状态
元件会得到所有所需的全局资源，这些全局资源将被通过该元 件的数据流所使用。例如打开设备、分配缓存等。但在这种状态下，数据流仍未开始被处 理，所 以数据流的位置信息应该自动置 0。如果数据流先前被打开过，它应该被关闭，并且其位置信 息、特性信息应该被重新置为初始状态。
GST_STATE_PAUSED: 暂停状态
在这种状态下，元件已经对流开始了处理，但此刻暂停了处理。因此该 状态下元件可以修改流的位置信息，读取或者处理流数据，以及一旦状态变为 PLAYING，流可 以重放数据流。这种情况下，时钟是禁止运行的。总之， PAUSED 状态除了不能运行时钟外， 其它与 PLAYING 状态一模一样。处于 PAUSED 状态的元件会很快变换到 PLAYING 状态。举 例来说，视频或音频输出元件会等待数据的到来，并将它们压入队列。一旦状态改变，元件就会 处理接收到的数据。同样，视频接收元件能够播放数据的第 一帧。(因为这并不会影响时钟)。自 动加载器（Autopluggers）可以对已经加载进管道的插件进行这种状态转换。其它更多的像 codecs 或者 filters 这种元件不需要在这个状态上做任何事情。
GST_STATE_PLAYING:
PLAYING 状态除了当前运行时钟外，其它与 PAUSED 状态一模一 样。你可以通过函数 gst_element_set_state()来改变一个元件的状态。你如果显式地改变一个元件 的状态，GStreamer 可能会 使它在内部经过一些中间状态。例如你将一个元件从 NULL 状态设 置为 PLAYING 状态，GStreamer 在其内部会使得元件经历过 READY 以及 PAUSED 状态。 当处于 GST_STATE_PLAYING 状态，管道会自动处理数据。它们不需要任何形式的迭代。
7、查看插件
#include
#include
#include
int main(int argc,char *argv[])
{
GstElementFactory *factory;
//声明插件，插件是GstElementFactory
/* init GStreamer */
gst_init(&argc, &argv);
/* get factory */
factory = gst_element_factory_find("audiotestsrc");
//寻找系统里是否有这个插件
if (!factory) {
g_print("You don't have the 'audiotestsrc' element installed!
");
return -1;
}
/* display information */
g_print("The '%s' element is a member of the category %s.
"
"Description: %s
",
gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(factory)),
gst_element_factory_get_klass(factory),
gst_element_factory_get_description(factory));
//打印出插件的信息
return 0;
}
这个功能就像是命令行里的如下命令：
gst-inspect-1.0 audiotestsrc
#gst-inspect-1.0 加插件名
8、链接元件
#include
#include
#include
#include
int
main(int argc,
char *argv[])
{
GstElement *pipeline;
GstElement *source, *filter, *sink;
//声明一个源元件，过滤元件和接收元件
/* init */
gst_init(&argc, &argv);
/* create pipeline */
pipeline = gst_pipeline_new("my-pipeline");
/* create elements */
source = gst_element_factory_make("fakesrc", "source");
filter = gst_element_factory_make("identity", "filter");
sink = gst_element_factory_make("fakesink", "sink");
//选择3个插件创建3个不同的元件
/* must add elements to pipeline before linking them */
gst_bin_add_many(GST_BIN(pipeline), source, filter, sink, NULL);
/* link */
if (!gst_element_link_many(source, filter, sink, NULL)) {
g_warning("Failed to link elements!");
}
return 0;
}
9、箱柜（箱柜本身是一个元件，但是它内部还可以是一串链接起来的元件）
#include
#include
#include
#include
int
main(int argc,
char *argv[])
{
GstElement *bin, *pipeline, *source, *sink;
/* init */
gst_init(&argc, &argv);
/* create */
pipeline = gst_pipeline_new("my_pipeline");
bin = gst_pipeline_new("my_bin");
//创建箱柜：gst_pipeline_new和gst_bin_new
source = gst_element_factory_make("fakesrc", "source");
sink = gst_element_factory_make("fakesink", "sink");
/* set up pipeline */
gst_bin_add_many(GST_BIN(bin), source, sink, NULL);
gst_bin_add(GST_BIN(pipeline), bin);
//添加元件到箱柜
gst_bin_remove(GST_BIN(bin),sink);
//从箱柜中移除元件，移除的元件自动被销毁，
gst_element_link(source, sink);
//链接元件，因为sink元件被我移除了，所以可能实际上运行不起来
gst_object_unref(GST_OBJECT(source));
gst_object_unref(GST_OBJECT(sink));
return 0;
}
10、bus总线
获取bus总线：gst_pipeline_get_bus
在总线上添加一个回调函数（官方语言叫watch）：gst_bus_add_watch
#include
static GMainLoop *loop;
static gboolean
my_bus_callback(GstBus *bus,GstMessage *message,gpointer data)
{
g_print("Got %s message
", GST_MESSAGE_TYPE_NAME(message));
switch (GST_MESSAGE_TYPE(message)) {
case GST_MESSAGE_ERROR: {
GError *err;
gchar *debug;
gst_message_parse_error(message, &err, &debug);
g_print("Error: %s
", err->message);
g_error_free(err);
g_free(debug);
g_main_loop_quit(loop);
break;
}
case GST_MESSAGE_EOS:
/* end-of-stream */
g_main_loop_quit(loop);
break;
default:
/* unhandled message */
g_print("something happend!
");
break;
}
/* we want to be notified again the next time there is a message
* on the bus, so returning TRUE (FALSE means we want to stop watching
* for messages on the bus and our callback should not be called again)
*/
return TRUE;
}
gint main(gint argc,gchar *argv[])
{
GstElement *pipeline;
GstBus *bus;
/* init */
gst_init(&argc, &argv);
/* create pipeline, add handler */
pipeline = gst_pipeline_new("my_pipeline");
/* adds a watch for new message on our pipeline's message bus to
* the default GLib main context, which is the main context that our
* GLib main loop is attached to below
*/
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
//首先获取总线
gst_bus_add_watch(bus, my_bus_callback, NULL);
//然后添加一个消息处理器：设置消息处理器到管道的总线上gst_bus_add_watch ()
gst_object_unref(bus);
/* create a mainloop that runs/iterates the default GLib main context
* (context NULL), in other words: makes the context check if anything
* it watches for has happened. When a message has been posted on the
* bus, the default main context will automatically call our
* my_bus_callback() function to notify us of that message.
* The main loop will be run until someone calls g_main_loop_quit()
*/
loop = g_main_loop_new(NULL, FALSE);
g_main_loop_run(loop);
/* clean up */
gst_element_set_state(pipeline, GST_STATE_NULL);
/*gst_element_unref(pipeline);
gst_main_loop_unref(loop);*/
return 0;
}
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版权声明：本文为CSDN博主「Geek.Fan」的原创文章，遵循CC 4.0 BY-SA版权协议，转载请附上原文出处链接及本声明。
原文链接：https://blog.csdn.net/fanyun_01/article/details/125511163