【GStreamer开发】GStreamer基础教程03——动态pipeline
本教程介绍pipeline的一种新的创建方式——在运行中创建,而不是在运行前一次性的创建结束。
介绍
在这篇教程里的pipeline并非在运行前就全部创建结束的。放松一下,这样做没有任何问题。如果我们不进行更深入的处理,那么数据在到达pipeline的末尾时就直接丢弃了,当然,我们肯定会进行深入处理的。。。
在这个例子中,我们会打开一个已经包含了音视频的文件(Container file)。负责打开这样的容器文件的element叫做demuxer,我们常见的容器格式包括MKV、QT、MOV、Ogg还有ASF、WMV、WMA等等。
在一个容器中可能包含多个流(比如:一路视频,两路音频),demuxer会把他们分离开来,然后从不同的输出口送出来。这样在pipeline里面的不同的分支可以处理不同的数据。
在GStreamer里面有个术语描述这样的接口——pad(GstPad)。Pad分成sink pad——数据从这里进入一个element——和source pad——数据从这里流出element。很自然的,source element仅包含source pad,sink element仅包含sink pad,而过滤器两种pad都包含。
一个demuxer包含一个sink pad和多个source pad,数据从sink pad输入然后每个流都有一个source pad。
为了完整起见,给出一张示意图,图中有一个demuxer和两个分支,一个处理音频一个处理视频。请注意,这不是本教程pipeline的示意图。
这里主要复杂在demuxer在没有看到容器文件之前无法确定需要做的工作,不能生成对应的内容。也就是说,demuxer开始时是没有source pad给其他element连接用的。
解决方法是只管建立pipeline,让source和demuxer连接起来,然后开始运行。当demuxer接收到数据之后它就有了足够的信息生成source pad。这时我们就可以继续把其他部分和demuxer新生成的pad连接起来,生成一个完整的pipeline。
简单起见,在这个例子中,我们仅仅连接音频的pad而不处理视频的pad。
动态Hello World
- #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 *sink;
- } CustomData;
- /* Handler for the pad-added signal */
- static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
- int main(int argc, charchar *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.sink = gst_element_factory_make ("autoaudiosink", "sink");
- /* Create the empty pipeline */
- data.pipeline = gst_pipeline_new ("test-pipeline");
- if (!data.pipeline || !data.source || !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.convert , data.sink, NULL);
- 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 to play */
- g_object_set (data.source, "uri", "http://docs.gstreamer.com/media/sintel_trailer-480p.webm", 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 = 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 ("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 not raw audio. Ignoring.\n", new_pad_type);
- goto exit;
- }
- /* 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);
- }
- 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);
- }
工作流程
- /* Structure to contain all our information, so we can pass it to callbacks */
- typedef struct _CustomData {
- GstElement *pipeline;
- GstElement *source;
- GstElement *convert;
- GstElement *sink;
- } CustomData;
- /* Handler for the pad-added signal */
- static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
- /* 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");
我们像前面一样创建一个个element。uridecodebin自己会在内部初始化必要的element,然后把一个URI变成一个原始音视频流输出,它差不多做了playbin2的一半工作。因为它自己带着demuxer,所以它的source pad没有初始化,我们等会会用到。
audioconvert在不同的音频格式转换时很有用。这里用这个element是为了确保应用的平台无关性。
autoaudiosink和上一篇教程里面的autovideosink是非常相似的,只是操作的时音频流。这个element的输出就是直接送往声卡的音频流。
- 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 to play */
- g_object_set (data.source, "uri", "http://docs.gstreamer.com/media/sintel_trailer-480p.webm", NULL);
信号
- /* Connect to the pad-added signal */
- g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
在这段代码里面,我们使用g_signal_connect()方法把“pad-added”信号和我们的源(uridecodebin)联系了起来,并且注册了一个回调函数。GStreamer把&data这个指针的内容传给回调函数,这样CustomData这个数据结构中的数据也就传递了过去。
这个信号是有GstElement产生的,可以在相关的文档中找到或者用gst-inspect方法来查到。
我们现在准备开始运行了!和前面的教程一样,把pipeline置成PLAYING状态,然后开始监听ERROR或者EOS。
回调函数
当我们的source element最后获得足够的数据时,它就会自动生成source pad,并且触发“pad-added”信号。这样我们的回调就会被调用了:
- static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
new_pad是加到src上的pad。通常来说,是我们需要连接的pad。
data指针是跟随信号一起过来的参数,在这个例子中,我们传递的时CustomData指针。
- GstPad *sink_pad = gst_element_get_static_pad (data->convert, "sink");
- /* 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 not raw audio. Ignoring.\n", new_pad_type);
- goto exit;
- }
gst_pad_get_caps()方法会获得pad的capability(也就是pad支持的数据类型),是被封装起来的GstCaps结构。一个pad可以有多个capability,GstCaps可以包含多个GstStructure,每个都描述了一个不同的capability。
在这个例子中,我们知道我们的pad需要的capability是声音,我们使用gst_caps_get_structure()方法来获得GstStructure。
最后,我们用gst_structure_get_name()方法来获得structure的名字——最主要的描述部分。如果名字不是由audio/x-raw开始的,就意味着不是一个解码的音频数据,也就不是我们所需要的,反之,就是我们需要连接的:
- /* 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);
- }
到这儿我们的任务就完成了,当一个合适的pad出现后,它会和后面的audio处理部分相连,然后继续运行直到ERROR或者EOS。下面,我们再介绍一点点GStreamer状态的概念。
状态
我们介绍过不把pipeline置成PLAYING状态,播放是不会开始的。这里我们继续介绍一下其他的几种状态,在GStreamer里面有4种状态:
NULL | NULL状态或者初始化状态 |
READY | element已经READY或者PAUSED |
PAUSED | element已经PAUSED,准备接受数据 |
PLAYING | element在PLAYING,时钟在运行数据 |
- 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;
绝大多数应用都是在PLAYING状态开始播放,然后跳转到PAUSE状态来提供暂停功能,最后在退出时退到NULL状态。