摘要
在我们前面的文章中，我们的Pipline都是使用GStreamer自带的插件去产生/消费数据。在实际的情况中，我们的数据源可能没有相应的gstreamer插件，但我们又需要将数据发送到GStreamer Pipeline中。GStreamer为我们提供了Appsrc以及Appsink插件，用于处理这种情况，本文将介绍如何使用这些插件来实现数据与应用程序的交互。
Appsrc与Appsink
GStreamer提供了多种方法使得应用程序与GStreamer Pipeline之间可以进行数据交互，我们这里介绍的是最简单的一种方式：appsrc与appsink。
- appsrc：
用于将应用程序的数据发送到Pipeline中。应用程序负责数据的生成，并将其作为GstBuffer传输到Pipeline中。
appsrc有2中模式，拉模式和推模式。在拉模式下，appsrc会在需要数据时，通过指定接口从应用程序中获取相应数据。在推模式下，则需要由应用程序主动将数据推送到Pipeline中，应用程序可以指定在Pipeline的数据队列满时是否阻塞相应调用，或通过监听enough-data和need-data信号来控制数据的发送。
- appsink：
用于从Pipeline中提取数据，并发送到应用程序中。
appsrc和appsink需要通过特殊的API才能与Pipeline进行数据交互，相应的接口可以查看官方文档，在编译的时候还需连接gstreamer-app库。
GstBuffer
在GStreamer Pipeline中的plugin间传输的数据块被称为buffer，在GStreamer内部对应于GstBuffer。Buffer由Source Pad产生，并由Sink Pad消耗。一个Buffer只表示一块数据，不同的buffer可能包含不同大小，不同时间长度的数据。同时，某些Element中可能对Buffer进行拆分或合并，所以GstBuffer中可能包含不止一个内存数据，实际的内存数据在GStreamer系统中通过GstMemory对象进行描述，因此，GstBuffer可以包含多个GstMemory对象。
每个GstBuffer都有相应的时间戳以及时间长度，用于描述这个buffer的解码时间以及显示时间。
示例代码
本例在GStreamer基础教程08 - 多线程示例上进行扩展，首先使用appsrc替代audiotestsrc用于产生audio数据，另外增加一个新的分支，将tee产生的数据发送到应用程序，由应用程序决定如何处理收到的数据。Pipeline的示意图如下：
#include#include #include <string.h> #define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */ #define SAMPLE_RATE 44100 /* Samples per second we are sending */ /* Structure to contain all our information, so we can pass it to callbacks */ typedef struct _CustomData { GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink; GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink; GstElement *app_queue, *app_sink; guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */ gfloat a, b, c, d; /* For waveform generation */ guint sourceid; /* To control the GSource */ GMainLoop *main_loop; /* GLib's Main Loop */ } CustomData; /* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc. * The idle handler is added to the mainloop when appsrc requests us to start sending data (need-data signal) * and is removed when appsrc has enough data (enough-data signal). */ static gboolean push_data (CustomData *data) { GstBuffer *buffer; GstFlowReturn ret; int i; GstMapInfo map; gint16 *raw; gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */ gfloat freq; /* Create a new empty buffer */ buffer = gst_buffer_new_and_alloc (CHUNK_SIZE); /* Set its timestamp and duration */ GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE); GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (num_samples, GST_SECOND, SAMPLE_RATE); /* Generate some psychodelic waveforms */ gst_buffer_map (buffer, &map, GST_MAP_WRITE); raw = (gint16 *)map.data; data->c += data->d; data->d -= data->c / 1000; freq = 1100 + 1000 * data->d; for (i = 0; i < num_samples; i++) { data->a += data->b; data->b -= data->a / freq; raw[i] = (gint16)(500 * data->a); } gst_buffer_unmap (buffer, &map); data->num_samples += num_samples; /* Push the buffer into the appsrc */ g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret); /* Free the buffer now that we are done with it */ gst_buffer_unref (buffer); if (ret != GST_FLOW_OK) { /* We got some error, stop sending data */ return FALSE; } return TRUE; } /* This signal callback triggers when appsrc needs data. Here, we add an idle handler * to the mainloop to start pushing data into the appsrc */ static void start_feed (GstElement *source, guint size, CustomData *data) { if (data->sourceid == 0) { g_print ("Start feeding\n"); data->sourceid = g_idle_add ((GSourceFunc) push_data, data); } } /* This callback triggers when appsrc has enough data and we can stop sending. * We remove the idle handler from the mainloop */ static void stop_feed (GstElement *source, CustomData *data) { if (data->sourceid != 0) { g_print ("Stop feeding\n"); g_source_remove (data->sourceid); data->sourceid = 0; } } /* The appsink has received a buffer */ static GstFlowReturn new_sample (GstElement *sink, CustomData *data) { GstSample *sample; /* Retrieve the buffer */ g_signal_emit_by_name (sink, "pull-sample", &sample); if (sample) { /* The only thing we do in this example is print a * to indicate a received buffer */ g_print ("*"); gst_sample_unref (sample); return GST_FLOW_OK; } return GST_FLOW_ERROR; } /* This function is called when an error message is posted on the bus */ static void error_cb (GstBus *bus, GstMessage *msg, CustomData *data) { GError *err; gchar *debug_info; /* Print error details on the screen */ 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); } int main(int argc, char *argv[]) { CustomData data; GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad; GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad; GstAudioInfo info; GstCaps *audio_caps; GstBus *bus; /* Initialize cumstom data structure */ memset (&data, 0, sizeof (data)); data.b = 1; /* For waveform generation */ data.d = 1; /* Initialize GStreamer */ gst_init (&argc, &argv); /* Create the elements */ data.app_source = gst_element_factory_make ("appsrc", "audio_source"); data.tee = gst_element_factory_make ("tee", "tee"); data.audio_queue = gst_element_factory_make ("queue", "audio_queue"); data.audio_convert1 = gst_element_factory_make ("audioconvert", "audio_convert1"); data.audio_resample = gst_element_factory_make ("audioresample", "audio_resample"); data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink"); data.video_queue = gst_element_factory_make ("queue", "video_queue"); data.audio_convert2 = gst_element_factory_make ("audioconvert", "audio_convert2"); data.visual = gst_element_factory_make ("wavescope", "visual"); data.video_convert = gst_element_factory_make ("videoconvert", "video_convert"); data.video_sink = gst_element_factory_make ("autovideosink", "video_sink"); data.app_queue = gst_element_factory_make ("queue", "app_queue"); data.app_sink = gst_element_factory_make ("appsink", "app_sink"); /* Create the empty pipeline */ data.pipeline = gst_pipeline_new ("test-pipeline"); if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 || !data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual || !data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) { g_printerr ("Not all elements could be created.\n"); return -1; } /* Configure wavescope */ g_object_set (data.visual, "shader", 0, "style", 0, NULL); /* Configure appsrc */ gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL); audio_caps = gst_audio_info_to_caps (&info); g_object_set (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL); g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data); g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data); /* Configure appsink */ g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL); g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data); gst_caps_unref (audio_caps); /* Link all elements that can be automatically linked because they have "Always" pads */ gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, data.app_queue, data.app_sink, NULL); if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE || gst_element_link_many (data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE || gst_element_link_many (data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, NULL) != TRUE || gst_element_link_many (data.app_queue, data.app_sink, NULL) != TRUE) { g_printerr ("Elements could not be linked.\n"); gst_object_unref (data.pipeline); return -1; } /* Manually link the Tee, which has "Request" pads */ tee_audio_pad = gst_element_get_request_pad (data.tee, "src_%u"); g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad)); queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink"); tee_video_pad = gst_element_get_request_pad (data.tee, "src_%u"); g_print ("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad)); queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink"); tee_app_pad = gst_element_get_request_pad (data.tee, "src_%u"); g_print ("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad)); queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink"); if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK || gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK || gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) { g_printerr ("Tee could not be linked\n"); gst_object_unref (data.pipeline); return -1; } gst_object_unref (queue_audio_pad); gst_object_unref (queue_video_pad); gst_object_unref (queue_app_pad); /* Instruct the bus to emit signals for each received message, and connect to the interesting signals */ bus = gst_element_get_bus (data.pipeline); gst_bus_add_signal_watch (bus); g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, &data); gst_object_unref (bus); /* Start playing the pipeline */ gst_element_set_state (data.pipeline, GST_STATE_PLAYING); /* 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); /* Release the request pads from the Tee, and unref them */ gst_element_release_request_pad (data.tee, tee_audio_pad); gst_element_release_request_pad (data.tee, tee_video_pad); gst_element_release_request_pad (data.tee, tee_app_pad); gst_object_unref (tee_audio_pad); gst_object_unref (tee_video_pad); gst_object_unref (tee_app_pad); /* Free resources */ gst_element_set_state (data.pipeline, GST_STATE_NULL); gst_object_unref (data.pipeline); return 0; }
保存以上代码，执行下列编译命令即可得到可执行程序：
gcc basic-tutorial-9.c -o basic-tutorial-9 `pkg-config --cflags --libs gstreamer-1.0 gstreamer-audio-1.0 `
Note：本例在编译时没有连接gstreamer-app-1.0的库是因为我们使用的是通过信号的方式，由appsrc自动处理buffer，所以无需在编译时连接相应库。在源码分析部分会详述。
源码分析
与上一示例相同，首先对所需Element进行实例化，同时将Element的Always Pad连接起来，并与tee的Request Pad相连。此外我们还对appsrc及appsink进行了相应的配置：
/* Configure appsrc */ gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL); audio_caps = gst_audio_info_to_caps (&info); g_object_set (data.app_source, "caps", audio_caps, NULL); g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data); g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
首先需要对appsrc的caps进行设定，指定我们会产生何种类型的数据，这样GStreamer会在连接阶段检查后续的Element是否支持此数据类型。这里的 caps必须为GstCaps对象，我们可以通过gst_caps_from_string()或gst_audio_info_to_caps ()得到相应的实例。
我们同时监听了“need-data”与“enough-data”事件，这2个事件由appsrc在需要数据和缓冲区满时触发，使用这2个事件可以方便的控制何时产生数据与停止数据。
/* Configure appsink */ g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL); g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data); gst_caps_unref (audio_caps);
对于appsink，我们监听“new-sample”事件，用于appsink在收到数据时的处理。同时我们需要显式的使能“new-sample”事件，因为这个事件默认是处于关闭状态。
Pipeline的播放，停止及消息处理与其他示例相同，不再复述。我们接下来将查看我们监听事件的回调函数。
/* This signal callback triggers when appsrc needs data. Here, we add an idle handler * to the mainloop to start pushing data into the appsrc */ static void start_feed (GstElement *source, guint size, CustomData *data) { if (data->sourceid == 0) { g_print ("Start feeding\n"); data->sourceid = g_idle_add ((GSourceFunc) push_data, data); } }
appsrc会在其内部的数据队列即将缺乏数据时调用此回调函数，这里我们通过注册一个GLib的idle函数来向appsrc填充数据，GLib的主循环在“idle”状态时会循环调用 push_data，用于向appsrc填充数据。这只是一种向appsrc填充数据的方式，我们可以在任意线程中想appsrc填充数据。
我们保存了g_idle_add()的返回值，以便后续用于停止数据写入。
/* This callback triggers when appsrc has enough data and we can stop sending. * We remove the idle handler from the mainloop */ static void stop_feed (GstElement *source, CustomData *data) { if (data->sourceid != 0) { g_print ("Stop feeding\n"); g_source_remove (data->sourceid); data->sourceid = 0; } }
stop_feed函数会在appsrc内部数据队列满时被调用。这里我们仅仅通过g_source_remove() 将先前注册的idle处理函数从GLib的主循环中移除（idle处理函数是被实现为一个GSource）。
/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc. * The ide handler is added to the mainloop when appsrc requests us to start sending data (need-data signal) * and is removed when appsrc has enough data (enough-data signal). */ static gboolean push_data (CustomData *data) { GstBuffer *buffer; GstFlowReturn ret; int i; gint16 *raw; gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */ gfloat freq; /* Create a new empty buffer */ buffer = gst_buffer_new_and_alloc (CHUNK_SIZE); /* Set its timestamp and duration */ GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE); GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (num_samples, GST_SECOND, SAMPLE_RATE); /* Generate some psychodelic waveforms */ raw = (gint16 *)GST_BUFFER_DATA (buffer);
此函数会将真实的数据填充到appsrc的数据队列中，首先通过gst_buffer_new_and_alloc()分配一个GstBuffer对象，然后通过产生的采样数量计算这块buffre所对应的时间戳及事件长度。
gst_util_uint64_scale(val, num, denom)函数用于计算 val * num / denom，此函数内部会对数据范围进行检测，避免溢出的问题。
GstBuffer的数据指针可以通过GST_BUFFER_DATA 宏获取，在写数据时需要避免超出内存分配大小。本文将跳过audio波形生成的函数，其内容不是本文介绍的重点。
/* Push the buffer into the appsrc */ g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret); /* Free the buffer now that we are done with it */ gst_buffer_unref (buffer);
在我们准备好数据后，我们这里通过“push-buffer”事件通知appsrc数据就绪，并释放我们申请的buffer。 另外一种方式为通过调用gst_app_src_push_buffer() 向appsrc填充数据，这种方式就需要在编译时链接gstreamer-app-1.0库，同时gst_app_src_push_buffer() 会接管GstBuffer的所有权，调用者无需释放buffer。在所有数据都发送完成后，我们可以调用gst_app_src_end_of_stream()向Pipeline写入EOS事件。
/* The appsink has received a buffer */ static GstFlowReturn new_sample (GstElement *sink, CustomData *data) { GstSample *sample; /* Retrieve the buffer */ g_signal_emit_by_name (sink, "pull-sample", &sample); if (sample) { /* The only thing we do in this example is print a * to indicate a received buffer */ g_print ("*"); gst_sample_unref (sample); return GST_FLOW_OK; } return GST_FLOW_ERROR; }
当appsink得到数据时会调用new_sample函数，我们使用“pull-sample”信号提取sample，这里仅输出一个”*“表明此函数被调用。除此之外，我们同样可以使用gst_app_sink_pull_sample ()获取Sample。得到GstSample之后，我们可以通过gst_sample_get_buffer()得到Sample中所包含的GstBuffer，再使用GST_BUFFER_DATA， GST_BUFFER_SIZE 等接口访问其中的数据。使用完后，得到的GstSample同样需要通过gst_sample_unref()进行释放。
需要注意的是，在某些Pipeline里得到的GstBuffer可能会和source中填充的GstBuffer有所差异，因为Pipeline中的Element可能对Buffer进行各种处理（此例中不存在此种情况，因为在appsrc与appsink之间只存在一个tee）。
总结
在本文中，我们介绍了:
- 如何通过appsrc向Pipeline中写入数据
- 如何通过appsink取得Pipeline中的数据
- 如何获取/填充GstBuffer中对应的数据
后续我们将继续学习有关GStreamer的其他知识。
引用
https://gstreamer.freedesktop.org/documentation/tutorials/basic/short-cutting-the-pipeline.html?gi-language=c