gstreamer-基础教程8-appsrc和appsink的应用

基础教程 8：Short-cutting the pipeline goal

这个教程通过appsrc元素，将应用程序数据注入 GStreamer 管道，并且使用appsink元素将 GStreamer 数据提取回应用程序。通过本教程，可以学习到：

如何将外部数据注入通用 GStreamer 管道。
如何从通用 GStreamer 管道中提取数据。
如何访问和操作这些数据。

说明

appsrc将生成音频数据
tee分成三路：
- audio sink一路播放音频
- video sink一路显示波形图像
- app sink一路通知用户已收到数据

关于appsrc和appsink元素的配置：

/* 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);

在appsrc上，需要设置的第一个属性appsrc是caps，它指定元素将要生成的数据类型，因此 GStreamer 可以检查是否可以与下游元素链接（也就是说，下游元素是否会理解这种数据）。该属性必须是一个GstCaps对象，它可以很容易地从gst_caps_from_string()获得.

然后我们连接到need-data和enough-data信号，它们分别在appsrc内部数据队列运行不足或者满时触发。

/* 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)

在appsink上，我们连接new-sample信号到new_sample处理函数，appsink接收到新的buffer时会触发这个signal。此外，需要通过该emit-signals属性启用信号发射，因为默认情况下它是禁用的。

全部代码路径:
gst-docs/examples/tutorials

#include <gst/gst.h>
#include <gst/audio/audio.h>
#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_request_pad_simple (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_request_pad_simple (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_request_pad_simple (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;
}

编译

这里除了用到gstreamer-1.0的基础库，还需要指定audio的库，所以pkg-config后面要跟gstreamer-audio-1.0参数：

gcc basic-tutorial-8.c -o tutorial-8 `pkg-config --cflags --libs gstreamer-audio-1.0`

编译后，运行tutorial-8，就可以看到下图的效果：

pkg-config的用法

$ pkg-config --cflags --libs gstreamer-audio-1.0-pthread -I/usr/local/include/gstreamer-1.0 -I/usr/local/include/orc-0.4 -I/usr/local/include/gstreamer-1.0 -I/usr/include/glib-2.0 -I/usr/lib/x86_64-linux-gnu/glib-2.0/include -L/usr/local/lib/x86_64-linux-gnu -lgstaudio-1.0 -lgstbase-1.0 -lgstreamer-1.0 -lgobject-2.0 -lglib-2.0

pkg-config后面的参数可以跟那些，完全取决于安装或者编译生成的.pc文件有哪些，比如用gstreamer最近的版本编译的pc文件有下面这么多，那么都可以通过pkg-config命令查到。

gst-editing-services-1.0.pc  gstreamer-base-1.0.pc          gstreamer-gl-egl-1.0.pc         gstreamer-pbutils-1.0.pc       gstreamer-riff-1.0.pc         gstreamer-tag-1.0.pc         orc-0.4.pc
gstreamer-1.0.pc             gstreamer-check-1.0.pc         gstreamer-gl-prototypes-1.0.pc  gstreamer-photography-1.0.pc   gstreamer-rtp-1.0.pc          gstreamer-transcoder-1.0.pc  orc-test-0.4.pc
gstreamer-allocators-1.0.pc  gstreamer-codecparsers-1.0.pc  gstreamer-gl-x11-1.0.pc         gstreamer-play-1.0.pc          gstreamer-rtsp-1.0.pc         gstreamer-video-1.0.pc
gstreamer-app-1.0.pc         gstreamer-controller-1.0.pc    gstreamer-insertbin-1.0.pc      gstreamer-player-1.0.pc        gstreamer-rtsp-server-1.0.pc  gstreamer-webrtc-1.0.pc
gstreamer-audio-1.0.pc       gstreamer-fft-1.0.pc           gstreamer-mpegts-1.0.pc         gstreamer-plugins-bad-1.0.pc   gstreamer-sctp-1.0.pc         gst-validate-1.0.pc
gstreamer-bad-audio-1.0.pc   gstreamer-gl-1.0.pc            gstreamer-net-1.0.pc            gstreamer-plugins-base-1.0.pc  gstreamer-sdp-1.0.pc          nice.pc

比如:

pkg-config --cflags --libs gstreamer-play-1.0
pkg-config --cflags --libs gstreamer-rtsp-server-1.0
pkg-config --cflags --libs pkg-config --cflags --libs gstreamer-webrtc-1.0