【GStreamer开发】GStreamer播放教程03——pipeline的快捷访问
目的
《GStreamer08——pipeline的快捷访问》展示了一个应用如何用appsrc和appsink这两个特殊的element在pipeline中手动输入/提取数据。playbin2也允许使用这两个element,但连接它们的方法有所不同。连接appsink到playbin2的方法在后面还会提到。这里我们主要讲述:
如何把appsrc连接到playbin2
如何配置appsrc
一个playbin2波形发生器
- <span style="font-size:14px;">#include <gst/gst.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 */
- #define AUDIO_CAPS "audio/x-raw-int,channels=1,rate=%d,signed=(boolean)true,width=16,depth=16,endianness=BYTE_ORDER"
- /* Structure to contain all our information, so we can pass it to callbacks */
- typedef struct _CustomData {
- GstElement *pipeline;
- GstElement *app_source;
- 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 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 (CHUNK_SIZE, GST_SECOND, SAMPLE_RATE);
- /* Generate some psychodelic waveforms */
- raw = (gint16 *)GST_BUFFER_DATA (buffer);
- data->c += data->d;
- data->d -= data->c / 1000;
- freq = 1100 + 11000 * data->d;
- for (i = 0; i < num_samples; i++) {
- data->a += data->b;
- data->b -= data->a / freq;
- raw[i] = (gint16)(5500 * data->a);
- }
- 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;
- }
- }
- /* 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);
- }
- /* This function is called when playbin2 has created the appsrc element, so we have
- * a chance to configure it. */
- static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) {
- gchar *audio_caps_text;
- GstCaps *audio_caps;
- g_print ("Source has been created. Configuring.\n");
- data->app_source = source;
- /* Configure appsrc */
- audio_caps_text = g_strdup_printf (AUDIO_CAPS, SAMPLE_RATE);
- audio_caps = gst_caps_from_string (audio_caps_text);
- g_object_set (source, "caps", audio_caps, NULL);
- g_signal_connect (source, "need-data", G_CALLBACK (start_feed), data);
- g_signal_connect (source, "enough-data", G_CALLBACK (stop_feed), data);
- gst_caps_unref (audio_caps);
- g_free (audio_caps_text);
- }
- int main(int argc, charchar *argv[]) {
- CustomData data;
- 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 playbin2 element */
- data.pipeline = gst_parse_launch ("playbin2 uri=appsrc://", NULL);
- g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data);
- /* 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);
- /* Free resources */
- gst_element_set_state (data.pipeline, GST_STATE_NULL);
- gst_object_unref (data.pipeline);
- return 0;
- }
- </span>
把appsrc用作pipeline的source,仅仅把playbin2的UIR设置成appsrc://即可。
- <span style="font-size:14px;"> /* Create the playbin2 element */
- data.pipeline = gst_parse_launch ("playbin2 uri=appsrc://", NULL);</span>
playbin2创建一个内部的appsrc element并且发送source-setup信号来通知应用进行设置。
- <span style="font-size:14px;"> g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data);</span>
特别地,设置appsrc的caps属性是很重要的,因为一旦这个信号的处理返回,playbin2就会根据返回值来初始化下一个element。
- <span style="font-size:14px;">/* This function is called when playbin2 has created the appsrc element, so we have
- * a chance to configure it. */
- static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) {
- gchar *audio_caps_text;
- GstCaps *audio_caps;
- g_print ("Source has been created. Configuring.\n");
- data->app_source = source;
- /* Configure appsrc */
- audio_caps_text = g_strdup_printf (AUDIO_CAPS, SAMPLE_RATE);
- audio_caps = gst_caps_from_string (audio_caps_text);
- g_object_set (source, "caps", audio_caps, NULL);
- g_signal_connect (source, "need-data", G_CALLBACK (start_feed), data);
- g_signal_connect (source, "enough-data", G_CALLBACK (stop_feed), data);
- gst_caps_unref (audio_caps);
- g_free (audio_caps_text);
- }</span>
appsrc的配置和《GStreamer08——pipeline的快捷访问》里面一样:caps设置成audio/x-raw-int,注册两个回调,这样element可以在需要/停止给它推送数据时通知应用。具体细节请参考《GStreamer08——pipeline的快捷访问》。
在这个点之后,playbin2接管处理了剩下的pipeline,应用仅仅需要生成数据即可。
至于使用appsink来从从playbin2里面提取数据,在后面的教程里面再讲述。