gstreamer/subprojects/gst-docs/examples/tutorials/playback-tutorial-3.c
Piotr Brzeziński 0b65c0ead5 tutorials: Fix warning when compiling on macOS
GstMainFunc has one more arg than the standard main() to support bindings, let's cast to get rid of the warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6038>
2024-02-02 12:54:32 +01:00

186 lines
5.5 KiB
C

#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#ifdef __APPLE__
#include <TargetConditionals.h>
#endif
#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;
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;
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;
}
}
/* 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 playbin has created the appsrc element, so we have
* a chance to configure it. */
static void
source_setup (GstElement * pipeline, GstElement * source, CustomData * data)
{
GstAudioInfo info;
GstCaps *audio_caps;
g_print ("Source has been created. Configuring.\n");
data->app_source = source;
/* 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 (source, "caps", audio_caps, "format", GST_FORMAT_TIME, 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);
}
int
tutorial_main (int argc, char *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 playbin element */
data.pipeline = gst_parse_launch ("playbin 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;
}
int
main (int argc, char *argv[])
{
#if defined(__APPLE__) && TARGET_OS_MAC && !TARGET_OS_IPHONE
return gst_macos_main ((GstMainFunc) tutorial_main, argc, argv, NULL);
#else
return tutorial_main (argc, argv);
#endif
}