gstreamer/examples/tutorials/basic-tutorial-8.c

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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <string.h>
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#define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */
#define SAMPLE_RATE 44100 /* Samples per second we are sending */
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/* Structure to contain all our information, so we can pass it to callbacks */
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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;
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GstElement *app_queue, *app_sink;
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guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */
gfloat a, b, c, d; /* For waveform generation */
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guint sourceid; /* To control the GSource */
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GMainLoop *main_loop; /* GLib's Main Loop */
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} CustomData;
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/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
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* The idle handler is added to the mainloop when appsrc requests us to start sending data (need-data signal)
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* and is removed when appsrc has enough data (enough-data signal).
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*/
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static gboolean
push_data (CustomData * data)
{
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GstBuffer *buffer;
GstFlowReturn ret;
int i;
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GstMapInfo map;
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gint16 *raw;
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gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
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gfloat freq;
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/* Create a new empty buffer */
buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
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/* Set its timestamp and duration */
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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);
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/* Generate some psychodelic waveforms */
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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raw = (gint16 *) map.data;
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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;
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raw[i] = (gint16) (500 * data->a);
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}
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gst_buffer_unmap (buffer, &map);
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data->num_samples += num_samples;
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/* Push the buffer into the appsrc */
g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
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/* Free the buffer now that we are done with it */
gst_buffer_unref (buffer);
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if (ret != GST_FLOW_OK) {
/* We got some error, stop sending data */
return FALSE;
}
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return TRUE;
}
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/* This signal callback triggers when appsrc needs data. Here, we add an idle handler
* to the mainloop to start pushing data into the appsrc */
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static void
start_feed (GstElement * source, guint size, CustomData * data)
{
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if (data->sourceid == 0) {
g_print ("Start feeding\n");
data->sourceid = g_idle_add ((GSourceFunc) push_data, data);
}
}
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/* This callback triggers when appsrc has enough data and we can stop sending.
* We remove the idle handler from the mainloop */
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static void
stop_feed (GstElement * source, CustomData * data)
{
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if (data->sourceid != 0) {
g_print ("Stop feeding\n");
g_source_remove (data->sourceid);
data->sourceid = 0;
}
}
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/* The appsink has received a buffer */
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static GstFlowReturn
new_sample (GstElement * sink, CustomData * data)
{
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GstSample *sample;
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/* Retrieve the buffer */
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g_signal_emit_by_name (sink, "pull-sample", &sample);
if (sample) {
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/* The only thing we do in this example is print a * to indicate a received buffer */
g_print ("*");
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gst_sample_unref (sample);
return GST_FLOW_OK;
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}
return GST_FLOW_ERROR;
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}
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/* This function is called when an error message is posted on the bus */
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static void
error_cb (GstBus * bus, GstMessage * msg, CustomData * data)
{
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GError *err;
gchar *debug_info;
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/* Print error details on the screen */
gst_message_parse_error (msg, &err, &debug_info);
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g_printerr ("Error received from element %s: %s\n",
GST_OBJECT_NAME (msg->src), err->message);
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g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
g_clear_error (&err);
g_free (debug_info);
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g_main_loop_quit (data->main_loop);
}
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int
main (int argc, char *argv[])
{
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CustomData data;
GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
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GstAudioInfo info;
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GstCaps *audio_caps;
GstBus *bus;
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/* Initialize cumstom data structure */
memset (&data, 0, sizeof (data));
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data.b = 1; /* For waveform generation */
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data.d = 1;
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/* Initialize GStreamer */
gst_init (&argc, &argv);
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/* 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");
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data.audio_convert1 =
gst_element_factory_make ("audioconvert", "audio_convert1");
data.audio_resample =
gst_element_factory_make ("audioresample", "audio_resample");
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data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
data.video_queue = gst_element_factory_make ("queue", "video_queue");
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data.audio_convert2 =
gst_element_factory_make ("audioconvert", "audio_convert2");
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data.visual = gst_element_factory_make ("wavescope", "visual");
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data.video_convert =
gst_element_factory_make ("videoconvert", "video_convert");
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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");
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/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
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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) {
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g_printerr ("Not all elements could be created.\n");
return -1;
}
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/* Configure wavescope */
g_object_set (data.visual, "shader", 0, "style", 0, NULL);
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/* Configure appsrc */
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
audio_caps = gst_audio_info_to_caps (&info);
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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);
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/* Configure appsink */
g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
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g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample),
&data);
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gst_caps_unref (audio_caps);
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/* Link all elements that can be automatically linked because they have "Always" pads */
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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) {
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g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
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/* Manually link the Tee, which has "Request" pads */
tee_audio_pad = gst_element_request_pad_simple (data.tee, "src_%u");
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g_print ("Obtained request pad %s for audio branch.\n",
gst_pad_get_name (tee_audio_pad));
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queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink");
tee_video_pad = gst_element_request_pad_simple (data.tee, "src_%u");
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g_print ("Obtained request pad %s for video branch.\n",
gst_pad_get_name (tee_video_pad));
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queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink");
tee_app_pad = gst_element_request_pad_simple (data.tee, "src_%u");
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g_print ("Obtained request pad %s for app branch.\n",
gst_pad_get_name (tee_app_pad));
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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);
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/* 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);
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g_signal_connect (G_OBJECT (bus), "message::error", (GCallback) error_cb,
&data);
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gst_object_unref (bus);
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/* Start playing the pipeline */
gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
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/* 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);
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/* 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);
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/* Free resources */
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}