#include <gst/gst.h>
#include <gst/audio/audio.h>
#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 */
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;
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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 */
} CustomData;
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/* 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).
gint16 *raw;
gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
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 */
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);
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/* Generate some psychodelic waveforms */
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
raw = (gint16 *)map.data;
}
gst_buffer_unmap (buffer, &map);
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 */
static void start_feed (GstElement *source, guint size, CustomData *data) {
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 */
static void stop_feed (GstElement *source, CustomData *data) {
data->sourceid = 0;
}
}
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/* The appsink has received a buffer */
static void new_sample (GstElement *sink, CustomData *data) {
GstSample *sample;
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/* Retrieve the buffer */
g_signal_emit_by_name (sink, "pull-sample", &sample);
if (sample) {
gst_sample_unref (sample);
}
}
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/* 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;
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/* 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);
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g_main_loop_quit (data->main_loop);
}
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int main(int argc, char *argv[]) {
CustomData data;
GstPadTemplate *tee_src_pad_template;
GstAudioInfo info;
GstCaps *audio_caps;
GstBus *bus;
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/* Initialize cumstom data structure */
memset (&data, 0, sizeof (data));
data.b = 1; /* For waveform generation */
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.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) {
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 */
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);
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/* 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);
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/* 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,
+ 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_object_unref (data.pipeline);
return -1;
}
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/* Manually link the Tee, which has "Request" pads */
tee_src_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (data.tee), "src_%u");
tee_audio_pad = gst_element_request_pad (data.tee, tee_src_pad_template, NULL, NULL);
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);
g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, &data);
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_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);
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