问题
I'm writing a C application that uses gstreamer to record audio from a microphone. I want to be able to parse that audio and show a visualization of that audio.
I have the following code:
#include <gst/gst.h>
#include <glib.h>
static gboolean
bus_call (GstBus *bus,
GstMessage *msg,
gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
void create_loop()
{
GMainLoop *loop;
GstElement *pipeline, *source, *sink;
GstBus *bus;
guint bus_watch_id;
/* Initialisation */
loop = g_main_loop_new (NULL, FALSE);
/* Create gstreamer elements */
pipeline = gst_pipeline_new ("audio-player");
source = gst_element_factory_make ("alsasrc", "alsa-source");
sink = gst_element_factory_make ("autoaudiosink", "audio-output");
if (!pipeline || !source || !sink) {
g_printerr ("One element could not be created. Exiting.\n");
return;
}
g_object_set (G_OBJECT(source),"device","hw:3,0",NULL);
/* we add a message handler */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
gst_bin_add_many (GST_BIN (pipeline),
source, sink, NULL);
gst_element_link (source, sink);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* Iterate */
g_print ("Running...\n");
g_main_loop_run (loop);
/* Out of the main loop, clean up nicely */
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (pipeline));
g_source_remove (bus_watch_id);
g_main_loop_unref (loop);
}
int main(int argc, char** argv) {
gst_init(&argc,&argv);
create_loop();
return 0;
}
as you can see in my code i create an alsasrc and autoaudiosink. I tested and I can listen properly to that device.
how can I write something in the middle that parses the data in order to create visualization.
any information regarding the issue would be greatly appreciated.
回答1:
appsink element allows you to get data from the pipeline.
You have three options:
- pull: start pipeline, then get samples with gst_app_sink_pull_sample
- push with signals: enable
new-samplesignal emission with gst_app_src_set_emit_signals, and subscribe to that signals - push with callbacks: register callback on
new-samplewith gst_app_sink_set_callbacks
I think that the third option is the easiest and more efficient.
So, just replace autoaudiosink with appsink, register callback and handle your data in it.
You can read a bit about appsrc and appsink in the manual.
回答2:
You need to create a "plugin" or a pad for your gsteamer application. There is a complete description on how to create a pad : http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/chapter-building-boiler.html#section-boiler-source
In most case you need to create a class using gobject. https://developer.gnome.org/gobject/stable/
Here's how : https://developer.gnome.org/gobject/stable/chapter-gobject.html
And you need at least :
/* Definition of structure storing data for this element. */
typedef struct _GstMyFilter {
GstElement element;
GstPad *sinkpad, *srcpad;
gboolean silent;
} GstMyFilter;
static GstStaticPadTemplate sink_factory =
GST_STATIC_PAD_TEMPLATE (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("ANY")
);
//... likewise for your src
static void
gst_my_filter_class_init (GstMyFilterClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
//...
gst_element_class_set_static_metadata (element_klass,
"An example plugin",
"Example/FirstExample",
"Shows the basic structure of a plugin",
"your name <your.name@your.isp>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
}
Your also need description for your plugin
static gboolean
plugin_init (GstPlugin *plugin)
{
return gst_element_register (plugin, "my_filter",
GST_RANK_NONE,
GST_TYPE_MY_FILTER);
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
my_filter,
"My filter plugin",
plugin_init,
VERSION,
"LGPL",
"GStreamer",
"http://gstreamer.net/"
)
static void
gst_my_filter_init (GstMyFilter *filter)
{
/* pad through which data comes in to the element */
filter->sinkpad = gst_pad_new_from_static_template (
&sink_template, "sink");
/* pads are configured here with gst_pad_set_*_function () */
gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
/* pad through which data goes out of the element */
filter->srcpad = gst_pad_new_from_static_template (
&src_template, "src");
/* pads are configured here with gst_pad_set_*_function () */
gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
/* configure chain function on the pad before adding
* the pad to the element */
gst_pad_set_chain_function (filter->sinkpad,
gst_my_filter_chain);
/* properties initial value */
filter->silent = FALSE;
}
and your the magic is operated the chain
static GstFlowReturn
gst_my_filter_chain (GstPad *pad,
GstObject *parent,
GstBuffer *buf)
{
GstMyFilter *filter = GST_MY_FILTER (parent);
if (!filter->silent)
g_print ("Have data of size %" G_GSIZE_FORMAT" bytes!\n",
gst_buffer_get_size (buf));
return gst_pad_push (filter->srcpad, buf);
}
static gboolean
gst_my_filter_sink_event (GstPad *pad,
GstObject *parent,
GstEvent *event)
{
GstMyFilter *filter = GST_MY_FILTER (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
/* we should handle the format here */
break;
case GST_EVENT_EOS:
/* end-of-stream, we should close down all stream leftovers here */
gst_my_filter_stop_processing (filter);
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static GstFlowReturn
gst_my_filter_chain (GstPad *pad,
GstObject *parent,
GstBuffer *buf)
{
GstMyFilter *filter = GST_MY_FILTER (parent);
GstBuffer *outbuf;
outbuf = gst_my_filter_process_data (filter, buf);
gst_buffer_unref (buf);
if (!outbuf) {
/* something went wrong - signal an error */
GST_ELEMENT_ERROR (GST_ELEMENT (filter), STREAM, FAILED, (NULL), (NULL));
return GST_FLOW_ERROR;
}
return gst_pad_push (filter->srcpad, outbuf);
}
use case :
//... your code and elements init...
//Creating your filter
filter = gst_element_factory_make ("my_filter", "my_filter");
//Adding it
gst_bin_add_many (GST_BIN (pipeline), source, filter, sink, NULL);
//Linking it
if (!gst_element_link_many (source, filter, sink, NULL)) {
g_print ("Failed to link one or more elements!\n");
return -1;
}
//... here the rest of your code
Bonus * There is a simple example : http://siilo.dyndns.org/wiki/index.php/Creating_a_Plugin_template
来源：https://stackoverflow.com/questions/22620617/how-can-i-parse-audio-raw-data-recorder-with-gstreamer