the general goal is that i want to play an audio track on my RPi with aplay ("aplay example.mp3") and the output audio gets looped back into an gstreamer program. This program then does a spectrum analysis.
I got the spectrum analysis already working on a static file with this code as source:
data.source = gst_element_factory_make ("uridecodebin", "source");
g_object_set (data.source, "uri", "file:///home/pi/example.mp3", NULL);
ofc i want to use the overall output from my RPi as a source for the program but i dont know how. I know i need to loopback the audio from the output to the input and i found that snd-aloop looks promising. Problem is i still dont know how to use it. I tried to do:
data.source = gst_element_factory_make ("alsasrc", "source");
g_object_set(data.source, "device", XXX ,NULL);
where XXX =
- "alsa_output.platform-snd_aloop.0.analog-stereo.monitor"
- "hw:1"
- "hw:0"
Error -> Trying to dispose element sink, but it is in READY instead of the NULL state. You need to explicitly set Elements to the NULL state before dropping the final reference [...]
Bonus question: Is it possible to pipe audio into a gstreamer program? something like: "aplay example.mp3 > gstreamerCprogram".
Here is the code:
#include <gst/gst.h>
#define AUDIOFREQ 32000
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *convert;
GstElement *sink;
} CustomData;
/* Handler for the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
static gboolean message_handler (GstBus *bus, GstMessage *message, gpointer data){
if(message->type == GST_MESSAGE_EOS){
g_printerr("EOS\n");
}
if(message->type == GST_MESSAGE_ELEMENT){
const GstStructure *s = gst_message_get_structure (message);
const gchar *name = gst_structure_get_name(s);
if(strcmp(name, "spectrum") == 0){
const GValue *magnitudes;
gdouble freq;
magnitudes = gst_structure_get_value (s,"magnitude");
int i = 0;
for(i = 0; i < 20; ++i){
freq = (gdouble)((32000/2) * i + 32000 / 4 / 20);
if(freq > 10000){
g_printerr("%f\n",freq);
}else{
g_printerr("|");
}
}
}
}
}
int main(int argc, char *argv[]) {
CustomData data;
GstCaps *caps;
GstElement *spectrum;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
//____________________________HERE IS THE PROBLEM________________________
//data.source = gst_element_factory_make ("uridecodebin", "source");
//g_object_set (data.source, "uri", "file:///home/pi/example.mp3", NULL);
data.source = gst_element_factory_make ("alsasrc", "source");
g_object_set(data.source, "device", "alsa_output.platform-snd_aloop.0.analog-stereo.monitor",NULL);
//____________________________HERE ENDS THE PROBLEM________________________
data.convert = gst_element_factory_make ("audioconvert", "convert");
data.sink = gst_element_factory_make ("autoaudiosink", "sink");
spectrum = gst_element_factory_make ("spectrum", "spectrum");
caps = gst_caps_new_simple ("audio/x-raw", "rate",G_TYPE_INT, AUDIOFREQ, NULL);
//SET SOME VARIABLES ON SPECTRUM
g_object_set (G_OBJECT (spectrum), "bands", 20, "post-messages", TRUE, "message-phase", TRUE, NULL);
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.convert || !data.sink || !caps || !spectrum) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Build the pipeline. Note that we are NOT linking the source at this
* point. We will do it later. */
gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert , spectrum,data.sink, NULL);
if (!gst_element_link_many (data.convert, spectrum, data.sink, NULL)) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Connect to the pad-added signal */
g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
GMainLoop *loop;
/* Listen to the bus */
bus = gst_element_get_bus (data.pipeline);
gst_bus_add_watch(bus, message_handler, NULL);
loop = g_main_loop_new (NULL,FALSE);
g_main_loop_run(loop);
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
/* This function will be called by the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
GstPad *sink_pad = gst_element_get_static_pad (data->convert, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" We are already linked. Ignoring.\n");
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_query_caps (new_pad, NULL);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) {
g_print (" It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type);
goto exit;
}
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
