gstreamer/examples/mixer/mixer.c

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/*
* mixer.c - stereo audio mixer - thomas@apestaart.org
* example based on helloworld
* demonstrates the adder plugin and the volume envelope plugin
* work in progress but do try it out
*
* Latest change : 28/08/2001
* trying to adapt to incsched
* delayed start for channels > 1
* now works by quickhacking the
* adder plugin to set
* GST_ELEMENT_COTHREAD_STOPPING
* Version : 0.5.1
*/
#include <stdlib.h>
#include <gst/gst.h>
#include "mixer.h"
#include <unistd.h>
/*#define WITH_BUG */
/*#define WITH_BUG2 */
/*#define DEBUG */
/*#define AUTOPLUG * define if you want autoplugging of input channels * */
/* function prototypes */
input_channel_t* create_input_channel (int id, char* location);
void destroy_input_channel (input_channel_t *pipe);
void env_register_cp (GstElement *volenv, double cp_time, double cp_level);
gboolean playing;
/* eos will be called when the src element has an end of stream */
void eos(GstElement *element)
{
g_print("have eos, quitting ?\n");
/* playing = FALSE; */
}
G_GNUC_UNUSED static GstCaps*
gst_play_type_find (GstBin *bin, GstElement *element)
{
GstElement *typefind;
GstElement *pipeline;
GstCaps *caps = NULL;
GST_DEBUG (0,"GstPipeline: typefind for element \"%s\"",
GST_ELEMENT_NAME(element));
pipeline = gst_pipeline_new ("autoplug_pipeline");
typefind = gst_element_factory_make ("typefind", "typefind");
g_return_val_if_fail (typefind != NULL, FALSE);
gst_pad_connect (gst_element_get_pad (element, "src"),
gst_element_get_pad (typefind, "sink"));
gst_bin_add (bin, typefind);
gst_bin_add (GST_BIN (pipeline), GST_ELEMENT (bin));
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* push a buffer... the have_type signal handler will set the found flag */
gst_bin_iterate (GST_BIN (pipeline));
gst_element_set_state (pipeline, GST_STATE_NULL);
caps = gst_pad_get_caps (gst_element_get_pad (element, "src"));
gst_pad_disconnect (gst_element_get_pad (element, "src"),
gst_element_get_pad (typefind, "sink"));
gst_bin_remove (bin, typefind);
gst_bin_remove (GST_BIN (pipeline), GST_ELEMENT (bin));
gst_object_unref (GST_OBJECT (typefind));
gst_object_unref (GST_OBJECT (pipeline));
return caps;
}
int main(int argc,char *argv[])
{
int i, j;
int num_channels;
char buffer[20];
GList *input_channels; /* structure holding all the input channels */
input_channel_t *channel_in;
GstElement *main_bin;
GstElement *adder;
GstElement *audiosink;
GstPad *pad; /* to request pads for the adder */
gst_init(&argc,&argv);
if (argc == 1) {
g_print("usage: %s <filename1> <filename2> <...>\n", argv[0]);
exit(-1);
}
num_channels = argc - 1;
/* set up output channel and main bin */
/* create adder */
adder = gst_element_factory_make ("adder", "adderel");
/* create an audio sink */
audiosink = gst_element_factory_make ("esdsink", "play_audio");
/* create main bin */
main_bin = gst_pipeline_new("bin");
/* connect adder and output to bin */
GST_INFO (0, "main: adding adder to bin");
gst_bin_add (GST_BIN(main_bin), adder);
GST_INFO (0, "main: adding audiosink to bin");
gst_bin_add (GST_BIN(main_bin), audiosink);
/* connect adder and audiosink */
gst_pad_connect(gst_element_get_pad(adder,"src"),
gst_element_get_pad(audiosink,"sink"));
/* start looping */
input_channels = NULL;
for (i = 1; i < argc; ++i)
{
printf ("Opening channel %d from file %s...\n", i, argv[i]);
channel_in = create_input_channel (i, argv[i]);
input_channels = g_list_append (input_channels, channel_in);
if (i > 1) gst_element_set_state (main_bin, GST_STATE_PAUSED);
gst_bin_add (GST_BIN(main_bin), channel_in->pipe);
/* request pads and connect to adder */
GST_INFO (0, "requesting pad\n");
pad = gst_element_get_request_pad (adder, "sink%d");
printf ("\tGot new adder sink pad %s\n", gst_pad_get_name (pad));
sprintf (buffer, "channel%d", i);
gst_pad_connect (gst_element_get_pad (channel_in->pipe, buffer), pad);
/* register a volume envelope */
printf ("\tregistering volume envelope...\n");
/*
* this is the volenv :
* each song gets a slot of 5 seconds, with a 5 second fadeout
* at the end of that, all audio streams play simultaneously
* at a level ensuring no distortion
* example for three songs :
* song1 : starts at full level, plays 5 seconds, faded out at 10 seconds,
* sleep until 25, fade to end level at 30
* song2 : starts silent, fades in at 5 seconds, full blast at 10 seconds,
* full level until 15, faded out at 20, sleep until 25, fade to end at 30
* song3 : starts muted, fades in from 15, full at 20, until 25, fade to end level
*/
if (i == 1)
{
/* first song gets special treatment for end style */
env_register_cp (channel_in->volenv, 0.0, 1.0);
}
else
{
env_register_cp (channel_in->volenv, 0.0 , 0.0000001); /* start muted */
env_register_cp (channel_in->volenv, i * 10.0 - 15.0, 0.0000001); /* start fade in */
env_register_cp (channel_in->volenv, i * 10.0 - 10.0, 1.0);
}
env_register_cp (channel_in->volenv, i * 10.0 - 5.0, 1.0); /* end of full level */
if (i != num_channels)
{
env_register_cp (channel_in->volenv, i * 10.0 , 0.0000001); /* fade to black */
env_register_cp (channel_in->volenv, num_channels * 10.0 - 5.0, 0.0000001); /* start fade in */
}
env_register_cp (channel_in->volenv, num_channels * 10.0 , 1.0 / num_channels); /* to end level */
#ifndef GST_DISABLE_LOADSAVE
gst_xml_write_file (GST_ELEMENT (main_bin), fopen ("mixer.xml", "w"));
#endif
/* start playing */
gst_element_set_state(main_bin, GST_STATE_PLAYING);
/* write out the schedule */
- some fixes to int2float making automake 1.5 happy (gst now requires automake1.5). It's still not perfect but it bui... Original commit message from CVS: - added playondemand plugin by Leif Morgan Johnson <lmjohns3@eos.ncsu.edu> - some fixes to int2float - aplied a patch from wrobell <wrobell@ite.pl> that is a first attempt at making automake 1.5 happy (gst now requires automake1.5). It's still not perfect but it builds. - Made the schedulers plugable. The default scheduler now lives inside a plugin. - Added a new mpeg1/2 parser/demuxer. - Fixed some compiler warnings in the core libs. - substantial work to GstThread (hopefully less race conditions). simplified the code in GstThread a bit. A state change can now also happen in the thread context. - reworked the state semantics of a bin. it'll now automatically get the highest state of its children. - the autoplugger now nests the threads so that a state change failure of one thread doesn't make its upstream thread lock. - GstQueue refuses to go to PLAYING if the sinkpad is not connected. This way the queue will not wedge in the _get lock. - GstQueue unlocks its mutexes when going to PAUSED. - make sure that when all elements in a bin/thread go to PAUSED, the bin is set to PAUSED too. - make a parent bin wait for its children to PAUSE before ending the iteration with FALSE (EOS) - Some changes to GstPlay to deal with EOS. - aplied the latest patch from Zeenix to gstrtp. end result: GstPlay doesn't crash on EOS and the pipeline is now shut down properly.
2001-12-04 22:12:50 +00:00
gst_scheduler_show(GST_ELEMENT_SCHED(main_bin));
playing = TRUE;
j = 0;
/*printf ("main: start iterating from 0"); */
while (playing && j < 100)
{
/* printf ("main: iterating %d\n", j); */
gst_bin_iterate(GST_BIN(main_bin));
/*fprintf(stderr,"after iterate()\n"); */
++j;
}
}
printf ("main: all the channels are open\n");
while (playing)
{
gst_bin_iterate(GST_BIN(main_bin));
/*fprintf(stderr,"after iterate()\n"); */
}
/* stop the bin */
gst_element_set_state(main_bin, GST_STATE_NULL);
while (input_channels)
{
destroy_input_channel (input_channels->data);
input_channels = g_list_next (input_channels);
}
g_list_free (input_channels);
gst_object_destroy(GST_OBJECT(audiosink));
gst_object_destroy(GST_OBJECT(main_bin));
exit(0);
}
input_channel_t*
create_input_channel (int id, char* location)
{
/* create an input channel, reading from location
* return a pointer to the channel
* return NULL if failed
*/
input_channel_t *channel;
char buffer[20]; /* hold the names */
/* GstAutoplug *autoplug;
GstCaps *srccaps; */
GstElement *new_element;
GstElement *decoder;
GST_DEBUG (0, "c_i_p : creating channel with id %d for file %s",
id, location);
/* allocate channel */
channel = (input_channel_t *) malloc (sizeof (input_channel_t));
if (channel == NULL)
{
printf ("create_input_channel : could not allocate memory for channel !\n");
return NULL;
}
/* create channel */
GST_DEBUG (0, "c_i_p : creating pipeline");
sprintf (buffer, "pipeline%d", id);
channel->pipe = gst_bin_new (buffer);
g_assert(channel->pipe != NULL);
/* create elements */
GST_DEBUG(0, "c_i_p : creating filesrc");
sprintf (buffer, "filesrc%d", id);
channel->filesrc = gst_element_factory_make ("filesrc", buffer);
g_assert(channel->filesrc != NULL);
GST_DEBUG(0, "c_i_p : setting location");
g_object_set(G_OBJECT(channel->filesrc),"location", location, NULL);
/* add filesrc to the bin before autoplug */
gst_bin_add(GST_BIN(channel->pipe), channel->filesrc);
/* connect signal to eos of filesrc */
g_signal_connect (G_OBJECT(channel->filesrc),"eos",
G_CALLBACK(eos),NULL);
#ifdef DEBUG
printf ("DEBUG : c_i_p : creating volume envelope\n");
#endif
sprintf (buffer, "volenv%d", id);
channel->volenv = gst_element_factory_make ("volenv", buffer);
g_assert(channel->volenv != NULL);
/* autoplug the pipe */
#ifdef DEBUG
printf ("DEBUG : c_i_p : getting srccaps\n");
#endif
#ifdef WITH_BUG
srccaps = gst_play_type_find (GST_BIN (channel->pipe), channel->filesrc);
#endif
#ifdef WITH_BUG2
{
GstElement *pipeline;
pipeline = gst_pipeline_new ("autoplug_pipeline");
gst_bin_add (GST_BIN (pipeline), channel->pipe);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_bin_remove (GST_BIN (pipeline), channel->pipe);
}
#endif
#ifdef AUTOPLUG
if (!srccaps) {
g_print ("could not autoplug, unknown media type...\n");
exit (-1);
}
#ifdef DEBUG
printf ("DEBUG : c_i_p : creating autoplug\n");
#endif
autoplug = gst_autoplug_factory_make ("static");
g_assert (autoplug != NULL);
#ifdef DEBUG
printf ("DEBUG : c_i_p : autoplugging\n");
#endif
new_element = gst_autoplug_to_caps (autoplug, srccaps,
gst_caps_new ("audio", "audio/raw", NULL), NULL);
if (!new_element) {
g_print ("could not autoplug, no suitable codecs found...\n");
exit (-1);
}
#else
new_element = gst_bin_new ("autoplug_bin");
/* static plug, use mad plugin and assume mp3 input */
printf ("using static plugging for input channel\n");
decoder = gst_element_factory_make ("mad", "mpg123");
if (!decoder)
{
fprintf (stderr, "Could not get a decoder element !\n");
exit (1);
}
gst_bin_add (GST_BIN (new_element), decoder);
gst_element_add_ghost_pad (new_element,
gst_element_get_pad (decoder, "sink"), "sink");
gst_element_add_ghost_pad (new_element,
gst_element_get_pad (decoder, "src"), "src_00");
#endif
#ifndef GST_DISABLE_LOADSAVE
gst_xml_write_file (GST_ELEMENT (new_element), fopen ("mixer.gst", "w"));
#endif
gst_bin_add (GST_BIN(channel->pipe), channel->volenv);
gst_bin_add (GST_BIN (channel->pipe), new_element);
gst_element_connect_pads (channel->filesrc, "src", new_element, "sink");
gst_element_connect_pads (new_element, "src_00", channel->volenv, "sink");
/* add a ghost pad */
sprintf (buffer, "channel%d", id);
gst_element_add_ghost_pad (channel->pipe,
gst_element_get_pad (channel->volenv, "src"), buffer);
#ifdef DEBUG
printf ("DEBUG : c_i_p : end function\n");
#endif
return channel;
}
void
destroy_input_channel (input_channel_t *channel)
{
/*
* destroy an input channel
*/
#ifdef DEBUG
printf ("DEBUG : d_i_p : start\n");
#endif
/* destroy elements */
gst_object_destroy (GST_OBJECT (channel->pipe));
free (channel);
}
void env_register_cp (GstElement *volenv, double cp_time, double cp_level)
{
char buffer[30];
sprintf (buffer, "%f:%f", cp_time, cp_level);
g_object_set(G_OBJECT(volenv), "controlpoint", buffer, NULL);
}