#include <gst/gst_global.h>
#include <stdlib.h>
#include <gst/gst.h>
#include <gst/gstelement.h>
#include <string.h>
#define LOG_FILE "c:\\logs\\launch_logs.txt"
#include "std_log_result.h"
#define LOG_FILENAME_LINE __FILE__, __LINE__
static guint _bitrate = 128000;
static guint _channels = 1;
static guint _sample_rate = 8000;
static guint _aac_profile = 2; // default is LC
static guint _enable_logs = 1;
static guint _record_duration = 10000; // recording duration
#define REC_FILENAME_LEN 256
static char rec_filename[REC_FILENAME_LEN];
GstElement *pipeline;
#define ENABLE_LOGS
#ifdef ENABLE_LOGS
#define RET_GST_ERR_STR(var, level, str) \
if ( level == var )\
return str;
static inline const char* _gst_err_cat( GstDebugLevel level)
{
RET_GST_ERR_STR(level,GST_LEVEL_NONE,"");
RET_GST_ERR_STR(level,GST_LEVEL_ERROR,"E ");
RET_GST_ERR_STR(level,GST_LEVEL_WARNING,"W ");
RET_GST_ERR_STR(level,GST_LEVEL_INFO,"I ");
RET_GST_ERR_STR(level,GST_LEVEL_DEBUG,"D ");
RET_GST_ERR_STR(level,GST_LEVEL_LOG, "L ");
RET_GST_ERR_STR(level,GST_LEVEL_FIXME, "F ");
RET_GST_ERR_STR(level,GST_LEVEL_MEMDUMP, "M ");
return "";
}
static inline const char* _str_aac_profile()
{
if ( _aac_profile == 0) return "auto";
if ( _aac_profile == 2) return "lc";
if ( _aac_profile == 5) return "he";
return "unknown";
}
static FILE* log_fp = 0;
static void open_log_fp()
{
if (!log_fp)
{
snprintf(rec_filename, REC_FILENAME_LEN, "C://Data//gst_br%d_c%d_sr%d_%s.log", _bitrate, _channels, _sample_rate, _str_aac_profile());
log_fp = fopen(rec_filename, "w");
if (!log_fp)
return;
}
}
static void _gstLogFunction (GstDebugCategory *category,
GstDebugLevel level,
const gchar *file,
const gchar *function,
gint line,
GObject *object,
GstDebugMessage *message,
gpointer data)
{
// if ( (level != GST_LEVEL_ERROR) /*&& (level != GST_LEVEL_DEBUG)*/ && (level != GST_LEVEL_WARNING) )
// return;
open_log_fp();
fprintf(log_fp, "%s : %s \n", _gst_err_cat(level), gst_debug_message_get(message));
fflush(log_fp);
}
#endif // ENABLE_LOGS
// Local Functions
static gboolean
bus_call (GstBus *bus,
GstMessage *msg,
gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
open_log_fp();
fprintf(log_fp,"[msg] %s from %s\n", GST_MESSAGE_TYPE_NAME(msg), GST_MESSAGE_SRC_NAME (msg));
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (GST_OBJECT (pipeline));
g_main_loop_quit(loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *err;
gst_message_parse_error (msg, &err, &debug);
fprintf(log_fp, "[ERROR] %s\n", debug);
g_free (debug);
g_error_free (err);
g_main_loop_quit(loop);
break;
}
#if 0
case GST_MESSAGE_STATE_CHANGED:
{
GstState state;
// gst_element_get_state (GstElement * element,
// GstState * state,
// GstState * pending,
// GstClockTime timeout);
gst_element_get_state(GST_ELEMENT(pipeline),&state,NULL,-1);
if(state == GST_STATE_PLAYING)
{
}
}
break;
#endif
default:
break;
}
return TRUE;
}
static gboolean
quit_loop (gpointer data)
{
GST_DEBUG("quiting loop");
gst_element_send_event (pipeline, gst_event_new_eos ());
return TRUE;
}
static void parse_args(int argc, char** argv)
{
gint cur = 1;
while ( argv[cur] && cur < argc )
{
if( !strcmp(argv[cur],"-br") ) _bitrate = atoi(argv[cur+1]);
else if( !strcmp(argv[cur],"-c") ) _channels = atoi(argv[cur+1]);
else if( !strcmp(argv[cur],"-sr") ) _sample_rate = atoi(argv[cur+1]);
else if( !strcmp(argv[cur],"-p") ) _aac_profile = atoi(argv[cur+1]);
else if( !strcmp(argv[cur],"-l") ) _enable_logs = atoi(argv[cur+1]);
else if( !strcmp(argv[cur],"-d") ) _record_duration = atoi(argv[cur+1]);
cur+=2;
}
}
// Currently unused, TODO merge all recording usecases in this app.
#if 0
char gst_pipeline[4096];
static inline GstElement* __parse_wav_pipeline()
{
snprintf(gst_pipeline, 4096, "devsoundsrc ! audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)%d, channels=(int)%d, endianness=(int)1234 ! wavenc ! filesink location=C://Data//wav_c%d_sr%d.wav",
_sample_rate, _channels , _channels, _sample_rate );
pipeline = gst_parse_launch( gst_pipeline,0);
return pipeline;
}
static inline GstElement* __get_wav_pipeline()
{
GstCaps* caps;
GstElement *devsoundsrc,*filesink,*wavenc;
GError *error = NULL;
pipeline = gst_pipeline_new ("pipeline");
devsoundsrc = gst_element_factory_make ("devsoundsrc", "devsoundsrc");
wavenc = gst_element_factory_make ("wavenc", "wavenc");
filesink = gst_element_factory_make ("filesink", "filesink");
snprintf(rec_filename, REC_FILENAME_LEN, "C:\\\\data\\\\wav_c%d_sr%d.wav", _channels, _sample_rate);
g_object_set (G_OBJECT (filesink), "location", rec_filename, NULL);
GST_DEBUG("filtered linking...");
caps = gst_caps_new_simple ("audio/x-raw-int",
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"signed",G_TYPE_BOOLEAN, TRUE,
"endianness",G_TYPE_INT, G_BYTE_ORDER,
"rate", G_TYPE_INT, _sample_rate,
"channels", G_TYPE_INT, _channels, NULL);
gst_bin_add_many (GST_BIN (pipeline), devsoundsrc, wavenc, filesink, NULL);
gst_element_link_filtered (devsoundsrc, wavenc, caps);
gst_element_link (wavenc, filesink);
gst_caps_unref (caps);
return pipeline;
}
#endif
static inline GstElement* __get_aac_pipeline()
{
GstCaps* caps;
GstElement *devsoundsrc,*filesink,*nokiaaacenc,*mp4mux;
GstPad *qtsinkpad,*aacencsrcpad;
pipeline = gst_pipeline_new ("pipeline");
devsoundsrc = gst_element_factory_make ("devsoundsrc", "devsoundsrc");
nokiaaacenc = gst_element_factory_make ("nokiaaacenc", "nokiaaacenc");
mp4mux = gst_element_factory_make ("mp4mux", "mp4mux");
filesink = gst_element_factory_make ("filesink", "filesink");
snprintf(rec_filename, REC_FILENAME_LEN, "C:\\\\data\\\\rec-aac_br%d_c%d_sr%d_%s.mp4", _bitrate, _channels, _sample_rate,
_str_aac_profile());
g_object_set (G_OBJECT (filesink), "location", rec_filename, NULL);
//GST_DEBUG("set bitrate on aacenc");
g_object_set (G_OBJECT (nokiaaacenc), "bitrate", _bitrate, NULL);
//if ( _aac_profile )
g_object_set (G_OBJECT (nokiaaacenc), "profile", _aac_profile, NULL);
GST_DEBUG("filtered linking...");
caps = gst_caps_new_simple ("audio/x-raw-int",
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"signed",G_TYPE_BOOLEAN, TRUE,
"endianness",G_TYPE_INT, G_BYTE_ORDER,
"rate", G_TYPE_INT, _sample_rate,
"channels", G_TYPE_INT, _channels, NULL);
gst_bin_add_many (GST_BIN (pipeline), devsoundsrc, nokiaaacenc, mp4mux, filesink, NULL);
gst_element_link_filtered (devsoundsrc, nokiaaacenc, caps);
qtsinkpad = gst_element_get_request_pad( mp4mux, "audio_%d");
aacencsrcpad = gst_element_get_pad( nokiaaacenc, "src");
if (gst_pad_link (aacencsrcpad,qtsinkpad) != GST_PAD_LINK_OK) {
GST_ERROR("gst_pad_link (aacencsrcpad,qtsinkpad) failed");
return NULL;
}
gst_element_link (mp4mux, filesink);
gst_caps_unref (caps);
return pipeline;
}
int main (int argc, char *argv[])
{
GMainLoop *loop;
#ifdef ENABLE_LOGS
if ( _enable_logs )
setenv("GST_DEBUG","2",1);
#endif // ENABLE_LOGS
gst_init (NULL, NULL);
parse_args(argc, argv);
#ifdef ENABLE_LOGS
if ( _enable_logs )
gst_debug_add_log_function( _gstLogFunction, 0);
#endif // ENABLE_LOGS
GST_DEBUG("args : br %d chans %d sr %d ", _bitrate, _channels, _sample_rate);
loop = g_main_loop_new (NULL, FALSE);
//pipeline = __get_wav_pipeline();
//pipeline = __parse_wav_pipeline();
pipeline = __get_aac_pipeline();
/* start playing */
gst_bus_add_watch (gst_pipeline_get_bus (GST_PIPELINE (pipeline)), bus_call, loop);
// watchdog timer
//g_timeout_add (_record_duration * 1.5, quit_program, 0);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_timeout_add (_record_duration, quit_loop, loop);
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (GST_OBJECT (pipeline));
#ifdef ENABLE_LOGS
if ( _enable_logs )
fclose(log_fp);
#endif // ENABLE_LOGS
}