--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gstreamer_core/tsrc/examples/aac_record/src/aacrecord.c Wed Aug 18 10:04:13 2010 +0300
@@ -0,0 +1,313 @@
+
+#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
+}
+
+