gstreamer_core/tsrc/examples/aac_record/src/aacrecord.c
branchRCL_3
changeset 30 7e817e7e631c
parent 29 567bb019e3e3
--- a/gstreamer_core/tsrc/examples/aac_record/src/aacrecord.c	Tue Aug 31 15:30:33 2010 +0300
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,313 +0,0 @@
-
-#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
-}
-
-