1 

     2 #include <gst/gst_global.h>

     3 #include <stdlib.h>

     4 #include <gst/gst.h>

     5 #include <gst/gstelement.h>

     6 #include <string.h>

     7 #define LOG_FILE "c:\\logs\\amr_record_logs.txt" 

     8 #include "std_log_result.h" 

     9 #include <gst/interfaces/gstspeechencoderconfig.h>

    10 #define LOG_FILENAME_LINE __FILE__, __LINE__

    11 #define LOG_FILENAME_LEN 256

    12 #define _DEBUG 1

    13 

    14 static char log_filename[LOG_FILENAME_LEN];

    15 

    16 static guint bitrate = 0; 

    17 static guint record_duration = 10000;

    18 static guint _enable_logs = 0;

    19 

    20 static GstSpeechEncoderConfigIntfc* iface;

    21 

    22 

    23 #define ENABLE_LOGS

    24 

    25 #ifdef ENABLE_LOGS

    26 #define RET_GST_ERR_STR(var, level, str) \

    27     if ( level == var )\

    28 return str;

    29 

    30 static inline const char* _gst_err_cat( GstDebugLevel level)

    31 {

    32 

    33     RET_GST_ERR_STR(level,GST_LEVEL_NONE,"");

    34     RET_GST_ERR_STR(level,GST_LEVEL_ERROR,"E ");

    35     RET_GST_ERR_STR(level,GST_LEVEL_WARNING,"W ");

    36     RET_GST_ERR_STR(level,GST_LEVEL_INFO,"I ");

    37     RET_GST_ERR_STR(level,GST_LEVEL_DEBUG,"D ");

    38     RET_GST_ERR_STR(level,GST_LEVEL_LOG, "L ");

    39     RET_GST_ERR_STR(level,GST_LEVEL_FIXME, "F ");

    40     RET_GST_ERR_STR(level,GST_LEVEL_MEMDUMP, "M ");

    41     return "";

    42 }

    43 

    44 static FILE* log_fp = 0;

    45 

    46 static void open_log_fp()

    47 {

    48     if (!log_fp)

    49     {

    50         snprintf(log_filename, LOG_FILENAME_LEN, "C:\\logs\\testframework\\gst_amr_br%d.log", bitrate);

    51 

    52         log_fp = fopen(log_filename, "w");

    53         if (!log_fp)

    54             return;

    55     }

    56 }

    57 

    58 

    59 static void _gstLogFunction (GstDebugCategory *category,

    60         GstDebugLevel level,

    61         const gchar *file,

    62         const gchar *function,

    63         gint line,

    64         GObject *object,

    65         GstDebugMessage *message,

    66         gpointer data)

    67 {

    68     open_log_fp();

    69 

    70     fprintf(log_fp, "%s : %s \n", _gst_err_cat(level), gst_debug_message_get(message));

    71     fflush(log_fp);

    72 

    73 }

    74 #endif // ENABLE_LOGS

    75 

    76 static void parse_args(int argc, char** argv)

    77 {

    78 

    79     gint cur = 1;

    80     while ( argv[cur] && cur < argc )

    81     {

    82         GST_WARNING("br:%d, record_duration:%d \n",bitrate,record_duration);

    83         if( !strcmp(argv[cur],"-br") ) bitrate = atoi(argv[cur+1]);

    84         else if( !strcmp(argv[cur],"-l") ) _enable_logs = atoi(argv[cur+1]);

    85         else if( !strcmp(argv[cur],"-d") ) record_duration = atoi(argv[cur+1]);

    86 

    87         cur+=2;

    88     }

    89 }

    90 

    91 

    92 GstElement *pipeline, *source, *amrmux,*sink;

    93 GstBus *bus;

    94 GMainLoop *loop;

    95 

    96 static gboolean

    97 bus_call (GstBus     *bus,

    98         GstMessage *msg,

    99         gpointer    data)

   100     {

   101 #ifdef ENABLE_LOGS

   102     if(_enable_logs)

   103         {

   104         open_log_fp();

   105         fprintf(log_fp,"[msg] %s from %s\n", GST_MESSAGE_TYPE_NAME(msg), GST_MESSAGE_SRC_NAME (msg));

   106         }

   107 #endif

   108 

   109     switch (GST_MESSAGE_TYPE (msg)) {

   110         case GST_MESSAGE_EOS:

   111             gst_element_set_state (pipeline, GST_STATE_NULL);

   112             g_main_loop_quit(loop);

   113             gst_object_unref (GST_OBJECT (pipeline));

   114             break;

   115             

   116         case GST_MESSAGE_ERROR: {

   117         gchar *debug;

   118         GError *err;

   119         gst_message_parse_error (msg, &err, &debug);

   120         g_free (debug);

   121         g_print ("Error: %s\n", err->message);

   122         g_error_free (err);

   123         break;

   124         }

   125         default:

   126             break;

   127     }

   128 

   129     return TRUE;

   130     }

   131 

   132 static gboolean

   133 quit_loop (gpointer data)

   134 {

   135     GST_WARNING("Sending EOS\n");

   136     gst_element_send_event (pipeline, gst_event_new_eos ());

   137     return TRUE;

   138 }

   139 

   140 int main (int argc, char *argv[])

   141 {

   142     char filename[1024];

   143     GstCaps* caps;

   144 #ifdef ENABLE_LOGS

   145     if ( _enable_logs )

   146     setenv("GST_DEBUG","2",1);

   147 #endif // ENABLE_LOGS

   148     

   149     parse_args(argc,argv);

   150     

   151     gst_init (&argc, &argv);

   152 #ifdef ENABLE_LOGS

   153     if ( _enable_logs )

   154     gst_debug_add_log_function( _gstLogFunction, 0);

   155 #endif // ENABLE_LOGS

   156     

   157     loop = g_main_loop_new (NULL, FALSE);

   158  

   159     /* create elements */

   160     pipeline = gst_pipeline_new ("audio-player");

   161     source = gst_element_factory_make ("devsoundsrc", "audio-source");

   162     amrmux = gst_element_factory_make ("amrmux", "amrmux");

   163     sink = gst_element_factory_make ("filesink", "sink");    

   164     

   165     if (!pipeline || !source || !amrmux || !sink) {

   166         g_print ("One element could not be created\n");

   167         GST_ERROR("One element could not be created\n");

   168         return -1;

   169         }

   170     if(bitrate > 0)

   171         {

   172         snprintf(filename, 1024, "C:\\data\\amr_record_%d.amr", bitrate );

   173         }

   174     else

   175         {

   176         snprintf(filename, 1024, "C:\\data\\amr_record.amr" );

   177         }

   178     

   179     /* set filename property on the file source. Also add a message  handler. */

   180     g_object_set (G_OBJECT (sink), "location", filename, NULL);

   181             /* put all elements in a bin */

   182     gst_bin_add_many (GST_BIN (pipeline),source, amrmux,sink, NULL);

   183 

   184     caps = gst_caps_new_simple ("audio/amr",

   185               "width", G_TYPE_INT, 8,

   186               "depth", G_TYPE_INT, 8,

   187               "signed",G_TYPE_BOOLEAN, TRUE,

   188               "endianness",G_TYPE_INT, G_BYTE_ORDER,

   189               "rate", G_TYPE_INT, 8000,

   190               "channels", G_TYPE_INT, 1, NULL);  

   191     

   192     if(!gst_element_link_filtered (source, amrmux, caps))

   193         {

   194         GST_ERROR("Linking source, amrmux failed\n");

   195         return -1;

   196         }

   197     

   198     if(!gst_element_link (amrmux, sink))

   199         {

   200         GST_ERROR("Linking amrmux,sink failed\n");

   201         return -1;

   202         }

   203 

   204     gst_bus_add_watch (gst_pipeline_get_bus (GST_PIPELINE (pipeline)), bus_call, loop);

   205     iface = GST_SPEECH_ENCODER_CONFIG_GET_IFACE(source);

   206     if(!iface)

   207         {

   208         GST_ERROR("Speech Encoder Interface NULL \n");

   209         return -1;

   210         }

   211     if (bitrate > 0)

   212         {

   213         GST_WARNING("Setting bitrate %d \n",bitrate);

   214         iface->SetBitrate(bitrate);

   215         }

   216     

   217             /* Now set to playing and iterate. */

   218     g_print ("Setting to PLAYING\n");

   219     GST_WARNING("Setting to Playing \n");

   220     gst_element_set_state (pipeline, GST_STATE_PLAYING);

   221     iface->GetBitrate(&bitrate);

   222     

   223     g_timeout_add (record_duration, quit_loop, loop);

   224     

   225     g_main_loop_run (loop);

   226             /* clean up nicely */

   227     GST_WARNING("Returned, stopping playback\n");

   228     

   229     gst_element_set_state (pipeline, GST_STATE_NULL);

   230     gst_object_unref (GST_OBJECT (pipeline));

   231     

   232 #ifdef ENABLE_LOGS

   233     if ( _enable_logs )

   234     fclose(log_fp);

   235 #endif // ENABLE_LOGS

   236     

   237     g_print ("completed playing audio\n");

   238     return 0;

   239 }

   240