#include <gst/gst_global.h>
#include <stdlib.h>
#include <gst/gst.h>
#include <gst/gstelement.h>
#include <string.h>
#define LOG_FILE "c:\\logs\\amr_record_logs.txt"
#include "std_log_result.h"
#include <gst/interfaces/gstspeechencoderconfig.h>
#define LOG_FILENAME_LINE __FILE__, __LINE__
#define LOG_FILENAME_LEN 256
#define _DEBUG 1
static char log_filename[LOG_FILENAME_LEN];
static guint bitrate = 0;
static guint record_duration = 10000;
static guint _enable_logs = 0;
static GstSpeechEncoderConfigIntfc* iface;
#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 FILE* log_fp = 0;
static void open_log_fp()
{
if (!log_fp)
{
snprintf(log_filename, LOG_FILENAME_LEN, "C:\\logs\\testframework\\gst_amr_br%d.log", bitrate);
log_fp = fopen(log_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)
{
open_log_fp();
fprintf(log_fp, "%s : %s \n", _gst_err_cat(level), gst_debug_message_get(message));
fflush(log_fp);
}
#endif // ENABLE_LOGS
static void parse_args(int argc, char** argv)
{
gint cur = 1;
while ( argv[cur] && cur < argc )
{
GST_WARNING("br:%d, record_duration:%d \n",bitrate,record_duration);
if( !strcmp(argv[cur],"-br") ) bitrate = 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;
}
}
GstElement *pipeline, *source, *amrmux,*sink;
GstBus *bus;
GMainLoop *loop;
static gboolean
bus_call (GstBus *bus,
GstMessage *msg,
gpointer data)
{
#ifdef ENABLE_LOGS
if(_enable_logs)
{
open_log_fp();
fprintf(log_fp,"[msg] %s from %s\n", GST_MESSAGE_TYPE_NAME(msg), GST_MESSAGE_SRC_NAME (msg));
}
#endif
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_quit(loop);
gst_object_unref (GST_OBJECT (pipeline));
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *err;
gst_message_parse_error (msg, &err, &debug);
g_free (debug);
g_print ("Error: %s\n", err->message);
g_error_free (err);
break;
}
default:
break;
}
return TRUE;
}
static gboolean
quit_loop (gpointer data)
{
GST_WARNING("Sending EOS\n");
gst_element_send_event (pipeline, gst_event_new_eos ());
return TRUE;
}
int main (int argc, char *argv[])
{
char filename[1024];
GstCaps* caps;
#ifdef ENABLE_LOGS
if ( _enable_logs )
setenv("GST_DEBUG","2",1);
#endif // ENABLE_LOGS
parse_args(argc,argv);
gst_init (&argc, &argv);
#ifdef ENABLE_LOGS
if ( _enable_logs )
gst_debug_add_log_function( _gstLogFunction, 0);
#endif // ENABLE_LOGS
loop = g_main_loop_new (NULL, FALSE);
/* create elements */
pipeline = gst_pipeline_new ("audio-player");
source = gst_element_factory_make ("devsoundsrc", "audio-source");
amrmux = gst_element_factory_make ("amrmux", "amrmux");
sink = gst_element_factory_make ("filesink", "sink");
if (!pipeline || !source || !amrmux || !sink) {
g_print ("One element could not be created\n");
GST_ERROR("One element could not be created\n");
return -1;
}
if(bitrate > 0)
{
snprintf(filename, 1024, "C:\\data\\amr_record_%d.amr", bitrate );
}
else
{
snprintf(filename, 1024, "C:\\data\\amr_record.amr" );
}
/* set filename property on the file source. Also add a message handler. */
g_object_set (G_OBJECT (sink), "location", filename, NULL);
/* put all elements in a bin */
gst_bin_add_many (GST_BIN (pipeline),source, amrmux,sink, NULL);
caps = gst_caps_new_simple ("audio/amr",
"width", G_TYPE_INT, 8,
"depth", G_TYPE_INT, 8,
"signed",G_TYPE_BOOLEAN, TRUE,
"endianness",G_TYPE_INT, G_BYTE_ORDER,
"rate", G_TYPE_INT, 8000,
"channels", G_TYPE_INT, 1, NULL);
if(!gst_element_link_filtered (source, amrmux, caps))
{
GST_ERROR("Linking source, amrmux failed\n");
return -1;
}
if(!gst_element_link (amrmux, sink))
{
GST_ERROR("Linking amrmux,sink failed\n");
return -1;
}
gst_bus_add_watch (gst_pipeline_get_bus (GST_PIPELINE (pipeline)), bus_call, loop);
iface = GST_SPEECH_ENCODER_CONFIG_GET_IFACE(source);
if(!iface)
{
GST_ERROR("Speech Encoder Interface NULL \n");
return -1;
}
if (bitrate > 0)
{
GST_WARNING("Setting bitrate %d \n",bitrate);
iface->SetBitrate(bitrate);
}
/* Now set to playing and iterate. */
g_print ("Setting to PLAYING\n");
GST_WARNING("Setting to Playing \n");
gst_element_set_state (pipeline, GST_STATE_PLAYING);
iface->GetBitrate(&bitrate);
g_timeout_add (record_duration, quit_loop, loop);
g_main_loop_run (loop);
/* clean up nicely */
GST_WARNING("Returned, stopping playback\n");
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
g_print ("completed playing audio\n");
return 0;
}