MCL/sf/mw/gstreamer: comparison gst_plugins_symbian/gst/devsound/gstdevsoundsrc.c

equal deleted inserted replaced

-:9b2c3c7a1a9c
+:567bb019e3e3
 */
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
+#include <gst/gst.h>
+#include <e32err.h>
 #include <gqueue.h>
 #include "common.h"
 #include "gstdevsoundsrc.h"
 #include <glib_global.h>
 #include "gstg729encoderinterface.h"
 #include "gstilbcencoderinterface.h"
 #include "string.h"
 GST_DEBUG_CATEGORY_EXTERN (devsoundsrc_debug);
+#ifdef GST_CAT_DEFAULT
+#undef GST_CAT_DEFAULT
+#endif
 #define GST_CAT_DEFAULT devsoundsrc_debug
 /* elementfactory information */
 static const GstElementDetails gst_devsound_src_details=
 GST_ELEMENT_DETAILS ("Audio Src (DEVSOUND)",
 static gboolean gst_devsound_src_get_size(GstBaseSrc * src, guint64 * size);
 static GstFlowReturn gst_devsound_src_create(GstBaseSrc *src, guint64 offset,
 guint size, GstBuffer **buf);
 static void *StartDevSoundThread(void *threadid);
+static gboolean gst_devsound_src_event(GstBaseSrc * asrc, GstEvent * event);
+static GstStateChangeReturn gst_devsound_src_change_state (GstElement * element,
+GstStateChange transition);
 /*********************************
 * Speech Encoder Config Interface
 * ******************************/
 static void gst_speech_encoder_config_handler_init(gpointer g_iface,
 GList** aSupportedBitrates);
 static gint gst_get_speech_encoder_vad_mode(gboolean* aVadMode);
 static gint gst_set_speech_encoder_bit_rate(guint aBitrate);
 static gint gst_set_speech_encoder_vad_mode(gboolean aVadMode);
 static void gst_Apply_SpeechEncoder_Update(GstDevsoundSrc *devsoundsrc );
+static void gst_update_devsound_speech_bitrate(GstDevsoundSrc *devsoundsrc );
 /*************************
 * G711 Encoder Interface
 * ***********************/
 static void gst_g711_encoder_handler_init(gpointer g_iface,
 static void gst_Apply_IlbcEncoder_Update(GstDevsoundSrc *devsoundsrc );
 static gint gst_devsound_src_get_rate(gint rate);
+static void post_symbian_error ( GstBaseSrc* element, int symbian_error );
 typedef struct _GstCustomIfaceUpdate GstCustomIfaceUpdate;
 struct _GstCustomIfaceUpdate{
 gboolean  speechbitrateupdate;
 int dataCopied = 0;
 enum command_to_consumer_thread_enum
 {
 OPEN = 2,
-READDATA,
+RECORDING,
+PAUSE,
+RESUME,
+STOP,
 /*UPDATE,*/
 CLOSE
 };
 enum command_to_consumer_thread_enum cmd;
+int return_error;
 enum
 {
 LAST_SIGNAL
 };
 GST_STATIC_CAPS ("audio/x-raw-int, "
 "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
 "signed = (boolean) TRUE, "
 "width = (int) 16, "
 "depth = (int) 16, "
-"rate = (int) [ 8000, 48000 ],"
+"rate = (int) { 8000, 16000, 24000, 32000, 48000 },"
 "channels = (int) [ 1, 2 ]; "
 "audio/amr, "
-"rate = (int) [ 8000, 48000 ], "
+"rate = (int) 8000, "
-"channels = (int) [ 1, 2 ]; "
+"channels = (int) [ 1, 2 ]; ")
-"audio/x-alaw, "
-"rate = (int) [ 8000, 48000 ], "
-"channels = (int) [ 1, 2 ]; "
-"audio/g729, "
-"rate = (int) [ 8000, 48000 ], "
-"channels = (int) [ 1, 2 ]; "
-"audio/ilbc, "
-"rate = (int) [ 8000, 48000 ], "
-"channels = (int) [ 1, 2 ]; "
-"audio/x-mulaw, "
-"rate = (int) [ 8000, 48000 ], "
-"channels = (int) [ 1, 2 ]")
 );
 static GstElementClass *parent_class= NULL;
 static void gst_devsound_src_base_init(gpointer g_class)
 NULL
 };
 devsoundsrc_type =
 g_type_register_static (GST_TYPE_PUSH_SRC, "GstDevsoundSrc",
-&devsoundsrc_info, 0);
+&devsoundsrc_info, (GTypeFlags)0);
 g_type_add_interface_static (devsoundsrc_type, GST_TYPE_SPEECH_ENCODER_CONFIG,
 &speech_encoder_config_info);
 g_type_add_interface_static (devsoundsrc_type, GST_TYPE_G711_ENCODER,
 gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_devsound_src_dispose);
 gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_devsound_src_get_property);
 gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_devsound_src_set_property);
+gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_devsound_src_change_state);
 g_object_class_install_property(gobject_class, PROP_DEVICE,
 g_param_spec_string("device", "Device", "Devsound device ",
-DEFAULT_DEVICE, G_PARAM_READWRITE));
+DEFAULT_DEVICE, (GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, GAIN, g_param_spec_int(
 "gain", "Gain", "Devsound src gain", -1, G_MAXINT, -1,
-G_PARAM_READWRITE));
+(GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, MAXGAIN, g_param_spec_int(
 "maxgain", "MaxGain", "Devsound src max gain", -1, G_MAXINT, -1,
 G_PARAM_READABLE));
 g_object_class_install_property(gobject_class, LEFTBALANCE,
 g_param_spec_int("leftbalance", "Left Balance", "Left Balance",
--1, G_MAXINT, -1, G_PARAM_READWRITE));
+-1, G_MAXINT, -1, (GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, RIGHTBALANCE,
 g_param_spec_int("rightbalance", "Right Balance",
-"Right Balance", -1, G_MAXINT, -1, G_PARAM_READWRITE));
+"Right Balance", -1, G_MAXINT, -1, (GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, SAMPLESRECORDED,
 g_param_spec_int("samplesrecorded", "Samples Recorded",
 "Samples Recorded", -1, G_MAXINT, -1, G_PARAM_READABLE));
 g_object_class_install_property(gobject_class, PRIORITY,
 g_param_spec_int("priority", "Priority", "Priority ", -1,
 G_MAXINT, -1,
-G_PARAM_READWRITE));
+(GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, PREFERENCE,
 g_param_spec_int("preference", "Preference", "Preference ", -1,
 G_MAXINT, -1,
-G_PARAM_READWRITE));
+(GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, RATE,
 g_param_spec_int("rate", "Rate", "Rate ", -1,
 G_MAXINT, -1,
-G_PARAM_READWRITE));
+(GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, CHANNELS,
 g_param_spec_int("channels", "Channels", "Channels ", -1,
 G_MAXINT, -1,
-G_PARAM_READWRITE));
+(GParamFlags)G_PARAM_READWRITE));
 gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_devsound_src_start);
 gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_devsound_src_stop);
 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_devsound_src_getcaps);
 gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_devsound_src_setcaps);
+gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_devsound_src_event);
 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_devsound_src_create);
 }
 static void gst_devsound_src_init(GstDevsoundSrc * devsoundsrc)
 {
 GST_DEBUG_OBJECT(devsoundsrc, "initializing devsoundsrc");
+gst_base_src_set_live(GST_BASE_SRC(devsoundsrc), TRUE);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "gst_devsound_src_init ENTER ",NULL);
 devsoundsrc->device = g_strdup(DEFAULT_DEVICE);
 devsoundsrc->handle=NULL;
 devsoundsrc->preference = 0; //default=>EMdaPriorityPreferenceNone;
 devsoundsrc->priority = 0;   //default=>EMdaPriorityNormal;
-pthread_mutex_init(&create_mutex1, NULL);
+devsoundsrc->firstTimeInit = kUnInitialized;
-pthread_cond_init(&create_condition1, NULL);
+//    pthread_mutex_init(&create_mutex1, NULL);
+//    pthread_cond_init(&create_condition1, NULL);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "gst_devsound_src_init EXIT ",NULL);
 }
 static void *StartDevSoundThread(void *threadarg)
 {
 GstDevsoundSrc *devsoundsrc= GST_DEVSOUND_SRC (threadarg);
 int buffersize;
 TUint8* gBuffer;
 GstBuffer* pushBuffer= NULL;
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "StartDevSoundThread ",NULL);
-open_devsound(&(devsoundsrc->handle));
+int ret = open_devsound(&(devsoundsrc->handle));
+if( KErrNone != ret )
+{
+pthread_mutex_lock(&(create_mutex1));
+return_error = ret;
+pthread_cond_signal(&(create_condition1));
+pthread_mutex_unlock(&(create_mutex1));
+pthread_exit(NULL);
+}
 getsupporteddatatypes(devsoundsrc);
 // TODO obtain mutex to update variable here???
 consumer_thread_state = CONSUMER_THREAD_INITIALIZED;
 //pre_init_setconf(devsoundsrc);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "After Record Init ",NULL);
 pthread_mutex_lock(&(create_mutex1));
+return_error = ret;
 pthread_cond_signal(&(create_condition1));
 pthread_mutex_unlock(&(create_mutex1));
 // Wait until we receive a command from the main thread
 // TODO obtain mutex to read variable here???
 // This could happen if client creates sink and deletes it
 // without putting it to play state
 if ( cmd != CLOSE )
 {
-initialize_devsound(devsoundsrc);
+gst_update_devsound_speech_bitrate(devsoundsrc);
+ret = initialize_devsound(devsoundsrc);
-recordinit(devsoundsrc->handle);
+if( KErrNone == ret )
-initproperties(devsoundsrc);
-}
-//cmd = READDATA;
-while (1)
-{
-//set/get properties
-//***************************************
-pre_init_setconf(devsoundsrc);
-gst_Apply_SpeechEncoder_Update(devsoundsrc);
-gst_Apply_G711Encoder_Update(devsoundsrc);
-gst_Apply_G729Encoder_Update(devsoundsrc );
-gst_Apply_IlbcEncoder_Update(devsoundsrc );
-populateproperties(devsoundsrc);
-supportedbitrates = devsoundsrc->supportedbitrates;
-//numofbitrates = devsoundsrc->numofbitrates;
-speechbitrate = devsoundsrc->speechbitrate;
-speechvadmode = devsoundsrc->speechvadmode;
-g711vadmode = devsoundsrc->g711vadmode;
-g729vadmode  = devsoundsrc->g729vadmode;
-ilbcvadmode = devsoundsrc->ilbcvadmode;
-//****************************************
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "Before Buffer Alloc ",NULL);
-buffersize = get_databuffer_size(devsoundsrc->handle);
-get_databuffer(devsoundsrc->handle, &gBuffer);
-pushBuffer = gst_buffer_new_and_alloc(buffersize);
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "After Buffer Alloc ",NULL);
-if (GST_BUFFER_DATA(pushBuffer))
 {
-memcpy(GST_BUFFER_DATA(pushBuffer),gBuffer,buffersize);
+ret = recordinit(devsoundsrc->handle);
+if( KErrNone == ret )
+initproperties(devsoundsrc);
 }
-else
+/// if initialization of devsound failed, return error,
+/// on return other thread will send CLOSE cmd to exit from thread.
+if( ret )
 {
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "Push buffer alloc failed ",NULL);
+pthread_mutex_lock(&(create_mutex1));
+return_error = ret;
+pthread_cond_signal(&(create_condition1));
+pthread_mutex_unlock(&(create_mutex1));
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_wait(&create_condition1, &create_mutex1);
+pthread_mutex_unlock(&create_mutex1);
 }
+}
-if (dataqueue)
-{
+while (TRUE)
-GST_OBJECT_LOCK(devsoundsrc);
+{
-g_queue_push_head (dataqueue,pushBuffer);
-GST_OBJECT_UNLOCK(devsoundsrc);
-//signalmutex_create(devsoundsrc->handle);
-if(dataqueue->length == 1 && (cmd != CLOSE))
-{
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "Before signal in DevSoundt ",NULL);
-pthread_mutex_lock(&(create_mutex1));
-pthread_cond_signal(&(create_condition1));
-pthread_mutex_unlock(&(create_mutex1));
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "After signal in DevSoundt ",NULL);
-}
-//cmd = READDATA;
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "Before DevSnd Wait ",NULL);
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "After DevSnd Wait ",NULL);
-}
-else
-{
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "dataqueue is NULL, CLOSE now ",NULL);
-cmd = CLOSE;
-}
 switch (cmd)
 {
-case READDATA:
+case PAUSE:
+pause_devsound(devsoundsrc);
+break;
+case RESUME:
+resume_devsound(devsoundsrc);
+break;
+case STOP:
+stop_devsound(devsoundsrc);
+break;
+case RECORDING:
 {
+pre_init_setconf(devsoundsrc);
+//                gst_Apply_SpeechEncoder_Update(devsoundsrc);
+gst_Apply_G711Encoder_Update(devsoundsrc);
+gst_Apply_G729Encoder_Update(devsoundsrc );
+gst_Apply_IlbcEncoder_Update(devsoundsrc );
+populateproperties(devsoundsrc);
+supportedbitrates = devsoundsrc->supportedbitrates;
+//numofbitrates = devsoundsrc->numofbitrates;
+speechbitrate = devsoundsrc->speechbitrate;
+speechvadmode = devsoundsrc->speechvadmode;
+g711vadmode = devsoundsrc->g711vadmode;
+g729vadmode  = devsoundsrc->g729vadmode;
+ilbcvadmode = devsoundsrc->ilbcvadmode;
+buffersize = get_databuffer_size(devsoundsrc->handle);
+get_databuffer(devsoundsrc->handle, &gBuffer);
+pushBuffer = gst_buffer_new_and_alloc(buffersize);
+memcpy(GST_BUFFER_DATA(pushBuffer),gBuffer,buffersize);
+GST_OBJECT_LOCK(devsoundsrc);
+g_queue_push_head (dataqueue,pushBuffer);
+GST_OBJECT_UNLOCK(devsoundsrc);
 record_data(devsoundsrc->handle);
 }
 break;
 case CLOSE:
 {
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "After handle NULL ",NULL);
 pthread_mutex_lock(&(create_mutex1));
 pthread_cond_signal(&(create_condition1));
 pthread_mutex_unlock(&(create_mutex1));
-				// TODO obtain mutex here
+// TODO obtain mutex here
 consumer_thread_state = CONSUMER_THREAD_UNINITIALIZED;
 pthread_exit(NULL);
 }
 break;
 default:
 // TODO obtain mutex here
 consumer_thread_state = CONSUMER_THREAD_UNINITIALIZED;
 pthread_exit(NULL);
 break;
 }
-}
+pthread_mutex_lock(&(create_mutex1));
-// TODO obtain mutex here
+return_error = call_back_error(devsoundsrc->handle);
-consumer_thread_state = CONSUMER_THREAD_UNINITIALIZED;
+pthread_cond_signal(&(create_condition1));
-pthread_exit(NULL);
+pthread_mutex_unlock(&(create_mutex1));
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_wait(&create_condition1, &create_mutex1);
+pthread_mutex_unlock(&create_mutex1);
+}
 }
 static void gst_devsound_src_set_property(GObject * object, guint prop_id,
 const GValue * value, GParamSpec * pspec)
 {
 }
 static gboolean gst_devsound_src_start(GstBaseSrc * bsrc)
 {
 GstBuffer *tmp_gstbuffer=NULL;
+gboolean ret = TRUE;
 GstDevsoundSrc *src= GST_DEVSOUND_SRC(bsrc);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "gst_devsound_src_start ENTER ",NULL);
+pthread_mutex_init(&create_mutex1, NULL);
+pthread_cond_init(&create_condition1, NULL);
 if(dataqueue)
 {
-while (dataqueue->length)
+while (g_queue_get_length(dataqueue))
 {
 tmp_gstbuffer = (GstBuffer*)g_queue_pop_tail(dataqueue);
 gst_buffer_unref(tmp_gstbuffer);
 }
 g_queue_free(dataqueue);
 dataqueue = g_queue_new();
 }
 consumer_thread_state = CONSUMER_THREAD_INITIALIZING;
 cmd = OPEN;
+return_error = KErrNone;
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "Before Thread Create ",NULL);
 pthread_create(&ds_thread, NULL, StartDevSoundThread, (void *)src);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "After Thread Create ",NULL);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "Before Mutex Wait in START ",NULL);
 if (consumer_thread_state == CONSUMER_THREAD_INITIALIZING)
 {
 pthread_mutex_lock(&create_mutex1);
 pthread_cond_wait(&create_condition1, &create_mutex1);
+ret = !return_error; // should be TRUE on no Error, and FALSE at ERROR.
 pthread_mutex_unlock(&create_mutex1);
 }
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "AFter Mutex Wait in START ",NULL);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "gst_devsound_src_start EXIT ",NULL);
-return TRUE;
+if ( return_error )
+post_symbian_error( bsrc,return_error );
+return ret;
 /* ERRORS */
 }
 static gboolean gst_devsound_src_stop(GstBaseSrc * bsrc)
 {
 GstDevsoundSrc *src= GST_DEVSOUND_SRC(bsrc);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "gst_devsound_src_stop ENTER ");
 cmd = CLOSE;
+pthread_mutex_lock(&(create_mutex1));
+pthread_cond_signal(&(create_condition1));
+pthread_mutex_unlock(&(create_mutex1));
 //GST_OBJECT_LOCK (src);
 pthread_mutex_lock(&(create_mutex1));
 pthread_cond_wait(&(create_condition1), &(create_mutex1));
 pthread_mutex_unlock(&(create_mutex1));
 //GST_OBJECT_UNLOCK (src);
 dataleft = 0;   //Arun'S changes
 }
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "Before QUEUE Lock in STOP ");
 GST_OBJECT_LOCK(src);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "After QUEUE Lock in STOP ");
-while (dataqueue->length)
+while (g_queue_get_length(dataqueue))
 {
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "Removing DATAQUEUE elements ENTER ");
 popBuffer = (GstBuffer*)g_queue_pop_tail(dataqueue);
 gst_buffer_unref(popBuffer);
 popBuffer = NULL;
 GST_OBJECT_UNLOCK(src);
 pthread_mutex_destroy(&create_mutex1);
 pthread_cond_destroy(&(create_condition1));
 g_free(src->device);
+src->device = NULL;
+src->firstTimeInit = kUnInitialized;
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "gst_devsound_src_stop EXIT ");
 return TRUE;
 }
 static gboolean gst_devsound_src_unlock(GstBaseSrc * bsrc)
 static GstFlowReturn gst_devsound_src_create(GstBaseSrc *src, guint64 offset,
 guint size, GstBuffer **buf)
 {
 GstDevsoundSrc *dsrc= GST_DEVSOUND_SRC(src);
 int bufferpos=0;
+int ret = KErrNone;
+if(!g_queue_get_length(dataqueue) && (dsrc->eosreceived == TRUE))
+{
+pthread_mutex_lock(&(create_mutex1));
+pthread_cond_signal(&(create_condition1));
+pthread_mutex_unlock(&(create_mutex1));
+post_symbian_error( src,KErrCancel );
+return GST_FLOW_UNEXPECTED;
+}
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "gst_devsound_src_create ENTER ");
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "Before Buffer Alloc in CREATE ",NULL);
 *buf = gst_buffer_try_new_and_alloc(size);
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "AFter Buffer Alloc in CREATE ",NULL);
+if(*buf == NULL)
+{
+post_symbian_error( src,KErrNoMemory );
+return GST_FLOW_UNEXPECTED;
+}
 while (size > 0)
 {
 if (dataleft >= size)
 {
 // if there is some data left in the popped buffer previously whose size
 popBuffer = NULL;
 }
 // we wait here if the dataqueue length is 0 and we need data
 // to be filled in the queue from the DevSound Thread
-if (!dataqueue->length)
+if (!g_queue_get_length(dataqueue))
 {
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "Before WAIT in CREATE ",NULL);
-cmd = READDATA;
+if(dsrc->eosreceived == TRUE)
-pthread_mutex_lock(&(create_mutex1));
+{
-pthread_cond_signal(&(create_condition1));
+post_symbian_error( src,KErrCancel );
-pthread_mutex_unlock(&(create_mutex1));
+return GST_FLOW_UNEXPECTED;
+}
-pthread_mutex_lock(&(create_mutex1));
+else
-pthread_cond_wait(&(create_condition1), &(create_mutex1));
+{
-pthread_mutex_unlock(&(create_mutex1));
+cmd = RECORDING;
+return_error = KErrNone;
+pthread_mutex_lock(&(create_mutex1));
+pthread_cond_signal(&(create_condition1));
+pthread_mutex_unlock(&(create_mutex1));
+pthread_mutex_lock(&(create_mutex1));
+pthread_cond_wait(&(create_condition1), &(create_mutex1));
+ret = return_error;
+pthread_mutex_unlock(&(create_mutex1));
+}
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "AFTER WAIT in CREATE ",NULL);
 }
+if( ret )
+{
+post_symbian_error( src,ret );
+return GST_FLOW_UNEXPECTED;
+}
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "Before POP in CREATE ",NULL);
 GST_OBJECT_LOCK(dsrc);
 popBuffer = (GstBuffer*)g_queue_pop_tail(dataqueue);
 GST_OBJECT_UNLOCK(dsrc);
-//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "AFTER POP in CREATE ",NULL);
+if(!popBuffer )
+{
+post_symbian_error( src,KErrNoMemory );
+return GST_FLOW_UNEXPECTED;
+}
+if(dsrc->firstTimeInit != kPlayed)
+{
+dsrc->prevbuffersize = gst_base_src_get_blocksize(src);
+gst_base_src_set_blocksize (src, GST_BUFFER_SIZE(popBuffer));
+(*buf)->size = GST_BUFFER_SIZE(popBuffer);
+}
 // copy the data from the popped buffer based on how much of the incoming
 //buffer size is left to fill. we might have filled the fresh buffer somewhat
 // where the size of the fresh buffer is more then the data remaining in the
 // popped buffer.
 if (size < GST_BUFFER_SIZE(popBuffer))
 {
 gst_buffer_unref(popBuffer);
 popBuffer = NULL;
 }
 }
-}
+if (dsrc->firstTimeInit == kPlayBufferPreRoll)
+{
+gst_base_src_set_blocksize (src, dsrc->prevbuffersize);
+dsrc->firstTimeInit = kPlayed;
+return GST_FLOW_OK;
+}
+if (dsrc->firstTimeInit == kPausedToPlaying)
+{
+dsrc->firstTimeInit = kPlayBufferPreRoll;
+return GST_FLOW_OK;
+}
+}
 //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "gst_devsound_src_create EXIT ",NULL);
 return GST_FLOW_OK;
 }
+static GstStateChangeReturn gst_devsound_src_change_state (GstElement * element,
+GstStateChange transition)
+{
+GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+GstDevsoundSrc *src= GST_DEVSOUND_SRC (element);
+switch (transition) {
+case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+if (src->firstTimeInit != kPlayed)
+src->firstTimeInit = kPausedToPlaying;
+if(cmd == PAUSE)
+{
+cmd = RESUME;
+return_error = KErrNone;
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_signal(&create_condition1);
+pthread_mutex_unlock(&create_mutex1);
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_wait(&create_condition1, &create_mutex1);
+if( return_error )
+{
+post_symbian_error( GST_BASE_SRC(element),return_error );
+ret = GST_STATE_CHANGE_FAILURE;
+}
+pthread_mutex_unlock(&create_mutex1);
+}
+break;
+default:
+break;
+}
+ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+if (G_UNLIKELY (ret == GST_STATE_CHANGE_FAILURE))
+goto activate_failed;
+switch (transition) {
+case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+cmd = PAUSE;
+return_error = KErrNone;
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_signal(&create_condition1);
+pthread_mutex_unlock(&create_mutex1);
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_wait(&create_condition1, &create_mutex1);
+if( return_error )
+{
+post_symbian_error( GST_BASE_SRC(element),return_error );
+ret = GST_STATE_CHANGE_FAILURE;
+}
+pthread_mutex_unlock(&create_mutex1);
+break;
+default:
+break;
+}
+return ret;
+activate_failed:
+{
+GST_DEBUG_OBJECT (src,
+"element failed to change states -- activation problem?");
+return GST_STATE_CHANGE_FAILURE;
+}
+}
 static gboolean gst_devsound_src_is_seekable(GstBaseSrc * bsrc)
 {
 GstDevsoundSrc *src= GST_DEVSOUND_SRC(bsrc);
 return FALSE;
 speechvadmode = aVadMode;
 customInfaceUpdate.speechvadmodeupdate = TRUE;
 return 0;
 }
+static void gst_update_devsound_speech_bitrate(GstDevsoundSrc *devsoundsrc )
+{
+if(customInfaceUpdate.speechbitrateupdate == TRUE)
+{
+update_devsound_speech_bitrate(devsoundsrc->handle,speechbitrate);
+customInfaceUpdate.speechbitrateupdate = FALSE;
+}
+}
 static void gst_Apply_SpeechEncoder_Update(GstDevsoundSrc *devsoundsrc )
 {
 if(customInfaceUpdate.speechbitrateupdate == TRUE)
 {
 set_speech_encoder_bit_rate(devsoundsrc->handle,speechbitrate);
 }
 return result;
 }
+static gboolean gst_devsound_src_event(GstBaseSrc *asrc, GstEvent *event)
+{
+int retValue = FALSE;
+GstDevsoundSrc *src = GST_DEVSOUND_SRC(asrc);
+switch (GST_EVENT_TYPE (event))
+{
+case GST_EVENT_EOS:
+// end-of-stream, we should close down all stream leftovers here
+//reset_devsound(sink->handle);
+src->eosreceived = TRUE;
+cmd = STOP;
+return_error = KErrNone;
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_signal(&create_condition1);
+pthread_mutex_unlock(&create_mutex1);
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_wait(&create_condition1, &create_mutex1);
+pthread_mutex_unlock(&create_mutex1);
+if(g_queue_get_length(dataqueue))
+{
+pthread_mutex_lock(&create_mutex1);
+pthread_cond_wait(&create_condition1, &create_mutex1);
+pthread_mutex_unlock(&create_mutex1);
+}
+gst_pad_push_event (asrc->srcpad, gst_event_new_eos ());
+retValue = TRUE;
+break;
+default:
+retValue = FALSE;
+break;
+}
+return retValue;
+}
+static void
+post_symbian_error ( GstBaseSrc* element, int symbian_error )
+{
+switch (symbian_error)
+{
+case KErrNone:
+break;
+case KErrNotFound:
+GST_ELEMENT_ERROR (element, RESOURCE, NOT_FOUND,
+(("resource not found error.")),
+("streaming paused because resource not found, Err(%d)", symbian_error));
+break;
+case KErrCancel:
+GST_ELEMENT_ERROR (element, RESOURCE, CLOSE,
+(("resource request canceled.")),
+("streaming paused because resource request canceled, Err(%d)", symbian_error));
+break;
+case KErrNoMemory:
+GST_ELEMENT_ERROR (element, RESOURCE, NO_SPACE_LEFT,
+(("Error out of memory")),
+("streaming paused because of no memory, Err(%d)", symbian_error));
+break;
+case KErrNotSupported:
+GST_ELEMENT_ERROR (element, RESOURCE, READ,
+(("resource not supported.")),
+("streaming paused because resource not supported, Err(%d)", symbian_error));
+break;
+case KErrNotReady:
+GST_ELEMENT_ERROR (element, RESOURCE, TOO_LAZY,
+(("resource is not ready for request.")),
+("streaming paused because resource is not ready, Err(%d)", symbian_error));
+break;
+case KErrPermissionDenied:
+GST_ELEMENT_ERROR (element, RESOURCE, READ,
+(("Permission Denied.")),
+("streaming paused because permission denied, Err(%d)", symbian_error));
+break;
+case KErrTimedOut:
+GST_ELEMENT_ERROR (element, RESOURCE, TOO_LAZY,
+(("Timed Out.")),
+("streaming paused because timed out, Err(%d)", symbian_error));
+break;
+case KErrHardwareNotAvailable:
+GST_ELEMENT_ERROR (element, RESOURCE, NOT_FOUND,
+(("Hardware Not Available.")),
+("streaming paused because hardware not available, Err(%d)", symbian_error));
+break;
+case KErrCorrupt:
+GST_ELEMENT_ERROR (element, RESOURCE, FAILED,
+(("Corrupted data.")),
+("streaming paused because of corrupted data, Err(%d)", symbian_error));
+break;
+case KErrUnderflow:
+GST_ELEMENT_ERROR (element, STREAM, FAILED,
+(("Under flow.")),
+("streaming paused because of under flow, Err(%d)", symbian_error));
+break;
+case KErrOverflow:
+GST_ELEMENT_ERROR (element, STREAM, FAILED,
+(("Over flow.")),
+("streaming paused because of over flow, Err(%d)", symbian_error));
+break;
+case KErrInUse:
+GST_ELEMENT_ERROR (element, RESOURCE, BUSY,
+(("Resource In Use.")),
+("streaming paused because resource in use, Err(%d)", symbian_error));
+break;
+case KErrDied:
+GST_ELEMENT_ERROR (element, RESOURCE, BUSY,
+(("Error Resource Died.")),
+("streaming paused because resource died., Err(%d)", symbian_error));
+break;
+case KErrAccessDenied:
+GST_ELEMENT_ERROR (element, RESOURCE, BUSY,
+(("Resource Access Denied.")),
+("streaming paused because resource access denied, Err(%d)", symbian_error));
+break;
+default:
+GST_ELEMENT_ERROR (element, STREAM, FAILED,
+(("internal data flow error.")),
+("streaming paused because of internal data flow error., Err(%d)", symbian_error));
+break;
+}
+}

branch	RCL_3
changeset 7	567bb019e3e3
parent 0	0e761a78d257
child 8	7e817e7e631c