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

equal deleted inserted replaced

-:9b2c3c7a1a9c
+:567bb019e3e3
 #include "gstg711decoderinterface.h"
 #include "gstg729decoderinterface.h"
 #include "gstilbcdecoderinterface.h"
 #include "string.h"
 #include <glib_global.h>
+#ifdef AV_SYNC
+#include <gst/audio/gstaudioclock.h>
+#endif /*AV_SYNC*/
 GST_DEBUG_CATEGORY_EXTERN (devsound_debug);
+#ifdef GST_CAT_DEFAULT
+#undef GST_CAT_DEFAULT
+#endif
 #define GST_CAT_DEFAULT devsound_debug
 /* elementfactory information */
 static const GstElementDetails gst_devsound_sink_details=
 GST_ELEMENT_DETAILS ("Audio Sink (DEVSOUND)",
 const GValue * value, GParamSpec * pspec);
 static GstCaps *gst_devsound_sink_getcaps(GstBaseSink * bsink);
 static gboolean gst_devsound_sink_setcaps(GstBaseSink *bsink, GstCaps *caps);
 static gboolean gst_devsound_sink_event(GstBaseSink * asink, GstEvent * event);
+#ifdef AV_SYNC
+static void gst_devsound_sink_get_times(GstBaseSink * bsink, GstBuffer * buffer,
+GstClockTime * start, GstClockTime * end);
+static GstClock *gst_devsound_sink_provide_clock (GstElement * element);
+static GstClockTime gst_devsound_sink_get_time (GstClock * clock,
+gpointer user_data);
+#endif /*AV_SYNC*/
+static GstStateChangeReturn gst_devsound_sink_change_state (GstElement * element,
+GstStateChange transition);
 static void *StartDevSoundThread(void *threadid);
 //Error concealment interface impl
 static void gst_error_concealment_handler_init (gpointer g_iface,
 gpointer iface_data);
-static gint gst_ConcealErrorForNextBuffer();
+static gint gst_ConcealErrorForNextBuffer(void);
 static gint gst_SetFrameMode(gboolean aFrameMode);
 static gint gst_FrameModeRqrdForEC(gboolean* aFrameModeRqrd);
 static void gst_Apply_ErrorConcealment_Update(GstDevsoundSink* dssink);
 //G711 interface impl
 //G729 interface impl
 static void gst_g729_decoder_handler_init (gpointer g_iface,
 gpointer iface_data);
-static gint gst_BadLsfNextBuffer();
+static gint gst_BadLsfNextBuffer(void);
 static void gst_Apply_G729_Decoder_Update(GstDevsoundSink* dssink );
 //Ilbc interface impl
 static void gst_ilbc_decoder_handler_init (gpointer g_iface,
 gpointer iface_data);
 static gint gst_GetIlbcCng(gboolean* aCng);
 static gint gst_SetIlbcCng(gboolean aCng);
 static gint gst_SetIlbcDecoderMode(enum TIlbcDecodeMode aDecodeMode);
 static void  gst_Apply_Ilbc_Decoder_Update(GstDevsoundSink* dssink );
+static void get_PopulateIntfcProperties(GstDevsoundSink* dssink);
 static gboolean gst_sink_start (GstBaseSink * sink);
 static gboolean gst_sink_stop (GstBaseSink * sink);
 static GstFlowReturn gst_sink_render (GstBaseSink * sink,
 GstBuffer * buffer);
 PROP_0,
 PROP_DEVICE,
 VOLUME,
 MAXVOLUME,
 VOLUMERAMP,
-CHANNELS,
+/*    CHANNELS,*/
 LEFTBALANCE,
 RIGHTBALANCE,
-RATE,
+/*    RATE,*/
 PRIORITY,
 PREFERENCE,
-SAMPLESPLAYED,
+/*    SAMPLESPLAYED,*/
-FOURCC, //FOURCC is not needed
+/*    FOURCC, //FOURCC is not needed*/
-MIMETYPE,
+/*    MIMETYPE,*/
 OUTPUTDEVICE
 };
 enum command_to_consumer_thread_enum
 {
 OPEN = 2,
-WRITEDATA,
+PLAYING,
+PAUSE,
+RESUME,
 /*UPDATE,*/
+WAIT,
 CLOSE
 };
 enum command_to_consumer_thread_enum cmd;
 static GstStaticPadTemplate devsoundsink_sink_factory=
 GST_STATIC_PAD_TEMPLATE ("sink",
 GST_PAD_SINK,
 GST_PAD_ALWAYS,
-GST_STATIC_CAPS ("audio/x-raw-int, "
+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 ]," "channels = (int) [ 1, 2 ]; "
-"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+"audio/amr, " "rate = (int) 8000, " "channels = (int) 1 ; "
-"signed = (boolean) TRUE, "
+"audio/AMR, " "rate = (int) 8000, " "channels = (int) 1 ; "
-"width = (int) 16, "
+"audio/x-alaw, " "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ]; "
-"depth = (int) 16, "
+"audio/g729, " "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ]; "
-"rate = (int) [ 8000, 48000 ],"
+"audio/mpeg, mpegversion = (int) 1, layer = (int) [ 1, 3 ], rate = (int) [ 8000, 48000 ], channels = (int) [ 1, 2 ]; "
-"channels = (int) [ 1, 2 ]; "
+"audio/ilbc, " "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ]; "
-"audio/amr, "
+"audio/x-mulaw, " "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ]")
-//"width = (int) 8, "
-//"depth = (int) 8, "
-"rate = (int) 8000, "
-"channels = (int) 1 ; "
-"audio/x-alaw, "
-"rate = (int) [ 8000, 48000 ], "
-"channels = (int) [ 1, 2 ]; "
-"audio/g729, "
-"rate = (int) [ 8000, 48000 ], "
-"channels = (int) [ 1, 2 ]; "
-"audio/mp3, "
-"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;
 GType gst_devsound_sink_get_type(void)
 NULL
 };
 devsoundsink_type =
 g_type_register_static (GST_TYPE_BASE_SINK, "GstDevsoundSink",
-&devsoundsink_info, 0);
+&devsoundsink_info, (GTypeFlags)0);
 g_type_add_interface_static (devsoundsink_type, GST_TYPE_ERROR_CONCEALMENT,
 &error_concealment_info);
 return devsoundsink_type;
 }
 static void gst_devsound_sink_dispose(GObject * object)
 {
-GstDevsoundSink *devsoundsink= GST_DEVSOUND_SINK (object);
+GstDevsoundSink *devsoundsink = GST_DEVSOUND_SINK (object);
 if (devsoundsink->probed_caps)
 {
 gst_caps_unref(devsoundsink->probed_caps);
 devsoundsink->probed_caps = NULL;
 }
+#ifdef AV_SYNC
+if (devsoundsink->clock)
+{
+gst_object_unref (devsoundsink->clock);
+}
+devsoundsink->clock = NULL;
+#endif /*AV_SYNC*/
 G_OBJECT_CLASS (parent_class)->dispose (object);
 }
 static void gst_devsound_sink_base_init(gpointer g_class)
 {
 gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_devsound_sink_dispose);
 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_devsound_sink_finalise);
 gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_devsound_sink_get_property);
 gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_devsound_sink_set_property);
+gstelement_class->change_state = GST_DEBUG_FUNCPTR(gst_devsound_sink_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, VOLUME,
 g_param_spec_int("volume", "Volume", "Devsound volume",
--1, G_MAXINT, -1, G_PARAM_READWRITE));
+-1, G_MAXINT, -1, (GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, VOLUMERAMP,
 g_param_spec_int("volumeramp", "VolumeRamp", "Devsound volume ramp",
--1, G_MAXINT, -1, G_PARAM_READWRITE));
+-1, G_MAXINT, -1, (GParamFlags)G_PARAM_READWRITE));
 g_object_class_install_property(gobject_class, MAXVOLUME,
 g_param_spec_int("maxvolume", "MaxVolume", "Devsound max volume",
 -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));
+-1, G_MAXINT, -1, (GParamFlags)G_PARAM_READWRITE));
+/*
 g_object_class_install_property(gobject_class, SAMPLESPLAYED,
 g_param_spec_int("samplesplayed", "Samples Played", "Samples Played",
 -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));
 g_object_class_install_property(gobject_class, CHANNELS,
 g_param_spec_int("channels", "Channels", "Channels ", -1,
 G_MAXINT, -1,
 G_PARAM_READWRITE));
+*/
 g_object_class_install_property(gobject_class, OUTPUTDEVICE,
 g_param_spec_int("outputdevice", "Output Device", "Output Device ", -1,
 G_MAXINT, -1,
-G_PARAM_READWRITE));
+(GParamFlags)G_PARAM_READWRITE));
+#ifdef AV_SYNC
+gstelement_class->provide_clock = GST_DEBUG_FUNCPTR (gst_devsound_sink_provide_clock);
+#endif /*AV_SYNC*/
 gstbasesink_class->start = GST_DEBUG_FUNCPTR (gst_sink_start);
 gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_sink_stop);
 gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_sink_render);
 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_devsound_sink_getcaps);
 gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_devsound_sink_setcaps);
 gstbasesink_class->event    = GST_DEBUG_FUNCPTR (gst_devsound_sink_event);
-}
+#ifdef AV_SYNC
+gstbasesink_class->get_times = GST_DEBUG_FUNCPTR (gst_devsound_sink_get_times);
-static void gst_devsound_sink_init(GstDevsoundSink * devsoundsink)
+#endif /*AV_SYNC*/
-{
+}
-GST_DEBUG_OBJECT(devsoundsink, "initializing devsoundsink");
-devsoundsink->device = g_strdup(DEFAULT_DEVICE);
+static void gst_devsound_sink_init(GstDevsoundSink * dssink)
-devsoundsink->handle = NULL;
+{
-devsoundsink->preference = 0; //default=>EMdaPriorityPreferenceNone;
+GST_DEBUG_OBJECT(dssink, "initializing devsoundsink");
-devsoundsink->priority = 0;   //default=>EMdaPriorityNormal;
+dssink->device = g_strdup(DEFAULT_DEVICE);
+dssink->handle = NULL;
+dssink->preference = 0; //default=>EMdaPriorityPreferenceNone;
+dssink->priority = 0;   //default=>EMdaPriorityNormal;
+#ifdef AV_SYNC
+dssink->time_or_samples_played = 0;
+dssink->timeplayedavailable = FALSE;
+/* Create the provided clock. */
+dssink->clock = gst_audio_clock_new ("clock", gst_devsound_sink_get_time, dssink);
+#endif /*AV_SYNC*/
 pthread_mutex_init(&ds_mutex, NULL);
 pthread_cond_init(&ds_condition, NULL);
 }
 static void *StartDevSoundThread(void *threadarg)
 {
+GstDevsoundSink *dssink;
-GstDevsoundSink *devsound;
 gint remainingDataLen = 0;
 GstBuffer *buffer = NULL;
 gboolean lastBufferSet=FALSE;
-devsound = (GstDevsoundSink*) threadarg;
+dssink = (GstDevsoundSink*) threadarg;
-open_devsound(&(devsound->handle));
+// TODO handle error here
+open_devsound(&(dssink->handle));
+#ifdef AV_SYNC
+dssink->timeplayedavailable = is_timeplayed_supported(dssink->handle);
+#endif /*AV_SYNC*/
 //get supported (in/out)put datatypes
 //from devsound to build caps
-getsupporteddatatypes(devsound);
+getsupporteddatatypes(dssink);
 // TODO obtain mutex to update variable here???
 consumer_thread_state = CONSUMER_THREAD_INITIALIZED;
 // Signal any waiting threads that consumer thread creation was successful.
 // without putting it to play state
 if ( cmd != CLOSE )
 {
 //TODO if there is preemption we have to somehow signal
 //the pipeline in the render
-initialize_devsound(devsound);
+initialize_devsound(dssink);
-playinit(devsound->handle);
+playinit(dssink->handle);
-initproperties(devsound);
+dssink->eosreceived = FALSE;
+initproperties(dssink);
 }
 while (1)
 {
 switch (cmd)
 {
-case WRITEDATA:
+case PAUSE:
+pause_devsound(dssink);
+cmd = WAIT;
+break;
+case RESUME:
+resume_devsound(dssink);
+cmd = PLAYING;
+break;
+case WAIT:
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_signal(&ds_condition);
+pthread_mutex_unlock(&ds_mutex);
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_wait(&ds_condition, &ds_mutex);
+pthread_mutex_unlock(&ds_mutex);
+break;
+case PLAYING:
 {
-pre_init_setconf(devsound);
+pre_init_setconf(dssink);
-gst_Apply_ErrorConcealment_Update(devsound);
+gst_Apply_ErrorConcealment_Update(dssink);
-gst_Apply_G711_Decoder_Update(devsound);
+gst_Apply_G711_Decoder_Update(dssink);
-gst_Apply_G729_Decoder_Update(devsound);
+gst_Apply_G729_Decoder_Update(dssink);
-gst_Apply_Ilbc_Decoder_Update(devsound);
+gst_Apply_Ilbc_Decoder_Update(dssink);
 // TODO we could do this in BTBF callback
-populateproperties(devsound);
+populateproperties(dssink);
+get_PopulateIntfcProperties(dssink);
-framemodereq = devsound->framemodereq;
-g711cng = devsound->g711cng;
-ilbccng = devsound->ilbccng;
-output = devsound->output;
 if(buffer_queue->length > 0)
 {
 if (remainingDataLen == 0)
 {
 // TODO enable lock and unlock
-GST_OBJECT_LOCK (devsound);
+GST_OBJECT_LOCK (dssink);
 buffer = GST_BUFFER_CAST(g_queue_peek_head(buffer_queue));
-GST_OBJECT_UNLOCK(devsound);
+GST_OBJECT_UNLOCK(dssink);
 remainingDataLen = GST_BUFFER_SIZE(buffer);
 }
 lastBufferSet =  GST_BUFFER_FLAG_IS_SET(buffer,GST_BUFFER_FLAG_LAST);
-remainingDataLen = write_data(devsound->handle,
+remainingDataLen = write_data(dssink->handle,
 GST_BUFFER_DATA(buffer) + (GST_BUFFER_SIZE(buffer) - remainingDataLen),
 remainingDataLen,
 lastBufferSet);
 if (remainingDataLen == 0)
 {
-GST_OBJECT_LOCK (devsound);
+GST_OBJECT_LOCK (dssink);
 buffer = GST_BUFFER_CAST(g_queue_pop_head(buffer_queue));
-GST_OBJECT_UNLOCK(devsound);
+GST_OBJECT_UNLOCK(dssink);
 gst_buffer_unref(buffer);
 buffer = NULL;
 }
 if (lastBufferSet && remainingDataLen == 0)
 {
-// Last Buffer is already sent to DevSound
+lastBufferSet = FALSE;
-// and we have received PlayError so now we exit
+dssink->eosreceived = FALSE;
-// from the big loop next time
+playinit(dssink->handle);
-/*
+initproperties(dssink);
-pthread_mutex_lock(&ds_mutex);
+get_PopulateIntfcProperties(dssink);
-pthread_cond_signal(&ds_condition);
+cmd = WAIT;
-pthread_mutex_unlock(&ds_mutex);
+}
-*/
-cmd = CLOSE;
-}
 }
 else
 {
-pthread_mutex_lock(&ds_mutex);
+cmd = WAIT;
-pthread_cond_wait(&ds_condition, &ds_mutex);
-pthread_mutex_unlock(&ds_mutex);
 }
 }
 break;
 case CLOSE:
 {
-close_devsound(devsound);
+close_devsound(dssink);
-devsound->handle= NULL;
+dssink->handle= NULL;
 pthread_mutex_lock(&ds_mutex);
 pthread_cond_signal(&ds_condition);
 pthread_mutex_unlock(&ds_mutex);
 // TODO obtain mutex here
 consumer_thread_state = CONSUMER_THREAD_UNINITIALIZED;
 }
 static gboolean gst_sink_start (GstBaseSink * sink)
 {
 GstBuffer *tmp_gstbuffer=NULL;
-GstDevsoundSink *devsound = GST_DEVSOUND_SINK(sink);
+GstDevsoundSink *dssink = GST_DEVSOUND_SINK(sink);
 if(buffer_queue)
 {
 while (buffer_queue->length)
 {
 buffer_queue = g_queue_new();
 }
 consumer_thread_state = CONSUMER_THREAD_INITIALIZING;
 cmd = OPEN;
-pthread_create(&ds_thread,  NULL, StartDevSoundThread, (void *)devsound);
+pthread_create(&ds_thread,  NULL, StartDevSoundThread, (void *)dssink);
 // Wait until consumer thread is created
 // TODO : obtain mutex to retreive thread state?
 if (consumer_thread_state == CONSUMER_THREAD_INITIALIZING)
 {
 }
 static gboolean gst_sink_stop (GstBaseSink * sink)
 {
 GstBuffer *tmp_gstbuffer=NULL;
-GstDevsoundSink *devsound = GST_DEVSOUND_SINK(sink);
+GstDevsoundSink *dssink = GST_DEVSOUND_SINK(sink);
 cmd = CLOSE;
 pthread_mutex_lock(&ds_mutex);
 pthread_cond_signal(&ds_condition);
 pthread_mutex_unlock(&ds_mutex);
-GST_OBJECT_LOCK(devsound);
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_wait(&ds_condition, &ds_mutex);
+pthread_mutex_unlock(&ds_mutex);
+GST_OBJECT_LOCK(dssink);
 while (buffer_queue->length)
 {
 tmp_gstbuffer = (GstBuffer*)g_queue_pop_tail(buffer_queue);
 gst_buffer_unref(tmp_gstbuffer);
 }
 g_queue_free(buffer_queue);
 buffer_queue = NULL;
-GST_OBJECT_UNLOCK(devsound);
+GST_OBJECT_UNLOCK(dssink);
 return TRUE;
 }
 static GstFlowReturn gst_sink_render (GstBaseSink * sink,
 GstBuffer * buffer)
 {
-GstDevsoundSink *devsound = GST_DEVSOUND_SINK(sink);
+GstDevsoundSink *dssink = GST_DEVSOUND_SINK(sink);
 GstBuffer* tmp;
-if (get_ds_cb_error(devsound->handle))
+if (get_ds_cb_error(dssink->handle))
 {
 return GST_FLOW_CUSTOM_ERROR;
 }
 tmp = gst_buffer_copy(buffer);
-GST_OBJECT_LOCK (devsound);
+GST_OBJECT_LOCK (dssink);
 g_queue_push_tail (buffer_queue, tmp);
-GST_OBJECT_UNLOCK (devsound);
+GST_OBJECT_UNLOCK (dssink);
-cmd = WRITEDATA;
+cmd = PLAYING;
 pthread_mutex_lock(&ds_mutex);
 pthread_cond_signal(&ds_condition);
 pthread_mutex_unlock(&ds_mutex);
 return GST_FLOW_OK;
 }
 static void gst_devsound_sink_finalise(GObject * object)
 {
-GstDevsoundSink *devsoundsink= GST_DEVSOUND_SINK (object);
+GstDevsoundSink *devsoundsink = GST_DEVSOUND_SINK (object);
 g_free(devsoundsink->device);
 }
 static void gst_devsound_sink_set_property(GObject * object, guint prop_id,
 {
 gst_caps_unref(sink->probed_caps);
 sink->probed_caps = NULL;
 }
 break;
-case CHANNELS:
+/*        case CHANNELS:
 sink->channels = g_value_get_int(value);
 sink->pending.channelsupdate = TRUE;
 break;
 case RATE:
 sink->rate = g_value_get_int(value);
 //Convert rate to something devsound understands
 sink->rate = gst_devsound_sink_get_rate(sink->rate);
 sink->pending.rateupdate = TRUE;
 break;
-case VOLUME:
+*/        case VOLUME:
 sink->volume = g_value_get_int(value);
 sink->pending.volumeupdate = TRUE;
 break;
 case LEFTBALANCE:
 sink->leftbalance = g_value_get_int(value);
 break;
 case PREFERENCE:
 sink->preference = g_value_get_int(value);
 sink->pending.preferenceupdate = TRUE;
 break;
-case FOURCC: //FOURCC is not needed
+/*        case FOURCC: //FOURCC is not needed
 sink->fourcc = g_value_get_int(value);
 sink->pending.fourccupdate = TRUE;
 break;
 case MIMETYPE:
 sink->mimetype = g_value_dup_string(value);
-break;
+break;*/
 case OUTPUTDEVICE:
 sink->output = g_value_get_int(value);
 sink->pending.outputupdate = TRUE;
 break;
 default:
 switch (prop_id)
 {
 case PROP_DEVICE:
 g_value_set_string(value, sink->device);
 break;
-case CHANNELS:
+/*        case CHANNELS:
 g_value_set_int(value, sink->channels);
 break;
 case RATE:
 g_value_set_int(value, sink->rate);
-break;
+break;*/
 case VOLUME:
 g_value_set_int(value, sink->volume);
 break;
 case MAXVOLUME:
 g_value_set_int(value, sink->maxvolume);
 break;
-case SAMPLESPLAYED:
+/*        case SAMPLESPLAYED:
 g_value_set_int(value, sink->samplesplayed);
-break;
+break;*/
 case OUTPUTDEVICE:
 g_value_set_int(value, sink->output);
 break;
 default:
 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
 }
 else if (!strncmp(mimetype, "audio/g729", 10))
 {
 devsoundsink->fourcc = 0x39323747; //KMccFourCCIdG729
 }
-else if (!strncmp(mimetype, "audio/mp3", 9))
+else if (!strncmp(mimetype, "audio/mpeg", 10))
 {
 devsoundsink->fourcc = 0x33504d20; //KMMFFourCCCodeMP3
 }
 else
 {
 static gboolean gst_devsound_sink_event(GstBaseSink *asink, GstEvent *event)
 {
-GstDevsoundSink *sink= GST_DEVSOUND_SINK (asink);
+GstDevsoundSink *sink = GST_DEVSOUND_SINK (asink);
 GstBuffer* lastBuffer = NULL;
 switch (GST_EVENT_TYPE (event))
 {
 case GST_EVENT_EOS:
 // end-of-stream, we should close down all stream leftovers here
 //reset_devsound(sink->handle);
+sink->eosreceived = TRUE;
 if(buffer_queue->length)
 {
 GST_OBJECT_LOCK(sink);
 lastBuffer = GST_BUFFER(g_queue_peek_tail(buffer_queue));
 GST_BUFFER_FLAG_SET(lastBuffer,GST_BUFFER_FLAG_LAST);
 lastBuffer = gst_buffer_new();
 GST_BUFFER_FLAG_SET(lastBuffer,GST_BUFFER_FLAG_LAST);
 GST_OBJECT_LOCK(sink);
 g_queue_push_tail(buffer_queue,lastBuffer);
 GST_OBJECT_UNLOCK(sink);
-cmd = WRITEDATA;
+cmd = PLAYING;
 pthread_mutex_lock(&ds_mutex);
 pthread_cond_signal(&ds_condition);
 pthread_mutex_unlock(&ds_mutex);
 }
 pthread_mutex_lock(&ds_mutex);
 }
 return TRUE;
 }
+#ifdef AV_SYNC
+static void gst_devsound_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
+GstClockTime * start, GstClockTime * end)
+{
+/* Like GstBaseAudioSink, we set these to NONE */
+*start = GST_CLOCK_TIME_NONE;
+*end = GST_CLOCK_TIME_NONE;
+}
+static GstClock *gst_devsound_sink_provide_clock (GstElement * element)
+{
+GstDevsoundSink *sink = GST_DEVSOUND_SINK (element);
+return GST_CLOCK (gst_object_ref (sink->clock));
+}
+static GstClockTime gst_devsound_sink_get_time (GstClock * clock, gpointer user_data)
+{
+GstClockTime result = 0;
+GstDevsoundSink *sink = GST_DEVSOUND_SINK (user_data);
+/* The value returned must be in nano seconds. 1 sec = 1000000000 nano seconds (9 zeros)*/
+/*If time played is available from DevSound (a3f devsound onwards) get it*/
+if (sink->timeplayedavailable)
+{
+result = sink->time_or_samples_played;
+}
+else if ((sink->time_or_samples_played > 0 ) && (sink->rate > 0 ))/*This is a pre-a3f devsound. So calculate times played based on samples played*/
+{ /*GST_SECOND = 1000000000*/
+result = gst_util_uint64_scale_int (sink->time_or_samples_played, GST_SECOND, sink->rate);
+}
+GST_LOG_OBJECT (sink, "Time: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
+return result;
+}
+#endif /*AV_SYNC*/
+static GstStateChangeReturn gst_devsound_sink_change_state (GstElement * element, GstStateChange transition)
+{
+GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+GstDevsoundSink *sink= GST_DEVSOUND_SINK (element);
+switch (transition)
+{
+case GST_STATE_CHANGE_NULL_TO_READY:
+{
+#ifdef AV_SYNC
+sink->time_or_samples_played = 0;
+#endif /*AV_SYNC*/
+}
+break;
+case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+if(cmd == WAIT)
+{
+cmd = RESUME;
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_signal(&ds_condition);
+pthread_mutex_unlock(&ds_mutex);
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_wait(&ds_condition, &ds_mutex);
+pthread_mutex_unlock(&ds_mutex);
+}
+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;
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_signal(&ds_condition);
+pthread_mutex_unlock(&ds_mutex);
+pthread_mutex_lock(&ds_mutex);
+pthread_cond_wait(&ds_condition, &ds_mutex);
+pthread_mutex_unlock(&ds_mutex);
+break;
+default:
+break;
+}
+return ret;
+activate_failed:
+{
+GST_DEBUG_OBJECT (sink,
+"element failed to change states -- activation problem?");
+return GST_STATE_CHANGE_FAILURE;
+}
+}
 /************************************
 * Error Concealment Interface begins
 * **********************************/
 static void gst_error_concealment_handler_init (gpointer g_iface,
 }
 static gint gst_BadLsfNextBuffer()
 {
 customInfaceUpdate.g729badlsfnextbufferupdate = TRUE;
+return 0;
 }
 static void gst_Apply_G729_Decoder_Update(GstDevsoundSink* dssink )
 {
 if(customInfaceUpdate.g729badlsfnextbufferupdate)
 {
 set_ilbcdecodermode(dssink->handle,ilbcdecodemode);
 customInfaceUpdate.ilbcdecodermodeupdate = FALSE;
 }
 }
+static void get_PopulateIntfcProperties(GstDevsoundSink* dssink)
+{
+framemode_rqrd_for_ec(dssink->handle,&framemodereq);
+get_cng(dssink->handle,&g711cng);
+get_ilbccng(dssink->handle,&ilbccng);
+}

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