--- a/data/Create_GStreamer_STUB_SIS.bat Fri Jan 22 09:59:59 2010 +0200
+++ b/data/Create_GStreamer_STUB_SIS.bat Fri Mar 19 09:35:09 2010 +0200
@@ -1,16 +1,21 @@
rem
-rem Copyright (c) 2009 Nokia Corporation and/or its subsidiary(-ies).
-rem All rights reserved.
-rem This component and the accompanying materials are made available
-rem under the terms of the License "Symbian Foundation License v1.0"
-rem which accompanies this distribution, and is available
-rem at the URL "http://www.symbianfoundation.org/legal/sfl-v10.html".
+rem Copyright (c) 2009 Nokia Corporation and/or its subsidiary(-ies). All rights reserved.
+rem
+rem This library is free software; you can redistribute it and/or
+rem modify it under the terms of the GNU Lesser General Public
+rem License as published by the Free Software Foundation; either
+rem version 2 of the License, or (at your option) any later version.
rem
-rem Initial Contributors:
-rem Nokia Corporation - initial contribution.
+rem This library is distributed in the hope that it will be useful,
+rem but WITHOUT ANY WARRANTY; without even the implied warranty of
+rem MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+rem Lesser General Public License for more details.
rem
-rem Contributors:
-rem
+rem You should have received a copy of the GNU Lesser General Public
+rem License along with this library; if not, write to the
+rem Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+rem Boston, MA 02111-1307, USA.
+rem
rem Description: PKG for GStreamer
rem
--- a/data/Gstreamer_Stub.pkg Fri Jan 22 09:59:59 2010 +0200
+++ b/data/Gstreamer_Stub.pkg Fri Mar 19 09:35:09 2010 +0200
@@ -1,15 +1,20 @@
;
-; Copyright (c) 2009 Nokia Corporation and/or its subsidiary(-ies).
-; All rights reserved.
-; This component and the accompanying materials are made available
-; under the terms of the License "Symbian Foundation License v1.0"
-; which accompanies this distribution, and is available
-; at the URL "http://www.symbianfoundation.org/legal/sfl-v10.html".
+; Copyright (c) 2009 Nokia Corporation and/or its subsidiary(-ies). All rights reserved.
+;
+; This library is free software; you can redistribute it and/or
+; modify it under the terms of the GNU Lesser General Public
+; License as published by the Free Software Foundation; either
+; version 2 of the License, or (at your option) any later version.
;
-; Initial Contributors:
-; Nokia Corporation - initial contribution.
+; This library is distributed in the hope that it will be useful,
+; but WITHOUT ANY WARRANTY; without even the implied warranty of
+; MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+; Lesser General Public License for more details.
;
-; Contributors:
+; You should have received a copy of the GNU Lesser General Public
+; License along with this library; if not, write to the
+; Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+; Boston, MA 02111-1307, USA.
;
; Description: GStreamer multimedia framework
;
--- a/gst_plugins_good/group/gstcamerabin.mmp Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_good/group/gstcamerabin.mmp Fri Mar 19 09:35:09 2010 +0200
@@ -1,16 +1,25 @@
// Gstreamer.MMP
/*
- * Copyright © 2008 Nokia Corporation.
- * This material, including documentation and any related
- * computer progrs, is protected by copyright controlled by
- * Nokia Corporation. All rights are reserved. Copying,
- * including reproducing, storing, adapting or translating, any
- * or all of this material requires the prior written consent of
- * Nokia Corporation. This material also contains confidential
- * information which may not be disclosed to others without the
- * prior written consent of Nokia Corporation.
- * ============================================================================
- */
+* Copyright (c) 2009 Nokia Corporation and/or its subsidiary(-ies). All rights reserved.
+*
+* This library is free software; you can redistribute it and/or
+* modify it under the terms of the GNU Lesser General Public
+* License as published by the Free Software Foundation; either
+* version 2 of the License, or (at your option) any later version.
+*
+* This library is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+* Lesser General Public License for more details.
+*
+* You should have received a copy of the GNU Lesser General Public
+* License along with this library; if not, write to the
+* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+* Boston, MA 02111-1307, USA.
+*
+* Description:
+*
+*/
#include <platform_paths.hrh>
--- a/gst_plugins_good/group/gstphotography.mmp Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_good/group/gstphotography.mmp Fri Mar 19 09:35:09 2010 +0200
@@ -1,16 +1,24 @@
-// Gstreamer.MMP
/*
- * Copyright © 2008 Nokia Corporation.
- * This material, including documentation and any related
- * computer progrs, is protected by copyright controlled by
- * Nokia Corporation. All rights are reserved. Copying,
- * including reproducing, storing, adapting or translating, any
- * or all of this material requires the prior written consent of
- * Nokia Corporation. This material also contains confidential
- * information which may not be disclosed to others without the
- * prior written consent of Nokia Corporation.
- * ============================================================================
- */
+* Copyright (c) 2009 Nokia Corporation and/or its subsidiary(-ies). All rights reserved.
+*
+* This library is free software; you can redistribute it and/or
+* modify it under the terms of the GNU Lesser General Public
+* License as published by the Free Software Foundation; either
+* version 2 of the License, or (at your option) any later version.
+*
+* This library is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+* Lesser General Public License for more details.
+*
+* You should have received a copy of the GNU Lesser General Public
+* License along with this library; if not, write to the
+* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+* Boston, MA 02111-1307, USA.
+*
+* Description:
+*
+*/
#include <platform_paths.hrh>
--- a/gst_plugins_symbian/gst/devsound/devsoundsinkwrapper.cpp Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/devsoundsinkwrapper.cpp Fri Mar 19 09:35:09 2010 +0200
@@ -93,7 +93,7 @@
{
TRequestStatus* stat = &(AL->iStatus);
User::RequestComplete(stat, aError);
- iCallbackError = aError;
+ iCallbackError = 0;
}
/*******************************************************/
void DevSoundWrapper::BufferToBeEmptied(CMMFBuffer* /*aBuffer*/)
@@ -244,6 +244,41 @@
/************************************************************/
+int pause_devsound(GstDevsoundSink *ds)
+ {
+ TRACE_PRN_FN_ENT;
+ DevSoundWrapper* handle = (DevSoundWrapper*) ds->handle;
+ if(handle->dev_sound->IsResumeSupported())
+ {
+ handle->dev_sound->Pause();
+ }
+ else
+ {
+ handle->iSamplesPlayed = handle->dev_sound->SamplesPlayed();
+ handle->dev_sound->Stop();
+ }
+ TRACE_PRN_FN_EXT;
+ return 0;
+ }
+
+int resume_devsound(GstDevsoundSink *ds)
+ {
+ TRACE_PRN_FN_ENT;
+ DevSoundWrapper* handle = (DevSoundWrapper*) ds->handle;
+ if(handle->dev_sound->IsResumeSupported())
+ {
+ handle->dev_sound->Resume();
+ }
+ else
+ {
+ playinit(handle);
+ initproperties(ds);
+ }
+ TRACE_PRN_FN_EXT;
+ return 0;
+ }
+
+
int close_devsound(GstDevsoundSink *ds)
{
TRACE_PRN_FN_ENT;
@@ -560,6 +595,19 @@
{
return handle->iCallbackError;
}
+
+#ifdef AV_SYNC
+gboolean is_timeplayed_supported(DevSoundWrapper *handle)
+ {
+ gboolean retVal = FALSE;
+ if (handle->dev_sound && (handle->dev_sound)->IsGetTimePlayedSupported())
+ {
+ retVal = TRUE;
+ }
+ return retVal;
+ }
+#endif /*AV_SYNC*/
+
/*******************************************************************/
int playinit(DevSoundWrapper *handle)
@@ -569,7 +617,7 @@
((handle)->AL)->InitialiseActiveListener();
handle->eosReceived = false;
- TRAP(handle->iCallbackError,(handle->dev_sound)->PlayInitL());
+ TRAP(handle->iCallbackError,(handle->dev_sound)->PlayInitL());
if (handle->iCallbackError == KErrNone)
{
((handle)->AL)->StartActiveScheduler();
@@ -715,7 +763,19 @@
{
TRACE_PRN_FN_ENT;
DevSoundWrapper* dsPtr = STATIC_CAST(DevSoundWrapper*, ds->handle);
- ds->samplesplayed = (dsPtr->dev_sound)->SamplesPlayed();
+#ifdef AV_SYNC
+ if (dsPtr->dev_sound->IsGetTimePlayedSupported())
+ {
+ TTimeIntervalMicroSeconds timePlayedInMS = 0;
+ (dsPtr->dev_sound)->GetTimePlayed(timePlayedInMS);
+ /* store value in nano seconds */
+ ds->time_or_samples_played = timePlayedInMS.Int64() * 1000;
+ }
+ else
+ {
+ ds->time_or_samples_played += (dsPtr->dev_sound)->SamplesPlayed();
+ }
+#endif /*AV_SYNC*/
get_outputdevice(dsPtr,&ds->output);
TRACE_PRN_FN_EXT;
}
@@ -725,9 +785,6 @@
TRACE_PRN_FN_ENT;
DevSoundWrapper* dsPtr= STATIC_CAST(DevSoundWrapper*, ds->handle);
ds->maxvolume = (dsPtr->dev_sound)->MaxVolume();
- ds->volume = (dsPtr->dev_sound)->Volume();
- framemode_rqrd_for_ec(dsPtr,&ds->framemodereq);
- get_cng(dsPtr,&ds->g711cng);
- get_ilbccng(dsPtr,&ds->ilbccng);
+ ds->volume = (dsPtr->dev_sound)->Volume();
TRACE_PRN_FN_EXT;
}
--- a/gst_plugins_symbian/gst/devsound/devsoundsinkwrapper.h Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/devsoundsinkwrapper.h Fri Mar 19 09:35:09 2010 +0200
@@ -96,6 +96,7 @@
int dev_count;
TInt iCallbackError;
TUint32 fourcc;
+ TUint32 iSamplesPlayed;
bool eosReceived;
//sem_t mutex;
//RArray<TFourCC> supportedtypes;
@@ -142,10 +143,14 @@
int open_devsound(DevSoundWrapper **handle);
int open_device(DevSoundWrapper **handle);
int initialize_devsound(GstDevsoundSink* sink);
+ int pause_devsound(GstDevsoundSink *ds);
+ int resume_devsound(GstDevsoundSink *ds);
int close_devsound(GstDevsoundSink *ds);
int check_if_device_open(DevSoundWrapper *handle) ;
-
int get_ds_cb_error(DevSoundWrapper *handle);
+#ifdef AV_SYNC
+ gboolean is_timeplayed_supported(DevSoundWrapper *handle);
+#endif /*AV_SYNC*/
//Error Concealment custom interface
void conceal_error_for_next_buffer(DevSoundWrapper *handle);
@@ -175,7 +180,6 @@
int pre_init_setconf(GstDevsoundSink *ds);
void getsupporteddatatypes(GstDevsoundSink *ds);
-
#ifdef __cplusplus
}//extern c
#endif
--- a/gst_plugins_symbian/gst/devsound/devsoundsrcwrapper.cpp Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/devsoundsrcwrapper.cpp Fri Mar 19 09:35:09 2010 +0200
@@ -212,6 +212,49 @@
}
/*********************************************************/
+int stop_devsound(GstDevsoundSrc *ds)
+ {
+ TRACE_PRN_FN_ENT;
+ DevSoundWrapperSrc* handle = (DevSoundWrapperSrc*) ds->handle;
+ handle->dev_sound->Stop();
+ TRACE_PRN_FN_EXT;
+ return 0;
+ }
+
+int pause_devsound(GstDevsoundSrc *ds)
+ {
+ TRACE_PRN_FN_ENT;
+ DevSoundWrapperSrc* handle = (DevSoundWrapperSrc*) ds->handle;
+ if(handle->dev_sound->IsResumeSupported())
+ {
+ handle->dev_sound->Pause();
+ }
+ else
+ {
+ handle->iSamplesRecorded = handle->dev_sound->SamplesRecorded();
+ handle->dev_sound->Stop();
+ }
+ TRACE_PRN_FN_EXT;
+ return 0;
+ }
+
+int resume_devsound(GstDevsoundSrc *ds)
+ {
+ TRACE_PRN_FN_ENT;
+ DevSoundWrapperSrc* handle = (DevSoundWrapperSrc*) ds->handle;
+ if(handle->dev_sound->IsResumeSupported())
+ {
+ handle->dev_sound->Resume();
+ }
+ else
+ {
+ recordinit(handle);
+ initproperties(ds);
+ }
+ TRACE_PRN_FN_EXT;
+ return 0;
+ }
+
int open_device(DevSoundWrapperSrc **handle)
{
int retcode = KErrNone;
@@ -538,11 +581,13 @@
void set_rate(DevSoundWrapperSrc *handle, int rate)
{
handle->caps.iRate = rate;
+ TRACE_PRN_N1(_L("set_rate %d"),rate);
}
/******************************************************************/
void set_channels(DevSoundWrapperSrc *handle, int channels)
{
handle->caps.iChannels = channels;
+ TRACE_PRN_N1(_L("set_channels %d"),channels);
}
/****************************************************************/
void set_encoding(DevSoundWrapperSrc *handle, int encoding)
@@ -557,7 +602,10 @@
/*****************************************************************/
void set_fourcc(DevSoundWrapperSrc *handle, int fourcc)
{
+ TRACE_PRN_FN_ENT;
handle->fourcc = fourcc;
+ TRACE_PRN_N1(_L("set_fourcc %d"),fourcc);
+ TRACE_PRN_FN_EXT;
}
/*******************************************************************/
--- a/gst_plugins_symbian/gst/devsound/devsoundsrcwrapper.h Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/devsoundsrcwrapper.h Fri Mar 19 09:35:09 2010 +0200
@@ -97,6 +97,7 @@
TUint32 fourcc;
int bufferreadpos;
guint* supportedbitrates;
+ int iSamplesRecorded;
CSpeechEncoderConfig* iSpeechEncoderConfig;
CG711EncoderIntfc* iG711EncoderIntfc;
CG729EncoderIntfc* iG729EncoderIntfc;
@@ -141,6 +142,9 @@
int open_devsound(DevSoundWrapperSrc **handle);
int open_device(DevSoundWrapperSrc **handle);
int initialize_devsound(GstDevsoundSrc* ds);
+ int pause_devsound(GstDevsoundSrc *ds);
+ int stop_devsound(GstDevsoundSrc *ds);
+ int resume_devsound(GstDevsoundSrc *ds);
int close_devsound(GstDevsoundSrc* ds);
int SetConfigurations(DevSoundWrapperSrc *handle);
--- a/gst_plugins_symbian/gst/devsound/gstdevsoundsink.c Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/gstdevsoundsink.c Fri Mar 19 09:35:09 2010 +0200
@@ -32,6 +32,10 @@
#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);
#define GST_CAT_DEFAULT devsound_debug
@@ -59,12 +63,21 @@
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 gboolean gst_devsound_sink_event(GstBaseSink * asink, GstEvent * event);
+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);
@@ -97,6 +110,7 @@
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);
@@ -160,23 +174,26 @@
VOLUME,
MAXVOLUME,
VOLUMERAMP,
- CHANNELS,
+/* CHANNELS,*/
LEFTBALANCE,
RIGHTBALANCE,
- RATE,
+/* RATE,*/
PRIORITY,
PREFERENCE,
- SAMPLESPLAYED,
- FOURCC, //FOURCC is not needed
- MIMETYPE,
+/* SAMPLESPLAYED,*/
+/* FOURCC, //FOURCC is not needed*/
+/* MIMETYPE,*/
OUTPUTDEVICE
};
enum command_to_consumer_thread_enum
{
OPEN = 2,
- WRITEDATA,
+ PLAYING,
+ PAUSE,
+ RESUME,
/*UPDATE,*/
+ WAIT,
CLOSE
};
enum command_to_consumer_thread_enum cmd;
@@ -185,33 +202,14 @@
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- 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 ]; "
- "audio/amr, "
- //"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 ]")
+ 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 ]; "
+ "audio/amr, " "rate = (int) 8000, " "channels = (int) 1 ; "
+ "audio/AMR, " "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;
@@ -284,14 +282,20 @@
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);
}
@@ -324,7 +328,10 @@
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));
@@ -348,11 +355,11 @@
g_object_class_install_property(gobject_class, RIGHTBALANCE,
g_param_spec_int("rightbalance", "Right Balance", "Right Balance",
-1, G_MAXINT, -1, 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,
@@ -362,7 +369,7 @@
g_param_spec_int("preference", "Preference", "Preference ", -1,
G_MAXINT, -1,
G_PARAM_READWRITE));
-
+/*
g_object_class_install_property(gobject_class, RATE,
g_param_spec_int("rate", "Rate", "Rate ", -1,
G_MAXINT, -1,
@@ -372,12 +379,16 @@
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));
+#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);
@@ -385,35 +396,47 @@
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);
+#endif /*AV_SYNC*/
}
-static void gst_devsound_sink_init(GstDevsoundSink * devsoundsink)
+static void gst_devsound_sink_init(GstDevsoundSink * dssink)
{
- GST_DEBUG_OBJECT(devsoundsink, "initializing devsoundsink");
- devsoundsink->device = g_strdup(DEFAULT_DEVICE);
- devsoundsink->handle = NULL;
- devsoundsink->preference = 0; //default=>EMdaPriorityPreferenceNone;
- devsoundsink->priority = 0; //default=>EMdaPriorityNormal;
+ GST_DEBUG_OBJECT(dssink, "initializing devsoundsink");
+ 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 *devsound;
+ GstDevsoundSink *dssink;
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;
@@ -438,82 +461,94 @@
{
//TODO if there is preemption we have to somehow signal
//the pipeline in the render
- initialize_devsound(devsound);
+ initialize_devsound(dssink);
- playinit(devsound->handle);
- initproperties(devsound);
+ playinit(dssink->handle);
+ 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);
- gst_Apply_ErrorConcealment_Update(devsound);
- gst_Apply_G711_Decoder_Update(devsound);
- gst_Apply_G729_Decoder_Update(devsound);
- gst_Apply_Ilbc_Decoder_Update(devsound);
+ pre_init_setconf(dssink);
+ gst_Apply_ErrorConcealment_Update(dssink);
+ gst_Apply_G711_Decoder_Update(dssink);
+ gst_Apply_G729_Decoder_Update(dssink);
+ gst_Apply_Ilbc_Decoder_Update(dssink);
// TODO we could do this in BTBF callback
- populateproperties(devsound);
-
- framemodereq = devsound->framemodereq;
- g711cng = devsound->g711cng;
- ilbccng = devsound->ilbccng;
- output = devsound->output;
-
+ populateproperties(dssink);
+ get_PopulateIntfcProperties(dssink);
+
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
- // and we have received PlayError so now we exit
- // from the big loop next time
-/*
- pthread_mutex_lock(&ds_mutex);
- pthread_cond_signal(&ds_condition);
- pthread_mutex_unlock(&ds_mutex);
-*/
- cmd = CLOSE;
- }
+ lastBufferSet = FALSE;
+ dssink->eosreceived = FALSE;
+ playinit(dssink->handle);
+ initproperties(dssink);
+ get_PopulateIntfcProperties(dssink);
+ cmd = WAIT;
+ }
}
else
{
- pthread_mutex_lock(&ds_mutex);
- pthread_cond_wait(&ds_condition, &ds_mutex);
- pthread_mutex_unlock(&ds_mutex);
+ cmd = WAIT;
}
}
break;
case CLOSE:
{
- close_devsound(devsound);
- devsound->handle= NULL;
+ close_devsound(dssink);
+ dssink->handle= NULL;
pthread_mutex_lock(&ds_mutex);
pthread_cond_signal(&ds_condition);
pthread_mutex_unlock(&ds_mutex);
@@ -537,7 +572,7 @@
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)
{
@@ -557,7 +592,7 @@
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?
@@ -574,7 +609,7 @@
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;
@@ -582,7 +617,12 @@
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);
@@ -590,7 +630,7 @@
}
g_queue_free(buffer_queue);
buffer_queue = NULL;
- GST_OBJECT_UNLOCK(devsound);
+ GST_OBJECT_UNLOCK(dssink);
return TRUE;
}
@@ -598,21 +638,21 @@
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);
@@ -622,7 +662,7 @@
static void gst_devsound_sink_finalise(GObject * object)
{
- GstDevsoundSink *devsoundsink= GST_DEVSOUND_SINK (object);
+ GstDevsoundSink *devsoundsink = GST_DEVSOUND_SINK (object);
g_free(devsoundsink->device);
}
@@ -646,7 +686,7 @@
sink->probed_caps = NULL;
}
break;
- case CHANNELS:
+/* case CHANNELS:
sink->channels = g_value_get_int(value);
sink->pending.channelsupdate = TRUE;
break;
@@ -656,7 +696,7 @@
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;
@@ -680,14 +720,13 @@
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;
@@ -710,21 +749,21 @@
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;
@@ -923,14 +962,14 @@
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);
@@ -945,7 +984,7 @@
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);
@@ -962,6 +1001,103 @@
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
@@ -1142,3 +1278,13 @@
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);
+ }
+
--- a/gst_plugins_symbian/gst/devsound/gstdevsoundsink.h Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/gstdevsoundsink.h Fri Mar 19 09:35:09 2010 +0200
@@ -28,7 +28,6 @@
#ifndef __GST_DEVSOUNDSINK_H__
#define __GST_DEVSOUNDSINK_H__
-
#include <gst/gst.h>
#include <gst/base/gstbasesink.h>
@@ -41,7 +40,7 @@
#define GST_IS_DEVSOUND_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DEVSOUND_SINK))
#define GST_IS_DEVSOUND_SINK_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DEVSOUND_SINK))
-
+//#define AV_SYNC
typedef struct _GstDevsoundSink GstDevsoundSink;
typedef struct _GstDevsoundSinkClass GstDevsoundSinkClass;
@@ -49,51 +48,55 @@
typedef struct _GstDevsoundUpdate GstDevsoundUpdate;
struct _GstDevsoundUpdate{
-gboolean channelsupdate;
-gboolean rateupdate;
-gboolean volumeupdate;
-gboolean volumerampupdate;
-gboolean leftbalanceupdate;
-gboolean rightbalanceupdate;
-gboolean preferenceupdate;
-gboolean priorityupdate;
-gboolean fourccupdate;
-gboolean outputupdate;
+ gboolean channelsupdate;
+ gboolean rateupdate;
+ gboolean volumeupdate;
+ gboolean volumerampupdate;
+ gboolean leftbalanceupdate;
+ gboolean rightbalanceupdate;
+ gboolean preferenceupdate;
+ gboolean priorityupdate;
+ gboolean fourccupdate;
+ gboolean outputupdate;
};
struct _GstDevsoundSink {
- GstBaseSink sink;
-
- void *handle;
- void *dataptr;
- gchar *device;
- gint bytes_per_sample;
- GstCaps *probed_caps;
+ GstBaseSink sink;
- GstDevsoundUpdate pending;
+ void *handle;
+ void *dataptr;
+ gchar *device;
+ gint bytes_per_sample;
+ GstCaps *probed_caps;
+
+ GstDevsoundUpdate pending;
- //properties
- gint channels;
- gint rate;
- gint volume;
- gint volumeramp;
- gint maxvolume;
- gint leftbalance;
- gint rightbalance;
- gint priority;
- gint preference;
- gint samplesplayed;
- gint output;
- gulong fourcc;
- gchar *mimetype;
- GList *fmt;
- gboolean framemodereq;
- gboolean g711cng;
- gboolean ilbccng;
+ //properties
+ gint channels;
+ gint rate;
+ gint volume;
+ gint volumeramp;
+ gint maxvolume;
+ gint leftbalance;
+ gint rightbalance;
+ gint priority;
+ gint preference;
+ gint output;
+ gulong fourcc;
+ gchar *mimetype;
+ GList *fmt;
+
+ gboolean eosreceived;
+
+#ifdef AV_SYNC
+ gboolean timeplayedavailable;
+ gulong time_or_samples_played;
+ GstClock *clock; /* The clock for this element. */
+#endif /*AV_SYNC*/
};
struct _GstDevsoundSinkClass {
- GstBaseSinkClass parent_class;
+ GstBaseSinkClass parent_class;
};
GType gst_devsound_sink_get_type(void);
@@ -101,3 +104,4 @@
G_END_DECLS
#endif /* __GST_DEVSOUNDSINK_H__ */
+
--- a/gst_plugins_symbian/gst/devsound/gstdevsoundsrc.c Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/gstdevsoundsrc.c Fri Mar 19 09:35:09 2010 +0200
@@ -71,6 +71,11 @@
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
* ******************************/
@@ -164,7 +169,10 @@
enum command_to_consumer_thread_enum
{
OPEN = 2,
- READDATA,
+ RECORDING,
+ PAUSE,
+ RESUME,
+ STOP,
/*UPDATE,*/
CLOSE
};
@@ -319,6 +327,8 @@
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));
@@ -362,17 +372,19 @@
g_param_spec_int("channels", "Channels", "Channels ", -1,
G_MAXINT, -1,
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;
@@ -423,71 +435,50 @@
recordinit(devsoundsrc->handle);
initproperties(devsoundsrc);
}
- //cmd = READDATA;
- while (1)
+
+ while (TRUE)
{
- //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);
- }
- else
- {
- //gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) devsoundsrc, "Push buffer alloc failed ",NULL);
- }
-
- if (dataqueue)
- {
- 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;
@@ -502,21 +493,25 @@
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;
+ consumer_thread_state = CONSUMER_THREAD_UNINITIALIZED;
pthread_exit(NULL);
break;
}
+ 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);
}
- // TODO obtain mutex here
- consumer_thread_state = CONSUMER_THREAD_UNINITIALIZED;
- pthread_exit(NULL);
}
static void gst_devsound_src_set_property(GObject * object, guint prop_id,
@@ -692,7 +687,7 @@
if(dataqueue)
{
- while (dataqueue->length)
+ while (g_queue_get_length(dataqueue))
{
tmp_gstbuffer = (GstBuffer*)g_queue_pop_tail(dataqueue);
gst_buffer_unref(tmp_gstbuffer);
@@ -731,6 +726,9 @@
//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));
@@ -746,7 +744,7 @@
//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);
@@ -764,7 +762,6 @@
pthread_mutex_destroy(&create_mutex1);
pthread_cond_destroy(&(create_condition1));
-
g_free(src->device);
//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) src, "gst_devsound_src_stop EXIT ");
return TRUE;
@@ -794,6 +791,16 @@
{
GstDevsoundSrc *dsrc= GST_DEVSOUND_SRC(src);
int bufferpos=0;
+
+ if(!g_queue_get_length(dataqueue) && (dsrc->eosreceived == TRUE))
+ {
+ pthread_mutex_lock(&(create_mutex1));
+ pthread_cond_signal(&(create_condition1));
+ pthread_mutex_unlock(&(create_mutex1));
+
+ 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);
@@ -839,26 +846,36 @@
// 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;
- 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(dsrc->eosreceived == TRUE)
+ {
+ return GST_FLOW_UNEXPECTED;
+ }
+ else
+ {
+ cmd = RECORDING;
+ 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));
+ }
//gst_debug_log(devsound_debug, GST_LEVEL_LOG, "", "", 0, (GObject *) dsrc, "AFTER WAIT in CREATE ",NULL);
}
-
+
//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)
+ {
+ return GST_FLOW_UNEXPECTED;
+ }
// 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
@@ -893,6 +910,63 @@
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(cmd == PAUSE)
+ {
+ cmd = RESUME;
+ 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);
+ }
+ 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(&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);
+ 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);
@@ -1137,3 +1211,40 @@
}
+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;
+ 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;
+ }
+
--- a/gst_plugins_symbian/gst/devsound/gstdevsoundsrc.h Fri Jan 22 09:59:59 2010 +0200
+++ b/gst_plugins_symbian/gst/devsound/gstdevsoundsrc.h Fri Mar 19 09:35:09 2010 +0200
@@ -77,6 +77,7 @@
gint samplesrecorded;
GList* fmt;
GList* supportedbitrates;
+ gboolean eosreceived;
guint speechbitrate;
gboolean speechvadmode;
--- a/gstreamer_core/gst/gstinterface.h Fri Jan 22 09:59:59 2010 +0200
+++ b/gstreamer_core/gst/gstinterface.h Fri Mar 19 09:35:09 2010 +0200
@@ -71,6 +71,9 @@
#define GST_IMPLEMENTS_INTERFACE_CHECK_INSTANCE_TYPE(obj, type) \
(gst_implements_interface_check ((obj), (type)))
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
GType gst_implements_interface_get_type (void);
/* wrapper functions to check for functionality implementation */
--- a/gstregistrygenerator/group/gstregistrygenerator.mmp Fri Jan 22 09:59:59 2010 +0200
+++ b/gstregistrygenerator/group/gstregistrygenerator.mmp Fri Mar 19 09:35:09 2010 +0200
@@ -38,6 +38,8 @@
SOURCEPATH ../src
SOURCE gstregistrygenerator.cpp
+STATICLIBRARY libcrt0.lib
+
LIBRARY libc.lib
LIBRARY libpthread.lib
LIBRARY libdl.lib
@@ -52,6 +54,8 @@
LIBRARY libgstbase.lib
LIBRARY libgstcontroller.lib
+LIBRARY euser.lib
+
#ifdef ENABLE_ABIV2_MODE
DEBUGGABLE_UDEBONLY
#endif