# HG changeset patch # User Dremov Kirill (Nokia-D-MSW/Tampere) # Date 1268984109 -7200 # Node ID 71e347f905f200acb7ab39dc317ce83bd99797ab # Parent 5505e89089447d368cbe9331bfdcfde389f6c9b6 Revision: 201007 Kit: 201011 diff -r 5505e8908944 -r 71e347f905f2 data/Create_GStreamer_STUB_SIS.bat --- 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 diff -r 5505e8908944 -r 71e347f905f2 data/Gstreamer_Stub.pkg --- 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 ; diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_good/group/gstcamerabin.mmp --- 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 diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_good/group/gstphotography.mmp --- 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 diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/devsoundsinkwrapper.cpp --- 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; } diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/devsoundsinkwrapper.h --- 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 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 diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/devsoundsrcwrapper.cpp --- 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; } /*******************************************************************/ diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/devsoundsrcwrapper.h --- 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); diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/gstdevsoundsink.c --- 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 +#ifdef AV_SYNC +#include +#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); + } + diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/gstdevsoundsink.h --- 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 #include @@ -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__ */ + diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/gstdevsoundsrc.c --- 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; + } + diff -r 5505e8908944 -r 71e347f905f2 gst_plugins_symbian/gst/devsound/gstdevsoundsrc.h --- 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; diff -r 5505e8908944 -r 71e347f905f2 gstreamer_core/gst/gstinterface.h --- 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 */ diff -r 5505e8908944 -r 71e347f905f2 gstregistrygenerator/group/gstregistrygenerator.mmp --- 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