...
| Code Block | ||||
|---|---|---|---|---|
| ||||
diff --git a/partner/dpi/gstreamer/CMakeLists.txt b/partner/dpi/gstreamer/CMakeLists.txt index 7081489..c9072d01e5ac32 100644 --- a/partner/dpi/gstreamer/CMakeLists.txt +++ b/partner/dpi/gstreamer/CMakeLists.txt @@ -204,710 +204,714 @@ if(FORCE_HDR_DV_MODE) add_definitions("-DFORCE_HDR_DV_MODE=1") endif() -if(FP_FAKE_INITBUFFER) +if(FP_FAKE_INITBUFFER AND NOT USE_RIALTO) add_definitions("-DFP_FAKE_INITBUFFER=1") endif() diff --git a/partner/dpi/gstreamer/PlaybackGroupNative.cpp b/partner/dpi/gstreamer/PlaybackGroupNative.cpp index 3c7d557..652f4aa 100644 --- a+if(USE_RIALTO) + add_definitions("-DUSE_RIALTO=1") +endif() + if(USE_FIRST_VFRAME_CALLBACK) add_definitions("-DUSE_FIRST_VFRAME_CALLBACK=1") endif() diff --git a/partner/dpi/gstreamer/PlaybackGroupNativeESPlayerGstGenericPort.cpp +++ b/partner/dpi/gstreamer/PlaybackGroupNativeESPlayerGstGenericPort.cpp @@ -34,7 +34,7 @@ extern void ApplicationManagerContentStopped(void); #if defined(GST_GL) #include "VideoSinkGStreamer.h" #endif -#define VIDEO_SINK_NAME "westerossink" +#define VIDEO_SINK_NAME "rialtomsevideosink" #define VIDEO_DECODER_NAME "westerossink" #define TOO_FAR_AHEAD_THRESHOLD_SECONDS 20 #define UHD_WITDH 3840 @@ -304,7 +304,7 @@ PlaybackGroupNative::PlaybackGroupNative(ESManagerNative& ESManager, uint32_t pi index d66b1bc..65ad350 100644 --- a/partner/dpi/gstreamer/ESPlayerGstGenericPort.cpp +++ b/partner/dpi/gstreamer/ESPlayerGstGenericPort.cpp @@ -569,9 +569,16 @@ void ESPlayerGstGenericPort::ProcessFrame() { // Lock mutex to ensure state change does not happen until seamless audio switch is completed mPlaybackGroup->AudioSwitchMutexLock(); + #ifdef USE_RIALTO + Log::infowarn(TRACE_MEDIACONTROLMEDIAPLAYBACK, "EnablingUsing audioperformAudioTrackCodecChannelSwitch downmixingfor to stereoRialto"); + g_object_set(mGstPipeline, "audio-filter", capsfilter, NULL); } - mVideoSink = createWesterosSink(); + mVideoSink = nullptr; #if defined(GST_GL) mVideoSinkGst.reset(new VideoSinkGStreamer()); g_object_set(mGstPipeline, "video-sink", mVideoSinkGst->createVideoSink(), NULL); @@ -456,6 +456,7 @@ PlaybackGroupNative::~PlaybackGroupNative() if (performAudioTrackCodecChannelSwitchRialto(&(mPlaybackGroup->mGstUtilsPlaybackGroup), pSampleAttr, pAudioAttr, + if (mSource) gst_object_unref(mSource); GstServer::instance()->disconnect(); + #if defined(WESTEROS_SINK) && (defined(GIBBON_GRAPHICSENGINE_WAYLAND) || defined(GIBBON_EVENTLOOP_WAYLAND)) pstatus, &delay, &mAudioChangeTargetPts, &mCurrentDispPts, &_audio_change_stage, &mAppsrcCaps, + gibbon::WaylandDisplay::instance()->disconnect(); #endif diff --git a/partner/dpi/gstreamer/ocdm/CMakeLists.txt b/partner/dpi/gstreamer/ocdm/CMakeLists.txt index f9f8d78..3555d12 100644 --- a/partner/dpi/gstreamer/ocdm/CMakeLists.txt +++ b/partner/dpi/gstreamer/ocdm/CMakeLists.txt @@ -88,11 +88,11 @@ endif () if (DPI_REFERENCE_DRM_OCDM) set(CMAKE_LIBRARY_OUTPUT_DIRECTORY ${HAVE_DPI_DIRECTORY}/ocdm) add_library(drmocdm ${LIBRARY_TYPE} ${PLAYREADY_SOURCES}) - target_link_libraries(drmocdm ocdm gstreamer-1.0 glib-2.0 gobject-2.0 gstaudio-1.0 gstapp-1.0 gstvideo-1.0 gstbase-1.0 WPEFrameworkProtocols WPEFrameworkDefinitions WPEFrameworkCore WPEFrameworkTracing &mAudioAAC, netflix::device::sConfiguration->svpEnabled, mSrc, &retVal) == true) + target_link_libraries(drmocdm ocdmRialto gstreamer-1.0 glib-2.0 gobject-2.0 gstaudio-1.0 gstapp-1.0 gstvideo-1.0 gstbase-1.0 WPEFrameworkProtocols WPEFrameworkDefinitions WPEFrameworkCore WPEFrameworkTracing ) #else if target(rdk_include_directories(drmocdm PUBLIC ${CMAKE_CURRENT_LIST_DIR}) gstreamer_utils::performAudioTrackCodecChannelSwitch(&(mPlaybackGroup->mGstUtilsPlaybackGroup), pSampleAttr, pAudioAttr, target_compile_options(drmocdm PRIVATE -fvisibility=hidden) set_target_properties(drmocdm PROPERTIES OUTPUT_NAME "drmsystem") - + target_compile_options(drmocdm PUBLIC -DPLAYREADY -DPLAYREADY4_0 -DDRM_BUILD_PROFILE=900 -DDRM_SUPPORT_ASSEMBLY=0 -DDRM_SUPPORT_DATASTORE_PREALLOC=1 -DDRM_SUPPORT_ECCPROFILING=0 -DDRM_SUPPORT_FORCE_ALIGN=1 -DDRM_SUPPORT_INLINEDWORDCPY=0 -DDRM_SUPPORT_FILE_LOCKING=1 -DDRM_SUPPORT_MULTI_THREADING=0 -DDRM_SUPPORT_NATIVE_64BIT_TYPES=1 -DDRM_SUPPORT_PRECOMPUTE_GTABLE=1 -DDRM_SUPPORT_SECUREOEMHAL=0 -DDRM_SUPPORT_SECUREOEMHAL_PLAY_ONLY=0 -DDRM_SUPPORT_TRACING=1 -DDRM_SUPPORT_TOOLS_NET_IO=0 -DDRM_COMPILE_FOR_NORMAL_WORLD=0 -DDRM_COMPILE_FOR_SECURE_WORLD=1) set_target_properties(drmocdm PROPERTIES ADDITIONAL_CLEAN_FILES ${HAVE_DPI_DIRECTORY}/ocdm) if (DPI_REFERENCE_DRM_DO_PLUGIN) |
When patch is applied, bitbake -C configure netflix has to be executed.
Some modifications have to be applied in rdk-gstreamer-utils code:
pstatus, &delay, &mAudioChangeTargetPts, &mCurrentDispPts, &_audio_change_stage, &mAppsrcCaps,
&mAudioAAC, netflix::device::sConfiguration->svpEnabled, mSrc, &retVal) == true)
+ #endif
{
if (!retVal)
{
@@ -1363,3 +1370,80 @@ void ESPlayerGstGenericPort::firstSampleAvailable()
Signal(AVAIL_FIRST_SAMPLE);
pinch();
}
+
+bool ESPlayerGstGenericPort::performAudioTrackCodecChannelSwitchRialto(
+ rdk_gstreamer_utils::rdkGstreamerUtilsPlaybackGrp *pgstUtilsPlaybackGroup, const void *pSampleAttr, rdk_gstreamer_utils::AudioAttributes *
pAudioAttr,
+ uint32_t *pStatus, unsigned int *pui32Delay, llong *pAudioChangeTargetPts, const llong *currentDispPts,
+ unsigned int *audio_change_stage, GstCaps **appsrcCaps, bool *audioaac, bool svpenabled, GstElement *aSrc, bool *ret)
+{
+ Log::warn(TRACE_MEDIAPLAYBACK," performAudioTrackCodecChannelSwitchRialto()");
+ if (*pStatus != 0 || pSampleAttr == nullptr || pAudioAttr == nullptr ) {
+ Log::warn(TRACE_MEDIAPLAYBACK, "performAudioTrackCodecChannelSwitchRialto() No audio data ready yet");
+ *pui32Delay = 100; //delay of 100ms
+ return false;
+ }
+
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): Audio change current disp pts %lld\n",
+ *currentDispPts);
+
+ gint64 currentPts = GST_CLOCK_TIME_NONE;
+
+ // Rialto Server's pipeline should be in PAUSED state here. Query position before switching it to READY
+ GstQuery* query = gst_query_new_position(GST_FORMAT_TIME);
+ if (gst_element_query(pgstUtilsPlaybackGroup->gstPipeline, query))
+ gst_query_parse_position(query, 0, ¤tPts);
+ gst_query_unref(query);
+
+ GstEvent* flush_start = NULL;
+ GstEvent* flush_stop = NULL;
+ *pAudioChangeTargetPts = *currentDispPts;
+ *audio_change_stage = AUDCHG_ALIGN;
+ *pui32Delay = 0;
+ pgstUtilsPlaybackGroup->isAudioAAC = *audioaac;
+
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): Sending flush-start event to rialto sink");
+ flush_start = gst_event_new_flush_start();
+ *ret = gst_element_send_event(aSrc, flush_start);
+ if (!*ret)
+ Log::warn(TRACE_MEDIAPLAYBACK,"failed to send flush-start event");
+
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): Sending flush-stop event to rialto sink");
+ flush_stop = gst_event_new_flush_stop(TRUE);
+ *ret = gst_element_send_event(aSrc, flush_stop);
+ if (!*ret)
+ Log::warn(TRACE_MEDIAPLAYBACK,"failed to send flush-stop event");
+
+ if (pgstUtilsPlaybackGroup->curAudioPlaysinkBin)
+ {
+ GstState current_state, pending;
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): Switching audio sink to READY");
+ gst_element_set_state(pgstUtilsPlaybackGroup->curAudioPlaysinkBin, GST_STATE_READY);
+ gst_element_get_state(pgstUtilsPlaybackGroup->curAudioPlaysinkBin, ¤t_state, &pending, GST_CLOCK_TIME_NONE);
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto():Current AudioPlaySinkBin State = %d", current_state);
+
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): Switching audio sink to previous state");
+ gst_element_sync_state_with_parent(pgstUtilsPlaybackGroup->curAudioPlaysinkBin);
+ gst_element_get_state(pgstUtilsPlaybackGroup->curAudioPlaysinkBin, ¤t_state, &pending, GST_CLOCK_TIME_NONE);
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): AudioPlaysinkbin State = %d Pending = %d",
+ current_state, pending);
+ gchar *oldCapsString = gst_caps_to_string(*appsrcCaps);
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): Old caps: %s" , oldCapsString);
+ g_free(oldCapsString);
+ rdk_gstreamer_utils::configAudioCap(pAudioAttr, audioaac, svpenabled, appsrcCaps);
+ gchar *newCapsString = gst_caps_to_string(*appsrcCaps);
+ Log::warn(TRACE_MEDIAPLAYBACK,"performAudioTrackCodecChannelSwitchRialto(): New caps: %s" , newCapsString);
+ g_free(newCapsString);
+
+ if(GST_CLOCK_TIME_NONE != currentPts)
+ {
+ *pAudioChangeTargetPts = currentPts / (1000*1000);
+ Log::warn(TRACE_MEDIAPLAYBACK, "performAudioTrackCodecChannelSwitchRialto(): *pAudioChangeTargetPts: %lld", *pAudioChangeTargetPts);
+ }
+ else
+ {
+ Log::warn(TRACE_MEDIAPLAYBACK, "performAudioTrackCodecChannelSwitchRialto(): *pAudioChangeTargetPts use *currentDispPts: %lld", *pAudioChangeTargetPts);
+ }
+ }
+
+ return true;
+}
diff --git a/partner/dpi/gstreamer/ESPlayerGstGenericPort.h b/partner/dpi/gstreamer/ESPlayerGstGenericPort.h
index 3a725b0..bfb7ed2 100644
--- a/partner/dpi/gstreamer/ESPlayerGstGenericPort.h
+++ b/partner/dpi/gstreamer/ESPlayerGstGenericPort.h
@@ -126,6 +126,12 @@ public:
void UnregisterCallbacksToAppSrc();
uint32_t getMinBuffersBeforeReady();
void ProcessAudioGap(const ISampleAttributes *pSampleAttr);
+ bool performAudioTrackCodecChannelSwitchRialto(struct rdk_gstreamer_utils::rdkGstreamerUtilsPlaybackGrp *pgstUtilsPlaybackGroup,
+ const void *pSampleAttr, rdk_gstreamer_utils::AudioAttributes *pAudioAttr,
+ uint32_t *pStatus, unsigned int *pui32Delay,
+ llong *pAudioChangeTargetPts, const llong *pcurrentDispPts,
+ unsigned int *audio_change_stage, GstCaps **appsrcCaps,
+ bool *audioaac, bool svpenabled, GstElement *aSrc, bool *ret);
std::shared_ptr<SampleWriterNative> mSampleWriter;
MediaType mMediaType;
unsigned int _current_data;
diff --git a/partner/dpi/gstreamer/PlaybackGroupNative.cpp b/partner/dpi/gstreamer/PlaybackGroupNative.cpp
index 3c7d557..c2a06a7 100644
--- a/partner/dpi/gstreamer/PlaybackGroupNative.cpp
+++ b/partner/dpi/gstreamer/PlaybackGroupNative.cpp
@@ -34,7 +34,7 @@ extern void ApplicationManagerContentStopped(void);
#if defined(GST_GL)
#include "VideoSinkGStreamer.h"
#endif
-#define VIDEO_SINK_NAME "westerossink"
+#define VIDEO_SINK_NAME "rialtomsevideosink"
#define VIDEO_DECODER_NAME "westerossink"
#define TOO_FAR_AHEAD_THRESHOLD_SECONDS 20
#define UHD_WITDH 3840
@@ -304,7 +304,7 @@ PlaybackGroupNative::PlaybackGroupNative(ESManagerNative& ESManager, uint32_t pi
Log::info(TRACE_MEDIACONTROL, "Enabling audio downmixing to stereo");
g_object_set(mGstPipeline, "audio-filter", capsfilter, NULL);
}
- mVideoSink = createWesterosSink();
+ mVideoSink = nullptr;
#if defined(GST_GL)
mVideoSinkGst.reset(new VideoSinkGStreamer());
g_object_set(mGstPipeline, "video-sink", mVideoSinkGst->createVideoSink(), NULL);
diff --git a/partner/dpi/gstreamer/ocdm/CMakeLists.txt b/partner/dpi/gstreamer/ocdm/CMakeLists.txt
index f9f8d78..5212371 100644
--- a/partner/dpi/gstreamer/ocdm/CMakeLists.txt
+++ b/partner/dpi/gstreamer/ocdm/CMakeLists.txt
@@ -88,11 +88,12 @@ endif ()
if (DPI_REFERENCE_DRM_OCDM)
set(CMAKE_LIBRARY_OUTPUT_DIRECTORY ${HAVE_DPI_DIRECTORY}/ocdm)
add_library(drmocdm ${LIBRARY_TYPE} ${PLAYREADY_SOURCES})
- target_link_libraries(drmocdm ocdm gstreamer-1.0 glib-2.0 gobject-2.0 gstaudio-1.0 gstapp-1.0 gstvideo-1.0 gstbase-1.0 WPEFrameworkProtocols WPEFrameworkDefinitions WPEFrameworkCore WPEFrameworkTracing )
+ target_link_libraries(drmocdm ocdmRialto gstreamer-1.0 glib-2.0 gobject-2.0 gstaudio-1.0 gstapp-1.0 gstvideo-1.0 gstbase-1.0 WPEFrameworkProtocols WPEFrameworkDefinitions WPEFrameworkCore WPEFrameworkTracing )
target_include_directories(drmocdm PUBLIC ${CMAKE_CURRENT_LIST_DIR})
target_compile_options(drmocdm PRIVATE -fvisibility=hidden)
set_target_properties(drmocdm PROPERTIES OUTPUT_NAME "drmsystem")
+ target_compile_options(drmocdm PUBLIC -DPLAYREADY -DPLAYREADY4_0 -DDRM_BUILD_PROFILE=900 -DDRM_SUPPORT_ASSEMBLY=0 -DDRM_SUPPORT_DATASTORE_PREALLOC=1 -DDRM_SUPPORT_ECCPROFILING=0 -DDRM_SUPPORT_FORCE_ALIGN=1 -DDRM_SUPPORT_INLINEDWORDCPY=0 -DDRM_SUPPORT_FILE_LOCKING=1 -DDRM_SUPPORT_MULTI_THREADING=0 -DDRM_SUPPORT_NATIVE_64BIT_TYPES=1 -DDRM_SUPPORT_PRECOMPUTE_GTABLE=1 -DDRM_SUPPORT_SECUREOEMHAL=0 -DDRM_SUPPORT_SECUREOEMHAL_PLAY_ONLY=0 -DDRM_SUPPORT_TRACING=1 -DDRM_SUPPORT_TOOLS_NET_IO=0 -DDRM_COMPILE_FOR_NORMAL_WORLD=0 -DDRM_COMPILE_FOR_SECURE_WORLD=1)
set_target_properties(drmocdm PROPERTIES ADDITIONAL_CLEAN_FILES ${HAVE_DPI_DIRECTORY}/ocdm)
if (DPI_REFERENCE_DRM_DO_PLUGIN) |
When patch is applied, bitbake -C configure netflix has to be executed.
To build rdk-gstreamer-utils with Rialto, its bitbake has to be modified:
| Code Block | ||||
|---|---|---|---|---|
| ||||
diff --git a/recipes-extended/gstreamer-netflix-platform/rdk-gstreamer-utils.bb b/recipes-extended/gstreamer-netflix-platform/rdk-gstreamer-utils.bb
index 9dfe370..fc67ee9 100644
--- a/recipes-extended/gstreamer-netflix-platform/rdk-gstreamer-utils.bb
+++ b/recipes-extended/gstreamer-netflix-platform/rdk-gstreamer-utils.bb
@@ -5,6 +5,7 @@ LIC_FILES_CHKSUM = "file://LICENSE;md5=175792518e4ac015ab6696d16c4f607e"
SRCREV = "${AUTOREV}"
DEPENDS += " gstreamer1.0 gstreamer1.0-plugins-base"
+RDEPENDS_${PN} += " rialto-gstreamer"
SRC_URI = "${RDK_GENERIC_ROOT_GIT}/gstreamer-netflix-platform/generic;protocol=${RDK_GIT_PROTOCOL};branch=${RDK_GIT_BRANCH}" |
Some modifications have to be applied in rdk-gstreamer-utils code:
| Code Block | ||||
|---|---|---|---|---|
| ||||
diff --git a/rdk_gstreamer_utils.cpp b/rdk_gstreamer_utils.cpp
index d9248871..ff3248b5 100644
--- a/rdk_gstreamer_utils.cpp
+++ b/rdk_gstreamer_utils.cpp
@@ -78,7 +78,7 @@ namespace rdk_gstreamer_utils {
{
const char* audiodecodername = getAudioDecoderName_soc();
GstElement* audiodecoder = retrieveGstElementByName(pipeline, audiodecodername);
- GstElement* videodecoder = retrieveGstElementByName(pipeline, "westerossink"); //default on RDK platforms
+ GstElement* videodecoder = retrieveGstElementByName(pipeline, "rialtomsevideosink"); //default on RDK platforms
const char* AudioUnderflowSignal = getAudioUnderflowSignalName_soc();
const char* VideoUnderflowSignal = getVideoUnderflowSignalName_soc();
@@ -216,7 +216,7 @@ namespace rdk_gstreamer_utils {
{
return switchToAudioMasterMode_soc();
}
-
+
void setKeyFrameFlag(GstBuffer *gstBuffer,bool val)
{
setKeyFrameFlag_soc(gstBuffer,val);
diff --git a/realtek/rdk_gstreamer_utils_soc.cpp b/realtek/rdk_gstreamer_utils_soc.cpp
index 25f70ec1..ae5dd165 100644
--- a/realtek/rdk_gstreamer_utils_soc.cpp
+++ b/realtek/rdk_gstreamer_utils_soc.cpp
@@ -105,8 +105,8 @@ namespace rdk_gstreamer_utils
GstElement * configureUIAudioSink_soc(bool TTSenabled)
{
GstElement *audioSink = NULL;
- LOG_RGU("configureUIAudioSink_soc: : connecting rtkaudiosink");
- audioSink = gst_element_factory_make ("rtkaudiosink","rtkaudiosink");
+ LOG_RGU("configureUIAudioSink_soc: : connecting rialtowebaudiosink");
+ audioSink = gst_element_factory_make ("rialtowebaudiosink","rialtowebaudiosink" | ||||
| Code Block | ||||
| ||||
diff --git a/rdk_gstreamer_utils.cpp b/rdk_gstreamer_utils.cpp index d9248871..b98cb2b4 100644 --- a/rdk_gstreamer_utils.cpp +++ b/rdk_gstreamer_utils.cpp @@ -78,7 +78,7 @@ namespace rdk_gstreamer_utils { { const char* audiodecodername = getAudioDecoderName_soc(); GstElement* audiodecoder = retrieveGstElementByName(pipeline, audiodecodername); -g_object_set(G_OBJECT(audioSink), "media-tunnel", FALSE, NULL); GstElement* videodecoder = retrieveGstElementByName(pipeline, "westerossink"); //default on RDK platforms +g_object_set(G_OBJECT(audioSink), "audio-service", TRUE, NULL); @@ -870,7 +870,7 @@ namespace rdk_gstreamer_utils bool getDelayTimerEnabled_soc() GstElement* videodecoder = retrieveGstElementByName(pipeline, "rialtomsevideosink"); //default on RDK platforms { - return true; + const char* AudioUnderflowSignal = getAudioUnderflowSignalName_soc()return false; } const char* VideoUnderflowSignalvoid = getVideoUnderflowSignalNameswitchToAudioMasterMode_soc(); diff --git a/realtekrialto/rdk_gstreamer_utils_soc.cpp b/realtekrialto/rdk_gstreamer_utils_soc.cpp index 6324664526ea04d4..564a364d67b3c852 100644 --- a/realtekrialto/rdk_gstreamer_utils_soc.cpp +++ b/realtekrialto/rdk_gstreamer_utils_soc.cpp @@ -10677,78 +10677,78 @@ namespace rdk_gstreamer_utils { GstElement * configureUIAudioSink_soc(bool TTSenabled) GstElement *audioSink = NULL;{ LOG_RGU("configureUIAudioSink_soc: : connecting rtkaudiosink")GstElement *audioSink = NULL; - audioSink = gst_element_factory_make ("rtkaudiosink","rtkaudiosink"); +LOG_RGU("configureUIAudioSink_soc: : connecting rialtouiaudiosink"); - audioSink = gst_element_factory_make ("rialtomseaudiosinkrialtouiaudiosink","rialtomseaudiosink"); g_object_set(G_OBJECT(audioSink), "media-tunnel", FALSE, NULLrialtouiaudiosink"); + g_object_set(G_OBJECT(audioSink), "audio-service", TRUE, NULL); diff --git a/rialto/rdk_gstreamer_utils_soc.cpp b/rialto/rdk_gstreamer_utils_soc.cpp index 26ea04d4..e5cef981 100644 --- a/rialto/rdk_gstreamer_utils_soc.cpp +++ b/rialto/rdk_gstreamer_utils_soc.cpp @@ -77,8 +77,8LOG_RGU("configureUIAudioSink_soc: : connecting rialtowebaudiosink"); + audioSink = gst_element_factory_make ("rialtowebaudiosink","rialtowebaudiosink"); return audioSink; } @@ -181,7 +181,7 @@ namespace rdk_gstreamer_utils // no op. To be implemented if required later return; GstElement} - * configureUIAudioSink_soc(bool TTSenabled) + void setKeyFrameFlag_soc(GstBuffer {*gstBuffer,bool val) { GstElement *audioSink = NULL; - // no op. To be LOG_RGU("configureUIAudioSink_soc: : connecting rialtouiaudiosink"); - audioSink = gst_element_factory_make ("rialtouiaudiosink","rialtouiaudiosink"); + implemented if required later @@ -190,7 +190,7 @@ namespace rdk_gstreamer_utils bool LOGgetDelayTimerEnabled_RGU("configureUIAudioSink_soc: : connecting rialtomseaudiosink"); + soc() { - audioSink = gst_element_factory_make ("rialtomseaudiosink","rialtomseaudiosink"); return true; + return audioSinkfalse; } void SetAudioServerParam_soc(bool enabled) |
After applying those changes, component has to be rebuilt using bitbake -C configure rdk-gstreamer-utils
...