From: hj kim <backto.kim@samsung.com>
Date: Tue, 12 Jul 2022 06:57:56 +0000 (+0900)
Subject: webrtc_source: fix wrong payload type for file source
X-Git-Tag: submit/tizen/20220714.015855~4
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F88%2F277688%2F7;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source: fix wrong payload type for file source

plus, rearrange some included header files.

[Version] 0.3.151
[Issue Type] Bug fix

Change-Id: I4a23a38a1b203f05cd9abfe5dfa299c830ed931a
---

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index dc750f85..98aad0d6 100644
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -24,17 +24,17 @@
 #define GST_USE_UNSTABLE_API
 #include <gst/webrtc/webrtc.h>
 #endif
-#include <gst/video/videooverlay.h>
 #include <iniparser.h>
 #include <mm_display_interface.h>
 #ifndef TIZEN_TV
 #include <mm_resource_manager.h>
 #endif
-#include <sound_manager_internal.h>
 #include <tbm_bufmgr.h>
 #include <libwebsockets.h>
 #include <libsoup/soup.h>
 
+#include "webrtc.h"
+
 #ifdef __cplusplus
 extern "C" {
 #endif
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 7c76be7a..602c6426 100644
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    0.3.150
+Version:    0.3.151
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index af355137..193f5b90 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -18,6 +18,7 @@
 #include "webrtc_private.h"
 #include <json-glib/json-glib.h>
 #include <pulse/proplist.h>
+#include <sound_manager_internal.h>
 
 #define DEFAULT_DOT_FILE_NAME_PREFIX   "webrtc"
 
diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index eab48e35..7e5a6496 100644
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -17,6 +17,8 @@
 #include "webrtc.h"
 #include "webrtc_private.h"
 #include <media_packet_internal.h>
+#include <sound_manager_internal.h>
+#include <gst/video/videooverlay.h>
 
 //LCOV_EXCL_START
 bool _is_owner_of_track_build_context(webrtc_s *webrtc, unsigned int track_id)
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index ea488322..c5d4bb22 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -23,8 +23,8 @@
 #include "webrtc_source_mediapacket.h"
 
 #include <tbm_surface_internal.h>
-#include <media_packet_internal.h>
 #include <Elementary.h>
+#include <sound_manager_internal.h>
 
 #define MIN_DYNAMIC_PAYLOAD_TYPE             96
 #define MAX_DYNAMIC_PAYLOAD_TYPE             127
@@ -1470,7 +1470,6 @@ static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s
 {
 	GstElement *capsfilter = NULL;
 	GstCaps *sink_caps = NULL;
-	unsigned int payload_type = 0;
 
 	RET_VAL_IF(source == NULL, NULL, "source is NULL");
 
@@ -1482,7 +1481,7 @@ static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s
 		return NULL;
 	}
 
-	if ((sink_caps = _make_rtp_caps(GET_MEDIA_TYPE_NAME(is_audio), payload_type, source))) {
+	if ((sink_caps = _make_rtp_caps(GET_MEDIA_TYPE_NAME(is_audio), source->av[GET_AV_IDX(is_audio)].pt, source))) {
 		g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
 		gst_caps_unref(sink_caps);
 	}
@@ -2755,7 +2754,8 @@ static int __complete_mediapacketsrc_from_raw_format(webrtc_s *webrtc, webrtc_gs
 		goto exit;
 
 	SAFE_G_LIST_FREE(element_list);
-	return ret;
+
+	return WEBRTC_ERROR_NONE;
 
 exit:
 	SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
@@ -2814,7 +2814,7 @@ int _set_media_format(webrtc_s *webrtc, unsigned int source_id, media_format_h f
 
 	LOG_INFO("webrtc[%p] source_id[%u] format[%p]", webrtc, source_id, format);
 
-	return ret;
+	return WEBRTC_ERROR_NONE;
 
 error:
 	media_format_unref(format);
diff --git a/src/webrtc_source_file.c b/src/webrtc_source_file.c
index b9fa67aa..06865a4c 100644
--- a/src/webrtc_source_file.c
+++ b/src/webrtc_source_file.c
@@ -14,7 +14,6 @@
  * limitations under the License.
  */
 
-#include "webrtc_internal.h"
 #include "webrtc_private.h"
 #include "webrtc_source_common.h"
 
diff --git a/src/webrtc_source_mediapacket.c b/src/webrtc_source_mediapacket.c
index 130952b5..a9b7edc8 100644
--- a/src/webrtc_source_mediapacket.c
+++ b/src/webrtc_source_mediapacket.c
@@ -14,7 +14,6 @@
  * limitations under the License.
  */
 
-#include "webrtc_internal.h"
 #include "webrtc_private.h"
 #include "webrtc_source_common.h"
 #include <gst/audio/audio.h>
@@ -586,7 +585,7 @@ int _complete_rest_of_mediapacketsrc(webrtc_gst_slot_s *source, GstPad **src_pad
 	if ((ret = _set_ghost_pad_target(*src_pad, capsfilter, true)) != WEBRTC_ERROR_NONE)
 		goto error;
 
-	return ret;
+	return WEBRTC_ERROR_NONE;
 
 error:
 	_remove_elements_from_bin(source->bin, element_list);
@@ -629,7 +628,8 @@ int _complete_mediapacketsrc_from_encoded_format(webrtc_s *webrtc, webrtc_gst_sl
 		goto exit;
 
 	SAFE_G_LIST_FREE(element_list);
-	return ret;
+
+	return WEBRTC_ERROR_NONE;
 
 exit:
 	SAFE_G_LIST_FREE_FULL(element_list, gst_object_unref);
diff --git a/src/webrtc_source_screen.c b/src/webrtc_source_screen.c
index e0818b32..fe51bed1 100644
--- a/src/webrtc_source_screen.c
+++ b/src/webrtc_source_screen.c
@@ -14,7 +14,6 @@
  * limitations under the License.
  */
 
-#include "webrtc_internal.h"
 #include "webrtc_private.h"
 #include "webrtc_source_common.h"