webrtc_test: Add sub-menu to use mic only in room case

Parameter of _webrtc_add_media_source() is revised.

[Version] 0.3.8
[Issue Type] Add

Change-Id: I8312d9f19701b2859bc736745fc54d1c79b8ab57
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 09828ba541733a45c6f8ebf3bfb59ae1f1ba5a40..fdacaa48fed3621f4c050c0fee67c4e30dee8cc8 100644 (file)
--- 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:    0.3.7
+Version:    0.3.8
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index 76e253ad7a27a55d46cca4ba7a56806a8511ee5e..b971fb92d17bb1ed83814e68b1d1311386ba8118 100644 (file)
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -751,61 +751,48 @@ static int __get_sound_stream_info(sound_stream_info_h *stream_info)
        return 0;
 }
 
-static void _webrtc_add_media_source(int index, int value, unsigned int *source_id)
+static void _webrtc_add_media_source(int index, int type, unsigned int *source_id)
 {
        int ret = WEBRTC_ERROR_NONE;
        unsigned int _source_id = 0;
        int i;
 
-       if (value - 1 <= WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET) {
-               webrtc_media_source_type_e type = value - 1;
-               switch (type) {
-               case WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST ... WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET:
-                       if (type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET) {
-                               i = _get_empty_packet_sources_index(index);
-                               RET_IF(i == -1, "media packet source can be added up to %d", MAX_MEDIA_PACKET_SOURCE_LEN);
-                       }
-                       ret = webrtc_add_media_source(g_conns[index].webrtc, type, &_source_id);
-                       RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
-
-                       if (type == WEBRTC_MEDIA_SOURCE_TYPE_MIC) {
-                               if (g_conns[index].source.stream_info) {
-                                       sound_manager_destroy_stream_information(g_conns[index].source.stream_info);
-                                       g_conns[index].source.stream_info = NULL;
-                               }
+       switch (type) {
+       case WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST ... WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET:
+               if (type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET) {
+                       i = _get_empty_packet_sources_index(index);
+                       RET_IF(i == -1, "media packet source can be added up to %d", MAX_MEDIA_PACKET_SOURCE_LEN);
+               }
+               ret = webrtc_add_media_source(g_conns[index].webrtc, type, &_source_id);
+               RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
 
-                               if (__get_sound_stream_info(&g_conns[index].source.stream_info) < 0) {
-                                       g_printerr("failed to __get_sound_stream_info()\n");
+               if (type == WEBRTC_MEDIA_SOURCE_TYPE_MIC) {
+                       if (g_conns[index].source.stream_info) {
+                               sound_manager_destroy_stream_information(g_conns[index].source.stream_info);
+                               g_conns[index].source.stream_info = NULL;
+                       }
 
-                               } else {
-                                       ret = webrtc_mic_source_set_sound_stream_info(g_conns[index].webrtc, _source_id, g_conns[index].source.stream_info);
-                                       if (ret != WEBRTC_ERROR_NONE)
-                                               g_printerr("failed to webrtc_mic_source_set_sound_stream_info(), ret[0x%x]\n", ret);
-                               }
+                       if (__get_sound_stream_info(&g_conns[index].source.stream_info) < 0) {
+                               g_printerr("failed to __get_sound_stream_info()\n");
+                       } else {
+                               ret = webrtc_mic_source_set_sound_stream_info(g_conns[index].webrtc, _source_id, g_conns[index].source.stream_info);
+                               if (ret != WEBRTC_ERROR_NONE)
+                                       g_printerr("failed to webrtc_mic_source_set_sound_stream_info(), ret[0x%x]\n", ret);
                        }
-                       break;
-               default:
-                       g_printerr("failed to _webrtc_add_media_source(), invalid type(%d)\n", type);
-                       return;
-               }
-               if (type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET) {
+
+               } else if (type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET) {
                        g_conns[index].packet_sources[i].source_id = _source_id;
                        g_mutex_init(&g_conns[index].packet_sources[i].mutex);
                        g_cond_init(&g_conns[index].packet_sources[i].cond);
                }
-
-       } else {
-               webrtc_media_source_type_internal_e type = value - 1;
-               switch (type) {
-               case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_AUDIO:
-               case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_VIDEO:
-                       ret = webrtc_add_media_source_internal(g_conns[index].webrtc, type, &_source_id);
-                       RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
-                       break;
-               default:
-                       g_printerr("failed to _webrtc_add_media_source(), invalid type(%d)\n", type);
-                       return;
-               }
+               break;
+       case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_AUDIO ... WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_VIDEO:
+               ret = webrtc_add_media_source_internal(g_conns[index].webrtc, type, &_source_id);
+               RET_IF(ret != WEBRTC_ERROR_NONE, "ret[0x%x]", ret);
+               break;
+       default:
+               g_printerr("failed to _webrtc_add_media_source(), invalid type(%d)\n", type);
+               return;
        }
 
        g_print("_webrtc_add_media_source() success, source_id[%u]\n", _source_id);
@@ -3403,17 +3390,20 @@ static void __auto_configure_add_peer(gchar *peer_id, bool is_offer)
 
                switch (g_conns[i].room_source_type) {
                case 1:
-                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST + 1, NULL);
-                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_VIDEOTEST + 1, &source_id);
+                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST, NULL);
+                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_VIDEOTEST, &source_id);
                        if (i == 0) /* only set loopback video of the first one */
                                _webrtc_media_source_set_video_loopback(i, source_id);
                        break;
                case 2:
-                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_MIC + 1, NULL);
-                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_CAMERA + 1, &source_id);
+                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_MIC, NULL);
+                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_CAMERA, &source_id);
                        if (i == 0) /* only set loopback video of the first one */
                                _webrtc_media_source_set_video_loopback(i, source_id);
                        break;
+               case 3:
+                       _webrtc_add_media_source(i, WEBRTC_MEDIA_SOURCE_TYPE_MIC, NULL);
+                       break;
                default:
                        return;
                }
@@ -4382,7 +4372,7 @@ static void displaymenu()
 
        } else if (g_menu_state == CURRENT_STATUS_REQUEST_JOIN_ROOM) {
                if (g_cnt == 0)
-                       g_print("*** input source type.(1:videotest/audiotest 2:camera/mic)\n");
+                       g_print("*** input source type.(1:audiotest/videotest 2:mic/camera 3:mic only)\n");
                else if (g_cnt == 1)
                        g_print("*** input room name to join.\n");
 
@@ -4445,7 +4435,7 @@ static void interpret(char *cmd)
                break;
        case CURRENT_STATUS_ADD_MEDIA_SOURCE: {
                value = atoi(cmd);
-               _webrtc_add_media_source(0, value, NULL);
+               _webrtc_add_media_source(0, value - 1, NULL);
                reset_menu_state();
                break;
        }
@@ -4733,7 +4723,7 @@ static void interpret(char *cmd)
                switch (g_cnt) {
                case 0:
                        value = atoi(cmd);
-                       if (value <= 0 || value > 2) {
+                       if (value <= 0 || value > 3) {
                                g_printerr("invalid value[%d]\n", value);
                                reset_menu_state();
                                break;