webrtc_source: Fix to return previous payload type before getting new one

[Version] 0.3.141
[Issue Type] Bug fix

Change-Id: I1680cac9e845da25102c88bcab8365d1df450b5c

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 3ac7b139227cbfab78b35c3dd605d6d9f729d64f..49308471ce01dae54dfb4fb6e5d83ef57c8abaee 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:    0.3.140
+Version:    0.3.141
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 60d1ff0c51573606f73c5764cb01e318592d9261..c3b04717c6e9b1c640d6ba784a00079a1d223117 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -63,6 +63,9 @@
 #define ELEMENT_NAME_AUDIO_APPSRC            "audioAppsrc"
 #define ELEMENT_NAME_VIDEO_APPSRC            "videoAppsrc"
 
+#define MIN_DYNAMIC_PAYLOAD_TYPE             96
+#define MAX_DYNAMIC_PAYLOAD_TYPE             127
+
 #define APPEND_ELEMENT(x_list, x_element) \
 do { \
        if (!(x_element)) \
@@ -774,7 +777,7 @@ static unsigned int __get_available_payload_type(webrtc_s *webrtc)
                }
                webrtc->payload_types |= bitmask;
                LOG_DEBUG("found available payload type[%d]", count + 95);
-               return count + 95; /* 96 ~ 127 */
+               return count + (MIN_DYNAMIC_PAYLOAD_TYPE - 1); /* 96 ~ 127 */
        }
 
        LOG_ERROR("could not assign payload type");
@@ -787,9 +790,10 @@ static void __return_payload_type(webrtc_s *webrtc, unsigned int payload_type)
        int bitmask = 0x1;
 
        RET_IF(webrtc == NULL, "webrtc is NULL");
-       RET_IF(payload_type < 96 || payload_type > 127, "invalid payload_type(%u)", payload_type);
+       RET_IF(payload_type < MIN_DYNAMIC_PAYLOAD_TYPE || payload_type > MAX_DYNAMIC_PAYLOAD_TYPE, "invalid payload_type(%u)", payload_type);
+
+       i = payload_type - MIN_DYNAMIC_PAYLOAD_TYPE;
 
-       i = payload_type - 96;
        while (i-- > 0)
                bitmask <<= 1;
 
@@ -1735,6 +1739,34 @@ exit:
        return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
+static int __add_transceiver(webrtc_gst_slot_s *source, webrtc_media_type_e media_type, rtp_payload_info_s *payload_info)
+{
+       GstWebRTCRTPTransceiver *trans = NULL;
+       GstCaps *caps = NULL;
+       int av_idx = (media_type == WEBRTC_MEDIA_TYPE_AUDIO) ? AV_IDX_AUDIO : AV_IDX_VIDEO;
+
+       RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+       RET_VAL_IF(payload_info == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "payload_info is NULL");
+
+       caps = __make_transceiver_caps(payload_info, source->av[av_idx].pt);
+       PRINT_CAPS(caps, "transceiver");
+
+       if (!source->av[av_idx].transceiver) {
+               g_signal_emit_by_name(source->webrtc->gst.webrtcbin, "add-transceiver",
+                       GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, caps, &trans, NULL);
+               gst_object_unref(trans);
+       } else {
+               g_object_set(G_OBJECT(source->av[av_idx].transceiver), "codec-preferences", caps, NULL);
+       }
+
+       gst_caps_unref(caps);
+
+       LOG_DEBUG("webrtc[%p] source_id[%u] [%s] transceiver[%p] codec[%s]",
+               source->webrtc, source->id, payload_info->media_type, source->av[av_idx].transceiver, payload_info->encoding_name);
+
+       return WEBRTC_ERROR_NONE;
+}
+
 int _complete_sources(webrtc_s *webrtc)
 {
        int i;
@@ -2713,7 +2745,7 @@ static unsigned int __get_unoccupied_id(GHashTable *slots)
 
        RET_VAL_IF(slots == NULL, 0, "slot is NULL");
 
-       /* Payload identifiers 96–127 are used for payloads defined dynamically during a session,
+       /* Payload identifiers 96 ~ 127 are used for payloads defined dynamically during a session,
         * hence the id range is limited here to 1-32. */
        for (i = 1; i < MAX_SOURCE_NUM + 1; i++) {
                key = g_strdup_printf("media_source_%u", i);
@@ -3294,29 +3326,19 @@ int _set_transceiver_codec(webrtc_s *webrtc, unsigned int source_id, webrtc_medi
        source->av[av_idx].codec = payload_info->encoding_name;
 
        if (source->type == WEBRTC_MEDIA_SOURCE_TYPE_NULL) {
-               GstWebRTCRTPTransceiver *trans = NULL;
-               GstCaps *caps;
                int payload_type;
 
+               if (source->av[av_idx].pt >= MIN_DYNAMIC_PAYLOAD_TYPE)
+                       __return_payload_type(webrtc, source->av[av_idx].pt);
+
                if ((payload_type = __get_fixed_payload_type(payload_info->gst_media_type)) == -1)
                        if ((payload_type = __get_available_payload_type(webrtc)) == 0)
                                return WEBRTC_ERROR_NONE;
-               source->av[av_idx].pt = payload_type;
-
-               caps = __make_transceiver_caps(payload_info, payload_type);
-               PRINT_CAPS(caps, "transceiver");
 
-               if (!source->av[av_idx].transceiver) {
-                       g_signal_emit_by_name(webrtc->gst.webrtcbin, "add-transceiver",
-                               GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, caps, &trans, NULL);
-                       gst_object_unref(trans);
-               } else {
-                       g_object_set(G_OBJECT(source->av[av_idx].transceiver), "codec-preferences", caps, NULL);
-               }
-               LOG_DEBUG("webrtc[%p] source_id[%u] [%s] transceiver[%p] codec[%s]",
-                       webrtc, source_id, payload_info->media_type, source->av[av_idx].transceiver, payload_info->encoding_name);
+               source->av[av_idx].pt = payload_type;
 
-               gst_caps_unref(caps);
+               ret = __add_transceiver(source, media_type, payload_info);
+               RET_VAL_IF(ret != WEBRTC_ERROR_NONE, ret, "failed to __add_transceiver");
        }
 
        /* FIXME: to utilize 'codec-preferences' of trans object, we need to re-create and re-link elements again */