Rename payload id to payload type(pt)

New one is the term the most commonly used.

[Version] 0.3.35
[Issue Type] Refactoring

Change-Id: Ic0b070cab1fd445ae0bd327f382a4f3d349356ff
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 0b783b737afdfa7f26bb77f86adf72dc80ae6ddf..eea7ebf641e70c75259da03c4e57c44aaec4c806 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -240,7 +240,7 @@ do { \
 #define MEDIA_TYPE_VIDEO_JPEG           "image/jpeg"
 
 #define WEBRTC_DISPLAY_TYPE_ECORE_WL    2
-#define PAYLOAD_ID_BITS                 32 /* 96 ~ 127 */
+#define PAYLOAD_TYPE_BITS               32 /* 96 ~ 127 */
 #define TRACK_ID_THRESHOLD_OF_LOOPBACK  100
 #define MAX_MLINE_NUM                   16
 
@@ -425,7 +425,7 @@ typedef struct _webrtc_s {
        GCond desc_cond;
 
        webrtc_gst_s gst;
-       unsigned int payload_ids;
+       unsigned int payload_types;
 
        gchar *stun_server_url;
        GList *turn_server_urls;
@@ -480,7 +480,7 @@ typedef struct _webrtc_gst_slot_s {
                gulong src_pad_probe_id;
                bool pause;
                bool inbandfec;
-               unsigned int payload_id;
+               unsigned int pt;
                struct {
                        unsigned int track_id;
                        bool need_decoding;
@@ -709,7 +709,7 @@ int _remove_filesrc_pad_block_probe(webrtc_s *webrtc);
 
 gchar * _get_media_type_from_pad(GstPad *pad);
 gchar * _get_mime_type_from_pad(GstPad *pad);
-int _get_payload_id_from_pad(GstPad *pad);
+int _get_payload_type_from_pad(GstPad *pad);
 bool _is_supported_media_type(const char *media_type);
 bool _is_audio_media_type(const char *media_type);
 

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index ccc2294e2eee1f30a8251738236f3311f25d9368..b55b8b84d985d1ed8bc22a495a1d3839ae61839f 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:    0.3.34
+Version:    0.3.35
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index c49e97455285965be280a9abd0c730cd20723392..9bb26f9f58840555296af109360c1c2671da9111 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1271,24 +1271,24 @@ gchar * _get_mime_type_from_pad(GstPad *pad)
        return mime_type;
 }
 
-int _get_payload_id_from_pad(GstPad *pad)
+int _get_payload_type_from_pad(GstPad *pad)
 {
        GstCaps *caps = NULL;
-       gint id = -1;
+       gint pt = -1;
 
        RET_VAL_IF(pad == NULL, -1, "pad is NULL");
 
        caps = gst_pad_get_current_caps(pad);
        RET_VAL_IF(caps == NULL, -1, "caps is NULL");
 
-       if (!gst_structure_get_int(gst_caps_get_structure(caps, 0), "payload", &id))
+       if (!gst_structure_get_int(gst_caps_get_structure(caps, 0), "payload", &pt))
                LOG_ERROR("failed to gst_structure_get_int()");
        else
-               LOG_DEBUG("payload id[%d]", id);
+               LOG_DEBUG("payload type[%d]", pt);
 
        gst_caps_unref(caps);
 
-       return id;
+       return pt;
 }
 //LCOV_EXCL_STOP
 

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index ab131a61adce2a52224bab107fe64c9b428b0ab6..6bd6be49632b33e0919cfd55068b65254ea9bae6 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -362,7 +362,7 @@ static void __decodebin_element_added_cb(GstElement *decodebin, GstElement *elem
                for (i = 0; i < MAX_MLINE_NUM; i ++) { /* FIXME: we don't know mline in this situation, so retrieve the array here. */
                        if (!sink->webrtc->data_recovery_types[i].inbandfec.use)
                                continue;
-                       if (sink->webrtc->data_recovery_types[i].inbandfec.pt != (int)sink->av[AV_IDX_AUDIO].payload_id)
+                       if (sink->webrtc->data_recovery_types[i].inbandfec.pt != (int)sink->av[AV_IDX_AUDIO].pt)
                                continue;
                        if (!g_object_class_find_property(G_OBJECT_GET_CLASS(G_OBJECT(element)), "use-inband-fec")) {
                                LOG_WARNING("could not find 'use-inband-fec' property");
@@ -550,7 +550,7 @@ int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
        if (!decodebin)
                goto error_before_insert;
 
-       sink->av[GET_AV_IDX(is_audio)].payload_id = _get_payload_id_from_pad(src_pad);
+       sink->av[GET_AV_IDX(is_audio)].pt = _get_payload_type_from_pad(src_pad);
 
        gst_bin_add(sink->bin, decodebin);
 

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 4ac01d3bb7e357d3729fb1cfe012af2dec480d3e..0bee0756c8a05a1ec6cc303e3806238d22ed2329 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -616,7 +616,7 @@ static GstCaps *__make_encoded_caps_from_media_format(webrtc_gst_slot_s *source,
 }
 //LCOV_EXCL_STOP
 
-static GstCaps *__make_rtp_caps(const gchar *media_type, unsigned int payload_id, webrtc_gst_slot_s *source)
+static GstCaps *__make_rtp_caps(const gchar *media_type, unsigned int payload_type, webrtc_gst_slot_s *source)
 {
        GstCaps *caps;
        bool is_audio;
@@ -627,7 +627,7 @@ static GstCaps *__make_rtp_caps(const gchar *media_type, unsigned int payload_id
 
        caps = gst_caps_new_simple("application/x-rtp",
                                "media", G_TYPE_STRING, GET_MEDIA_TYPE_NAME(is_audio),
-                               "payload", G_TYPE_INT, payload_id,
+                               "payload", G_TYPE_INT, payload_type,
                                NULL);
 
        if (is_audio && source->av[AV_IDX_AUDIO].inbandfec)
@@ -723,7 +723,7 @@ static bool __is_encoded_format_supported(webrtc_media_source_type_e type, webrt
        return ini_source->v_encoded_fmt_support;
 }
 
-static int __get_fixed_payload_id(const gchar *media_type)
+static int __get_fixed_payload_type(const gchar *media_type)
 {
        RET_VAL_IF(media_type == NULL, -1, "media_type is NULL");
 
@@ -736,40 +736,40 @@ static int __get_fixed_payload_id(const gchar *media_type)
        return -1;
 }
 
-static unsigned int __get_available_payload_id(webrtc_s *webrtc)
+static unsigned int __get_available_payload_type(webrtc_s *webrtc)
 {
        int bitmask = 0x1;
        int count = 0;
 
        RET_VAL_IF(webrtc == NULL, 0, "webrtc is NULL");
 
-       while (count++ < PAYLOAD_ID_BITS) {
-               if (webrtc->payload_ids & bitmask) {
+       while (count++ < PAYLOAD_TYPE_BITS) {
+               if (webrtc->payload_types & bitmask) {
                        bitmask <<= 1;
                        continue;
                }
-               webrtc->payload_ids |= bitmask;
-               LOG_DEBUG("found available payload id[%d]", count + 95);
+               webrtc->payload_types |= bitmask;
+               LOG_DEBUG("found available payload type[%d]", count + 95);
                return count + 95; /* 96 ~ 127 */
        }
 
-       LOG_ERROR("could not assign payload id");
+       LOG_ERROR("could not assign payload type");
        return 0;
 }
 
-static void __return_payload_id(webrtc_s *webrtc, unsigned int payload_id)
+static void __return_payload_type(webrtc_s *webrtc, unsigned int payload_type)
 {
        int i;
        int bitmask = 0x1;
 
        RET_IF(webrtc == NULL, "webrtc is NULL");
-       RET_IF(payload_id < 96 || payload_id > 127, "invalid payload_id(%u)", payload_id);
+       RET_IF(payload_type < 96 || payload_type > 127, "invalid payload_type(%u)", payload_type);
 
-       i = payload_id - 96;
+       i = payload_type - 96;
        while (i-- > 0)
                bitmask <<= 1;
 
-       webrtc->payload_ids ^= bitmask;
+       webrtc->payload_types ^= bitmask;
 }
 
 static GstPadProbeReturn __source_data_probe_cb(GstPad *pad, GstPadProbeInfo *info, gpointer user_data)
@@ -913,7 +913,7 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source
        GstCaps *sink_caps;
        element_info_s elem_info;
        gchar *media_type = NULL;
-       int payload_id;
+       int payload_type;
        int idx;
 
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
@@ -983,13 +983,13 @@ skip_encoder:
                goto error;
        APPEND_ELEMENT(*element_list, capsfilter2);
 
-       if ((payload_id = __get_fixed_payload_id(media_type)) == -1)
-               if ((payload_id = __get_available_payload_id(webrtc)) == 0)
+       if ((payload_type = __get_fixed_payload_type(media_type)) == -1)
+               if ((payload_type = __get_available_payload_type(webrtc)) == 0)
                        goto error;
 
-       source->av[idx].payload_id = payload_id;
+       source->av[idx].pt = payload_type;
 
-       if ((sink_caps = __make_rtp_caps(media_type, payload_id, source))) {
+       if ((sink_caps = __make_rtp_caps(media_type, payload_type, source))) {
                g_object_set(G_OBJECT(capsfilter2), "caps", sink_caps, NULL);
                gst_caps_unref(sink_caps);
        }
@@ -1015,7 +1015,7 @@ static int __create_rest_of_elements_for_encoded_format(webrtc_s *webrtc, webrtc
        GstElement *payloader;
        GstElement *queue;
        GstElement *capsfilter;
-       unsigned int payload_id;
+       unsigned int payload_type;
 
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -1038,13 +1038,13 @@ static int __create_rest_of_elements_for_encoded_format(webrtc_s *webrtc, webrtc
                goto error;
        APPEND_ELEMENT(*element_list, capsfilter);
 
-       payload_id = __get_available_payload_id(webrtc);
-       if (payload_id == 0)
+       payload_type = __get_available_payload_type(webrtc);
+       if (payload_type == 0)
                goto error;
 
-       source->av[GET_AV_IDX_BY_TYPE(source->media_types)].payload_id = payload_id;
+       source->av[GET_AV_IDX_BY_TYPE(source->media_types)].pt = payload_type;
 
-       if ((sink_caps = __make_rtp_caps(media_type, payload_id, source))) {
+       if ((sink_caps = __make_rtp_caps(media_type, payload_type, source))) {
                g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
                gst_caps_unref(sink_caps);
        }
@@ -1957,22 +1957,22 @@ static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s
 {
        GstElement *capsfilter = NULL;
        GstCaps *sink_caps = NULL;
-       unsigned int payload_id = 0;
+       unsigned int payload_type = 0;
 
        RET_VAL_IF(source == NULL, NULL, "source is NULL");
 
        if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, _av_tbl[GET_AV_IDX(is_audio)].capsfilter_name)))
                return NULL;
 
-       payload_id = __get_available_payload_id(source->webrtc);
-       if (payload_id == 0) {
+       payload_type = __get_available_payload_type(source->webrtc);
+       if (payload_type == 0) {
                SAFE_GST_OBJECT_UNREF(capsfilter);
                return NULL;
        }
 
-       source->av[GET_AV_IDX(is_audio)].payload_id = payload_id;
+       source->av[GET_AV_IDX(is_audio)].pt = payload_type;
 
-       if ((sink_caps = __make_rtp_caps(GET_MEDIA_TYPE_NAME(is_audio), payload_id, source))) {
+       if ((sink_caps = __make_rtp_caps(GET_MEDIA_TYPE_NAME(is_audio), payload_type, source))) {
                g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
                gst_caps_unref(sink_caps);
        }
@@ -2619,8 +2619,8 @@ void _source_slot_destroy_cb(gpointer data)
                __remove_probe_from_pad_for_pause(source, i);
                __remove_probe_from_pad_for_render(source, i);
 
-               if (source->av[i].payload_id > 0)
-                       __return_payload_id(source->webrtc, source->av[i].payload_id);
+               if (source->av[i].pt > 0)
+                       __return_payload_type(source->webrtc, source->av[i].pt);
 
                if (source->av[i].render.pipeline) {
                        gst_element_set_state(source->av[i].render.pipeline, GST_STATE_NULL);
@@ -3207,8 +3207,8 @@ static void __release_filesrc_resources(webrtc_gst_slot_s *source)
                __remove_probe_from_pad_for_pause(source, av_idx);
                __remove_probe_from_pad_for_render(source, av_idx);
 
-               if (source->av[av_idx].payload_id > 0)
-                       __return_payload_id(source->webrtc, source->av[av_idx].payload_id);
+               if (source->av[av_idx].pt > 0)
+                       __return_payload_type(source->webrtc, source->av[av_idx].pt);
 
                if (source->av[av_idx].render.pipeline) {
                        gst_element_set_state(source->av[av_idx].render.pipeline, GST_STATE_NULL);