From: Sangchul Lee Date: Tue, 9 Nov 2021 06:56:05 +0000 (+0900) Subject: webrtc_source: Fix typos X-Git-Tag: submit/tizen/20211110.084340~1 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F08%2F266208%2F1;p=platform%2Fcore%2Fapi%2Fwebrtc.git webrtc_source: Fix typos DEFAULT_NAME_XXX should be ELEMENT_NAME_XXX. [Version] 0.3.6 [Issue Type] Typo fix Change-Id: Id365803d8e92272aec050e9bccc1a4b2075bfa28 Signed-off-by: Sangchul Lee --- diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec index a093534e..130df8b3 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: 0.3.5 +Version: 0.3.6 Release: 0 Group: Multimedia/API License: Apache-2.0 diff --git a/src/webrtc_source.c b/src/webrtc_source.c index ea2aa916..0af8e4ab 100644 --- a/src/webrtc_source.c +++ b/src/webrtc_source.c @@ -48,21 +48,21 @@ #define ELEMENT_NAME_VIDEO_MUTE_SRC "videoMuteSrc" #define ELEMENT_NAME_VOLUME "volume" #define ELEMENT_NAME_MIC_SRC "micSrc" -#define DEFAULT_NAME_FILE_SRC "fileSrc" -#define DEFAULT_NAME_AUDIO_QUEUE "audioQueue" -#define DEFAULT_NAME_VIDEO_QUEUE "videoQueue" -#define DEFAULT_NAME_AUDIO_CAPSFILTER "audioCapsfilter" -#define DEFAULT_NAME_VIDEO_CAPSFILTER "videoCapsfilter" -#define DEFAULT_NAME_AUDIO_PAYLOADER "audioPayloader" -#define DEFAULT_NAME_VIDEO_PAYLOADER "videoPayloader" -#define DEFAULT_NAME_VIDEOCROP "videoCrop" -#define DEFAULT_NAME_SCREENSRC "waylandSrc" -#define DEFAULT_NAME_AUDIO_FAKESINK "audioFakeSink" -#define DEFAULT_NAME_VIDEO_FAKESINK "videoFakeSink" -#define DEFAULT_NAME_AUDIO_APPSRC "audioAppsrc" -#define DEFAULT_NAME_VIDEO_APPSRC "videoAppsrc" -#define DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR "audioNetSim" -#define DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR "videoNetSim" +#define ELEMENT_NAME_FILE_SRC "fileSrc" +#define ELEMENT_NAME_AUDIO_QUEUE "audioQueue" +#define ELEMENT_NAME_VIDEO_QUEUE "videoQueue" +#define ELEMENT_NAME_AUDIO_CAPSFILTER "audioCapsfilter" +#define ELEMENT_NAME_VIDEO_CAPSFILTER "videoCapsfilter" +#define ELEMENT_NAME_AUDIO_PAYLOADER "audioPayloader" +#define ELEMENT_NAME_VIDEO_PAYLOADER "videoPayloader" +#define ELEMENT_NAME_VIDEOCROP "videoCrop" +#define ELEMENT_NAME_SCREENSRC "waylandSrc" +#define ELEMENT_NAME_AUDIO_FAKESINK "audioFakeSink" +#define ELEMENT_NAME_VIDEO_FAKESINK "videoFakeSink" +#define ELEMENT_NAME_AUDIO_APPSRC "audioAppsrc" +#define ELEMENT_NAME_VIDEO_APPSRC "videoAppsrc" +#define ELEMENT_NAME_AUDIO_NETWORK_SIMULATOR "audioNetSim" +#define ELEMENT_NAME_VIDEO_NETWORK_SIMULATOR "videoNetSim" #define APPEND_ELEMENT(x_list, x_element) \ do { \ @@ -116,20 +116,20 @@ typedef struct { static av_mapping_table_s _av_tbl[AV_IDX_MAX] = { { - DEFAULT_NAME_AUDIO_APPSRC, - DEFAULT_NAME_AUDIO_QUEUE, - DEFAULT_NAME_AUDIO_PAYLOADER, - DEFAULT_NAME_AUDIO_CAPSFILTER, - DEFAULT_NAME_AUDIO_FAKESINK, - DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR + ELEMENT_NAME_AUDIO_APPSRC, + ELEMENT_NAME_AUDIO_QUEUE, + ELEMENT_NAME_AUDIO_PAYLOADER, + ELEMENT_NAME_AUDIO_CAPSFILTER, + ELEMENT_NAME_AUDIO_FAKESINK, + ELEMENT_NAME_AUDIO_NETWORK_SIMULATOR }, { - DEFAULT_NAME_VIDEO_APPSRC, - DEFAULT_NAME_VIDEO_QUEUE, - DEFAULT_NAME_VIDEO_PAYLOADER, - DEFAULT_NAME_VIDEO_CAPSFILTER, - DEFAULT_NAME_VIDEO_FAKESINK, - DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR + ELEMENT_NAME_VIDEO_APPSRC, + ELEMENT_NAME_VIDEO_QUEUE, + ELEMENT_NAME_VIDEO_PAYLOADER, + ELEMENT_NAME_VIDEO_CAPSFILTER, + ELEMENT_NAME_VIDEO_FAKESINK, + ELEMENT_NAME_VIDEO_NETWORK_SIMULATOR } }; @@ -905,7 +905,7 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source } if (source->type == WEBRTC_MEDIA_SOURCE_TYPE_SCREEN && !source->zerocopy_enabled) { - if (!(videocrop = _create_element(DEFAULT_ELEMENT_VIDEOCROP, DEFAULT_NAME_VIDEOCROP))) + if (!(videocrop = _create_element(DEFAULT_ELEMENT_VIDEOCROP, ELEMENT_NAME_VIDEOCROP))) goto error; APPEND_ELEMENT(*element_list, videocrop); } @@ -1313,7 +1313,7 @@ static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source) source->media_types = MEDIA_TYPE_VIDEO; source->zerocopy_enabled = __is_hw_encoder_used(webrtc, source->type, source->media_types); - if (!(screensrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_SCREEN), DEFAULT_NAME_SCREENSRC))) + if (!(screensrc = _create_element(__get_source_element(webrtc, WEBRTC_MEDIA_SOURCE_TYPE_SCREEN), ELEMENT_NAME_SCREENSRC))) return WEBRTC_ERROR_INVALID_OPERATION; APPEND_ELEMENT(switch_src_list, screensrc); @@ -1830,7 +1830,7 @@ static void __filesrc_pipeline_audio_stream_handoff_cb(GstElement *object, GstBu webrtc_gst_slot_s *source = data; GstFlowReturn gst_ret = GST_FLOW_OK; - g_signal_emit_by_name(gst_bin_get_by_name(source->bin, DEFAULT_NAME_AUDIO_APPSRC), "push-buffer", buffer, &gst_ret, NULL); + g_signal_emit_by_name(gst_bin_get_by_name(source->bin, ELEMENT_NAME_AUDIO_APPSRC), "push-buffer", buffer, &gst_ret, NULL); if (gst_ret != GST_FLOW_OK) LOG_ERROR("failed to 'push-buffer', gst_ret[0x%x]", gst_ret); } @@ -1840,7 +1840,7 @@ static void __filesrc_pipeline_video_stream_handoff_cb(GstElement *object, GstBu webrtc_gst_slot_s *source = data; GstFlowReturn gst_ret = GST_FLOW_OK; - g_signal_emit_by_name(gst_bin_get_by_name(source->bin, DEFAULT_NAME_VIDEO_APPSRC), "push-buffer", buffer, &gst_ret, NULL); + g_signal_emit_by_name(gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_APPSRC), "push-buffer", buffer, &gst_ret, NULL); if (gst_ret != GST_FLOW_OK) LOG_ERROR("failed to 'push-buffer', gst_ret[0x%x]", gst_ret); } @@ -1855,9 +1855,9 @@ static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad, GstPadProbeInfo *info gst_structure_get(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0), "media", G_TYPE_STRING, &media, NULL); if (g_strrstr(media, "audio")) - appsrc = gst_bin_get_by_name(source->bin, DEFAULT_NAME_AUDIO_APPSRC); + appsrc = gst_bin_get_by_name(source->bin, ELEMENT_NAME_AUDIO_APPSRC); else - appsrc = gst_bin_get_by_name(source->bin, DEFAULT_NAME_VIDEO_APPSRC); + appsrc = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_APPSRC); RET_VAL_IF(appsrc == NULL, GST_PAD_PROBE_OK, "There is no appsrc for [%s]", media); @@ -2232,7 +2232,7 @@ static int __build_filesrc_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s *source) goto error; } - if (!(filesrc = _create_element(DEFAULT_ELEMENT_FILESRC, DEFAULT_NAME_FILE_SRC))) + if (!(filesrc = _create_element(DEFAULT_ELEMENT_FILESRC, ELEMENT_NAME_FILE_SRC))) goto error; if (!(decodebin = _create_element("decodebin", NULL))) { @@ -3213,7 +3213,7 @@ int _set_media_path(webrtc_s *webrtc, unsigned int source_id, const char *path) } } - filesrc = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), DEFAULT_NAME_FILE_SRC); + filesrc = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), ELEMENT_NAME_FILE_SRC); RET_VAL_IF(filesrc == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc is NULL"); g_object_get(G_OBJECT(filesrc), "location", &location, NULL); @@ -3237,7 +3237,7 @@ static gboolean __check_path_is_not_set_cb(gpointer key, gpointer value, gpointe if (source->type == GPOINTER_TO_INT(user_data)) { LOG_INFO("found file source[%p, id:%u]", source, source->id); - g_object_get(G_OBJECT(gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), DEFAULT_NAME_FILE_SRC)), "location", &location, NULL); + g_object_get(G_OBJECT(gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), ELEMENT_NAME_FILE_SRC)), "location", &location, NULL); if (!location) return TRUE; @@ -4287,10 +4287,10 @@ int _set_screen_source_crop(webrtc_s *webrtc, unsigned int source_id, int x, int RET_VAL_IF(width == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "width is NULL"); RET_VAL_IF(height == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "height is NULL"); - screen_source = gst_bin_get_by_name(source->bin, DEFAULT_NAME_SCREENSRC); + screen_source = gst_bin_get_by_name(source->bin, ELEMENT_NAME_SCREENSRC); RET_VAL_IF(screen_source == NULL, WEBRTC_ERROR_INVALID_OPERATION, "sreen_source is NULL"); - videocrop = gst_bin_get_by_name(source->bin, DEFAULT_NAME_VIDEOCROP); + videocrop = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEOCROP); RET_VAL_IF(videocrop == NULL, WEBRTC_ERROR_INVALID_OPERATION, "videocrop is NULL"); LOG_INFO("set source crop x:%d, y:%d, width:%d, height:%d, mode:%s", x, y, w, h, (portrait_mode) ? "portrait" : "landscape"); @@ -4342,10 +4342,10 @@ int _unset_screen_source_crop(webrtc_s *webrtc, unsigned int source_id) RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL"); RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_SCREEN, WEBRTC_ERROR_INVALID_PARAMETER, "source type is not screen"); - screen_source = gst_bin_get_by_name(source->bin, DEFAULT_NAME_SCREENSRC); + screen_source = gst_bin_get_by_name(source->bin, ELEMENT_NAME_SCREENSRC); RET_VAL_IF(screen_source == NULL, WEBRTC_ERROR_INVALID_OPERATION, "sreen_source is NULL"); - videocrop = gst_bin_get_by_name(source->bin, DEFAULT_NAME_VIDEOCROP); + videocrop = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEOCROP); RET_VAL_IF(videocrop == NULL, WEBRTC_ERROR_INVALID_OPERATION, "videocrop is NULL"); g_object_get(G_OBJECT(videocrop), "left", &left, "right", &right, "top", &top, "bottom", &bottom, NULL); @@ -4435,12 +4435,12 @@ int _set_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, f if (source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE) { int count = 0; - if ((netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR))) { + if ((netsim = gst_bin_get_by_name(bin, ELEMENT_NAME_AUDIO_NETWORK_SIMULATOR))) { g_object_set(G_OBJECT(netsim), "drop-probability", probability, NULL); LOG_INFO("webrtc[%p] source_id[%u] probability[%f] applies to [%s]", webrtc, source_id, probability, GST_ELEMENT_NAME(netsim)); count++; } - if ((netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR))) { + if ((netsim = gst_bin_get_by_name(bin, ELEMENT_NAME_VIDEO_NETWORK_SIMULATOR))) { g_object_set(G_OBJECT(netsim), "drop-probability", probability, NULL); LOG_INFO("webrtc[%p] source_id[%u] probability[%f] applies to [%s]", webrtc, source_id, probability, GST_ELEMENT_NAME(netsim)); count++; @@ -4475,8 +4475,8 @@ int _get_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, f RET_VAL_IF(bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL"); if (source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE) { - if (!(netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_AUDIO_NETWORK_SIMULATOR)) && - !(netsim = gst_bin_get_by_name(bin, DEFAULT_NAME_VIDEO_NETWORK_SIMULATOR))) { + if (!(netsim = gst_bin_get_by_name(bin, ELEMENT_NAME_AUDIO_NETWORK_SIMULATOR)) && + !(netsim = gst_bin_get_by_name(bin, ELEMENT_NAME_VIDEO_NETWORK_SIMULATOR))) { LOG_ERROR("could not find any element for network simulator"); return WEBRTC_ERROR_INVALID_OPERATION; }