webrtc_source: Save video mute value if it does not meet the required condition

This is a preparation for setting option values regardless of elements
creation or linking pads.

[Version] 0.3.119
[Issue Type] Improvement

Change-Id: Ifbac64fd9c7300e5887840dcf556378387286e5d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index cd9dfd7123844e2b82cf603b8ed341ad3a38e6a6..f6c1cbc3c0cd960c91a3542b6e9f006244664bc7 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -495,6 +495,7 @@ typedef struct _webrtc_gst_slot_s {
                GstPad *src_pad;
                gulong src_pad_probe_id;
                bool pause;
+               bool mute;
                bool inbandfec;
                int packet_loss_percentage;
                unsigned int pt;
@@ -523,7 +524,6 @@ typedef struct _webrtc_gst_slot_s {
        webrtc_callbacks_s buffer_state_changed_cb;
        webrtc_callbacks_s *encoded_frame_cb;
        gulong camerasrc_probe_id;
-       bool video_muted;
 
        GstElement *filesrc_pipeline;
        GstBus *filesrc_bus;

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 1f4ef8e38a965f1f067a1a62cf6af8371cc1e3af..3163f10995e3f712f698b9a1310b5468523c4ed2 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:    0.3.118
+Version:    0.3.119
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 307f008d722cdf4b4e0f367bee9b67986d910d8d..fed43ab3148b3e0b5a8d4d12b606ffb2aca8c583 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1387,6 +1387,14 @@ static int __build_screensrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
        if (!__link_switch_srcs(videoswitch, switch_src_list))
                goto exit_with_remove_from_bin;
 
+       if (source->av[AV_IDX_VIDEO].mute) {
+               GstPad *pad = gst_element_get_static_pad(videoswitch, "sink_1");
+               if (!pad)
+                       goto exit_with_remove_from_bin;
+               g_object_set(G_OBJECT(videoswitch), "active-pad", pad, NULL);
+               gst_object_unref(pad);
+       }
+
        if (!__link_elements(element_list))
                goto exit_with_remove_from_bin;
 
@@ -1435,6 +1443,10 @@ static int __build_rest_of_videosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
        }
        _gst_set_element_properties(videosrc_element, ini_source->source_element_properties);
 
+       /* NOTE: in case of videotestsrc with mute operation */
+       if (g_object_class_find_property(G_OBJECT_GET_CLASS(videosrc_element), "pattern") && source->av[AV_IDX_VIDEO].mute)
+               g_object_set(G_OBJECT(videosrc_element), "pattern", 2, NULL); /* 2: black */
+
        /* NOTE: in case of an element that supports tizen zerocopy format, not to emit an error in GST_STATE_PLAYING
         * without buffer consumption before finishing negotiation, set this property to 0 here. */
        if (g_object_class_find_property(G_OBJECT_GET_CLASS(G_OBJECT(videosrc_element)), "empty-buffer-timeout"))
@@ -3593,42 +3605,35 @@ static GstPadProbeReturn __camerasrc_probe_cb(GstPad *pad,  GstPadProbeInfo *inf
 
 static int __mute_camerasrc(webrtc_gst_slot_s *source, bool mute)
 {
-       int ret = WEBRTC_ERROR_NONE;
        GstElement *camerasrc = NULL;
-       GstPad *src_pad = NULL;
-
-       LOG_DEBUG_ENTER();
+       g_autoptr(GstPad) src_pad = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
-       RET_VAL_IF(source->video_muted == mute, WEBRTC_ERROR_NONE, "Already %s", mute ? "muted" : "unmuted");
+       RET_VAL_IF(source->av[AV_IDX_VIDEO].mute == mute, WEBRTC_ERROR_NONE, "Already %s", mute ? "muted" : "unmuted");
 
-       camerasrc = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_SRC);
-       src_pad = gst_element_get_static_pad(camerasrc, "src");
+       if ((camerasrc = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_SRC))) {
+               src_pad = gst_element_get_static_pad(camerasrc, "src");
 
-       if (mute) {
-               if (source->camerasrc_probe_id != 0) {
-                       LOG_ERROR("fail to change to mute");
-                       ret = WEBRTC_ERROR_INVALID_OPERATION;
-               } else {
+               if (mute && source->camerasrc_probe_id == 0) {
                        source->camerasrc_probe_id = gst_pad_add_probe(src_pad, GST_PAD_PROBE_TYPE_BUFFER, __camerasrc_probe_cb, NULL, NULL);
                        if (source->camerasrc_probe_id == 0) {
                                LOG_ERROR("failed to gst_pad_add_probe()");
-                               ret = WEBRTC_ERROR_INVALID_OPERATION;
+                               return WEBRTC_ERROR_INVALID_OPERATION;
                        }
-               }
-       } else {
-               if (source->camerasrc_probe_id == 0) {
-                       LOG_ERROR("fail to change to unmute");
-                       ret = WEBRTC_ERROR_INVALID_OPERATION;
-               } else {
+
+               } else if (!mute && source->camerasrc_probe_id != 0) {
                        gst_pad_remove_probe(src_pad, source->camerasrc_probe_id);
                        source->camerasrc_probe_id = 0;
+
+               } else {
+                       LOG_ERROR("failed to change mute to (%d)", mute);
+                       return WEBRTC_ERROR_INVALID_OPERATION;
                }
        }
 
-       gst_object_unref(src_pad);
+       source->av[AV_IDX_VIDEO].mute = mute;
 
-       return ret;
+       return WEBRTC_ERROR_NONE;
 }
 
 static int __mute_videosrc(webrtc_gst_slot_s *source, bool mute)
@@ -3637,15 +3642,17 @@ static int __mute_videosrc(webrtc_gst_slot_s *source, bool mute)
        GstPad *new_pad = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
-       RET_VAL_IF(source->video_muted == mute, WEBRTC_ERROR_NONE, "Already %s", mute ? "muted" : "unmuted");
+       RET_VAL_IF(source->av[AV_IDX_VIDEO].mute == mute, WEBRTC_ERROR_NONE, "Already %s", mute ? "muted" : "unmuted");
 
-       video_switch = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_SWITCH);
+       if ((video_switch = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_SWITCH))) {
+               new_pad = gst_element_get_static_pad(video_switch, mute ? "sink_1" : "sink_0");
+               RET_VAL_IF(new_pad == NULL, WEBRTC_ERROR_INVALID_OPERATION, "new_pad is NULL");
 
-       new_pad = gst_element_get_static_pad(video_switch, mute ? "sink_1" : "sink_0");
-       RET_VAL_IF(new_pad == NULL, WEBRTC_ERROR_INVALID_OPERATION, "new_pad is NULL");
+               g_object_set(G_OBJECT(video_switch), "active-pad", new_pad, NULL);
+               gst_object_unref(new_pad);
+       }
 
-       g_object_set(G_OBJECT(video_switch), "active-pad", new_pad, NULL);
-       gst_object_unref(new_pad);
+       source->av[AV_IDX_VIDEO].mute = mute;
 
        return WEBRTC_ERROR_NONE;
 }
@@ -3656,17 +3663,18 @@ static int __mute_videotestsrc(webrtc_gst_slot_s *source, bool mute)
        GstElement *src_element = NULL;
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
-       RET_VAL_IF(source->video_muted == mute, WEBRTC_ERROR_NONE, "Already %s", mute ? "muted" : "unmuted");
+       RET_VAL_IF(source->av[AV_IDX_VIDEO].mute == mute, WEBRTC_ERROR_NONE, "Already %s", mute ? "muted" : "unmuted");
 
-       src_element = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_SRC);
-       RET_VAL_IF(src_element == NULL, WEBRTC_ERROR_INVALID_OPERATION, "src_element is NULL");
-
-       if (!g_object_class_find_property(G_OBJECT_GET_CLASS(src_element), "pattern")) {
-               LOG_ERROR("there is no pattern property");
-               return WEBRTC_ERROR_INVALID_OPERATION;
+       if ((src_element = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VIDEO_SRC))) {
+               if (!g_object_class_find_property(G_OBJECT_GET_CLASS(src_element), "pattern")) {
+                       LOG_ERROR("there is no pattern property");
+                       return WEBRTC_ERROR_INVALID_OPERATION;
+               }
+               /* FIXME: get original value from ini file */
+               g_object_set(G_OBJECT(src_element), "pattern", mute ? 2 : 18, NULL); /* 2: black 18: ball */
        }
 
-       g_object_set(G_OBJECT(src_element), "pattern", mute ? 2 : 18, NULL); /* 2: black 18: ball */
+       source->av[AV_IDX_VIDEO].mute = mute;
 
        return WEBRTC_ERROR_NONE;
 }
@@ -3717,10 +3725,7 @@ int _set_video_mute(webrtc_s *webrtc, unsigned int source_id, bool mute)
 
        LOG_DEBUG("webrtc[%p] source_id[%u] mute[%d]", webrtc, source_id, mute);
 
-       if ((ret = (videosrc_mute_funcs[source->type])(source, mute)) == WEBRTC_ERROR_NONE)
-               source->video_muted = mute;
-
-       return ret;
+       return (videosrc_mute_funcs[source->type])(source, mute);
 }
 
 //LCOV_EXCL_START
@@ -3787,7 +3792,7 @@ int _get_video_mute(webrtc_s *webrtc, unsigned int source_id, bool *muted)
        if ((ret = __validate_video_source_for_mute(webrtc, source_id, &source)) != WEBRTC_ERROR_NONE)
                return ret;
 
-       *muted = source->video_muted;
+       *muted = source->av[AV_IDX_VIDEO].mute;
 
        LOG_INFO("webrtc[%p] source_id[%u] muted[%d]", webrtc, source_id, *muted);