webrtc_source: Revise caps information of appsrc in case of H264 format

To make negotiation with incoming media packet source of H264 byte stream data
caps information set to appsrc is revised.

[Version] 0.1.100
[Issue Type] Improvement

Change-Id: I0a95e4f5bdb1a02e195095ce1b862409fd65c64d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 741833ca0dd1789741c1d72df30e440369e827b2..525930fdbd9de19d9b3b17dcb49c3e3493796afa 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:    0.1.99
+Version:    0.1.100
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 1760758756dde84affcd6eb15108c736f9f8a5f8..3aaf42fea88f13c65804642f330518d6b34b15d0 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -156,6 +156,8 @@ static const char *__get_format_name(media_format_mimetype_e mime_type)
        case MEDIA_FORMAT_VP9:
                return "VP9";
        case MEDIA_FORMAT_H264_SP: /* baseline profile */
+       case MEDIA_FORMAT_H264_MP: /* main profile */
+       case MEDIA_FORMAT_H264_HP: /* high profile */
                return "H264";
        default:
                LOG_ERROR("not supported mime_type(0x%x)", mime_type);
@@ -198,6 +200,8 @@ static bool __is_supported_mime_type(media_format_mimetype_e mime_type)
        case MEDIA_FORMAT_VP8:
        case MEDIA_FORMAT_VP9:
        case MEDIA_FORMAT_H264_SP:
+       case MEDIA_FORMAT_H264_MP:
+       case MEDIA_FORMAT_H264_HP:
                LOG_INFO("[%s][ENCODED/%s] mime_type[0x%x]", mime_type & MEDIA_FORMAT_AUDIO ? "AUDIO" : "VIDEO",
                        __get_format_name(mime_type), mime_type);
                return true;
@@ -958,6 +962,30 @@ static int __complete_mediapacketsrc_from_raw_format(webrtc_s *webrtc, webrtc_gs
        return _set_ghost_pad_target(source->src_pad, capsfilter, true);
 }
 
+static GstCaps *__make_encoded_caps_for_appsrc(webrtc_gst_slot_s *source)
+{
+       GstCaps *caps;
+       media_format_mimetype_e mime_type;
+       const char *_media_type;
+
+       RET_VAL_IF(source == NULL, NULL, "source is NULL");
+       RET_VAL_IF(source->media_format == NULL, NULL, "media_format is NULL");
+       RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET, NULL, "invalid media source type(%d)", source->type);
+       RET_VAL_IF(media_format_get_video_info(source->media_format, &mime_type, NULL, NULL, NULL, NULL) != MEDIA_FORMAT_ERROR_NONE,
+                               NULL, "failed to media_format_get_video_info()");
+
+       _media_type = __get_video_media_type(__get_format_name(mime_type));
+       RET_VAL_IF(g_strcmp0(_media_type, MEDIA_TYPE_VIDEO_H264), NULL, "not supported type(%s)", _media_type);
+       /* FIXME: prepare other types */
+
+       caps = gst_caps_new_simple(_media_type,
+               "stream-format", G_TYPE_STRING, "byte-stream",
+               "alignment", G_TYPE_STRING, "au",
+               NULL);
+
+       return caps;
+}
+
 static int __complete_mediapacketsrc_from_encoded_format(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
        int ret = WEBRTC_ERROR_NONE;
@@ -976,8 +1004,8 @@ static int __complete_mediapacketsrc_from_encoded_format(webrtc_s *webrtc, webrt
        appsrc = __find_element_in_bin(source->bin, "appsrc");
        RET_VAL_IF(appsrc == NULL, WEBRTC_ERROR_INVALID_OPERATION, "appsrc is NULL");
 
-       if (!(sink_caps = __make_encoded_caps_from_media_format(source, NULL))) {
-               LOG_ERROR("failed to __make_encoded_caps_from_media_format()");
+       if (!(sink_caps = __make_encoded_caps_for_appsrc(source))) {
+               LOG_ERROR("failed to __make_encoded_caps_for_appsrc()");
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
        caps_str = gst_caps_to_string(sink_caps);
@@ -1228,7 +1256,7 @@ int _set_media_format(webrtc_s *webrtc, unsigned int source_id, media_format_h f
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
        RET_VAL_IF(format == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "format is NULL");
        RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
-       RET_VAL_IF(!__find_element_in_bin(source->bin, "appsrc"), WEBRTC_ERROR_INVALID_PARAMETER, "source_id[%u] is not for media packet source", source_id);
+       RET_VAL_IF(source->type != WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET, WEBRTC_ERROR_INVALID_PARAMETER, "invalid media source type(%d)", source->type);
        RET_VAL_IF(source->media_format != NULL, WEBRTC_ERROR_INVALID_OPERATION, "format is already set to this media packet source");
 
        RET_VAL_IF(media_format_get_type(format, &format_type) != MEDIA_FORMAT_ERROR_NONE, WEBRTC_ERROR_INVALID_OPERATION,
@@ -1363,6 +1391,7 @@ int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h
        GstElement *appsrc;
        GstBuffer *new_buffer;
        guint64 pts = 0;
+       guint64 dts = 0;
        guint64 duration = 0;
        guint64 size = 0;
        GstFlowReturn gst_ret = GST_FLOW_OK;
@@ -1410,11 +1439,13 @@ int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h
 
        media_packet_get_pts(packet, &pts);
        GST_BUFFER_PTS(new_buffer) = pts;
+       media_packet_get_dts(packet, &dts);
+       GST_BUFFER_DTS(new_buffer) = dts;
 
        media_packet_get_duration(packet, &duration);
        GST_BUFFER_DURATION(new_buffer) = duration;
 
-       LOG_DEBUG("new gst buffer[%p, pts:%llu, duration:%llu]", new_buffer, pts, duration);
+       LOG_DEBUG("new gst buffer[%p, pts:%llu, dts:%llu, duration:%llu]", new_buffer, pts, dts, duration);
 
        g_signal_emit_by_name(G_OBJECT(appsrc), "push-buffer", new_buffer, &gst_ret, NULL);
        gst_buffer_unref(new_buffer);