webrtc_source: Improve to get caps for encoded format

A bug making invalid caps in __make_default_encoded_caps()
is fixed.

Tainted array index is also fixed in webrtc_test.

[Version] 0.2.28
[Issue Type] Bug fix

Change-Id: I49fd509fa04836199baa19b25b36f59e45040222
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 553858e691abd85f69a1412fbaaad44cb9cc39b1..e53d9c74cf114dc33659faa8795c88269b718bad 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:    0.2.27
+Version:    0.2.28
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 3560b406037387c4089af553748c6954cff870fa..3e71a9f945a655cdf53ae7f22af1957c156b36c6 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -365,6 +365,48 @@ static GstCaps *__make_raw_caps_from_media_format(webrtc_gst_slot_s *source)
        return caps;
 }
 
+static GstCaps *__get_caps_from_encoded_audio_media_type(const char *media_type, int channels, int samplerate)
+{
+       RET_VAL_IF(media_type == NULL, NULL, "media_type is NULL");
+
+       if (!strcmp(media_type, MEDIA_TYPE_AUDIO_OPUS) ||
+               !strcmp(media_type, MEDIA_TYPE_AUDIO_VORBIS))
+               return gst_caps_new_simple(media_type, NULL, NULL);
+
+       LOG_ERROR_IF_REACHED("invalid media_type(%s)", media_type);
+
+       return NULL;
+}
+
+static GstCaps *__get_caps_from_encoded_video_media_type(const char *media_type, int width, int height)
+{
+       RET_VAL_IF(media_type == NULL, NULL, "media_type is NULL");
+
+       if (!strcmp(media_type, MEDIA_TYPE_VIDEO_H264) ||
+               !strcmp(media_type, MEDIA_TYPE_VIDEO_H265))
+               return gst_caps_new_simple(media_type,
+                       "stream-format", G_TYPE_STRING, "byte-stream",
+                       "alignment", G_TYPE_STRING, "au",
+                       "width", G_TYPE_INT, width,
+                       "height", G_TYPE_INT, height,
+                       NULL);
+
+       if (!strcmp(media_type, MEDIA_TYPE_VIDEO_VP8) ||
+               !strcmp(media_type, MEDIA_TYPE_VIDEO_VP9) ||
+               !strcmp(media_type, MEDIA_TYPE_VIDEO_THEORA))
+               return gst_caps_new_simple(media_type, NULL, NULL); /* NOTE: need to verify these codecs */
+
+       if (!strcmp(media_type, MEDIA_TYPE_VIDEO_JPEG))
+               return gst_caps_new_simple(media_type,
+                       "width", G_TYPE_INT, width,
+                       "height", G_TYPE_INT, height,
+                       NULL);
+
+       LOG_ERROR_IF_REACHED("invalid media_type(%s)", media_type);
+
+       return NULL;
+}
+
 /* Use g_free() to free the media_type parameter. */
 static GstCaps *__make_default_encoded_caps(webrtc_gst_slot_s *source, webrtc_ini_s *ini, gchar **media_type)
 {
@@ -386,12 +428,8 @@ static GstCaps *__make_default_encoded_caps(webrtc_gst_slot_s *source, webrtc_in
        case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_VIDEO:
                _media_type = __get_video_media_type(ini_source->v_codec);
                RET_VAL_IF(_media_type == NULL, NULL, "_media_type is NULL");
+               caps = __get_caps_from_encoded_video_media_type(_media_type, ini_source->v_width, ini_source->v_height);
 
-               caps = gst_caps_new_simple(_media_type,
-                                               "framerate", GST_TYPE_FRACTION, ini_source->v_framerate, 1,
-                                               "width", G_TYPE_INT, ini_source->v_width,
-                                               "height", G_TYPE_INT, ini_source->v_height,
-                                               NULL);
                source->video_info.width = ini_source->v_width;
                source->video_info.height = ini_source->v_height;
                break;
@@ -401,35 +439,23 @@ static GstCaps *__make_default_encoded_caps(webrtc_gst_slot_s *source, webrtc_in
        case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_AUDIO:
                _media_type = __get_audio_media_type(ini_source->a_codec);
                RET_VAL_IF(_media_type == NULL, NULL, "_media_type is NULL");
-
-               caps = gst_caps_new_simple(_media_type,
-                                               "channels", G_TYPE_INT, ini_source->a_channels,
-                                               "rate", G_TYPE_INT, ini_source->a_samplerate,
-                                               NULL);
+               caps = __get_caps_from_encoded_audio_media_type(_media_type, ini_source->a_channels, ini_source->a_samplerate);
                break;
 
-       case WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET: {
+       case WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET:
                if (source->media_types == MEDIA_TYPE_AUDIO) {
                        _media_type = __get_audio_media_type(ini_source->a_codec);
-                       caps = gst_caps_new_simple(_media_type,
-                                                       "channels", G_TYPE_INT, ini_source->a_channels,
-                                                       "rate", G_TYPE_INT, ini_source->a_samplerate,
-                                                       NULL);
-
+                       RET_VAL_IF(_media_type == NULL, NULL, "_media_type is NULL");
+                       caps = __get_caps_from_encoded_audio_media_type(_media_type, ini_source->a_channels, ini_source->a_samplerate);
                } else if (source->media_types == MEDIA_TYPE_VIDEO) {
                        _media_type = __get_video_media_type(ini_source->v_codec);
-                       caps = gst_caps_new_simple(_media_type,
-                                                       "framerate", GST_TYPE_FRACTION, ini_source->v_framerate, 1,
-                                                       "width", G_TYPE_INT, ini_source->v_width,
-                                                       "height", G_TYPE_INT, ini_source->v_height,
-                                                       NULL);
-
+                       RET_VAL_IF(_media_type == NULL, NULL, "_media_type is NULL");
+                       caps = __get_caps_from_encoded_video_media_type(_media_type, ini_source->v_width, ini_source->v_height);
                } else {
                        LOG_ERROR_IF_REACHED("source->media_types(0x%x)", source->media_types);
                        return NULL;
                }
                break;
-       }
 
        default:
                LOG_ERROR_IF_REACHED("type(%d)", source->type);
@@ -465,10 +491,7 @@ static GstCaps *__make_encoded_caps_from_media_format(webrtc_gst_slot_s *source,
                        _media_type = __get_audio_media_type(__get_format_name(mime_type, source->zerocopy_enabled));
                        RET_VAL_IF(_media_type == NULL, NULL, "media_type is NULL");
 
-                       caps = gst_caps_new_simple(_media_type,
-                                                       "channels", G_TYPE_INT, channels,
-                                                       "rate", G_TYPE_INT, samplerate,
-                                                       NULL);
+                       caps = __get_caps_from_encoded_audio_media_type(_media_type, channels, samplerate);
 
                } else if (source->media_types == MEDIA_TYPE_VIDEO) {
                        int width;
@@ -483,11 +506,7 @@ static GstCaps *__make_encoded_caps_from_media_format(webrtc_gst_slot_s *source,
                        _media_type = __get_video_media_type(__get_format_name(mime_type, source->zerocopy_enabled));
                        RET_VAL_IF(_media_type == NULL, NULL, "media_type is NULL");
 
-                       caps = gst_caps_new_simple(_media_type,
-                                                       "framerate", GST_TYPE_FRACTION, framerate, 1,
-                                                       "width", G_TYPE_INT, width,
-                                                       "height", G_TYPE_INT, height,
-                                                       NULL);
+                       caps = __get_caps_from_encoded_video_media_type(_media_type, width, height);
 
                } else {
                        LOG_ERROR_IF_REACHED("source->media_types(0x%x)", source->media_types);
@@ -663,9 +682,19 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
        RET_VAL_IF(element_list == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "element_list is NULL");
 
+       if (source->media_types == MEDIA_TYPE_VIDEO) {
+               encoder_klass_name = GST_KLASS_NAME_ENCODER_VIDEO;
+       } else if (source->media_types == MEDIA_TYPE_AUDIO) {
+               encoder_klass_name = GST_KLASS_NAME_ENCODER_AUDIO;
+       } else {
+               LOG_ERROR("not ready for this media_types[0x%x]", source->media_types);
+               return WEBRTC_ERROR_INVALID_OPERATION;
+       }
+
        if (need_capsfilter) {
                if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_FIRST_CAPSFILTER)))
                        return WEBRTC_ERROR_INVALID_OPERATION;
+               APPEND_ELEMENT(*element_list, capsfilter);
 
                if (__is_encoded_format_supported(source->type, &webrtc->ini)) {
                        if ((sink_caps = __make_default_encoded_caps(source, &webrtc->ini, NULL))) {
@@ -683,15 +712,6 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source
                }
        }
 
-       if (source->media_types == MEDIA_TYPE_VIDEO) {
-               encoder_klass_name = GST_KLASS_NAME_ENCODER_VIDEO;
-       } else if (source->media_types == MEDIA_TYPE_AUDIO) {
-               encoder_klass_name = GST_KLASS_NAME_ENCODER_AUDIO;
-       } else {
-               LOG_ERROR("not ready for this media_types[0x%x]", source->media_types);
-               return WEBRTC_ERROR_INVALID_OPERATION;
-       }
-
        if (source->zerocopy_enabled)
                encoder = __get_hw_encoder_element(webrtc, source);
        else
@@ -701,6 +721,7 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source
                                                        encoder);
        if (encoder == NULL)
                return WEBRTC_ERROR_INVALID_OPERATION;
+       APPEND_ELEMENT(*element_list, encoder);
 
        encoder_name = gst_element_get_name(encoder);
        if (encoder_name && (!g_strcmp0(encoder_name, "vp8enc") || !g_strcmp0(encoder_name, "vp9enc"))) {
@@ -724,16 +745,19 @@ skip_encoder:
                g_free(media_type);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
+       APPEND_ELEMENT(*element_list, payloader);
 
        if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, NULL))) {
                g_free(media_type);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
+       APPEND_ELEMENT(*element_list, queue);
 
        if (!(capsfilter2 = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_RTP_CAPSFILTER))) {
                g_free(media_type);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
+       APPEND_ELEMENT(*element_list, capsfilter2);
 
        payload_id = __get_available_payload_id(webrtc);
        if (payload_id == 0) {
@@ -749,12 +773,6 @@ skip_encoder:
 
        g_free(media_type);
 
-       APPEND_ELEMENT(*element_list, capsfilter);
-       APPEND_ELEMENT(*element_list, encoder);
-       APPEND_ELEMENT(*element_list, payloader);
-       APPEND_ELEMENT(*element_list, queue);
-       APPEND_ELEMENT(*element_list, capsfilter2);
-
        return WEBRTC_ERROR_NONE;
 }
 
@@ -1550,7 +1568,6 @@ exit:
 
 static GstCaps *__make_encoded_caps_for_appsrc(webrtc_gst_slot_s *source)
 {
-       GstCaps *caps;
        media_format_mimetype_e mime_type;
        const char *_media_type;
 
@@ -1562,14 +1579,8 @@ static GstCaps *__make_encoded_caps_for_appsrc(webrtc_gst_slot_s *source)
 
        _media_type = __get_video_media_type(__get_format_name(mime_type, source->zerocopy_enabled));
        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;
+       return __make_encoded_caps_from_media_format(source, NULL);
 }
 
 static int __complete_mediapacketsrc_from_encoded_format(webrtc_s *webrtc, webrtc_gst_slot_s *source)

diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index 1b0eeef1d460b576e38c8861dc56c24a9d76df90..949615e2736f02f4fbce00bbb1a5dc511aad66d6 100644 (file)
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -3282,7 +3282,7 @@ static void displaymenu()
                g_print("*** input proxy URL.\n");
 
        } else if (g_conns[g_conn_index].menu_state == CURRENT_STATUS_CHANGE_CONNECTION) {
-               g_print("*** input to connection index.(0 ~ 3)\n");
+               g_print("*** input to connection index.(0 ~ 2)\n");
 
        } else if (g_conns[g_conn_index].menu_state == CURRENT_STATUS_REQUEST_SESSION) {
                g_print("*** input remote peer id.\n");
@@ -3554,7 +3554,7 @@ static void interpret(char *cmd)
        }
        case CURRENT_STATUS_CHANGE_CONNECTION: {
                value = atoi(cmd);
-               if (value < 0 || value > 3)
+               if (value < 0 || value >= MAX_CONNECTION_LEN)
                        g_print("invalid value[%d]", value);
                else
                        g_conn_index = value;