webrtc_source_private: move _get_payload_info() and related code to webrtc_source_pri...

[Version] 0.3.167
[Issue Type] Refactoring

Change-Id: I6aba8972a7d42a2bbe10ef6fbf092a3b780da404

diff --git a/include/webrtc_source_private.h b/include/webrtc_source_private.h
index d2c332aa39986026a1c574a9a22f84295cf5c46a..7221cc88a61f02de69f500323b7ab44bab1aac19 100644 (file)
--- a/include/webrtc_source_private.h
+++ b/include/webrtc_source_private.h
@@ -72,6 +72,14 @@ typedef enum {
        ELEMENT_FAKESINK
 } gst_element_e;
 
+typedef struct {
+       webrtc_transceiver_codec_e codec;
+       const char *gst_media_type;
+       const char *media_type;
+       const char *encoding_name;
+       int clock_rate;
+} rtp_payload_info_s;
+
 const char *_get_audio_format_name(media_format_mimetype_e mime_type);
 const char *_get_video_format_name(media_format_mimetype_e mime_type, bool zerocopy_enabled);
 const char *_get_source_element(webrtc_s *webrtc, int type);
@@ -95,5 +103,7 @@ void _remove_probe_from_pad_for_pause(webrtc_gst_slot_s *source, unsigned int id
 GstPadProbeReturn _source_data_probe_cb(GstPad *pad, GstPadProbeInfo *info, gpointer user_data);
 void _add_probe_to_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx, GstPad *pad, void *probe_cb);
 void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsigned int idx);
+rtp_payload_info_s * _get_payload_info(webrtc_transceiver_codec_e codec);
+rtp_payload_info_s * _get_payload_info_by_encoding_name(const char *encoding_name);
 
 #endif /* __TIZEN_MEDIA_WEBRTC_SOURCE_COMMON_H__ */

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index b2df18c86e3dbecf4a2b14f5f7fa55bd22b79e12..44645c19a814980fd538ef5fc2a52d201246b1b0 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:    0.3.166
+Version:    0.3.167
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 816b76c01f8a66ac2781c32ff30e218b88184586..5d1126c9486c68d457ab9036b147cf3514c78674 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -33,26 +33,6 @@ static direction_info_s __direction_info[] = {
        [WEBRTC_TRANSCEIVER_DIRECTION_SENDRECV] = { "SENDRECV", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV }
 };
 
-typedef struct {
-       webrtc_transceiver_codec_e codec;
-       const char *gst_media_type;
-       const char *media_type;
-       const char *encoding_name;
-       int clock_rate;
-} rtp_payload_info_s;
-
-static rtp_payload_info_s __payload_info[] = {
-       { WEBRTC_TRANSCEIVER_CODEC_PCMU, MEDIA_TYPE_AUDIO_MULAW, "audio", "PCMU", 8000 },
-       { WEBRTC_TRANSCEIVER_CODEC_PCMA, MEDIA_TYPE_AUDIO_ALAW, "audio", "PCMA", 8000 },
-       { WEBRTC_TRANSCEIVER_CODEC_OPUS, MEDIA_TYPE_AUDIO_OPUS, "audio", "OPUS", 48000 },
-       { WEBRTC_TRANSCEIVER_CODEC_VP8, MEDIA_TYPE_VIDEO_VP8, "video", "VP8", 90000 },
-       { WEBRTC_TRANSCEIVER_CODEC_VP9, MEDIA_TYPE_VIDEO_VP9, "video", "VP9", 90000 },
-       { WEBRTC_TRANSCEIVER_CODEC_H264, MEDIA_TYPE_VIDEO_H264, "video", "H264", 90000 },
-       { 0 /* TBD */, MEDIA_TYPE_AUDIO_VORBIS, "audio", "VORBIS", 8000 },
-       { 0 /* TBD */, MEDIA_TYPE_VIDEO_JPEG, "video", "JPEG", 90000 },
-       { 0, NULL, NULL, NULL, 0 }
-};
-
 static int __link_source_with_webrtcbin(webrtc_gst_slot_s *source, GstElement *webrtcbin);
 static GstPadProbeReturn __camerasrc_probe_cb(GstPad *pad,  GstPadProbeInfo *info, gpointer u_data);
 
@@ -902,25 +882,6 @@ static int __add_transceiver(webrtc_gst_slot_s *source, webrtc_media_type_e medi
        return WEBRTC_ERROR_NONE;
 }
 
-static rtp_payload_info_s * __get_payload_info(const char *codec)
-{
-       int i = 0;
-       size_t n = 0;
-
-       RET_VAL_IF(codec == NULL, NULL, "codec is NULL");
-
-       n = ARRAY_SIZE(__payload_info);
-
-       for (i = 0; i < (int)n; i++) {
-               if ((__payload_info[i].encoding_name) && !strcasecmp(__payload_info[i].encoding_name, codec))
-                       return &__payload_info[i];
-       }
-
-       LOG_ERROR("could not find payload_info. codec[%s]", codec);
-
-       return NULL;
-}
-
 static void __check_and_add_recvonly_transceiver(webrtc_gst_slot_s *source)
 {
        rtp_payload_info_s *payload_info = NULL;
@@ -928,13 +889,13 @@ static void __check_and_add_recvonly_transceiver(webrtc_gst_slot_s *source)
        RET_IF(source == NULL, "source is NULL");
 
        if (source->av[AV_IDX_AUDIO].direction == WEBRTC_TRANSCEIVER_DIRECTION_RECVONLY && source->av[AV_IDX_AUDIO].codec) {
-               payload_info = __get_payload_info(source->av[AV_IDX_AUDIO].codec);
+               payload_info = _get_payload_info_by_encoding_name(source->av[AV_IDX_AUDIO].codec);
                if (payload_info)
                        __add_transceiver(source, WEBRTC_MEDIA_TYPE_AUDIO, payload_info);
        }
 
        if (source->av[AV_IDX_VIDEO].direction == WEBRTC_TRANSCEIVER_DIRECTION_RECVONLY && source->av[AV_IDX_VIDEO].codec) {
-               payload_info = __get_payload_info(source->av[AV_IDX_VIDEO].codec);
+               payload_info = _get_payload_info_by_encoding_name(source->av[AV_IDX_VIDEO].codec);
                if (payload_info)
                        __add_transceiver(source, WEBRTC_MEDIA_TYPE_VIDEO, payload_info);
        }
@@ -2274,7 +2235,6 @@ int _set_transceiver_codec(webrtc_s *webrtc, unsigned int source_id, webrtc_medi
        int ret;
        webrtc_gst_slot_s *source;
        int av_idx = (media_type == WEBRTC_MEDIA_TYPE_AUDIO) ? AV_IDX_AUDIO : AV_IDX_VIDEO;
-       int i;
        rtp_payload_info_s *payload_info = NULL;
 
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
@@ -2284,12 +2244,7 @@ int _set_transceiver_codec(webrtc_s *webrtc, unsigned int source_id, webrtc_medi
        RET_VAL_IF((source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE), WEBRTC_ERROR_INVALID_PARAMETER, "this API does not support the file source");
        RET_VAL_IF((source->type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET), WEBRTC_ERROR_INVALID_PARAMETER, "this API does not support the media packet source");
 
-       for (i = 0; i != (int)__payload_info[i].codec; i++) {
-               if (__payload_info[i].codec != codec)
-                       continue;
-               payload_info = &__payload_info[i];
-               break;
-       }
+       payload_info = _get_payload_info(codec);
        RET_VAL_IF(payload_info == NULL, WEBRTC_ERROR_INVALID_OPERATION, "could not find payload_info");
 
        /* NOTE: for null source, set media_types here exceptionally. */

diff --git a/src/webrtc_source_private.c b/src/webrtc_source_private.c
index 2e9d453e5302e93617bedcec553a1c70547b2938..b1f7a0a4da6579439e508d2b0f70d218c1c107e1 100644 (file)
--- a/src/webrtc_source_private.c
+++ b/src/webrtc_source_private.c
@@ -23,6 +23,18 @@ typedef struct {
        webrtc_gst_slot_s *source;
 } probe_userdata_s;
 
+static rtp_payload_info_s __payload_info[] = {
+       { WEBRTC_TRANSCEIVER_CODEC_PCMU, MEDIA_TYPE_AUDIO_MULAW, "audio", "PCMU", 8000 },
+       { WEBRTC_TRANSCEIVER_CODEC_PCMA, MEDIA_TYPE_AUDIO_ALAW, "audio", "PCMA", 8000 },
+       { WEBRTC_TRANSCEIVER_CODEC_OPUS, MEDIA_TYPE_AUDIO_OPUS, "audio", "OPUS", 48000 },
+       { WEBRTC_TRANSCEIVER_CODEC_VP8, MEDIA_TYPE_VIDEO_VP8, "video", "VP8", 90000 },
+       { WEBRTC_TRANSCEIVER_CODEC_VP9, MEDIA_TYPE_VIDEO_VP9, "video", "VP9", 90000 },
+       { WEBRTC_TRANSCEIVER_CODEC_H264, MEDIA_TYPE_VIDEO_H264, "video", "H264", 90000 },
+       { 0 /* TBD */, MEDIA_TYPE_AUDIO_VORBIS, "audio", "VORBIS", 8000 },
+       { 0 /* TBD */, MEDIA_TYPE_VIDEO_JPEG, "video", "JPEG", 90000 },
+       { 0, NULL, NULL, NULL, 0 }
+};
+
 const char *_get_audio_format_name(media_format_mimetype_e mime_type)
 {
        switch (mime_type) {
@@ -561,3 +573,36 @@ void _remove_probe_from_pad_for_render(webrtc_gst_slot_s *source, unsigned int i
                source->av[idx].render.appsrc_caps = NULL;
        }
 }
+
+rtp_payload_info_s * _get_payload_info(webrtc_transceiver_codec_e codec)
+{
+       int i = 0;
+
+       for (i = 0; i != (int)__payload_info[i].codec; i++) {
+               if (__payload_info[i].codec == codec)
+                       return &__payload_info[i];
+       }
+
+       LOG_ERROR("could not find payload_info. codec[%d]", codec);
+
+       return NULL;
+}
+
+rtp_payload_info_s * _get_payload_info_by_encoding_name(const char *encoding_name)
+{
+       int i = 0;
+       size_t n = 0;
+
+       RET_VAL_IF(encoding_name == NULL, NULL, "encoding_name is NULL");
+
+       n = ARRAY_SIZE(__payload_info);
+
+       for (i = 0; i < (int)n; i++) {
+               if ((__payload_info[i].encoding_name) && !strcasecmp(__payload_info[i].encoding_name, encoding_name))
+                       return &__payload_info[i];
+       }
+
+       LOG_ERROR("could not find payload_info. encoding_name[%s]", encoding_name);
+
+       return NULL;
+}