webrtc_source: refactor audio/video branches using static mapping table

[Version] 0.2.118
[Issue Type] Refactoring

Change-Id: I5046a03cf070e54a8e7624b273219ac4099e0d3b

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 12ae164673e890983dd45c3f5647e99324aea3e6..0ffebebe33a4abd90c3ce5d8ba4267cb2edbfd03 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:    0.2.117
+Version:    0.2.118
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 7fb0d64a4107278ed61d446f7abe024bfad2925e..f8fd51846ae1094767d508382557e820c1959d2d 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -76,6 +76,10 @@ do { \
        LOG_DEBUG("%s is prepended", GST_ELEMENT_NAME(x_element)); \
 } while (0)
 
+#define GET_MEDIA_TYPE_NAME(x_is_audio) (x_is_audio) ? "audio" : "video"
+#define GET_AV_IDX(x_is_audio) (x_is_audio) ? AV_IDX_AUDIO : AV_IDX_VIDEO
+#define GET_AV_IDX_BY_TYPE(x_media_type) GET_AV_IDX(x_media_type == MEDIA_TYPE_AUDIO)
+
 typedef struct {
        media_packet_h packet;
        GstBuffer *buffer;
@@ -97,6 +101,31 @@ static direction_info_s __direction_info[] = {
        [WEBRTC_TRANSCEIVER_DIRECTION_SENDRECV] = { "SENDRECV", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV }
 };
 
+typedef struct {
+       const char *appsrc_name;
+       const char *queue_name;
+       const char *payload_name;
+       const char *capsfilter_name;
+       const char *fakesink_name;
+} av_mapping_table_s;
+
+static av_mapping_table_s _av_tbl[AV_IDX_MAX] = {
+       {
+               DEFAULT_NAME_AUDIO_APPSRC,
+               DEFAULT_NAME_AUDIO_QUEUE,
+               DEFAULT_NAME_AUDIO_PAYLOAD,
+               DEFAULT_NAME_AUDIO_CAPSFILTER,
+               DEFAULT_NAME_AUDIO_FAKESINK,
+       },
+       {
+               DEFAULT_NAME_VIDEO_APPSRC,
+               DEFAULT_NAME_VIDEO_QUEUE,
+               DEFAULT_NAME_VIDEO_PAYLOAD,
+               DEFAULT_NAME_VIDEO_CAPSFILTER,
+               DEFAULT_NAME_VIDEO_FAKESINK,
+       }
+};
+
 static int __link_source_with_webrtcbin(webrtc_gst_slot_s *source, GstElement *webrtcbin);
 
 static const char * __get_audio_media_type(const char *codec_name)
@@ -589,14 +618,14 @@ static GstCaps *__make_encoded_caps_from_media_format(webrtc_gst_slot_s *source,
 static GstCaps *__make_rtp_caps(const gchar *media_type, unsigned int payload_id)
 {
        GstCaps *caps;
-       bool is_video;
+       bool is_audio;
 
        RET_VAL_IF(media_type == NULL, NULL, "media_type is NULL");
 
-       is_video = (g_strrstr(media_type, "video") || g_strrstr(media_type, "image")) ? TRUE : FALSE;
+       is_audio = !(g_strrstr(media_type, "video") || g_strrstr(media_type, "image"));
 
        caps = gst_caps_new_simple("application/x-rtp",
-                               "media", G_TYPE_STRING, is_video ? "video" : "audio",
+                               "media", G_TYPE_STRING, GET_MEDIA_TYPE_NAME(is_audio),
                                "payload", G_TYPE_INT, payload_id, NULL);
 
        PRINT_CAPS(caps, "RTP");
@@ -843,7 +872,7 @@ static int __create_rest_of_elements(webrtc_s *webrtc, webrtc_gst_slot_s *source
                LOG_ERROR("not ready for this media_types[0x%x]", source->media_types);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
-       idx = (source->media_types == MEDIA_TYPE_AUDIO) ? AV_IDX_AUDIO : AV_IDX_VIDEO;
+       idx = GET_AV_IDX_BY_TYPE(source->media_types);
 
        if (need_capsfilter) {
                if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, ELEMENT_NAME_FIRST_CAPSFILTER)))
@@ -983,7 +1012,7 @@ static int __create_rest_of_elements_for_encoded_format(webrtc_s *webrtc, webrtc
                g_free(media_type);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
-       source->av[source->media_types == MEDIA_TYPE_VIDEO ? AV_IDX_VIDEO : AV_IDX_AUDIO].payload_id = payload_id;
+       source->av[GET_AV_IDX_BY_TYPE(source->media_types)].payload_id = payload_id;
 
        if ((sink_caps = __make_rtp_caps(media_type, payload_id))) {
                g_object_set(G_OBJECT(*capsfilter), "caps", sink_caps, NULL);
@@ -1051,7 +1080,7 @@ static GstPadProbeReturn __payloaded_data_probe_cb(GstPad *pad, GstPadProbeInfo
        if (probe_data->source->av[probe_data->av_idx].pause) {
                if (counts[probe_data->av_idx]++ % 10 == 0)
                        LOG_DEBUG("paused, drop [%s] buffer[%p] of pad[%p], source[%p], count[%u]",
-                               probe_data->av_idx == AV_IDX_AUDIO ? "AUDIO" : "VIDEO",
+                               GET_MEDIA_TYPE_NAME(probe_data->av_idx == AV_IDX_AUDIO),
                                buffer, pad, probe_data->source, counts[probe_data->av_idx]);
                return GST_PAD_PROBE_DROP;
        }
@@ -1059,7 +1088,7 @@ static GstPadProbeReturn __payloaded_data_probe_cb(GstPad *pad, GstPadProbeInfo
        if (counts[probe_data->av_idx] > 0) {
                counts[probe_data->av_idx] = 0;
                LOG_DEBUG("play again, [%s] buffer[%p] of pad[%p], source[%p]",
-                       probe_data->av_idx == AV_IDX_AUDIO ? "AUDIO" : "VIDEO", buffer, pad, probe_data->source);
+                       GET_MEDIA_TYPE_NAME(probe_data->av_idx == AV_IDX_AUDIO), buffer, pad, probe_data->source);
        }
 
        return GST_PAD_PROBE_OK;
@@ -1646,23 +1675,21 @@ exit:
 static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_type)
 {
        int ret = WEBRTC_ERROR_NONE;
-       gboolean is_audio;
        GstPad *src_pad = NULL;
        GstElement *appsrc = NULL;
        GstElement *queue = NULL;
        GstElement *capsfilter = NULL;
+       int idx = GET_AV_IDX_BY_TYPE(media_type);
 
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
        RET_VAL_IF(source->bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL");
 
-       is_audio = (media_type == MEDIA_TYPE_AUDIO) ? TRUE : FALSE;
-
        source->media_types |= media_type;
 
        ret = _add_no_target_ghostpad_to_slot(source, true, &src_pad);
        RET_VAL_IF(ret != WEBRTC_ERROR_NONE, ret, "failed to _add_no_target_ghostpad_to_slot()");
 
-       if (!(appsrc = _create_element("appsrc", is_audio ? DEFAULT_NAME_AUDIO_APPSRC : DEFAULT_NAME_VIDEO_APPSRC)))
+       if (!(appsrc = _create_element("appsrc", _av_tbl[idx].appsrc_name)))
                return WEBRTC_ERROR_INVALID_OPERATION;
 
        g_object_set(G_OBJECT(appsrc),
@@ -1670,12 +1697,12 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
                        "format", GST_FORMAT_TIME,
                        NULL);
 
-       if (!(queue = _create_element("queue", is_audio ? DEFAULT_NAME_AUDIO_QUEUE : DEFAULT_NAME_VIDEO_QUEUE))) {
+       if (!(queue = _create_element("queue", _av_tbl[idx].queue_name))) {
                SAFE_GST_OBJECT_UNREF(appsrc);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
-       if (!(capsfilter = _create_element("capsfilter", is_audio ? DEFAULT_NAME_AUDIO_CAPSFILTER : DEFAULT_NAME_VIDEO_CAPSFILTER))) {
+       if (!(capsfilter = _create_element("capsfilter", _av_tbl[idx].capsfilter_name))) {
                SAFE_GST_OBJECT_UNREF(appsrc);
                SAFE_GST_OBJECT_UNREF(queue);
                return WEBRTC_ERROR_INVALID_OPERATION;
@@ -1707,7 +1734,7 @@ static int __build_filesrc_bin(webrtc_gst_slot_s *source, media_type_e media_typ
        if (ret != WEBRTC_ERROR_NONE)
                goto error;
 
-       __add_probe_to_pad_for_pause(source, is_audio ? AV_IDX_AUDIO : AV_IDX_VIDEO, src_pad, __payloaded_data_probe_cb);
+       __add_probe_to_pad_for_pause(source, idx, src_pad, __payloaded_data_probe_cb);
 
        return WEBRTC_ERROR_NONE;
 
@@ -1719,22 +1746,22 @@ error:
 static void __remove_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, bool is_audio)
 {
        GstBin *bin = NULL;
-
        GstElement *payload = NULL;
        GstElement *capsfilter = NULL;
        GstElement *fakesink = NULL;
+       int idx = GET_AV_IDX(is_audio);
 
        RET_IF(source == NULL, "pad is NULL");
 
        bin = GST_BIN(source->filesrc_pipeline);
 
-       payload = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD: DEFAULT_NAME_VIDEO_PAYLOAD);
+       payload = gst_bin_get_by_name(bin, _av_tbl[idx].payload_name);
        RET_IF(payload == NULL, "payload is NULL");
 
-       capsfilter = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_CAPSFILTER : DEFAULT_NAME_VIDEO_CAPSFILTER);
+       capsfilter = gst_bin_get_by_name(bin, _av_tbl[idx].capsfilter_name);
        RET_IF(capsfilter == NULL, "capsfilter is NULL");
 
-       fakesink = gst_bin_get_by_name(bin, is_audio ? DEFAULT_NAME_AUDIO_FAKESINK: DEFAULT_NAME_VIDEO_FAKESINK);
+       fakesink = gst_bin_get_by_name(bin, _av_tbl[idx].fakesink_name);
        RET_IF(fakesink == NULL, "fakesink is NULL");
 
        gst_bin_remove_many(bin, payload, capsfilter, fakesink, NULL);
@@ -1799,7 +1826,7 @@ static GstElement * __create_payload_for_filesrc_pipeline(GstPad *pad, bool is_a
 
        RET_VAL_IF(payload == NULL, NULL, "payload is NULL");
 
-       gst_element_set_name(payload, is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+       gst_element_set_name(payload, _av_tbl[GET_AV_IDX(is_audio)].payload_name);
 
        return payload;
 }
@@ -1812,7 +1839,7 @@ static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s
 
        RET_VAL_IF(source == NULL, NULL, "source is NULL");
 
-       if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, is_audio ? DEFAULT_NAME_AUDIO_CAPSFILTER : DEFAULT_NAME_VIDEO_CAPSFILTER)))
+       if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, _av_tbl[GET_AV_IDX(is_audio)].capsfilter_name)))
                return NULL;
 
        payload_id = __get_available_payload_id(source->webrtc);
@@ -1821,9 +1848,9 @@ static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s
                return NULL;
        }
 
-       source->av[is_audio ? AV_IDX_AUDIO : AV_IDX_VIDEO].payload_id = payload_id;
+       source->av[GET_AV_IDX(is_audio)].payload_id = payload_id;
 
-       if ((sink_caps = __make_rtp_caps(is_audio ? "audio" : "video", payload_id))) {
+       if ((sink_caps = __make_rtp_caps(GET_MEDIA_TYPE_NAME(is_audio), payload_id))) {
                g_object_set(G_OBJECT(capsfilter), "caps", sink_caps, NULL);
                gst_caps_unref(sink_caps);
        }
@@ -1838,7 +1865,7 @@ static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *s
 
        RET_VAL_IF(source == NULL, NULL, "source is NULL");
 
-       if (!(fakesink = _create_element("fakesink", is_audio ? DEFAULT_NAME_AUDIO_FAKESINK : DEFAULT_NAME_VIDEO_FAKESINK)))
+       if (!(fakesink = _create_element("fakesink", _av_tbl[GET_AV_IDX(is_audio)].fakesink_name)))
                return NULL;
 
        sink_pad = gst_element_get_static_pad(fakesink, "sink");
@@ -1850,7 +1877,10 @@ static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *s
                "signal-handoffs", TRUE,
                NULL);
 
-       g_signal_connect(fakesink, "handoff", is_audio ? G_CALLBACK(__filesrc_pipeline_audio_stream_handoff_cb) : G_CALLBACK(__filesrc_pipeline_video_stream_handoff_cb), (gpointer)source);
+       g_signal_connect(fakesink,
+               "handoff",
+               G_CALLBACK(is_audio ? __filesrc_pipeline_audio_stream_handoff_cb : __filesrc_pipeline_video_stream_handoff_cb),
+               (gpointer)source);
 
        return fakesink;
 }
@@ -1917,9 +1947,9 @@ static int __link_decodebin_with_payload(GstPad *pad, webrtc_gst_slot_s *source,
        RET_VAL_IF(pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "pad is NULL");
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
 
-       payload = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+       payload = gst_bin_get_by_name(GST_BIN(source->filesrc_pipeline), _av_tbl[GET_AV_IDX(is_audio)].payload_name);
        if (!payload) {
-               LOG_ERROR("failed to get element [%s]", is_audio ? DEFAULT_NAME_AUDIO_PAYLOAD : DEFAULT_NAME_VIDEO_PAYLOAD);
+               LOG_ERROR("failed to get element [%s]", _av_tbl[GET_AV_IDX(is_audio)].payload_name);
                return WEBRTC_ERROR_INVALID_OPERATION;
        }
 
@@ -1964,7 +1994,7 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        }
 
        is_audio = (g_strrstr(media_type, "audio")) ? TRUE : FALSE;
-       av_idx = (is_audio) ? AV_IDX_AUDIO : AV_IDX_VIDEO;
+       av_idx = GET_AV_IDX(is_audio);
 
        if (source->av[av_idx].src_pad_probe_id > 0) {
                LOG_INFO("Pipeline already built");
@@ -2251,7 +2281,7 @@ static int __complete_mediapacketsrc_from_raw_format(webrtc_s *webrtc, webrtc_gs
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
        RET_VAL_IF(source->bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL");
 
-       src_pad = (source->media_types == MEDIA_TYPE_AUDIO) ? &source->av[AV_IDX_AUDIO].src_pad : &source->av[AV_IDX_VIDEO].src_pad;
+       src_pad = &source->av[GET_AV_IDX_BY_TYPE(source->media_types)].src_pad;
        ret = _add_no_target_ghostpad_to_slot(source, true, src_pad);
        RET_VAL_IF(ret != WEBRTC_ERROR_NONE, ret, "failed to _add_no_target_ghostpad_to_slot()");
 
@@ -2317,7 +2347,7 @@ static int __complete_mediapacketsrc_from_encoded_format(webrtc_s *webrtc, webrt
        RET_VAL_IF(source->bin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bin is NULL");
        RET_VAL_IF(source->media_format == NULL, WEBRTC_ERROR_INVALID_OPERATION, "media_format is NULL");
 
-       src_pad = (source->media_types == MEDIA_TYPE_AUDIO) ? &source->av[AV_IDX_AUDIO].src_pad : &source->av[AV_IDX_VIDEO].src_pad;
+       src_pad = &source->av[GET_AV_IDX_BY_TYPE(source->media_types)].src_pad;
        ret = _add_no_target_ghostpad_to_slot(source, true, src_pad);
        RET_VAL_IF(ret != WEBRTC_ERROR_NONE, ret, "failed to _add_no_target_ghostpad_to_slot()");
 
@@ -3564,7 +3594,7 @@ int _set_video_mute(webrtc_s *webrtc, unsigned int source_id, bool mute)
        }
 
        if (ret == WEBRTC_ERROR_NONE)
-               source->video_muted = mute ? true : false;
+               source->video_muted = mute;
 
        return ret;
 }
@@ -3859,9 +3889,9 @@ static void __loopback_decodebin_pad_added_cb(GstElement *decodebin, GstPad *new
 
 static int __build_loopback_render_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s *source, media_type_e type, unsigned int *track_id)
 {
-       int idx = (type == MEDIA_TYPE_AUDIO) ? AV_IDX_AUDIO : AV_IDX_VIDEO;
        GstElement *appsrc;
        gchar *pipeline_name;
+       int idx = GET_AV_IDX_BY_TYPE(type);
 
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
        RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -3870,7 +3900,7 @@ static int __build_loopback_render_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s
                RET_VAL_IF(source->display->surface == NULL, WEBRTC_ERROR_INVALID_OPERATION, "display->surface is NULL");
        }
 
-       pipeline_name = g_strdup_printf("webrtc-source_%u-%s-render-pipeline", source->id, type == MEDIA_TYPE_AUDIO ? "audio" : "video");
+       pipeline_name = g_strdup_printf("webrtc-source_%u-%s-render-pipeline", source->id, GET_MEDIA_TYPE_NAME(type == MEDIA_TYPE_AUDIO));
        source->av[idx].render.pipeline = gst_pipeline_new(pipeline_name);
        g_free(pipeline_name);
        RET_VAL_IF(source->av[idx].render.pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "render_pipeline is NULL");