webrtc_source_file: Encoding audio stream from filesrc with default codec for compati...

gstreamer does not support payloaders for some encoding audio formats.
In this case, after decoding, encode it again with the corresponding format in ini.

[Version] 0.3.188
[Issue Type] Improvement

Change-Id: I5c2f19bfe0056986e0128770ef9038966b7e3989

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 71fc81be7376ed1ed85e9de0cfc6636952cf3f07..00556f90bdfdf39465f3fab18afbc172ab0e33cf 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:    0.3.187
+Version:    0.3.188
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
@@ -86,6 +86,8 @@ cp %{SOURCE1003} .
 export CFLAGS+=" -DSYSCONFDIR=\\\"%{_hal_sysconfdir}\\\""
 export CXXFLAGS+=" -DSYSCONFDIR=\\\"%{_hal_sysconfdir}\\\""
 
+export CFLAGS+=" -DSUPPORT_FILESRC_AUDIO_FORMAT_CHANGE"
+
 %if 0%{?gcov:1}
 export CFLAGS+=" -fprofile-arcs -ftest-coverage"
 export CXXFLAGS+=" -fprofile-arcs -ftest-coverage"

diff --git a/src/webrtc_source_file.c b/src/webrtc_source_file.c
index 623953e673546013d1e023dfbc2c320155943684..b410d5dd009f6bed0d8dc4b36c82751adf8be1bf 100644 (file)
--- a/src/webrtc_source_file.c
+++ b/src/webrtc_source_file.c
@@ -456,7 +456,22 @@ static GstAutoplugSelectResult __filesrc_pipeline_decodebin_autoplug_select_cb(G
        const gchar *klass = gst_element_factory_get_metadata(factory, GST_ELEMENT_METADATA_KLASS);
 
        if (g_strrstr(klass, "Codec/Decoder")) {
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+               const gchar *media_type = gst_structure_get_name(gst_caps_get_structure(caps, 0));
+               RET_VAL_IF(media_type == NULL, GST_AUTOPLUG_SELECT_TRY, "media_type is NULL");
+
+               if (g_strrstr(media_type, "audio")) {
+                       GstElement *payloader = __create_payloader_for_filesrc_pipeline(pad, true);
+                       if (!payloader) {
+                               LOG_INFO("failed to find payloader. decode it");
+                               return GST_AUTOPLUG_SELECT_TRY;
+                       }
+
+                       gst_object_unref(payloader);
+               }
+#endif
                LOG_INFO("expose [%s]", klass);
+
                return GST_AUTOPLUG_SELECT_EXPOSE;
        }
 
@@ -560,6 +575,62 @@ static int __link_decodebin_with_queue(GstPad *pad, webrtc_gst_slot_s *source, b
        return WEBRTC_ERROR_NONE;
 }
 
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+static int __create_rest_of_raw_audio_elements_for_filesrc_pipeline(webrtc_s *webrtc, webrtc_gst_slot_s *source, GList **element_list)
+{
+       GstElement *queue = NULL;
+       GstElement *resample = NULL;
+       GstElement *conv = NULL;
+       GstElement *fakesink = NULL;
+       GList *_element_list = NULL;
+
+       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+       RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+       RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
+
+       if (!(queue = _create_element(DEFAULT_ELEMENT_QUEUE, _get_element_name(GET_AV_IDX(true), ELEMENT_QUEUE))))
+               return WEBRTC_ERROR_INVALID_OPERATION;
+       APPEND_ELEMENT(_element_list, queue);
+
+       if (!(resample = _create_element(DEFAULT_ELEMENT_AUDIORESAMPLE, NULL)))
+               goto exit;
+       APPEND_ELEMENT(_element_list, resample);
+
+       if (!(conv = _create_element(DEFAULT_ELEMENT_AUDIOCONVERT, NULL)))
+               goto exit;
+       APPEND_ELEMENT(_element_list, conv);
+
+       if (_create_rest_of_elements(webrtc, source, false, &_element_list, true) != WEBRTC_ERROR_NONE)
+               goto exit;
+
+       if(!(fakesink = __prepare_fakesink_for_filesrc_pipeline(source, true)))
+               goto exit;
+       APPEND_ELEMENT(_element_list, fakesink);
+
+       if (!_add_elements_to_bin(GST_BIN(source->filesrc_pipeline), _element_list))
+               goto exit;
+
+       if (!_link_elements(_element_list))
+               goto exit_with_remove_from_bin;
+
+       if (!_sync_elements_state_with_parent(_element_list))
+               goto exit_with_remove_from_bin;
+
+       *element_list = _element_list;
+
+       return WEBRTC_ERROR_NONE;
+
+exit_with_remove_from_bin:
+       _remove_elements_from_bin(GST_BIN(source->filesrc_pipeline), _element_list);
+       SAFE_G_LIST_FREE(_element_list);
+       return WEBRTC_ERROR_INVALID_OPERATION;
+
+exit:
+       SAFE_G_LIST_FREE_FULL(_element_list, gst_object_unref);
+       return WEBRTC_ERROR_INVALID_OPERATION;
+}
+#endif
+
 static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPad *pad, gpointer data)
 {
        int ret = WEBRTC_ERROR_NONE;
@@ -570,6 +641,10 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        GList *filesrc_element_list = NULL;
        GList *bin_element_list = NULL;
        GstElement *queue = NULL;
+       bool need_decoding = true;
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       GList *audio_pipeline_element_list = NULL;
+#endif
 
        RET_IF(source == NULL, "source is NULL");
        RET_IF(source->filesrc_pipeline == NULL, "filesrc_pipeline is NULL");
@@ -587,6 +662,17 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        is_audio = _is_audio_media_type(media_type);
        av_idx = GET_AV_IDX(is_audio);
 
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       if (is_audio && g_strrstr(media_type, "raw")) {
+               need_decoding = false;
+
+               const ini_item_media_source_s *ini_source = _ini_get_source_by_type(&source->webrtc->ini, source->type);
+               if (ini_source == NULL)
+                       ini_source = &(source->webrtc->ini.media_source);
+
+               source->av[AV_IDX_AUDIO].codec = ini_source->a_codecs[0];
+       }
+#endif
        g_free(media_type);
 
        if (source->av[av_idx].src_pad_probe_id > 0) {
@@ -597,7 +683,14 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
                return;
        }
 
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       if (need_decoding)
+               ret = __create_rest_of_elements_for_filesrc_pipeline(source, pad, &filesrc_element_list, is_audio);
+       else
+               ret = __create_rest_of_raw_audio_elements_for_filesrc_pipeline(source->webrtc, source, &audio_pipeline_element_list);
+#else
        ret = __create_rest_of_elements_for_filesrc_pipeline(source, pad, &filesrc_element_list, is_audio);
+#endif
        if (ret != WEBRTC_ERROR_NONE) {
                LOG_ERROR("failed to __create_rest_of_elements_for_filesrc_pipeline()");
                return;
@@ -627,12 +720,15 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
                goto exit_with_remove_from_bin;
        }
 
-       source->av[av_idx].render.need_decoding = true;
+       source->av[av_idx].render.need_decoding = need_decoding;
        source->av[av_idx].render.appsrc_caps = gst_pad_get_current_caps(pad);
        _add_probe_to_pad_for_render(source, av_idx, gst_element_get_static_pad(queue, "src"), _source_data_probe_cb);
 
        SAFE_G_LIST_FREE(filesrc_element_list);
        SAFE_G_LIST_FREE(bin_element_list);
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       SAFE_G_LIST_FREE(audio_pipeline_element_list);
+#endif
 
        GENERATE_DOT(source->webrtc, source->filesrc_pipeline, "%s.%s-%s",
                GST_ELEMENT_NAME(source->filesrc_pipeline), GST_ELEMENT_NAME(element), GST_PAD_NAME(pad));
@@ -646,8 +742,18 @@ exit_with_remove_from_bin:
        _remove_elements_from_bin(GST_BIN(source->bin), bin_element_list);
        SAFE_G_LIST_FREE(bin_element_list);
 exit:
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       if (filesrc_element_list)
+               _remove_elements_from_bin(GST_BIN(source->filesrc_pipeline), filesrc_element_list);
+       SAFE_G_LIST_FREE(filesrc_element_list);
+
+       if (audio_pipeline_element_list)
+               _remove_elements_from_bin(GST_BIN(source->filesrc_pipeline), audio_pipeline_element_list);
+       SAFE_G_LIST_FREE(audio_pipeline_element_list);
+#else
        _remove_elements_from_bin(GST_BIN(source->filesrc_pipeline), filesrc_element_list);
        SAFE_G_LIST_FREE(filesrc_element_list);
+#endif
        return;
 }
 

diff --git a/src/webrtc_source_private.c b/src/webrtc_source_private.c
index e694bb976abf8070be2fdd08d7b028cfea52c178..cb576d8e981333fb89849af34c11e8c64e4f3fdb 100644 (file)
--- a/src/webrtc_source_private.c
+++ b/src/webrtc_source_private.c
@@ -762,6 +762,9 @@ static GstCaps *__make_default_raw_caps(webrtc_gst_slot_s *source, webrtc_ini_s
        case WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST:
        case WEBRTC_MEDIA_SOURCE_TYPE_MIC:
        case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_AUDIO:
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       case WEBRTC_MEDIA_SOURCE_TYPE_FILE:
+#endif
                format = _get_gst_audio_raw_format_from_string(ini_source->a_raw_format);
                RET_VAL_IF(format == GST_AUDIO_FORMAT_UNKNOWN, NULL, "not supported raw format");
 
@@ -815,6 +818,9 @@ static GstCaps *__make_default_encoded_caps(webrtc_gst_slot_s *source, webrtc_in
        case WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST:
        case WEBRTC_MEDIA_SOURCE_TYPE_MIC:
        case WEBRTC_MEDIA_SOURCE_TYPE_CUSTOM_AUDIO:
+#ifdef SUPPORT_FILESRC_AUDIO_FORMAT_CHANGE
+       case WEBRTC_MEDIA_SOURCE_TYPE_FILE:
+#endif
                _media_type = _get_audio_media_type(source->av[AV_IDX_AUDIO].codec);
                RET_VAL_IF(_media_type == NULL, NULL, "_media_type is NULL");