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diff --git a/abi_symbols b/abi_symbols
--- a/abi_symbols
+++ b/abi_symbols
@@ -1,4 +1,5 @@
libgstallocators-1.0.so.0:gst_dmabuf_allocator_alloc
+libgstallocators-1.0.so.0:gst_dmabuf_allocator_get_type
libgstallocators-1.0.so.0:gst_dmabuf_allocator_new
libgstallocators-1.0.so.0:gst_dmabuf_memory_get_fd
libgstallocators-1.0.so.0:gst_fd_allocator_alloc
@@ -7,6 +8,7 @@
libgstallocators-1.0.so.0:gst_fd_memory_get_fd
libgstallocators-1.0.so.0:gst_is_dmabuf_memory
libgstallocators-1.0.so.0:gst_is_fd_memory
+libgstapp-1.0.so.0:gst_app_sink_get_buffer_list_support
libgstapp-1.0.so.0:gst_app_sink_get_caps
libgstapp-1.0.so.0:gst_app_sink_get_drop
libgstapp-1.0.so.0:gst_app_sink_get_emit_signals
@@ -16,6 +18,7 @@
libgstapp-1.0.so.0:gst_app_sink_is_eos
libgstapp-1.0.so.0:gst_app_sink_pull_preroll
libgstapp-1.0.so.0:gst_app_sink_pull_sample
+libgstapp-1.0.so.0:gst_app_sink_set_buffer_list_support
libgstapp-1.0.so.0:gst_app_sink_set_callbacks
libgstapp-1.0.so.0:gst_app_sink_set_caps
libgstapp-1.0.so.0:gst_app_sink_set_drop
@@ -47,6 +50,7 @@
libgstapp-1.0.so.0:gst_app_stream_type_get_type
libgstaudio-1.0.so.0:_gst_audio_decoder_error
libgstaudio-1.0.so.0:gst_audio_base_sink_create_ringbuffer
+libgstaudio-1.0.so.0:gst_audio_base_sink_discont_reason_get_type
libgstaudio-1.0.so.0:gst_audio_base_sink_get_alignment_threshold
libgstaudio-1.0.so.0:gst_audio_base_sink_get_discont_wait
libgstaudio-1.0.so.0:gst_audio_base_sink_get_drift_tolerance
@@ -102,6 +106,7 @@
libgstaudio-1.0.so.0:gst_audio_converter_new
libgstaudio-1.0.so.0:gst_audio_converter_reset
libgstaudio-1.0.so.0:gst_audio_converter_samples
+libgstaudio-1.0.so.0:gst_audio_converter_supports_inplace
libgstaudio-1.0.so.0:gst_audio_converter_update_config
libgstaudio-1.0.so.0:gst_audio_decoder_allocate_output_buffer
libgstaudio-1.0.so.0:gst_audio_decoder_finish_frame
@@ -242,6 +247,7 @@
libgstaudio-1.0.so.0:gst_audio_ring_buffer_release
libgstaudio-1.0.so.0:gst_audio_ring_buffer_samples_done
libgstaudio-1.0.so.0:gst_audio_ring_buffer_set_callback
+libgstaudio-1.0.so.0:gst_audio_ring_buffer_set_callback_full
libgstaudio-1.0.so.0:gst_audio_ring_buffer_set_channel_positions
libgstaudio-1.0.so.0:gst_audio_ring_buffer_set_flushing
libgstaudio-1.0.so.0:gst_audio_ring_buffer_set_sample
@@ -376,6 +382,7 @@
libgstpbutils-1.0.so.0:gst_encoding_container_profile_new
libgstpbutils-1.0.so.0:gst_encoding_list_all_targets
libgstpbutils-1.0.so.0:gst_encoding_list_available_categories
+libgstpbutils-1.0.so.0:gst_encoding_profile_copy
libgstpbutils-1.0.so.0:gst_encoding_profile_find
libgstpbutils-1.0.so.0:gst_encoding_profile_from_discoverer
libgstpbutils-1.0.so.0:gst_encoding_profile_get_allow_dynamic_output
@@ -420,6 +427,7 @@
libgstpbutils-1.0.so.0:gst_encoding_video_profile_set_pass
libgstpbutils-1.0.so.0:gst_encoding_video_profile_set_variableframerate
libgstpbutils-1.0.so.0:gst_install_plugins_async
+libgstpbutils-1.0.so.0:gst_install_plugins_context_copy
libgstpbutils-1.0.so.0:gst_install_plugins_context_free
libgstpbutils-1.0.so.0:gst_install_plugins_context_get_type
libgstpbutils-1.0.so.0:gst_install_plugins_context_new
@@ -618,7 +626,12 @@
libgstrtp-1.0.so.0:gst_rtp_payload_info_for_name
libgstrtp-1.0.so.0:gst_rtp_payload_info_for_pt
libgstrtp-1.0.so.0:gst_rtp_profile_get_type
+libgstrtsp-1.0.so.0:gst_rtsp_auth_credential_get_type
+libgstrtsp-1.0.so.0:gst_rtsp_auth_credentials_free
libgstrtsp-1.0.so.0:gst_rtsp_auth_method_get_type
+libgstrtsp-1.0.so.0:gst_rtsp_auth_param_copy
+libgstrtsp-1.0.so.0:gst_rtsp_auth_param_free
+libgstrtsp-1.0.so.0:gst_rtsp_auth_param_get_type
libgstrtsp-1.0.so.0:gst_rtsp_connection_accept
libgstrtsp-1.0.so.0:gst_rtsp_connection_clear_auth_params
libgstrtsp-1.0.so.0:gst_rtsp_connection_close
@@ -673,6 +686,7 @@
libgstrtsp-1.0.so.0:gst_rtsp_family_get_type
libgstrtsp-1.0.so.0:gst_rtsp_find_header_field
libgstrtsp-1.0.so.0:gst_rtsp_find_method
+libgstrtsp-1.0.so.0:gst_rtsp_generate_digest_auth_response
libgstrtsp-1.0.so.0:gst_rtsp_header_allow_multiple
libgstrtsp-1.0.so.0:gst_rtsp_header_as_text
libgstrtsp-1.0.so.0:gst_rtsp_header_field_get_type
@@ -694,6 +708,7 @@
libgstrtsp-1.0.so.0:gst_rtsp_message_new_data
libgstrtsp-1.0.so.0:gst_rtsp_message_new_request
libgstrtsp-1.0.so.0:gst_rtsp_message_new_response
+libgstrtsp-1.0.so.0:gst_rtsp_message_parse_auth_credentials
libgstrtsp-1.0.so.0:gst_rtsp_message_parse_data
libgstrtsp-1.0.so.0:gst_rtsp_message_parse_request
libgstrtsp-1.0.so.0:gst_rtsp_message_parse_response
@@ -1085,6 +1100,7 @@
libgstvideo-1.0.so.0:gst_video_decoder_add_to_frame
libgstvideo-1.0.so.0:gst_video_decoder_allocate_output_buffer
libgstvideo-1.0.so.0:gst_video_decoder_allocate_output_frame
+libgstvideo-1.0.so.0:gst_video_decoder_allocate_output_frame_with_params
libgstvideo-1.0.so.0:gst_video_decoder_drop_frame
libgstvideo-1.0.so.0:gst_video_decoder_finish_frame
libgstvideo-1.0.so.0:gst_video_decoder_get_allocator
@@ -1144,6 +1160,9 @@
libgstvideo-1.0.so.0:gst_video_event_parse_downstream_force_key_unit
libgstvideo-1.0.so.0:gst_video_event_parse_still_frame
libgstvideo-1.0.so.0:gst_video_event_parse_upstream_force_key_unit
+libgstvideo-1.0.so.0:gst_video_field_order_from_string
+libgstvideo-1.0.so.0:gst_video_field_order_get_type
+libgstvideo-1.0.so.0:gst_video_field_order_to_string
libgstvideo-1.0.so.0:gst_video_filter_get_type
libgstvideo-1.0.so.0:gst_video_flags_get_type
libgstvideo-1.0.so.0:gst_video_format_flags_get_type
@@ -1266,6 +1285,7 @@
libgstvideo-1.0.so.0:gst_video_tile_mode_get_type
libgstvideo-1.0.so.0:gst_video_tile_type_get_type
libgstvideo-1.0.so.0:gst_video_time_code_add_frames
+libgstvideo-1.0.so.0:gst_video_time_code_add_interval
libgstvideo-1.0.so.0:gst_video_time_code_clear
libgstvideo-1.0.so.0:gst_video_time_code_compare
libgstvideo-1.0.so.0:gst_video_time_code_copy
@@ -1274,11 +1294,21 @@
libgstvideo-1.0.so.0:gst_video_time_code_get_type
libgstvideo-1.0.so.0:gst_video_time_code_increment_frame
libgstvideo-1.0.so.0:gst_video_time_code_init
+libgstvideo-1.0.so.0:gst_video_time_code_init_from_date_time
+libgstvideo-1.0.so.0:gst_video_time_code_interval_clear
+libgstvideo-1.0.so.0:gst_video_time_code_interval_copy
+libgstvideo-1.0.so.0:gst_video_time_code_interval_free
+libgstvideo-1.0.so.0:gst_video_time_code_interval_get_type
+libgstvideo-1.0.so.0:gst_video_time_code_interval_init
+libgstvideo-1.0.so.0:gst_video_time_code_interval_new
+libgstvideo-1.0.so.0:gst_video_time_code_interval_new_from_string
libgstvideo-1.0.so.0:gst_video_time_code_is_valid
libgstvideo-1.0.so.0:gst_video_time_code_meta_api_get_type
libgstvideo-1.0.so.0:gst_video_time_code_meta_get_info
libgstvideo-1.0.so.0:gst_video_time_code_new
libgstvideo-1.0.so.0:gst_video_time_code_new_empty
+libgstvideo-1.0.so.0:gst_video_time_code_new_from_date_time
+libgstvideo-1.0.so.0:gst_video_time_code_new_from_string
libgstvideo-1.0.so.0:gst_video_time_code_nsec_since_daily_jam
libgstvideo-1.0.so.0:gst_video_time_code_to_date_time
libgstvideo-1.0.so.0:gst_video_time_code_to_string
diff --git a/package.yml b/package.yml
--- a/package.yml
+++ b/package.yml
@@ -1,8 +1,8 @@
name : gstreamer-1.0-plugins-base
-version : 1.10.4
-release : 16
+version : 1.12.2
+release : 17
source :
- - https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.10.4.tar.xz : f6d245b6b3d4cb733f81ebb021074c525ece83db0c10e932794b339b8d935eb7
+ - https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.12.2.tar.xz : 5067dce3afe197a9536fea0107c77213fab536dff4a213b07fc60378d5510675
license :
- LGPL-2.0
component : multimedia.gstreamer
@@ -26,9 +26,10 @@
The GStreamer Base Plug-ins is a well-groomed and well-maintained collection of GStreamer plug-ins and elements, spanning the range of possible types of elements one would want to write for GStreamer. You will need at least one of Good, Bad, Ugly or Libav plugins for GStreamer applications to function properly.
setup : |
%configure --disable-static \
- --with-package-name="GStreamer Base Plugins 1.10.4 Solus" \
+ --with-package-name="GStreamer Base Plugins 1.12.2 Solus" \
--with-package-origin="https://solus-project.com"
build : |
%make
install : |
%make_install
+
diff --git a/pspec_x86_64.xml b/pspec_x86_64.xml
--- a/pspec_x86_64.xml
+++ b/pspec_x86_64.xml
@@ -2,8 +2,8 @@
<Source>
<Name>gstreamer-1.0-plugins-base</Name>
<Packager>
- <Name>Ikey Doherty</Name>
- <Email>ikey@solus-project.com</Email>
+ <Name>Pierre-Yves</Name>
+ <Email>pyu@riseup.net</Email>
</Packager>
<License>LGPL-2.0</License>
<PartOf>multimedia.gstreamer</PartOf>
@@ -37,12 +37,14 @@
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstaudiorate.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstaudioresample.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstaudiotestsrc.so</Path>
- <Path fileType="library">/usr/lib64/gstreamer-1.0/libgstencodebin.so</Path>
+ <Path fileType="library">/usr/lib64/gstreamer-1.0/libgstencoding.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstgio.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstogg.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstopus.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstpango.so</Path>
+ <Path fileType="library">/usr/lib64/gstreamer-1.0/libgstpbtypes.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstplayback.so</Path>
+ <Path fileType="library">/usr/lib64/gstreamer-1.0/libgstrawparse.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgstsubparse.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgsttcp.so</Path>
<Path fileType="library">/usr/lib64/gstreamer-1.0/libgsttheora.so</Path>
@@ -78,7 +80,7 @@
</Description>
<PartOf>programming.devel</PartOf>
<RuntimeDependencies>
- <Dependency release="16">gstreamer-1.0-plugins-base</Dependency>
+ <Dependency release="17">gstreamer-1.0-plugins-base</Dependency>
</RuntimeDependencies>
<Files>
<Path fileType="header">/usr/include/</Path>
@@ -97,12 +99,12 @@
</Files>
</Package>
<History>
- <Update release="16">
- <Date>2017-04-12</Date>
- <Version>1.10.4</Version>
+ <Update release="17">
+ <Date>2017-07-15</Date>
+ <Version>1.12.2</Version>
<Comment>Packaging update</Comment>
- <Name>Ikey Doherty</Name>
- <Email>ikey@solus-project.com</Email>
+ <Name>Pierre-Yves</Name>
+ <Email>pyu@riseup.net</Email>
</Update>
</History>
</PISI>
\ No newline at end of file

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