gstreamer/gst-libs/gst
Wim Taymans c36ac3ce45 audioconvert: move audio quantize code to libs
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
2015-11-06 12:10:48 +01:00
..
allocators fdmemory: remove 'allow-none' annotation in gst_fd_allocator_alloc() doc 2015-09-23 10:01:41 -04:00
app app: pass PKG_CONFIG_PATH for gir files for libgstapp as well 2015-09-29 13:10:35 +01:00
audio audioconvert: move audio quantize code to libs 2015-11-06 12:10:48 +01:00
fft gi: Skip fft constructor for now 2015-06-16 14:36:44 -04:00
pbutils codec-utils: Add utilities for Opus caps and the OpusHead header 2015-11-03 20:35:33 +02:00
riff gi: Use INTROSPECTION_INIT for --add-init-section 2015-06-16 18:04:57 -04:00
rtp rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1 2015-10-20 12:10:52 +03:00
rtsp Update GLib dependency to 2.40.0 2015-10-02 22:19:52 +03:00
sdp sdp: prevent the sdp message parser from reading past the end of the buffer 2015-05-29 15:05:31 +02:00
tag gstreamer: base: Fix memory leaks when context parse fails. 2015-10-02 17:26:27 +03:00
video video: blend using OVER operation 2015-11-04 21:58:32 +01:00
gettext.h Fix FSF address 2012-11-03 23:05:09 +00:00
glib-compat-private.h Fix FSF address 2012-11-03 23:05:09 +00:00
gst-i18n-app.h tools: add simple command-line gst-play utility for testing purposes 2013-08-16 15:45:23 +01:00
gst-i18n-plugin.h Fix FSF address 2012-11-03 23:05:09 +00:00
Makefile.am libs: build rtp after audio 2015-09-04 09:53:42 +01:00