gstreamer/gst-libs/gst
Wim Taymans de37491662 audio-converter: simplify API
Remove the consumed/produced output fields from the resampler and
converter. Let the caler specify the right number of input/output
samples so we can be more optimal.
Use just one function to update the converter configuration.
Simplify some things internally.
Make it possible to use writable input as temp space in audioconvert.
2016-03-28 13:25:50 +02:00
..
allocators base: Add g_autoptr() support to all types 2015-12-14 13:39:43 -05:00
app base: use new gst_element_class_add_static_pad_template() 2016-03-24 14:25:41 +02:00
audio audio-converter: simplify API 2016-03-28 13:25:50 +02:00
fft Drop usage of deprecated g-ir-scanner --strip-prefix flag 2015-12-02 20:19:43 -08:00
pbutils codec-utils: Add utilities for AAC and the AACHead header 2016-03-24 14:27:21 +02:00
riff win32: remove outdated build cruft 2016-02-20 10:05:17 +00:00
rtp rtcpbuffer: Add API for APP packets 2016-03-24 14:24:11 +02:00
rtsp base: Add g_autoptr() support to all types 2015-12-14 13:39:43 -05:00
sdp docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc 2016-01-03 17:21:18 +00:00
tag base: use new gst_element_class_add_static_pad_template() 2016-03-24 14:25:41 +02:00
video video: update disted orc backup file 2016-02-27 00:13:03 +00:00
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 rtp: build audio library before rtp 2016-02-16 17:42:44 +02:00