Carlos Rafael Giani 9adaeb0a71 riff: Remove sample rate and channel count boundaries in caps ... WAV is too generic to impose more-or-less arbitrary boundaries on the sample rate and channel count caps. For example, there are 384 kHz WAV files. Another example: it is in theory possible that somebody puts DSD data into a WAV file, which will then have a sample rate of ~2.8 MHz. For this reason, get rid of the rate and channel caps unless they are fixed values. Downstream anyway usually knows the limitations better. https://bugzilla.gnome.org/show_bug.cgi?id=761514		2016-08-02 15:25:53 +03:00
..
allocators	g-i: pass compiler env to g-ir-scanner	2016-05-24 00:44:21 +01:00
app	appsink: add _pull_sample/preroll() variants with timeout	2016-07-18 16:55:16 +01:00
audio	audioclock: use GST_STIME_FORMAT for the correct argument	2016-07-20 12:28:54 +01:00
fft	g-i: pass compiler env to g-ir-scanner	2016-05-24 00:44:21 +01:00
pbutils	pbutils: Add more h264 scalable profiles	2016-08-02 14:25:57 +03:00
riff	riff: Remove sample rate and channel count boundaries in caps	2016-08-02 15:25:53 +03:00
rtp	rtpbuffer: Add some const qualifiers	2016-07-26 17:46:38 +03:00
rtsp	rtspconnection: Fix potential deadlock caused by blocking read forever	2016-07-07 19:15:18 +03:00
sdp	g-i: pass compiler env to g-ir-scanner	2016-05-24 00:44:21 +01:00
tag	tagdemux: fix handling of very short files in push mode	2016-06-30 18:53:07 +01:00
video	videoorientation: Use G_DEFINE_INTERFACE instead of a manually written get_type()	2016-07-12 09:05:32 +03:00
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