mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
audiorate: add documentation
This commit is contained in:
parent
60635a9fbc
commit
b5fe63ed79
4 changed files with 48 additions and 0 deletions
|
@ -107,6 +107,7 @@ EXTRA_HFILES = \
|
|||
$(top_srcdir)/gst/gdp/gstgdppay.h \
|
||||
$(top_srcdir)/gst/playback/gstplay-enum.h \
|
||||
$(top_srcdir)/gst/playback/gstsubtitleoverlay.h \
|
||||
$(top_srcdir)/gst/audiorate/gstaudiorate.h \
|
||||
$(top_srcdir)/gst/audioresample/gstaudioresample.h \
|
||||
$(top_srcdir)/gst/tcp/gstmultifdsink.h \
|
||||
$(top_srcdir)/gst/tcp/gsttcpclientsrc.h \
|
||||
|
|
|
@ -24,6 +24,7 @@
|
|||
<xi:include href="xml/element-alsasink.xml" />
|
||||
<xi:include href="xml/element-alsasrc.xml" />
|
||||
<xi:include href="xml/element-audioconvert.xml" />
|
||||
<xi:include href="xml/element-audiorate.xml" />
|
||||
<xi:include href="xml/element-audioresample.xml" />
|
||||
<xi:include href="xml/element-audiotestsrc.xml" />
|
||||
<xi:include href="xml/element-cdparanoiasrc.xml" />
|
||||
|
|
|
@ -153,6 +153,20 @@ audio_convert_get_sizes
|
|||
audio_convert_prepare_context
|
||||
</SECTION>
|
||||
|
||||
<SECTION>
|
||||
<FILE>element-audiorate</FILE>
|
||||
<TITLE>audiorate</TITLE>
|
||||
GstAudioRate
|
||||
<SUBSECTION Standard>
|
||||
GST_AUDIO_RATE
|
||||
GST_IS_AUDIO_RATE
|
||||
GST_TYPE_AUDIO_RATE
|
||||
gst_audio_rate_get_type
|
||||
GST_AUDIO_RATE_CLASS
|
||||
GST_IS_AUDIO_RATE_CLASS
|
||||
GstAudioRateClass
|
||||
</SECTION>
|
||||
|
||||
<SECTION>
|
||||
<FILE>element-audioresample</FILE>
|
||||
<TITLE>audioresample</TITLE>
|
||||
|
|
|
@ -17,6 +17,38 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
/**
|
||||
* SECTION:element-audiorate
|
||||
* @see_also: #GstVideoRate
|
||||
*
|
||||
* This element takes an incoming stream of timestamped raw audio frames and
|
||||
* produces a perfect stream by inserting or dropping samples as needed.
|
||||
*
|
||||
* This operation may be of use to link to elements that require or otherwise
|
||||
* implicitly assume a perfect stream as they do not store timestamps,
|
||||
* but derive this by some means (e.g. bitrate for some AVI cases).
|
||||
*
|
||||
* The properties #GstAudioRate:in, #GstAudioRate:out, #GstAudioRate:add
|
||||
* and #GstAudioRate:drop can be read to obtain information about number of
|
||||
* input samples, output samples, dropped samples (i.e. the number of unused input
|
||||
* samples) and inserted samples (i.e. the number of samples added to stream).
|
||||
*
|
||||
* When the #GstAudioRate:silent property is set to FALSE, a GObject property
|
||||
* notification will be emitted whenever one of the #GstAudioRate:add or
|
||||
* #GstAudioRate:drop values changes.
|
||||
* This can potentially cause performance degradation.
|
||||
* Note that property notification will happen from the streaming thread, so
|
||||
* applications should be prepared for this.
|
||||
*
|
||||
* <refsect2>
|
||||
* <title>Example pipelines</title>
|
||||
* |[
|
||||
* gst-launch -v alsasrc ! audiorate ! wavenc ! filesink location=alsa.wav
|
||||
* ]| Capture audio from an ALSA device, and turn it into a perfect stream
|
||||
* for saving in a raw audio file.
|
||||
* </refsect2>
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include "config.h"
|
||||
#endif
|
||||
|
|
Loading…
Reference in a new issue