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audioencoder: clean up some documentation

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This commit is contained in:
Mark Nauwelaerts 2011-09-27 15:31:20 +02:00
parent 7d577d955c
commit 9a9541ff35
1 changed files with 3 additions and 10 deletions
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13
gst-libs/gst/audio/gstaudioencoder.c
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@ -457,20 +457,13 @@ gst_audio_encoder_finalize (GObject * object)
* @buffer: encoded data
* @samples: number of samples (per channel) represented by encoded data
*
* Collects encoded data and/or pushes encoded data downstream.
* Source pad caps must be set when this is called. Depending on the nature
* of the (framing of) the format, subclass can decide whether to push
* encoded data directly or to collect various "frames" in a single buffer.
* Note that the latter behaviour is recommended whenever the format is allowed,
* as it incurs no additional latency and avoids otherwise generating a
* a multitude of (small) output buffers. If not explicitly pushed,
* any available encoded data is pushed at the end of each processing cycle,
* i.e. which encodes as much data as available input data allows.
* Collects encoded data and pushes encoded data downstream.
* Source pad caps must be set when this is called.
*
* If @samples < 0, then best estimate is all samples provided to encoder
* (subclass) so far. @buf may be NULL, in which case next number of @samples
* are considered discarded, e.g. as a result of discontinuous transmission,
* and a discontinuity is marked (note that @buf == NULL => push == TRUE).
* and a discontinuity is marked.
*
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
*