From d8ef396fb023a9328c2ec2e189fd1872a19e4f9d Mon Sep 17 00:00:00 2001 From: Stefan Sauer Date: Thu, 5 Oct 2017 17:52:37 +0200 Subject: [PATCH] audioaggreator: update docs Remove wrote references to collectpads. Document the units. --- gst-libs/gst/audio/gstaudioaggregator.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) diff --git a/gst-libs/gst/audio/gstaudioaggregator.c b/gst-libs/gst/audio/gstaudioaggregator.c index c7f7583348..fab852de53 100644 --- a/gst-libs/gst/audio/gstaudioaggregator.c +++ b/gst-libs/gst/audio/gstaudioaggregator.c @@ -47,18 +47,20 @@ struct _GstAudioAggregatorPadPrivate { /* All members are protected by the pad object lock */ - GstBuffer *buffer; /* current buffer we're mixing, - for comparison with collect.buffer - to see if we need to update our + GstBuffer *buffer; /* current input buffer we're mixing, for + comparison with a new input buffer from + aggregator to see if we need to update our cached values. */ - guint position, size; + + guint position, size; /* position in the input buffer and size of the + input buffer in number of samples */ guint64 output_offset; /* Sample offset in output segment relative to - segment.start that collect.pos refers to in the - current buffer. */ + pad.segment.start that position refers to + in the current buffer. */ - guint64 next_offset; /* Next expected sample offset in the input segment - relative to segment.start */ + guint64 next_offset; /* Next expected sample offset relative to + pad.segment.start */ /* Last time we noticed a discont */ GstClockTime discont_time; @@ -151,13 +153,15 @@ struct _GstAudioAggregatorPrivate GstClockTime discont_wait; /* Protected by srcpad stream clock */ - /* Buffer starting at offset containing block_size frames */ + /* Output buffer starting at offset containing blocksize frames (calculated + * from output_buffer_duration) */ GstBuffer *current_buffer; /* counters to keep track of timestamps */ /* Readable with object lock, writable with both aag lock and object lock */ - gint64 offset; /* Sample offset starting from 0 at segment.start */ + /* Sample offset starting from 0 at aggregator.segment.start */ + gint64 offset; }; #define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex);