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audioaggreator: update docs
Remove wrote references to collectpads. Document the units.
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1 changed files with 14 additions and 10 deletions
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@ -47,18 +47,20 @@ struct _GstAudioAggregatorPadPrivate
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{
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{
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/* All members are protected by the pad object lock */
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/* All members are protected by the pad object lock */
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GstBuffer *buffer; /* current buffer we're mixing,
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GstBuffer *buffer; /* current input buffer we're mixing, for
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for comparison with collect.buffer
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comparison with a new input buffer from
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to see if we need to update our
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aggregator to see if we need to update our
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cached values. */
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cached values. */
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guint position, size;
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guint position, size; /* position in the input buffer and size of the
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input buffer in number of samples */
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guint64 output_offset; /* Sample offset in output segment relative to
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guint64 output_offset; /* Sample offset in output segment relative to
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segment.start that collect.pos refers to in the
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pad.segment.start that position refers to
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current buffer. */
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in the current buffer. */
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guint64 next_offset; /* Next expected sample offset in the input segment
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guint64 next_offset; /* Next expected sample offset relative to
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relative to segment.start */
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pad.segment.start */
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/* Last time we noticed a discont */
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/* Last time we noticed a discont */
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GstClockTime discont_time;
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GstClockTime discont_time;
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@ -151,13 +153,15 @@ struct _GstAudioAggregatorPrivate
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GstClockTime discont_wait;
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GstClockTime discont_wait;
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/* Protected by srcpad stream clock */
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/* Protected by srcpad stream clock */
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/* Buffer starting at offset containing block_size frames */
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/* Output buffer starting at offset containing blocksize frames (calculated
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* from output_buffer_duration) */
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GstBuffer *current_buffer;
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GstBuffer *current_buffer;
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/* counters to keep track of timestamps */
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/* counters to keep track of timestamps */
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/* Readable with object lock, writable with both aag lock and object lock */
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/* Readable with object lock, writable with both aag lock and object lock */
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gint64 offset; /* Sample offset starting from 0 at segment.start */
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/* Sample offset starting from 0 at aggregator.segment.start */
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gint64 offset;
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};
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};
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#define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex);
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#define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex);
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