[gst-devel] Buffer/Latency time settable preperty query

[Wim Taymans]

On Thu, 2007-11-29 at 07:27 -0500, kash wrote:
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>  hi all,
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> What is the significance of buffer time and latency time for audio buffers with respect to audio sinks/sources in 
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> case of live pipeline ,?

Buffer time is the size of the audio buffer in microseconds, typically
the amount of data you allocate for the hardware DMA buffer.
latency-time is the period at which you request the hardware to generate
interrupts. The bigger the buffer, the more you can compensate for
fluctuations in the scheduling of the producer threads, the smaller the
buffer, the lower the latency beween producer and actual playback. when
pausing/playing the latency time determines the reaction speed (we write
full segments of latency-time).

For the way the audiosink base class works, you will always introduce
the latency of the buffer-time for live sources (the playback of the
ringbuffer, or DMA, is started after the buffer is completely filled).
Custom implementations of GstBaseAudioSink can do something else and
start playback after filling less of the buffer.

Wim

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>   how to set this value properly for live pipeline ?
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>   do they depend on sampling rate , channels ?
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> Thanks in advance .
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