audiomixer
The audiomixer allows to mix several streams into one by adding the data. Mixed data is clamped to the min/max values of the data format.
Unlike the adder element audiomixer properly synchronises all input streams and also handles live inputs such as capture sources or RTP properly.
The audiomixer element can accept any sort of raw audio data, it will be converted to the target format if necessary, with the exception of the sample rate, which has to be identical to either what downstream expects, or the sample rate of the first configured pad. Use a capsfilter after the audiomixer element if you want to precisely control the format that comes out of the audiomixer, which supports changing the format of its output while playing.
If you want to control the manner in which incoming data gets converted, see the converter-config property, which will let you for example change the way in which channels may get remapped.
The input pads are from a GstPad subclass and have additional properties to mute each pad individually and set the volume:
- "mute": Whether to mute the pad or not (#gboolean)
- "volume": The volume of the pad, between 0.0 and 10.0 (#gdouble)
Example launch line
gst-launch-1.0 audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
This pipeline produces two sine waves mixed together.
Hierarchy
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstElement ╰──GstAggregator ╰──GstAudioAggregator ╰──GstAudioMixer
Factory details
Authors: – Sebastian Dröge
Classification: – Generic/Audio
Rank – none
Plugin – gstaudiomixer
Package – GStreamer Base Plug-ins
Pad Templates
sink_%u
audio/x-raw:
format: { F64LE, F64BE, F32LE, F32BE, S32LE, S32BE, U32LE, U32BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, S16LE, S16BE, U16LE, U16BE, S8, U8 }
rate: [ 1, 2147483647 ]
channels: [ 1, 2147483647 ]
layout: interleaved
src
audio/x-raw:
format: { S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }
rate: [ 1, 2147483647 ]
channels: [ 1, 2147483647 ]
layout: interleaved
Properties
alignment-threshold
“alignment-threshold” guint64
Timestamp alignment threshold in nanoseconds
Flags : Read / Write
Default value : 40000000
discont-wait
“discont-wait” guint64
Window of time in nanoseconds to wait before creating a discontinuity
Flags : Read / Write
Default value : 1000000000
latency
“latency” guint64
Additional latency in live mode to allow upstream to take longer to produce buffers for the current position (in nanoseconds)
Flags : Read / Write
Default value : 0
min-upstream-latency
“min-upstream-latency” guint64
When sources with a higher latency are expected to be plugged in dynamically after the aggregator has started playing, this allows overriding the minimum latency reported by the initial source(s). This is only taken into account when larger than the actually reported minimum latency. (nanoseconds)
Flags : Read / Write
Default value : 0
output-buffer-duration
“output-buffer-duration” guint64
Output block size in nanoseconds
Flags : Read / Write
Default value : 10000000
output-buffer-duration-fraction
“output-buffer-duration-fraction” GstFraction *
Output block size in nanoseconds, expressed as a fraction
Flags : Read / Write
Default value : 1/100
start-time
“start-time” guint64
Start time to use if start-time-selection=set
Flags : Read / Write
Default value : 18446744073709551615
start-time-selection
“start-time-selection” GstAggregatorStartTimeSelection *
Decides which start time is output
Flags : Read / Write
Default value : zero (0)
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