# pulsesrc Source: [https://docs.qualcomm.com/doc/80-70022-50/topic/pulsesrc.html](https://docs.qualcomm.com/doc/80-70022-50/topic/pulsesrc.html) The pulsesrc plugin interacts with the underlying PulseAudio sound server to capture the pulse-coded modulation (PCM) samples. The PulseAudio server interacts with the underlying hardware to provide the capture capability and is based on the [GstAudioSrc](https://gstreamer.freedesktop.org/documentation/audio/gstaudiosrc.html#GstAudioSrc) base class. ## Inheritance chain [GObject](https://docs.gtk.org/gobject/) → [GstObject](https://gstreamer.freedesktop.org/documentation/gstreamer/gstobject.html?gi-language=c) → [GstElement](https://gstreamer.freedesktop.org/documentation/gstreamer/gstelement.html?gi-language=c) → [GstBaseSrc](https://gstreamer.freedesktop.org/documentation/base/gstbasesrc.html#GstBaseSrc) → [GstPushSrc](https://gstreamer.freedesktop.org/documentation/base/gstpushsrc.html#GstPushSrc) → [GstAudioBaseSrc](https://gstreamer.freedesktop.org/documentation/audio/gstaudiobasesrc.html#GstAudioBaseSrc) → [GstAudioSrc](https://gstreamer.freedesktop.org/documentation/audio/gstaudiosrc.html#GstAudioSrc) → GstPulseSrc The following tables provide information on pad templates and element properties of pulsesrc. For use cases, see [Audio (only) capture](https://docs.qualcomm.com/doc/80-70022-50/topic/audio-use-cases.html#audio-only-capture). ## Pad configuration | Pad Name | Capabilities | Capabilities | Capabilities | | --- | --- | --- | --- | | SRC template: 'src'

Availability: Always

Direction: src

| audio/x-raw | format: | { (string)S16LE, (string)S16BE, (string)F32LE, (string)F32BE,
(string)S32LE, (string)S32BE, (string)S24LE, (string)S24BE,
(string)S24\_32LE, (string)S24\_32BE, (string)U8 } | | SRC template: 'src'

Availability: Always

Direction: src

Availability: Always

Direction: src

Availability: Always

Direction: src

Availability: Always

Direction: src

Availability: Always

Direction: src

Availability: Always

Direction: src

Availability: Always

Direction: src

| audio/x-mulaw | channels: | [ 1, 32 ] | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ## Element configuration Table : Element properties of pulsesrc | Property | Description | | --- | --- | | name | The name of the object.

flags: readable, writable

String. Default: "pulsesrc0"

| | parent | The parent of the object.

flags: readable, writable

Object of type "GstObject"

| | blocksize | Size in bytes to read per buffer (-1 = default).

flags: readable, writable

Unsigned Integer. Range: 0 - 4294967295 Default: 0

| | num-buffers | Number of buffers to output before sending EOS (-1 =
unlimited).

flags: readable, writable

Integer. Range: ˗1 to 2147483647, default: ˗1

| | typefind | Run type find before negotiating (deprecated, non-functional).

flags: readable, writable, deprecated

Boolean. Default: false

| | do-timestamp | Apply current stream time to buffers.

flags: readable, writable

Boolean. Default: true

| | buffer-time | Size of audio buffer in microseconds. This size is the maximum
amount of data that's buffered in the device and the maximum latency
that the source reports. The pulsesrc plugin can ignore this
value.

flags: readable, writable

Integer64. Range: 1 - 9223372036854775807 Default:
200000

| | latency-time | The minimum amount of data to read in each iteration is in
microseconds. This is the minimum latency that the source reports.
The plugin can ignore this value.

flags: readable, writable

Integer64. Range: 1 - 9223372036854775807 Default:
10000

| | actual-buffer-time | Actual configured size of audio buffer in microseconds.

flags: readable

Integer64. Range: ˗1 to 9223372036854775807, default:
˗1

| | actual-latency-time | Actual configured audio latency in microseconds.

flags: readable

Integer64. Range: ˗1 to 9223372036854775807, default:
˗1

| | provide-clock | Provide a clock to be used as the global pipeline clock.

flags: readable, writable

Boolean. Default: true

| | slave-method | Algorithm used to match the rate of the primary clock.

flags: readable, writable

Enum "GstAudioBaseSrcSlaveMethod" Default: 2, "skew"
- (0): resample -
  GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE
- (1): re-timestamp -
  GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP
- (2): skew - GST_AUDIO_BASE_SRC_SLAVE_SKEW
- (3): none - GST_AUDIO_BASE_SRC_SLAVE_NONE

| | server | The PulseAudio server for the connection.

flags: readable, writable

String. Default: null

| | device | The PulseAudio source device for connection.

flags: readable, writable

String. Default: null

| | current-device | The current PulseAudio source device.

flags: readable

String. Default: ""

| | device-name | Human-readable name of the sound device.

flags: readable

String. Default: null

| | volume | Linear volume of this stream, 1.0=100%.

flags: readable, writable

Double. Range: 0 - 10 Default: 1

| | mute | Mute state of this stream.

flags: readable, writable

Boolean. Default: false

| | client-name | The PulseAudio client\_name\_to\_use.

flags: readable, writable, changeable only in NULL or READY
state

String. Default: "gst-inspect-1.0"

| | stream-properties | A list of PulseAudio stream properties.

flags: readable, writable

Boxed pointer of type "GstStructure"

| | source-output-index | The index of the PulseAudio source output corresponding to this
record stream.

flags: readable

Unsigned Integer. Range: 0 - 4294967295 Default:
4294967295

| | stream-flags | Stream flags (not all default flags can be overridden).

flags: writable

Flags "GstPulseStreamFlags" Default: 0x0000000f, "Timing
update requests are issued periodically and automatically.
Don't force the time to increase monotonically and
interpolate the latency for this stream.
- ((0x00000000): Flag to pass when no specific options
  are needed - GST_PULSE_STREAM_NOFLAGS
- (0x00000001): Create the corked stream-
  GST_PULSE_STREAM_START_CORKED
- (0x00000002): Interpolate the latency for this
  stream - GST_PULSE_STREAM_INTERPOLATE_TIMING
- (0x00000004): Don't force the time to increase
  monotonically - GST_PULSE_STREAM_NOT_MONOTONIC
- (0x00000008): Timing update requests are issued
  periodically and automatically -
  GST_PULSE_STREAM_AUTO_TIMING_UPDATE
- (0x00000010): Don't remap channels by their name,
  instead map them simply by their index -
  GST_PULSE_STREAM_NO_REMAP_CHANNELS
- (0x00000020): When remapping channels by name, don't
  upmix or downmix them to related channels -
  GST_PULSE_STREAM_NO_REMIX_CHANNELS
- (0x00000040): Use the sample format of the
  sink/device this stream is being connected to -
  GST_PULSE_STREAM_FIX_FORMAT
- (0x00000080): Use the sample rate of the sink -
  GST_PULSE_STREAM_FIX_RATE
- (0x00000100): Use the number of channels and the
  channel map of the sink -
  GST_PULSE_STREAM_FIX_CHANNELS
- (0x00000200): Don't allow moving of this stream to
  another sink/device -
  GST_PULSE_STREAM_DONT_MOVE
- (0x00000400): Allow dynamic changing of the sampling
  rate during playback -
  GST_PULSE_STREAM_VARIABLE_RATE
- (0x00000800): Find peaks instead of resampling -
  GST_PULSE_STREAM_PEAK_DETECT
- (0x00001000): Create in muted state -
  GST_PULSE_STREAM_START_MUTED
- (0x00002000): Try to adjust the latency of the
  sink/source based on the requested buffer metrics
  and adjust buffer metrics accordingly -
  GST_PULSE_STREAM_ADJUST_LATENCY
- (0x00004000): Enable compatibility mode for legacy
  clients that rely on a classic hardware device
  fragment-style playback model -
  GST_PULSE_STREAM_EARLY_REQUESTS
- (0x00008000): If set, this stream isn't considered
  when it's checked whether the stream is connected.
  It's auto-suspended.
  GST_PULSE_STREAM_DONT_INHIBIT_AUTO_SUSPEND
- (0x00010000): Create in unmuted state -
  GST_PULSE_STREAM_START_UNMUTED
- (0x00020000): If the sink/source of this stream is
  connected, it's suspended during the creation of
  this stream, causing it to fail -
  GST_PULSE_STREAM_FAIL_ON_SUSPEND
- (0x00040000): If a volume is passed when this stream
  is created, creation of this stream causes it to
  fail - GST_PULSE_STREAM_RELATIVE_VOLUME
- ((0x00080000): Used to tag content that's rendered
  by pass-through sinks - GST_PULSE_STREAM_PASSTHROUGH
  write-only

| **Parent Topic:** [Configure audio plugins](https://docs.qualcomm.com/doc/80-70022-50/topic/audio-plugins.html) Last Published: Feb 20, 2026 [Previous Topic Configure audio plugins](https://docs.qualcomm.com/bundle/publicresource/80-70022-50/topics/audio-plugins.md) [Next Topic pulsesink](https://docs.qualcomm.com/bundle/publicresource/80-70022-50/topics/pulsesink.md)