Clocks and synchronization in GStreamer
When playing complex media, each sound and video sample must be played in a specific order at a specific time. For this purpose, GStreamer provides a synchronization mechanism.
GStreamer provides support for the following use cases:
Non-live sources with access faster than playback rate. This is the case where one is reading media from a file and playing it back in a synchronized fashion. In this case, multiple streams need to be synchronized, like audio, video and subtitles.
Capture and synchronized muxing/mixing of media from multiple live sources. This is a typical use case where you record audio and video from a microphone/camera and mux it into a file for storage.
Streaming from (slow) network streams with buffering. This is the typical web streaming case where you access content from a streaming server using HTTP.
Capture from live source and playback with configurable latency. This is used, for example, when capturing from a camera, applying an effect and displaying the result. It is also used when streaming low latency content over a network with UDP.
Simultaneous live capture and playback from prerecorded content. This is used in audio recording cases where you play a previously recorded audio and record new samples, the purpose is to have the new audio perfectly in sync with the previously recorded data.
GStreamer uses a
GstClock object, buffer timestamps and a
event to synchronize streams in a pipeline as we will see in the next
In a typical computer, there are many sources that can be used as a time
source, e.g., the system time, soundcards, CPU performance counters, etc.
For this reason, GStreamer has many
GstClock implementations available.
Note that clock time doesn't have to start from 0 or any other known
value. Some clocks start counting from a particular start date, others
from the last reboot, etc.
GstClock returns the absolute-time according to that clock with
gst_clock_get_time (). The absolute-time (or clock time) of a clock
is monotonically increasing.
A running-time is the difference between a previous snapshot of the absolute-time called the base-time, and any other absolute-time.
running-time = absolute-time - base-time
GstPipeline object maintains a
GstClock object and a
base-time when it goes to the
PLAYING state. The pipeline gives a handle
to the selected
GstClock to each element in the pipeline along with
selected base-time. The pipeline will select a base-time in such a way
that the running-time reflects the total time spent in the
state. As a result, when the pipeline is
PAUSED, the running-time stands
Because all objects in the pipeline have the same clock and base-time, they can thus all calculate the running-time according to the pipeline clock.
To calculate a buffer running-time, we need a buffer timestamp and the
SEGMENT event that preceded the buffer. First we can convert the
SEGMENT event into a
GstSegment object and then we can use the
gst_segment_to_running_time () function to perform the calculation of
the buffer running-time.
Synchronization is now a matter of making sure that a buffer with a certain running-time is played when the clock reaches the same running-time. Usually, this task is performed by sink elements. These elements also have to take into account the configured pipeline's latency and add it to the buffer running-time before synchronizing to the pipeline clock.
Non-live sources timestamp buffers with a running-time starting from 0. After a flushing seek, they will produce buffers again from a running-time of 0.
Live sources need to timestamp buffers with a running-time matching the pipeline running-time when the first byte of the buffer was captured.
The buffer stream-time, also known as the position in the stream, is
a value between 0 and the total duration of the media and it's calculated
from the buffer timestamps and the preceding
The stream-time is used in:
Report the current position in the stream with the
The position used in the seek events and queries.
The position used to synchronize controlled values.
The stream-time is never used to synchronize streams, this is only done with the running-time.
Here is an overview of the various timelines used in GStreamer.
The image below represents the different times in the pipeline when playing a 100ms sample and repeating the part between 50ms and 100ms.
You can see how the running-time of a buffer always increments monotonically along with the clock-time. Buffers are played when their running-time is equal to the clock-time - base-time. The stream-time represents the position in the stream and jumps backwards when repeating.
A clock provider is an element in the pipeline that can provide a
GstClock object. The clock object needs to report an absolute-time
that is monotonically increasing when the element is in the
state. It is allowed to pause the clock while the element is
Clock providers exist because they play back media at some rate, and this rate is not necessarily the same as the system clock rate. For example, a soundcard may play back at 44.1 kHz, but that doesn't mean that after exactly 1 second according to the system clock, the soundcard has played back 44100 samples. This is only true by approximation. In fact, the audio device has an internal clock based on the number of samples played that we can expose.
If an element with an internal clock needs to synchronize, it needs to estimate when a time according to the pipeline clock will take place according to the internal clock. To estimate this, it needs to slave its clock to the pipeline clock.
If the pipeline clock is exactly the internal clock of an element, the element can skip the slaving step and directly use the pipeline clock to schedule playback. This can be both faster and more accurate. Therefore, generally, elements with an internal clock like audio input or output devices will be a clock provider for the pipeline.
When the pipeline goes to the
PLAYING state, it will go over all
elements in the pipeline from sink to source and ask each element if
they can provide a clock. The last element that can provide a clock will
be used as the clock provider in the pipeline. This algorithm prefers a
clock from an audio sink in a typical playback pipeline and a clock from
source elements in a typical capture pipeline.
There exist some bus messages to let you know about the clock and clock
providers in the pipeline. You can see what clock is selected in the
pipeline by looking at the
NEW_CLOCK message on the bus. When a clock
provider is removed from the pipeline, a
CLOCK_LOST message is posted
and the application should go to
PAUSED and back to
PLAYING to select a
The latency is the time it takes for a sample captured at timestamp X to reach the sink. This time is measured against the clock in the pipeline. For pipelines where the only elements that synchronize against the clock are the sinks, the latency is always 0 since no other element is delaying the buffer.
For pipelines with live sources, a latency is introduced, mostly because
of the way a live source works. Consider an audio source, it will start
capturing the first sample at time 0. If the source pushes buffers with
44100 samples at a time at 44100Hz it will have collected the buffer at
second 1. Since the timestamp of the buffer is 0 and the time of the
clock is now
>= 1 second, the sink will drop this buffer because it is
too late. Without any latency compensation in the sink, all buffers will
Before the pipeline goes to the
PLAYING state, it will, in addition to
selecting a clock and calculating a base-time, calculate the latency in
the pipeline. It does this by doing a
LATENCY query on all the sinks in
the pipeline. The pipeline then selects the maximum latency in the
pipeline and configures this with a
All sink elements will delay playback by the value in the
Since all sinks delay with the same amount of time, they will be
relatively in sync.
Adding/removing elements to/from a pipeline or changing element
properties can change the latency in a pipeline. An element can request
a latency change in the pipeline by posting a
LATENCY message on the
bus. The application can then decide to query and redistribute a new
latency or not. Changing the latency in a pipeline might cause visual or
audible glitches and should therefore only be done by the application
when it is allowed.
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