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. Time in &GStreamer; is defined as the value returned from a particular GstClock object from the method gst_clock_get_time (). 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, ... For this reason, there are many GstClock implementations available in &GStreamer;. The clock time doesn't always start from 0 or from some known value. Some clocks start counting from some known start date, other clocks start counting since last reboot, etc... As clocks return an absolute measure of time, they are not usually used directly. Instead, differences between two clock times are used to measure elapsed time according to a clock. A clock returns the absolute-time according to that clock with gst_clock_get_time (). From the absolute-time is a running-time calculated, which is simply the difference between a previous snapshot of the absolute-time called the base-time. So: running-time = absolute-time - base-time A &GStreamer; 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 PLAYING state. As a result, when the pipeline is PAUSED, the running-time stands still. 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 preceeded 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 done by sink elements. Sink also have to take into account the latency configured in the pipeline and add this to the buffer running-time before synchronizing to the pipeline clock. Let us clarify the contract between GStreamer and each element in the pipeline. Non-live source elements must place a timestamp in each buffer that they deliver when this is possible. They must choose the timestamps and the values of the SEGMENT event in such a way that the running-time of the buffer starts from 0. Some sources, such as filesrc, is not able to generate timestamps on all buffers. It can and must however create a timestamp on the first buffer (with a running-time of 0). The source then pushes out the SEGMENT event followed by the timestamped buffers. Live source elements must place a timestamp in each buffer that they deliver. They must choose the timestamps and the values of the SEGMENT event in such a way that the running-time of the buffer matches exactly the running-time of the pipeline clock when the first byte in the buffer was captured. Parser/Decoder elements must use the incoming timestamps and transfer those to the resulting output buffers. They are allowed to interpolate or reconstruct timestamps on missing input buffers when they can. Demuxer elements can usually set the timestamps stored inside the media file onto the outgoing buffers. They need to make sure that outgoing buffers that are to be played at the same time have the same running-time. Demuxers also need to take into account the incoming timestamps on buffers and use that to calculate an offset on the outgoing buffer timestamps. Muxer elements should use the incoming buffer running-time to mux the different streams together. They should copy the incoming running-time to the outgoing buffers. If the element is intended to emit samples at a specific time (real time playing), the element should require a clock, and thus implement the method set_clock. The sink should then make sure that the sample with running-time is played exactly when the pipeline clock reaches that running-time + latency. Some elements might use the clock API such as gst_clock_id_wait() to perform this action. Other sinks might need to use other means of scheduling timely playback of the data.