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. There are two kinds of time in GStreamer. Clock time is an absolute time. By contrast, element time is the relative time, usually to the start of the current media stream. The element time represents the time that should have a media sample that is being processed by the element at this time. The element time is calculated by adding an offset to the clock time. GStreamer can use different clocks. Though the system time can be used as a clock, soundcards and other devices provides a better time source. For this reason some elements provide a clock. The method get_clock is implemented in elements that provide one. As clocks return an absolute measure of time, they are not usually used directly. Instead, a reference to a clock is stored in any element that needs it, and it is used internally by GStreamer to calculate the element time. Now we will see how time information travels the pipeline in different states. The pipeline starts playing. The source element typically knows the time of each sample. Sometimes it is a parser element the one that knows the time, for instance if a pipeline contains a filesrc element connected to a MPEG decoder element, the former is the one that knows the time of each sample, because the knowledge of when to play each sample is embedded in the MPEG format. In this case this element will be regarded as the source element for this discussion. First, the source element sends a discontinuous event. This event carries information about the current relative time of the next sample. This relative time is arbitrary, but it must be consistent with the timestamp that will be placed in buffers. It is expected to be the relative time to the start of the media stream, or whatever makes sense in the case of each media. When receiving it, the other elements adjust their offset of the element time so that this time matches the time written in the event. Then the source element sends media samples in buffers. This element places a timestamp in each buffer saying when the sample should be played. When the buffer reaches the sink pad of the last element, this element compares the current element time with the timestamp of the buffer. If the timestamp is higher or equal it plays the buffer, otherwise it waits until the time to place the buffer arrives with gst_element_wait(). If the stream is seeked, the next samples sent will have a timestamp that is not adjusted with the element time. Therefore, the source element must send a discontinuous event. Let us clarify the contract between GStreamer and each element in the pipeline. Source elements (or parsers of formats that provide notion of time, such as MPEG, as explained above) must place a timestamp in each buffer that they deliver. The origin of the time used is arbitrary, but it must match the time delivered in the discontinuous event (see below). However, it is expected that the origin is the origin of the media stream. In order to initialize the element time of the rest of the pipeline, a source element must send a discontinuous event before starting to play. In addition, after seeking, a discontinuous event must be sent, because the timestamp of the next element does not match the element time of the rest of the pipeline. 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. In addition, before playing each sample, if the current element time is less than the timestamp in the sample, it wait until the current time arrives should call gst_element_wait() With some schedulers, gst_element_wait() blocks the pipeline. For instance, if there is one audio sink element and one video sink element, while the audio element is waiting for a sample the video element cannot play other sample. This behaviour is under discussion, and might change in a future release.