Quality of service is about measuring and adjusting the real-time performance of a pipeline.
The real-time performance is always measured relative to the pipeline clock and typically happens in the sinks when they synchronize buffers against the clock.
The measurements result in QOS events that aim to adjust the datarate in one or more upstream elements. Two types of adjustments can be made:
short time "emergency" corrections based on latest observation in the sinks.
long term rate corrections based on trends observed in the sinks.
It is also possible for the application to artificially introduce delay between synchronized buffers, this is called throttling. It can be used to reduce the framerate, for example.
Sources of quality problems
High CPU load
Other resource problems such as disk load, memory bottlenecks etc.
application level throttling
The QoS event is generated by an element that synchronizes against the clock. It travels upstream and contains the following fields:
GST_TYPE_QOS_TYPE:The type of the QoS event, we have the following types and the default type is
GST_QOS_TYPE_OVERFLOW: an element is receiving buffers too fast and can't keep up processing them. Upstream should reduce the rate.
GST_QOS_TYPE_UNDERFLOW: an element is receiving buffers too slowly and has to drop them because they are too late. Upstream should increase the processing rate.
GST_QOS_TYPE_THROTTLE: the application is asking to add extra delay between buffers, upstream is allowed to drop buffers
G_TYPE_UINT64: The timestamp on the buffer that generated the QoS event. These timestamps are expressed in total
running_timein the sink so that the value is ever increasing.
G_TYPE_INT64: The difference of that timestamp against the current clock time. Negative values mean the timestamp was on time. Positive values indicate the timestamp was late by that amount. When buffers are received in time and throttling is not enabled, the QoS type field is set to OVERFLOW. When throttling, the jitter contains the throttling delay added by the application and the type is set to THROTTLE.
G_TYPE_DOUBLE: Long term prediction of the ideal rate relative to normal rate to get optimal quality.
The rest of this document deals with how these values can be calculated in a sink and how the values can be used by other elements to adjust their operations.
A QOS message is posted on the bus whenever an element decides to:
drop a buffer because of QoS reasons
change its processing strategy because of QoS reasons (quality)
It should be expected that creating and posting the QoS message is reasonably fast and does not significantly contribute to the QoS problems. Options to disable this feature could also be presented on elements.
This message can be posted by a sink/src that performs synchronisation against the clock (live) or it could be posted by an upstream element that performs QoS because of QOS events received from a downstream element (!live).
GST_MESSAGE_QOS contains at least the following info:
G_TYPE_BOOLEAN: If the QoS message was dropped by a live element such as a sink or a live source. If the live property is FALSE, the QoS message was generated as a response to a QoS event in a non-live element.
running_timeof the buffer that generated the QoS message.
stream_timeof the buffer that generated the QoS message.
G_TYPE_UINT64: The timestamp of the buffer that generated the QoS message.
G_TYPE_UINT64: The duration of the buffer that generated the QoS message.
G_TYPE_INT64: The difference of the running-time against the deadline. Negative values mean the timestamp was on time. Positive values indicate the timestamp was late (and dropped) by that amount. The deadline can be a realtime
running_timeor an estimated
G_TYPE_DOUBLE: Long term prediction of the ideal rate relative to normal rate to get optimal quality.
G_TYPE_INT: An element dependent integer value that specifies the current quality level of the element. The default maximum quality is 1000000.
GST_TYPE_FORMATUnits of the processed and dropped fields. Video sinks and video filters will use
GST_FORMAT_BUFFERS(frames). Audio sinks and audio filters will likely use
G_TYPE_UINT64: Total number of units correctly processed since the last state change to READY or a flushing operation.
G_TYPE_UINT64: Total number of units dropped since the last state change to READY or a flushing operation.
The running-time and processed fields can be used to estimate the average processing rate (framerate for video).
Elements might add additional fields in the message which are documented in the relevant elements or baseclasses.
A buffer with timestamp B1 arrives in the sink at time T1. The buffer timestamp is then synchronized against the clock which yields a jitter J1 return value from the clock. The jitter J1 is simply calculated as
J1 = CT - B1
Where CT is the clock time when the entry arrives in the sink. This
value is calculated inside the clock when we perform
If the jitter is negative, the entry arrived in time and can be rendered after waiting for the clock to reach time B1 (which is also CT - J1).
If the jitter is positive however, the entry arrived too late in the sink and should therefore be dropped. J1 is the amount of time the entry was late.
Any buffer that arrives in the sink should generate a QoS event upstream.
Using the jitter we can calculate the time when the buffer arrived in the sink:
T1 = B1 + J1. (1)
The time the buffer leaves the sink after synchronisation is measured as:
T2 = B1 + (J1 < 0 ? 0 : J1) (2)
For buffers that arrive in time (J1 < 0) the buffer leaves after
synchronisation which is exactly B1. Late buffers (J1 >= 0) leave the
sink when they arrive, whithout any synchronisation, which is
T2 = T1 = B1 + J1.
Using a previous T0 and a new T1, we can calculate the time it took for upstream to generate a buffer with timestamp B1.
PT1 = T1 - T0 (3)
We call PT1 the processing time needed to generate buffer with timestamp B1.
Moreover, given the duration of the buffer D1, the current data rate (DR1) of the upstream element is given as:
PT1 T1 - T0 DR1 = --- = ------- (4) D1 D1
For values 0.0 < DR1 ⇐ 1.0 the upstream element is producing faster than real-time. If DR1 is exactly 1.0, the element is running at a perfect speed.
Values DR1 > 1.0 mean that the upstream element cannot produce buffers of duration D1 in real-time. It is exactly DR1 that tells the amount of speedup we require from upstream to regain real-time performance.
An element that is not receiving enough data is said to be underflowed.
In addition to the measurements of the datarate of the upstream element, a typical element must also measure its own performance. Global pipeline performance problems can indeed also be caused by the element itself when it receives too much data it cannot process in time. The element is then said to be overflowed.
Short term correction
The timestamp and jitter serve as short term correction information for upstream elements. Indeed, given arrival time T1 as given in (1) we can be certain that buffers with a timestamp B2 < T1 will be too late in the sink.
In case of a positive jitter we can therefore send a QoS event with a timestamp B1, jitter J1 and proportion given by (4).
This allows an upstream element to not generate any data with timestamps B2 < T1, where the element can derive T1 as B1 + J1.
This will effectively result in frame drops.
The element can even do a better estimation of the next valid timestamp it should output.
Indeed, given the element generated a buffer with timestamp B0 that arrived in time in the sink but then received a QoS event stating B1 arrived J1 too late. This means generating B1 took (B1 + J1) - B0 = T1 - T0 = PT1, as given in (3). Given the buffer B1 had a duration D1 and assuming that generating a new buffer B2 will take the same amount of processing time, a better estimation for B2 would then be:
B2 = T1 + D2 * DR1
B2 = (B1 + J1) + D2 * (B1 + J1 - B0) -------------- D1
assuming the durations of the frames are equal and thus D1 = D2:
B2 = (B1 + J1) + (B1 + J1 - B0) B2 = 2 * (B1 + J1) - B0
B0 = B1 - D1
B2 = 2 * (B1 + J1) - (B1 - D1)
Which yields a more accurate prediction for the next buffer given as:
B2 = B1 + 2 * J1 + D1 (5)
Long term correction
The datarate used to calculate (5) for the short term prediction is based on a single observation. A more accurate datarate can be obtained by creating a running average over multiple datarate observations.
This average is less susceptible to sudden changes that would only influence the datarate for a very short period.
A running average is calculated over the observations given in (4) and is used as the proportion member in the QoS event that is sent upstream.
Receivers of the QoS event should permanently reduce their datarate as given by the proportion member. Failure to do so will certainly lead to more dropped frames and a generally worse QoS.
In throttle mode, the time distance between buffers is kept to a configurable throttle interval. This means that effectively the buffer rate is limited to 1 buffer per throttle interval. This can be used to limit the framerate, for example.
When an element is configured in throttling mode (this is usually only implemented on sinks) it should produce QoS events upstream with the jitter field set to the throttle interval. This should instruct upstream elements to skip or drop the remaining buffers in the configured throttle interval.
The proportion field is set to the desired slowdown needed to get the desired throttle interval. Implementations can use the QoS Throttle type, the proportion and the jitter member to tune their implementations.
Several strategies exist to reduce processing delay that might affect real time performance.
dropping frames (reduce CPU/bandwidth usage)
switch to a lower decoding/encoding quality (reduce algorithmic complexity)
switch to a lower quality source (reduce network usage)
increasing thread priorities
switch to real-time scheduling
assign more CPU cycles to critial pipeline parts
assign more CPU(s) to critical pipeline parts
Here follows a small overview of how QoS can be implemented in a range of different types of elements.
The primary implementor of QoS is GstBaseSink. It will calculate the following values:
upstream running average of processing time (5) in stream time.
running average of buffer durations.
running average of render time (in system time)
The processing time and the average buffer durations will be used to calculate a proportion.
The processing time in system time is compared to render time to decide if the majority of the time is spend upstream or in the sink itself. This value is used to decide overflow or underflow.
The number of rendered and dropped buffers is used to query stats on the sink.
A QoS event with the most current values is sent upstream for each buffer that was received by the sink.
Normally QoS is only enabled for video pipelines. The reason being that drops in audio are more disturbing than dropping video frames. Also video requires in general more processing than audio.
Normally there is a threshold for when buffers get dropped in a video sink. Frames that arrive 20 milliseconds late are still rendered as it is not noticeable for the human eye.
A QoS message is posted whenever a (part of a) buffer is dropped.
In throttle mode, the sink sends QoS event upstream with the timestamp
set to the
running_time of the latest buffer and the jitter set to the
throttle interval. If the throttled buffer is late, the lateness is
subtracted from the throttle interval in order to keep the desired
Transform elements can entirely skip the transform based on the timestamp and jitter values of recent QoS event since these buffers will certainly arrive too late.
With any intermediate element, the element should measure its performance to decide if it is responsible for the quality problems or any upstream/downstream element.
some transforms can reduce the complexity of their algorithms. Depending on the algorithm, the changes in quality may have disturbing visual or audible effect that should be avoided.
A QoS message should be posted when a frame is dropped or when the quality of the filter is reduced. The quality member in the QOS message should reflect the quality setting of the filter.
A video decoder can, based on the codec in use, decide to not decode intermediate frames. A typical codec can for example skip the decoding of B-frames to reduce the CPU usage and framerate.
If each frame is independantly decodable, any arbitrary frame can be skipped based on the timestamp and jitter values of the latest QoS event. In addition can the proportion member be used to permanently skip frames.
It is suggested to adjust the quality field of the QoS message with the
expected amount of dropped frames (skipping B and/or P frames). This
depends on the particular spacing of B and P frames in the stream. If
the quality control would result in half of the frames to be dropped
(typical B frame skipping), the quality field would be set to
1000000 * 1/2 = 500000. If a typical I frame spacing of 18 frames is used,
skipping B and P frames would result in 17 dropped frames or 1 decoded
frame every 18 frames. The quality member should be set to
1000000 * 1/18 = 55555.
skipping B frames: quality = 500000
skipping P/B frames: quality = 55555 (for I-frame spacing of 18 frames)
Demuxers usually cannot do a lot regarding QoS except for skipping frames to the next keyframe when a lateness QoS event arrives on a source pad.
A demuxer can however measure if the performance problems are upstream or downstream and forward an updated QoS event upstream.
Most demuxers that have multiple output pads might need to combine the QoS events on all the pads and derive an aggregated QoS event for the upstream element.
The QoS events only apply to push based sources since pull based sources are entirely controlled by another downstream element.
Sources can receive a overflow or underflow event that can be used to switch to less demanding source material. In case of a network stream, a switch could be done to a lower or higher quality stream or additional enhancement layers could be used or ignored.
Live sources will automatically drop data when it takes too long to process the data that the element pushes out.
Live sources should post a QoS message when data is dropped.
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