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gstreamer/docs/design/part-overview.txt
Wim Taymans b9dbb55105 docs/design/part-overview.txt: Make upsteam/downstream concepts more clear. 
Original commit message from CVS:
* docs/design/part-overview.txt:
Make upsteam/downstream concepts more clear.
Give an example of serialized/non-serialized events.
* docs/design/part-events.txt:
* docs/design/part-streams.txt:
Mention applied_rate.
* docs/design/part-trickmodes.txt:
Mention applied rate, flesh out some more use cases.
* gst/gstevent.c: (gst_event_new_new_segment),
(gst_event_parse_new_segment), (gst_event_new_new_segment_full),
(gst_event_parse_new_segment_full), (gst_event_new_tag),
(gst_event_parse_tag), (gst_event_new_buffer_size),
(gst_event_parse_buffer_size), (gst_event_new_qos),
(gst_event_parse_qos), (gst_event_parse_seek),
(gst_event_new_navigation):
* gst/gstevent.h:
Add applied_rate field to NEWSEGMENT event.
API: gst_event_new_new_segment_full()
API: gst_event_parse_new_segment_full()
* gst/gstsegment.c: (gst_segment_init), (gst_segment_set_seek),
(gst_segment_set_newsegment), (gst_segment_set_newsegment_full),
(gst_segment_to_stream_time), (gst_segment_to_running_time):
* gst/gstsegment.h:
Add applied_rate to GstSegment structure.
Make calculation of stream_time and running_time more correct
wrt rate/applied_rate.
Add some more docs.
API: GstSegment::applied_rate field
API: gst_segment_set_newsegment_full();
* libs/gst/base/gstbasesink.c: (gst_base_sink_configure_segment),
(gst_base_sink_get_sync_times), (gst_base_sink_get_position):
* libs/gst/base/gstbasetransform.c:
(gst_base_transform_sink_eventfunc),
(gst_base_transform_handle_buffer):
Parse and use applied_rate in the GstSegment field.
* tests/check/gst/gstevent.c: (GST_START_TEST):
Add check for applied_rate field.
* tests/check/gst/gstsegment.c: (GST_START_TEST),
(gstsegments_suite):
Add more checks for various GstSegment operations.
2006-05-08 09:52:33 +00:00

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Overview
--------
This part gives an overview of the design of GStreamer with references to
the more detailed explanations of the different topics.
This document is intented for people that want to have a global overview of
the inner workings of GStreamer.
Introduction
------------
GStreamer is a set of libraries and plugins that can be used to implement various
multimedia applications ranging from desktop players, audio/video recorders,
multimedia servers, transcoders, etc.
Applications are built by constructing a pipeline composed of elements. An element
is an object that performs some action on a multimedia stream such as:
- read a file
- decode or encode between formats
- capture from a hardware device
- render to a hardware device
- mix or multiplex multiple streams
Elements have input and output pads called sink and source pads in GStreamer. An
application links elements together on pads to construct a pipeline. Below is
an example of an ogg/vorbis playback pipeline.
+-----------------------------------------------------------+
| ----------> downstream -------------------> |
| |
| pipeline |
| +---------+ +----------+ +-----------+ +----------+ |
| | filesrc | | oggdemux | | vorbisdec | | alsasink | |
| | src-sink src-sink src-sink | |
| +---------+ +----------+ +-----------+ +----------+ |
| |
| <---------< upstream <-------------------< |
+-----------------------------------------------------------+
The filesrc element reads data from a file on disk. The oggdemux element parses
the data and sends the compressed audio data to the vorbisdec element. The
vorbisdec element decodes the compressed data and sends it to the alsasink
element. The alsasink element sends the samples to the audio card for playback.
Downstream and upstream are the terms used to describe the direction in the
Pipeline. From source to sink is called "downstream" and "upstream" is
from sink to source.
The task of the application is to construct a pipeline as above using existing
elements. This is further explained in the pipeline building topic.
The application does not have to manage any of the complexities of the
actual dataflow/decoding/conversions/synchronsiation etc. but only calls high
level functions on the pipeline object such as PLAY/PAUSE/STOP.
The application also receives messages and notifications from the pipeline such
as metadata, warning, error and EOS messages.
If the application needs more control over the graph it is possible to directly
access the elements and pads in the pipeline.
Design overview
---------------
GStreamer design goals include:
- Process large amounts of data quickly
- Allow fully multithreaded processing
- Ability to deal with multiple formats
- Synchronize different dataflows
- Ability to deal with multiple devices
The capabilities presented to the application depends on the number of elements
installed on the system and their functionality.
The GStreamer core is designed to be media agnostic but provides many features
to elements to describe media formats.
Elements
--------
The smallest building blocks in a pipeline are elements. An element provides a
number of pads which can be source or sinkpads. Sourcepads provide data and
sinkpads consume data. Below is an example of an ogg demuxer element that has
one pad that takes (sinks) data and two source pads that produce data.
+-----------+
| oggdemux |
| src0
sink src1
+-----------+
An element can be in four different states: NULL, READY, PAUSED, PLAYING. In the
NULL and READY state, the element is not processing any data. In the PLAYING state
it is processing data. The intermediate PAUSED state is used to preroll data in
the pipeline. A state change can be performed with gst_element_set_state().
An element always goes through all the intermediate state changes. This means that
when en element is in the READY state and is put to PLAYING, it will first go
through the intermediate PAUSED state.
An element state change to PAUSED will activate the pads of the element. First the
source pads are activated, then the sinkpads. When the pads are activated, the
pad activate function is called. Some pads will start a thread (GstTask) or some
other mechanism to start producing or consuming data.
The PAUSED state is special as it is used to preroll data in the pipeline. The purpose
is to fill all connected elements in the pipeline with data so that the subsequent
PLAYING state change happens very quickly. Some elements will therefore not complete
the state change to PAUSED before they have received enough data. Sink elements are
required to only complete the state change to PAUSED after receiving the first data.
Normally the state changes of elements are coordinated by the pipeline as explained
in [part-states.txt].
Different categories of elements exist:
- source elements, these are elements that do not consume data but only provide data
for the pipeline.
- sink elements, these are elements that do not produce data but renders data to
an output device.
- transform elements, these elements transform an input stream in a certain format
into a stream of another format. Encoder/decoder/converters are examples.
- demuxer elements, these elements parse a stream and produce several output streams.
- mixer/muxer elements, combine several input streams into one output stream.
Other categories of elements can be constructed (see part-klass.txt).
Bins
----
A bin is an element subclass and acts as a container for other elements so that multiple
elements can be combined into one element.
A bin coordinates its children's state changes as explained later. It also distributes
events and various other functionality to elements.
A bin can have its own source and sinkpads by ghostpadding one or more of its children's
pads to itself.
Below is a picture of a bin with two elements. The sinkpad of one element is ghostpadded
to the bin.
+---------------------------+
| bin |
| +--------+ +-------+ |
| | | | | |
| /sink src-sink | |
sink +--------+ +-------+ |
+---------------------------+
Pipeline
--------
A pipeline is a special bin subclass that provides the following features to its
children:
- Select and manage a clock.
- Manage stream time based on the selected clock.
- Provide means for elements to comunicate with the application by the GstBus.
- Manage the global state of the elements such as Errors and end-of-stream.
Normally the application creates one pipeline that will manage all the elements
in the application.
Dataflow and buffers
--------------------
GStreamer supports two possible types of dataflow, the push and pull model. In the
push model, an upstream element sends data to a downstream element by calling a
method on a sinkpad. In the pull model, a downstream element requests data from
an upstream element by calling a method on a source pad.
The most common dataflow is the push model. The pull model can be used in specific
circumstances by demuxer elements. The pull model can also be used by low latency
audio applications.
The data passed between pads is encapsulated in Buffers. The buffer contains a
pointer to the actual data and also metadata describing the data. This metadata
includes:
- timestamp of the data, this is the time instance at which the data was captured
or the time at which the data should be played back.
- offset of the data: a media specific offset, this could be samples for audio or
frames for video.
- the duration of the data in time.
- the media type of the data described with caps, these are key/value pairs that
describe the media type in a unique way.
- additional flags describing special properties of the data such as
discontinuities or delta units.
When an element whishes to send a buffer to another element is does this using one
of the pads that is linked to a pad of the other element. In the push model, a
buffer is pushed to the peer pad with gst_pad_push(). In the pull model, a buffer
is pulled from the peer with the gst_pad_pull_range() function.
Before an element pushes out a buffer, it should make sure that the peer element
can understand the buffer contents. It does this by querying the peer element
for the supported formats and by selecting a suitable common format. The selected
format is then attached to the buffer with gst_buffer_set_caps() before pushing
out the buffer.
When an element pad receives a buffer, if has to check if it understands the media
type of the buffer before starting processing it. The GStreamer core does this
automatically and will call the gst_pad_set_caps() function of the element before
sending the buffer to the element.
Both gst_pad_push() and gst_pad_pull_range() have a return value indicating wether
the operation succeeded. An error code means that no more data should be send
to that pad. A source element that initiates the data flow in a thread typically
pauses the producing thread when this happens.
A buffer can be created with gst_buffer_new() or by requesting a usable buffer
from the peer pad using gst_pad_alloc_buffer(). Using the second method, it is
possible for the peer element to suggest the element to produce data in another
format by attaching another media type caps to the buffer.
The process of selecting a media type and attaching it to the buffers is called
caps negotiation.
Caps
----
A media type (Caps) is described using a generic list of key/value pairs. The key is
a string and the value can be a single/list/range of int/float/string.
Caps that have no ranges/list or other variable parts are said to be fixed and
can be used to put on a buffer.
Caps with variables in them are used to describe possible media types that can be
handled by a pad.
Dataflow and events
-------------------
Parallel to the dataflow is a flow of events. Unlike the buffers, events can pass
upstream and downstream. Some events only travel upstream others only downstream.
the events are used to denote special conditions in the dataflow such as EOS or
to inform plugins of special events such as flushing or seeking.
Some events must be serialized with the buffer flow, others don't. Serialized
events are inserted between the buffers. Non serialized events jump in front
of any buffers current being processed.
An example of a serialized event is a TAG event that is inserted between buffers
to mark metadata for those buffers.
An example of a non serialized event is the FLUSH event.
Pipeline construction
---------------------
The application starts by creating a Pipeline element using gst_pipeline_new ().
Elements are added to and removed from the pipeline with gst_bin_add() and
gst_bin_remove().
After adding the elements, the pads of an element can be retrieved with
gst_element_get_pad(). Pads can then be linked together with gst_pad_link().
Some elements create new pads when actual dataflow is happening in the pipeline.
With g_signal_connect() one can receive a notification when an element has created
a pad. These new pads can then be linked to other unlinked pads.
Some elements cannot be linked together because they operate on different
incompatible data types. The possible datatypes a pad can provide or consume can
be retrieved with gst_pad_get_caps().
Below is a simple mp3 playback pipeline that we constructed. We will use this
pipeline in further examples.
+-------------------------------------------+
| pipeline |
| +---------+ +----------+ +----------+ |
| | filesrc | | mp3dec | | alsasink | |
| | src-sink src-sink | |
| +---------+ +----------+ +----------+ |
+-------------------------------------------+
Pipeline clock
--------------
One of the important functions of the pipeline is to select a global clock
for all the elements in the pipeline.
The purpose of the clock is to provide a stricly increasing value at the rate
of one GST_SECOND per second. Clock values are expressed in nanoseconds.
Elements use the clock time to synchronize the playback of data.
Before the pipeline is set to PLAYING, the pipeline asks each element if they can
provide a clock. The clock is selected in the following order:
- If the application selected a clock, use that one.
- If a source element provides a clock, use that clock.
- Select a clock from any other element that provides a clock, start with the
sinks.
- If no element provides a clock a default system clock is used for the pipeline.
In a typical playback pipeline this algorithm will select the clock provided by
a sink element such as an audio sink.
In capture pipelines, this will typically select the clock of the data producer, which
can in most cases not control the rate at which it produces data.
Pipeline states
---------------
When all the pads are linked or signals have been connected, the pipeline can
be put in the PAUSED state to start dataflow.
When a bin (and hence a pipeline) performs a state change, it will change the state
of all its children. The pipeline will change the state of its children from the
sink elements to the source elements, this to make sure that no upstream element
produces data to an element that is not yet ready to accept it.
In the mp3 playback pipeline, the state of the elements is changed in the order
alsasink, mp3dec, filesrc.
All intermediate states are traversed for each element resulting in the following
chain of state changes:
alsasink to READY: the audio device is probed
mp3dec to READY: nothing happens.
filesrc to READY: the file is probed
alsasink to PAUSED: the audio device is opened. alsasink is a sink and returns
ASYNC because it did not receive data yet.
mp3dec to PAUSED: the decoding library is initialized
filesrc to PAUSED: the file is opened and a thread is started to push data to
mp3dec
At this point data flows from filesrc to mp3dec and alsasink. Since mp3dec is PAUSED,
it accepts the data from filesrc on the sinkpad and starts decoding the compressed
data to raw audio samples.
The mp3 decoder figures out the samplerate, the number of channels and other audio
properties of the raw audio samples, puts the decoded samples into a Buffer,
attaches the media type caps to the buffer and pushes this buffer to the next
element.
Alsasink then receives the buffer, inspects the caps and reconfigures itself to process
the buffer. Since it received the first buffer of samples, it completes the state change
to the PAUSED state. At this point the pipeline is prerolled and all elements have
samples. Alsasink is now also capable of providing a clock to the pipeline.
Since alsasink is now in the PAUSED state it blocks while receiving the first buffer. This
effectively blocks both mp3dec and filesrc in their gst_pad_push().
Since all elements now return SUCCESS from the gst_element_get_state() function,
the pipeline can be put in the PLAYING state.
Before going to PLAYING, the pipeline select a clock and samples the current time of
the clock. This is the base time. It then distributes this time to all elements.
Elements can then synchronize against the clock using the buffer timestamp+base time.
The following chain of state changes then takes place:
alsasink to PLAYING: the samples are played to the audio device
mp3dec to PLAYING: nothing happens
filesrc to PLAYING: nothing happens
Pipeline status
---------------
The pipeline informs the application of any special events that occur in the
pipeline with the bus. The bus is an object that the pipeline provides and that
can be retrieved with gst_pipeline_get_bus().
The bus can be polled or added to the glib mainloop.
The bus is distributed to all elements added to the pipeline. The elements use the bus
to post messages on. Various message types exist such as ERRORS, WARNINGS, EOS,
STATE_CHANGED, etc..
The pipeline handles EOS messages received from elements in a special way. It will
only forward the message to the application when all sink elements have posted an
EOS message.
Other methods for obtaining the pipeline status include the Query functionality that
can be performed with gst_element_query() on the pipeline. This type of query
is useful for obtaining information about the current position and total time of
the pipeline. It can also be used to query for the supported seeking formats and
ranges.
Pipeline EOS
------------
When the source filter encounters the end of the stream, it sends an EOS event to
the peer element. This event will then travel downstream to all of the connected
elements to inform them of the EOS. The element is not supposed to accept any more
data after receiving an EOS event on a sinkpad.
The element providing the streaming thread stops sending data after sending the
EOS message.
The EOS even will eventually arrive in the sink element. The sink will then post
an EOS message on the bus to inform the pipeline that a particular stream has
finished. When all sinks have reported EOS, the pipeline forwards the EOS message
to the application.
When in EOS, the pipeline remains in the playing state, it is the applications
responsability to PAUSE or READY the pipeline. The application can also issue
a seek, for example.
Pipeline READY
--------------
When a running pipeline is set from the PLAYING to READY state, the following
actions occur in the pipeline:
alsasink to PAUSED: alsasink blocks and completes the state change on the
next sample. If the element was EOS, it does not wait for
a sample to complete the state change.
mp3dec to PAUSED: nothing
filesrc to PAUSED: nothing
Going to the intermediate PAUSED state will block all elements in the _push()
functions. This happens because the sink element blocks on the first buffer
it receives.
Some elements might be performing blocking operations in the PLAYING state that
must be unblocked when they go into the PAUSED state. This makes sure that the
state change happens very fast.
In the next PAUSED to READY state change the pipeline has to shut down and all
streaming threads must stop sending data. This happens in the following sequence:
alsasink to READY: alsasink unblocks from the _chain() function and returns a
WRONG_STATE return value to the peer element. The sinkpad is
deactivated and becomes unusable for sending more data.
mp3dec to READY: the pads are deactivated and the state change completes when
mp3dec leaves its _chain() function.
filesrc to READY: the pads are deactivated and the thread is paused.
The upstream elements finish their chain() function because the downstream element
returned an error code (WRONG_STATE) from the _push() functions. These error codes
are eventually returned to the element that started the streaming thread (filesrc),
which pauses the thread and completes the state change.
This sequence of events ensure that all elements are unblocked and all streaming
threads stopped.
Pipeline seeking
----------------
Seeking in the pipeline requires a very specific order of operations to make
sure that the elements remain synchronized and that the seek is performed with
a minimal amount of latency.
An application issues a seek event on the pipeline using gst_element_send_event()
on the pipeline element. The event can be a seek event in any of the formats
supported by the elements.
The pipeline first pauses the pipeline to speed up the seek operations.
The pipeline then issues the seek event to all sink elements. The sink then forwards
the seek event upstream until some element can perform the seek operation, which is
typically the source or demuxer element. All intermediate elements can transform the
requested seek offset to another format, this way a decoder element can transform a
seek to a frame number to a timestamp, for example.
When the seek event reaches an element that will perform the seek operation, that
element performs the following steps.
1) send a FLUSH_START event to all downstream and upstream peer elements.
2) make sure the streaming thread is not running. The streaming thread will
always stop because of step 1).
3) perform the seek operation
4) send a FLUSH done event to all downstream and upstream peer elements.
5) send NEWSEGMENT event to inform all elements of the new position and to complete
the seek.
In step 1) all dowstream elements have to return from any blocking operations
and have to refuse any further buffers or events different from a FLUSH done.
The first step ensures that the streaming thread eventually unblocks and that
step 2) can be performed. At this point, dataflow is completely stopped in the
pipeline.
In step 3) the element performs the seek to the requested position.
In step 4) all peer elements are allowed to accept data again and streaming
can continue from the new position. A FLUSH done event is sent to all the peer
elements so that they accept new data again and restart their streaming threads.
Step 5) informs all elements of the new position in the stream. After that the
event function returns back to the application. and the streaming threads start
to produce new data.
Since the pipeline is still PAUSED, this will preroll the next media sample in the
sinks. The application can wait for this preroll to complete by performing a
_get_state() on the pipeline.
The last step in the seek operation is then to adjust the stream time of the pipeline
to 0 and to set the pipeline back to PLAYING.
The sequence of events in our mp3 playback example.
| a) seek on pipeline
| b) PAUSE pipeline
+----------------------------------V--------+
| pipeline | c) seek on sink
| +---------+ +----------+ +---V------+ |
| | filesrc | | mp3dec | | alsasink | |
| | src-sink src-sink | |
| +---------+ +----------+ +----|-----+ |
+-----------------------------------|-------+
<------------------------+
d) seek travels upstream
--------------------------> 1) FLUSH event
| 2) stop streaming
| 3) perform seek
--------------------------> 4) FLUSH done event
--------------------------> 5) NEWSEGMENT event
| e) update stream time to 0
| f) PLAY pipeline
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