gstreamer/markdown/design/buffering.md

Buffering

This document outlines the buffering policy used in the GStreamer core that can be used by plugins and applications.

The purpose of buffering is to accumulate enough data in a pipeline so that playback can occur smoothly and without interruptions. It is typically done when reading from a (slow) non-live network source but can also be used for live sources.

We want to be able to implement the following features:

• buffering up to a specific amount of data, in memory, before starting playback so that network fluctuations are minimized.

• download of the network file to a local disk with fast seeking in the downloaded data. This is similar to the quicktime/youtube players.

• caching of semi-live streams to a local, on disk, ringbuffer with seeking in the cached area. This is similar to tivo-like timeshifting.

• progress report about the buffering operations

• the possibility for the application to do more complex buffering

Some use cases

Stream buffering

+---------+     +---------+     +-------+
| httpsrc |     | buffer  |     | demux |
|        src - sink      src - sink     ....
+---------+     +---------+     +-------+

In this case we are reading from a slow network source into a buffer element (such as queue2).

The buffer element has a low and high watermark expressed in bytes. The buffer uses the watermarks as follows:

• The buffer element will post BUFFERING messages until the high watermark is hit. This instructs the application to keep the pipeline PAUSED, which will eventually block the srcpad from pushing while data is prerolled in the sinks.

• When the high watermark is hit, a BUFFERING message with 100% will be posted, which instructs the application to continue playback.

• When the low watermark is hit during playback, the queue will start posting BUFFERING messages again, making the application PAUSE the pipeline again until the high watermark is hit again. This is called the rebuffering stage.

• During playback, the queue level will fluctuate between the high and low watermarks as a way to compensate for network irregularities.

This buffering method is usable when the demuxer operates in push mode. Seeking in the stream requires the seek to happen in the network source. It is mostly desirable when the total duration of the file is not known, such as in live streaming or when efficient seeking is not possible/required.

Incremental download

+---------+     +---------+     +-------+
| httpsrc |     | buffer  |     | demux |
|        src - sink      src - sink     ....
+---------+     +----|----+     +-------+
                     V
                    file

In this case, we know the server is streaming a fixed length file to the client. The application can choose to download the file to disk. The buffer element will provide a push or pull based srcpad to the demuxer to navigate in the downloaded file.

This mode is only suitable when the client can determine the length of the file on the server.

In this case, buffering messages will be emitted as usual when the requested range is not within the downloaded area + buffersize. The buffering message will also contain an indication that incremental download is being performed. This flag can be used to let the application control the buffering in a more intelligent way, using the BUFFERING query, for example.

The application can use the BUFFERING query to get the estimated download time and match this time to the current/remaining playback time to control when playback should start to have a non-interrupted playback experience.

Timeshifting

+---------+     +---------+     +-------+
| httpsrc |     | buffer  |     | demux |
|        src - sink      src - sink     ....
+---------+     +----|----+     +-------+
                     V
              file-ringbuffer

In this mode, a fixed size ringbuffer is kept to download the server content. This allows for seeking in the buffered data. Depending on the size of the buffer one can seek further back in time.

This mode is suitable for all live streams.

As with the incremental download mode, buffering messages are emitted along with an indication that timeshifting download is in progress.

Live buffering

In live pipelines we usually introduce some latency between the capture and the playback elements. This latency can be introduced by a queue (such as a jitterbuffer) or by other means (in the audiosink).

Buffering messages can be emitted in those live pipelines as well and serve as an indication to the user of the latency buffering. The application usually does not react to these buffering messages with a state change.

Messages

A GST_MESSAGE_BUFFERING must be posted on the bus when playback temporarily stops to buffer and when buffering finishes. When the percentage field in the BUFFERING message is 100, buffering is done. Values less than 100 mean that buffering is in progress.

The BUFFERING message should be intercepted and acted upon by the application. The message contains at least one field that is sufficient for basic functionality:

• buffer-percent, G_TYPE_INT: between 0 and 100

Several more clever ways of dealing with the buffering messages can be used when in incremental or timeshifting download mode. For this purpose additional fields are added to the buffering message:

• buffering-mode, GST_TYPE_BUFFERING_MODE: enum { "stream", "download", "timeshift", "live" }: Buffering mode in use. See above for an explanation of the different alternatives. This field can be used to let the application have more control over the buffering process.

• avg-in-rate, G_TYPE_INT: Average input buffering speed in bytes/second. -1 is unknown. This is the average number of bytes per second that is received on the buffering element input (sink) pads. It is a measurement of the network speed in most cases.

• avg-out-rate, G_TYPE_INT: Average consumption speed in bytes/second. -1 is unknown. This is the average number of bytes per second that is consumed by the downstream element of the buffering element.

• buffering-left, G_TYPE_INT64: Estimated time that buffering will take in milliseconds. -1 is unknown. This is measured based on the avg-in-rate and the filled level of the queue. The application can use this hint to update the GUI about the estimated remaining time that buffering will take.

Application

While data is buffered the pipeline should remain in the PAUSED state. It is also possible that more data should be buffered while the pipeline is PLAYING, in which case the pipeline should be PAUSED until the buffering finishes.

BUFFERING messages can be posted while the pipeline is prerolling. The application should not set the pipeline to PLAYING before a BUFFERING message with a 100 percent value is received, which might only happen after the pipeline prerolls.

An exception is made for live pipelines. The application may not change the state of a live pipeline when a buffering message is received. Usually these buffering messages contain the "buffering-mode" = "live".

The buffering message can also instruct the application to switch to a periodical BUFFERING query instead, so it can more precisely control the buffering process. The application can, for example, choose not to act on the BUFFERING complete message (buffer-percent = 100) to resume playback but use the estimated download time instead, resuming playback when it has determined that it should be able to provide uninterrupted playback.

Buffering Query

In addition to the BUFFERING messages posted by the buffering elements, we want to be able to query the same information from the application. We also want to be able to present the user with information about the downloaded range in the file so that the GUI can react on it.

In addition to all the fields present in the buffering message, the BUFFERING query contains the following field, which indicates the available downloaded range in a specific format and the estimated time to complete:

• busy, G_TYPE_BOOLEAN: if buffering was busy. This flag allows the application to pause the pipeline by using the query only.

• format, GST_TYPE_FORMAT: the format of the "start" and "stop" values below

• start, G_TYPE_INT64, -1 unknown: the start position of the available data. If there are multiple ranges, this field contains the start position of the currently downloading range.

• stop, G_TYPE_INT64, -1 unknown: the stop position of the available data. If there are multiple ranges, this field contains the stop position of the currently downloading range.

• estimated-total, G_TYPE_INT64: gives the estimated download time in milliseconds. -1 unknown. When the size of the downloaded file is known, this value will contain the latest estimate of the remaining download time of the currently downloading range. This value is usually only filled for the "download" buffering mode. The application can use this information to estimate the amount of remaining time to download till the end of the file.

• buffering-ranges, G_TYPE_ARRAY of GstQueryBufferingRange: contains optionally the downloaded areas in the format given above. One of the ranges contains the same start/stop position as above:

    typedef struct
    {
      gint64 start;
      gint64 stop;
    } GstQueryBufferingRange;

For the download and timeshift buffering-modes, the start and stop positions specify the ranges where efficient seeking in the downloaded media is possible. Seeking outside of these ranges might be slow or not at all possible.

For the stream and live mode the start and stop values describe the oldest and newest item (expressed in format) in the buffer.

Defaults

Some defaults for common elements:

A GstBaseSrc with random access replies to the BUFFERING query with:

"buffer-percent" = 100
"buffering-mode" = "stream"
"avg-in-rate" = -1
"avg-out-rate" = -1
"buffering-left" = 0
"format" = GST_FORMAT_BYTES
"start" = 0
"stop" = the total filesize
"estimated-total" = 0
"buffering-ranges" = NULL

A GstBaseSrc in push mode replies to the BUFFERING query with:

"buffer-percent" = 100
"buffering-mode" = "stream"
"avg-in-rate" = -1
"avg-out-rate" = -1
"buffering-left" = 0
"format" = a valid GST_TYPE_FORMAT
"start" = current position
"stop" = current position
"estimated-total" = -1
"buffering-ranges" = NULL

Buffering strategies

Buffering strategies are specific implementations based on the buffering message and query described above.

Most strategies have to balance buffering time versus maximal playback experience.

Simple buffering

NON-live pipelines are kept in the paused state while buffering messages with a percent < 100% are received.

This buffering strategy relies on the buffer size and low/high watermarks of the element. It can work with a fixed size buffer in memory or on disk.

The size of the buffer is usually expressed in a fixed amount of time units and the estimated bitrate of the upstream source is used to convert this time to bytes.

All GStreamer applications must implement this strategy. Failure to do so will result in starvation at the sink.

No-rebuffer strategy

This strategy tries to buffer as much data as possible so that playback can continue without any further rebuffering.

This strategy is initially similar to simple buffering, the difference is in deciding on the condition to continue playback. When a 100% buffering message has been received, the application will not yet start the playback but it will start a periodic buffering query, which will return the estimated amount of buffering time left. When the estimated time left is less than the remaining playback time, playback can continue.

This strategy requires a unlimited buffer size in memory or on disk, such as provided by elements that implement the incremental download buffering mode.

Usually, the application can choose to start playback even before the remaining buffer time elapsed in order to more quickly start the playback at the expense of a possible rebuffering phase.

Incremental rebuffering

The application implements the simple buffering strategy but with each rebuffering phase, it increases the size of the buffer.

This strategy has quick, fixed time startup times but incrementally longer rebuffering times if the network is slower than the media bitrate.