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204 lines
8.8 KiB
Text
204 lines
8.8 KiB
Text
Trickmodes
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==========
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GStreamer provides API for performing various trickmode playback. This includes:
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- server side trickmodes
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- client side fast/slow forward playback
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- client side fast/slow backwards playback
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Server side trickmodes mean that a source (network source) can provide a
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stream with different playback speed and direction. The client does not have to
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perform any special algorithms to decode this stream.
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Client side trickmodes mean that the decoding client (GStreamer) performs the
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needed algorithms to change the direction and speed of the media file.
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Seeking can both be done in a playback pipeline and a transcoding pipeline.
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General seeking overview
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------------------------
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Consider a typical playback pipeline:
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.---------. .------.
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.-------. | decoder |->| sink |
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.--------. | |-->'---------' '------'
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| source |->| demux |
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'--------' | |-->.---------. .------.
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'-------' | decoder |->| sink |
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'---------' '------'
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The pipeline is initially configured to play back at speed 1.0 starting from
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position 0 and stopping at the total duration of the file.
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When performing a seek, the following steps have to be taken by the application:
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1) Create a seek event:
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The seek event contains:
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- various flags describing:
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- where to seek to (KEY_UNIT)
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- how accurate the seek should be (ACCURATE)
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- how to perform the seek (FLUSH)
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- what to do when the stop position is reached (SEGMENT).
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- extra playback options (SKIP)
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- a format to seek in, this can be time, bytes, units (frames, samples), ...
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- a playback rate, 1.0 is normal playback speed, positive values bigger than 1.0
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mean fast playback. negative values mean reverse playback. A playback speed of
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0.0 is not allowed (but is equivalent to PAUSING the pipeline).
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- a start position, this value has to be between 0 and the total duration of the
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file. It can also be relative to the previously configured start value.
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- a stop position, this value has to be between 0 and the total duration. It can
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also be relative to the previously configured stop value.
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See also gst_event_new_seek().
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2) Send the seek event to the pipeline with gst_element_send_event()
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By default the pipeline will send the event to all sink elements.
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By default an element will forward the event upstream on all sinkpads.
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Elements can modify the format of the seek event. The most common format is
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GST_FORMAT_TIME.
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One element will actually perform the seek, this is usually the demuxer or
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source element. For more information on how to perform the different seek
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types see part-seeking.txt.
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For client side trickmode a NEW_SEGMENT event will be sent downstream with
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the new rate and start/stop positions. All elements prepare themselves to
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handle the rate (see below). The applied rate of the NEW_SEGMENT event will
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be set to 1.0 to indicate that no rate adjustment has been done.
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for server side trick mode a NEW_SEGMENT event is sent downstream with a
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rate of 1.0 and the start/stop positions. The elements will configure themselves
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for normal playback speed since the server will perform the rate conversions.
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The applied rate will be set to the rate that will be applied by the server. This
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is done to insure that the position reporting performed in the sink is aware
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of the trick mode.
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When the seek succeeds, the _send_event() function will return TRUE.
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Server side trickmode
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---------------------
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The source element operates in push mode. It can reopen a server connection requesting
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a new byte or time position and a new playback speed. The capabilities can be queried
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from the server when the connection is opened.
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We assume the source element is derived from the GstPushSrc base class. The base source
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should be configured with gst_base_src_set_format (src, GST_FORMAT_TIME).
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The do_seek method will be called on the push src subclass with the seek information
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passed in the GstSegment argument.
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The rate value in the segment should be used to reopen the connection to the server
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requesting data at the new speed and possibly a new playback position.
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When the server connection was successfully reopened, set the rate of the segment
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to 1.0 so that the client side trickmode is not enabled. The applied rate in the
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segment is set to the rate transformation done by the server.
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Alternatively a combination of client side and serverside trickmode can be used, for
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example if the server does not support certain rates, the client can perform rate
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conversion for the remainder.
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source server
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do_seek | |
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----------->| |
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| reopen connection |
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|-------------------->|
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| .
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| success .
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|<--------------------|
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modify | |
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rate to 1.0 | |
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return | |
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TRUE | |
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After performing the seek, the source will inform the downstream elements of the
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new segment that is to be played back. Since the segment will have a rate of 1.0,
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no client side trick modes are enabled. The segment will have an applied rate
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different from 1.0 to indicate that the media contains data with non-standard
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playback speed or direction.
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client side forward trickmodes
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------------------------------
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The seek happens as stated above. a NEW_SEGMENT event is sent downstream with a rate
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different from 1.0. Plugins receiving the NEW_SEGMENT can decide to perform the
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rate conversion of the media data (retimestamp video frames, resample audio, ...).
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A plugin should not drop frames unless the SKIP flag is set. When the SKIP flag is
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set, a plugin can decide to drop frames in the case of fast playback or use a more
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efficient decoding algorithm (skip B frames, ...). If a plugin decides to resample
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or retimestamp, it should modify the NEW_SEGMENT with a rate of 1.0 and update the
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applied rate so that downstream elements don't resample again but are aware that the
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media has been modified.
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The GStreamer base audio and video sinks will resample automatically if they receive
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a NEW_SEGMENT event with a rate different from 1.0. The position reporting in the
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base audio and video sinks will also depend on the applied rate of the segment
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information.
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client side backwards trickmode
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-------------------------------
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For backwards playback the following rules apply:
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- the rate in the NEW_SEGMENT is less than 0.0.
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- the NEW_SEGMENT start position is less than the stop position, playback will
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however happen from stop to start in reverse.
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- the time member in the NEW_SEGMENT is set to the stream time of the start
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position.
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For plugins the following rules apply:
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- A source plugin sends data in chunks starting from the last chunk of the
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file. The actual bytes are not reversed. Each chunk that is not forward
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continuous with the previous chunk is marked with a DISCONT flag.
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- A demuxer accumulates the chunks. As soon as a keyframe is found, everything
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starting from the keyframe up to the accumulated data is sent downstream.
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Timestamps on the buffers are set starting from the stop position to start,
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effectively going backwards. Chunks are marked with DISCONT when they are not
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forward continuous with the previous buffer.
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- A video decoder decodes and accumulates all decoded frames. If a buffer with
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a DISCONT, accumulate NEWSEGMENT or EOS is received, all accumulated frames
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are sent downsteam in reverse.
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- An audio decoder decodes and accumulates all decoded audio. If a buffer with
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a DISCONT, accumulate NEWSEGMENT or EOS is received, all accumulated audio
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is sent downstream in reverse order. Some audio codecs need the previous
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data buffer to decode the current one, in that case, the previous DISCONT
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buffer needs to be combined with the last non-DISCONT buffer to generate the
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last bit of output.
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- A sink reverses (for audio) and retimestamps (audio, video) the buffers
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before playing them back. Retimestamping occurs relative to the stop
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position, making the timestamps increase again and suitable for synchronizing
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against the clock.
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Audio sinks also have to perform simple resampling before playing the
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samples.
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- for transcoding, audio and video resamplers can be used to reverse, resample
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and retimestamp the buffers. Any rate adjustments performed on the media must
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be added to the applied_rate and subtracted from the rate members in the
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NEWSEGMENT event.
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Notes:
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- The clock/running_time keeps running forward.
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- backwards playback potentially uses a lot of memory as frames and undecoded
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data gets buffered.
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