<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">The name of the template for the sink pad.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Gives the pointer to the sink #GstPad object of the element.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">The name of the template for the source pad.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Gives the pointer to the source #GstPad object of the element.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">This base class is for decoders which do not operate on a streaming model.
That is: they load the encoded media at once, as part of an initialization,
and afterwards can decode samples (sometimes referred to as "rendering the
samples").
This sets it apart from GstAudioDecoder, which is a base class for
streaming audio decoders.
The base class is conceptually a mix between decoder and parser. This is
unavoidable, since virtually no format that isn't streaming based has a
clear distinction between parsing and decoding. As a result, this class
also handles seeking.
Non-streaming audio formats tend to have some characteristics unknown to
more "regular" bitstreams. These include subsongs and looping.
Subsongs are a set of songs-within-a-song. An analogy would be a multitrack
recording, where each track is its own song. The first subsong is typically
the "main" one. Subsongs were popular for video games to enable context-
aware music; for example, subsong `#0` would be the "main" song, `#1` would be
an alternate song playing when a fight started, `#2` would be heard during
conversations etc. The base class is designed to always have at least one
subsong. If the subclass doesn't provide any, the base class creates a
"pseudo" subsong, which is actually the whole song.
Downstream is informed about the subsong using a table of contents (TOC),
but only if there are at least 2 subsongs.
Looping refers to jumps within the song, typically backwards to the loop
start (although bi-directional looping is possible). The loop is defined
by a chronological start and end; once the playback position reaches the
loop end, it jumps back to the loop start.
Depending on the subclass, looping may not be possible at all, or it
may only be possible to enable/disable it (that is, either no looping, or
an infinite amount of loops), or it may allow for defining a finite number
of times the loop is repeated.
Looping can affect output in two ways. Either, the playback position is
reset to the start of the loop, similar to what happens after a seek event.
Or, it is not reset, so the pipeline sees playback steadily moving forwards,
the playback position monotonically increasing. However, seeking must
always happen within the confines of the defined subsong duration; for
example, if a subsong is 2 minutes long, steady playback is at 5 minutes
(because infinite looping is enabled), then seeking will still place the
position within the 2 minute period.
Loop count 0 means no looping. Loop count -1 means infinite looping.
Nonzero positive values indicate how often a loop shall occur.
If the initial subsong and loop count are set to values the subclass does
not support, the subclass has a chance to correct these values.
@get_property then reports the corrected versions.
The base class operates as follows:
* Unloaded mode
- Initial values are set. If a current subsong has already been
defined (for example over the command line with gst-launch), then
the subsong index is copied over to current_subsong .
Same goes for the num-loops and output-mode properties.
Media is NOT loaded yet.
- Once the sinkpad is activated, the process continues. The sinkpad is
activated in push mode, and the class accumulates the incoming media
data in an adapter inside the sinkpad's chain function until either an
EOS event is received from upstream, or the number of bytes reported
by upstream is reached. Then it loads the media, and starts the decoder
output task.
- If upstream cannot respond to the size query (in bytes) of @load_from_buffer
fails, an error is reported, and the pipeline stops.
- If there are no errors, @load_from_buffer is called to load the media. The
subclass must at least call gst_nonstream_audio_decoder_set_output_format()
there, and is free to make use of the initial subsong, output mode, and
position. If the actual output mode or position differs from the initial
value,it must set the initial value to the actual one (for example, if
the actual starting position is always 0, set *initial_position to 0).
If loading is unsuccessful, an error is reported, and the pipeline
stops. Otherwise, the base class calls @get_current_subsong to retrieve
the actual current subsong, @get_subsong_duration to report the current
subsong's duration in a duration event and message, and @get_subsong_tags
to send tags downstream in an event (these functions are optional; if
set to NULL, the associated operation is skipped). Afterwards, the base
class switches to loaded mode, and starts the decoder output task.
* Loaded mode</title>
- Inside the decoder output task, the base class repeatedly calls @decode,
which returns a buffer with decoded, ready-to-play samples. If the
subclass reached the end of playback, @decode returns FALSE, otherwise
TRUE.
- Upon reaching a loop end, subclass either ignores that, or loops back
to the beginning of the loop. In the latter case, if the output mode is set
to LOOPING, the subclass must call gst_nonstream_audio_decoder_handle_loop()
*after* the playback position moved to the start of the loop. In
STEADY mode, the subclass must *not* call this function.
Since many decoders only provide a callback for when the looping occurs,
and that looping occurs inside the decoding operation itself, the following
mechanism for subclass is suggested: set a flag inside such a callback.
Then, in the next @decode call, before doing the decoding, check this flag.
If it is set, gst_nonstream_audio_decoder_handle_loop() is called, and the
flag is cleared.
(This function call is necessary in LOOPING mode because it updates the
current segment and makes sure the next buffer that is sent downstream
has its DISCONT flag set.)
- When the current subsong is switched, @set_current_subsong is called.
If it fails, a warning is reported, and nothing else is done. Otherwise,
it calls @get_subsong_duration to get the new current subsongs's
duration, @get_subsong_tags to get its tags, reports a new duration
(i.e. it sends a duration event downstream and generates a duration
message), updates the current segment, and sends the subsong's tags in
an event downstream. (If @set_current_subsong has been set to NULL by
the subclass, attempts to set a current subsong are ignored; likewise,
if @get_subsong_duration is NULL, no duration is reported, and if
@get_subsong_tags is NULL, no tags are sent downstream.)
- When an attempt is made to switch the output mode, it is checked against
the bitmask returned by @get_supported_output_modes. If the proposed
new output mode is supported, the current segment is updated
(it is open-ended in STEADY mode, and covers the (sub)song length in
LOOPING mode), and the subclass' @set_output_mode function is called
unless it is set to NULL. Subclasses should reset internal loop counters
in this function.
The relationship between (sub)song duration, output mode, and number of loops
is defined this way (this is all done by the base class automatically):
* Segments have their duration and stop values set to GST_CLOCK_TIME_NONE in
STEADY mode, and to the duration of the (sub)song in LOOPING mode.
* The duration that is returned to a DURATION query is always the duration
of the (sub)song, regardless of number of loops or output mode. The same
goes for DURATION messages and tags.
* If the number of loops is >0 or -1, durations of TOC entries are set to
the duration of the respective subsong in LOOPING mode and to G_MAXINT64 in
STEADY mode. If the number of loops is 0, entry durations are set to the
subsong duration regardless of the output mode.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Required if loads_from_sinkpad is set to TRUE (the default value).
Loads the media from the given buffer. The entire media is supplied at once,
so after this call, loading should be finished. This function
can also make use of a suggested initial subsong & subsong mode and initial
playback position (but isn't required to). In case it chooses a different starting
position, the function must pass this position to *initial_position.
The subclass does not have to unref the input buffer; the base class does that
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Required if loads_from_sinkpad is set to FALSE.
Loads the media in a way defined by the custom sink. Data is not supplied;
the derived class has to handle this on its own. Otherwise, this function is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Allocates an output buffer with the internally configured buffer pool.
This function may only be called from within @load_from_buffer,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Size of the output buffer, in bytes</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Gets sample format, sample rate, channel count from the allowed srcpad caps.
This is useful for when the subclass wishes to adjust one or more output
parameters to whatever downstream is supporting. For example, the output
sample rate is often a freely adjustable value in module players.
This function tries to find a value inside the srcpad peer's caps for
@format, @sample_rate, @num_chnanels . Any of these can be NULL; they
(and the corresponding downstream caps) are then skipped while retrieving
information. Non-fixated caps are fixated first; the value closest to
their present value is then chosen. For example, if the variables pointed
to by the arguments are GST_AUDIO_FORMAT_16, 48000 Hz, and 2 channels,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">#GstAudioFormat value to fill with a sample format</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Integer to fill with a sample rate</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Integer to fill with a channel count</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Reports that a loop has been completed and creates a new appropriate
segment for the next loop.
@new_position exists because a loop may not start at the beginning.
This function is only useful for subclasses which can be in the
GST_NONSTREAM_AUDIO_OUTPUT_MODE_LOOPING output mode, since in the
GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY output mode, this function
does nothing. See #GstNonstreamAudioOutputMode for more details.
The subclass calls this during playback when it loops. It produces
a new segment with updated base time and internal time values, to allow
for seamless looping. It does *not* check the number of elapsed loops;
this is up the subclass.
Note that if this function is called, then it must be done after the
last samples of the loop have been decoded and pushed downstream.
This function must be called with the decoder mutex lock held, since it
is typically called from within @decode (which in turn are called with
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Sets the output caps by means of a GstAudioInfo structure.
This must be called latest in the first @decode call, to ensure src caps are
set before decoded samples are sent downstream. Typically, this is called
from inside @load_from_buffer or @load_from_custom.
This function must be called with the decoder mutex lock held, since it
is typically called from within the aforementioned vfuncs (which in turn
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">TRUE if setting the output format succeeded, FALSE otherwise</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Valid audio info structure containing the output format</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Convenience function; sets the output caps by means of common parameters.
Internally, this fills a GstAudioInfo structure and calls
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">TRUE if setting the output format succeeded, FALSE otherwise</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Output sample rate to use, in Hz</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Number of output channels to use</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Subclasses can override any of the available optional virtual methods or not, as
needed. At minimum, @load_from_buffer (or @load_from_custom), @get_supported_output_modes,
and @decode need to be overridden.
All functions are called with a locked decoder mutex.
> If GST_ELEMENT_ERROR, GST_ELEMENT_WARNING, or GST_ELEMENT_INFO are called from
> inside one of these functions, it is strongly recommended to unlock the decoder mutex
> before and re-lock it after these macros to prevent potential deadlocks in case the
> application does something with the element when it receives an ERROR/WARNING/INFO
> message. Same goes for gst_element_post_message() calls and non-serialized events.
By default, this class works by reading media data from the sinkpad, and then commencing
playback. Some decoders cannot be given data from a memory block, so the usual way of
reading all upstream data and passing it to @load_from_buffer doesn't work then. In this case,
set the value of loads_from_sinkpad to FALSE. This changes the way this class operates;
it does not require a sinkpad to exist anymore, and will call @load_from_custom instead.
One example of a decoder where this makes sense is UADE (Unix Amiga Delitracker Emulator).
For some formats (such as TFMX), it needs to do the file loading by itself.
Since most decoders can read input data from a memory block, the default value of
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Required if loads_from_sinkpad is set to TRUE (the default value).
Loads the media from the given buffer. The entire media is supplied at once,
so after this call, loading should be finished. This function
can also make use of a suggested initial subsong & subsong mode and initial
playback position (but isn't required to). In case it chooses a different starting
position, the function must pass this position to *initial_position.
The subclass does not have to unref the input buffer; the base class does that
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Required if loads_from_sinkpad is set to FALSE.
Loads the media in a way defined by the custom sink. Data is not supplied;
the derived class has to handle this on its own. Otherwise, this function is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">The output mode defines how the output behaves with regards to looping. Either the playback position is
moved back to the beginning of the loop, acting like a backwards seek, or it increases steadily, as if
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Playback position is moved back to the beginning of the loop</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Playback position increases steadily, even when looping</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Only the current subsong is played</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">All subsongs are played (current subsong index is ignored)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">This class is similar to GstAdapter, but it is made to work with
non-interleaved (planar) audio buffers. Before using, an audio format
must be configured with gst_planar_audio_adapter_configure()</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Creates a new #GstPlanarAudioAdapter. Free with g_object_unref().</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Gets the maximum amount of samples available, that is it returns the maximum
value that can be supplied to gst_planar_audio_adapter_get_buffer() without
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">number of samples available in @adapter</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Removes all buffers from @adapter.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Sets up the @adapter to handle audio data of the specified audio format.
Note that this will internally clear the adapter and re-initialize it.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstAudioInfo describing the format of the audio data</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Get the DTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">The DTS at the last discont or GST_CLOCK_TIME_NONE.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Flushes the first @to_flush samples in the @adapter. The caller must ensure
that at least this many samples are available.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Returns a #GstBuffer containing the first @nsamples of the @adapter, but
does not flush them from the adapter.
Use gst_planar_audio_adapter_take_buffer() for flushing at the same time.
The map @flags can be used to give an optimization hint to this function.
When the requested buffer is meant to be mapped only for reading, it might
be possible to avoid copying memory in some cases.
Caller owns a reference to the returned buffer. gst_buffer_unref() after
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">hint the intended use of the returned buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Get the offset that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">The offset at the last discont or GST_BUFFER_OFFSET_NONE.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Get the dts that was before the current sample in the adapter. When
@distance is given, the amount of bytes between the dts and the current
position is returned.
The dts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
the adapter is first created or when it is cleared. This also means that
before the first sample with a dts is removed from the adapter, the dts
and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">pointer to location for distance, or %NULL</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Get the offset that was before the current sample in the adapter. When
@distance is given, the amount of samples between the offset and the current
position is returned.
The offset is reset to GST_BUFFER_OFFSET_NONE and the distance is set to 0
when the adapter is first created or when it is cleared. This also means that
before the first sample with an offset is removed from the adapter, the
offset and distance returned are GST_BUFFER_OFFSET_NONE and 0 respectively.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">pointer to a location for distance, or %NULL</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Get the pts that was before the current sample in the adapter. When
@distance is given, the amount of samples between the pts and the current
position is returned.
The pts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
the adapter is first created or when it is cleared. This also means that before
the first sample with a pts is removed from the adapter, the pts
and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">pointer to location for distance, or %NULL</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Get the PTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">The PTS at the last discont or GST_CLOCK_TIME_NONE.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Adds the data from @buf to the data stored inside @adapter and takes
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstBuffer to queue in the adapter</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Returns a #GstBuffer containing the first @nsamples bytes of the
@adapter. The returned bytes will be flushed from the adapter.
See gst_planar_audio_adapter_get_buffer() for more details.
Caller owns a reference to the returned buffer. gst_buffer_unref() after
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">hint the intended use of the returned buffer</doc>
