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Diffstat (limited to 'girs/GstBadAudio-1.0.gir')
| -rw-r--r-- | girs/GstBadAudio-1.0.gir | 1479 |
1 files changed, 1479 insertions, 0 deletions
diff --git a/girs/GstBadAudio-1.0.gir b/girs/GstBadAudio-1.0.gir new file mode 100644 index 0000000000..9ba916248b --- /dev/null +++ b/girs/GstBadAudio-1.0.gir @@ -0,0 +1,1479 @@ +<!-- This file was automatically generated from C sources - DO NOT EDIT! +To affect the contents of this file, edit the original C definitions, +and/or use gtk-doc annotations. --> +<repository xmlns="http://www.gtk.org/introspection/core/1.0" xmlns:c="http://www.gtk.org/introspection/c/1.0" xmlns:glib="http://www.gtk.org/introspection/glib/1.0" version="1.2"> + <include name="Gst" version="1.0"/> + <include name="GstAudio" version="1.0"/> + <include name="GstBase" version="1.0"/> + <package name="gstreamer-bad-audio-1.0"/> + <namespace name="GstBadAudio" version="1.0" shared-library="libgstbadaudio-1.0.so.0" c:identifier-prefixes="Gst" c:symbol-prefixes="gst"> + <function-macro name="IS_NONSTREAM_AUDIO_DECODER" c:identifier="GST_IS_NONSTREAM_AUDIO_DECODER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_NONSTREAM_AUDIO_DECODER_CLASS" c:identifier="GST_IS_NONSTREAM_AUDIO_DECODER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_PLANAR_AUDIO_ADAPTER" c:identifier="GST_IS_PLANAR_AUDIO_ADAPTER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_PLANAR_AUDIO_ADAPTER_CLASS" c:identifier="GST_IS_PLANAR_AUDIO_ADAPTER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="NONSTREAM_AUDIO_DECODER" c:identifier="GST_NONSTREAM_AUDIO_DECODER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="NONSTREAM_AUDIO_DECODER_CAST" c:identifier="GST_NONSTREAM_AUDIO_DECODER_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="NONSTREAM_AUDIO_DECODER_CLASS" c:identifier="GST_NONSTREAM_AUDIO_DECODER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="NONSTREAM_AUDIO_DECODER_GET_CLASS" c:identifier="GST_NONSTREAM_AUDIO_DECODER_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="NONSTREAM_AUDIO_DECODER_LOCK_MUTEX" c:identifier="GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Locks the decoder mutex. + +Internally, the mutex is locked before one of the class vfuncs are +called, when position and duration queries are handled, and when +properties are set/retrieved. + +Derived classes should call lock during decoder related modifications +(for example, setting/clearing filter banks), when at the same time +audio might get decoded. An example are configuration changes that +happen when properties are set. Properties might be set from another +thread, so while the derived decoder is reconfigured, the mutex +should be locked.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">base nonstream audio codec instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="NONSTREAM_AUDIO_DECODER_SINK_NAME" value="sink" c:type="GST_NONSTREAM_AUDIO_DECODER_SINK_NAME"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="NONSTREAM_AUDIO_DECODER_SINK_PAD" c:identifier="GST_NONSTREAM_AUDIO_DECODER_SINK_PAD" introspectable="0"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">base nonstream audio codec instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="NONSTREAM_AUDIO_DECODER_SRC_NAME" value="src" c:type="GST_NONSTREAM_AUDIO_DECODER_SRC_NAME"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="NONSTREAM_AUDIO_DECODER_SRC_PAD" c:identifier="GST_NONSTREAM_AUDIO_DECODER_SRC_PAD" introspectable="0"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">base nonstream audio codec instance</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX" c:identifier="GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <class name="NonstreamAudioDecoder" c:symbol-prefix="nonstream_audio_decoder" c:type="GstNonstreamAudioDecoder" parent="Gst.Element" abstract="1" glib:type-name="GstNonstreamAudioDecoder" glib:get-type="gst_nonstream_audio_decoder_get_type" glib:type-struct="NonstreamAudioDecoderClass"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <virtual-method name="decide_allocation"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="decode"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer**"/> + </parameter> + <parameter name="num_samples" transfer-ownership="none"> + <type name="guint" c:type="guint*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="get_current_subsong"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="get_main_tags"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.TagList" c:type="GstTagList*"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="get_num_loops"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="get_num_subsongs"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="get_subsong_duration"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="get_subsong_tags"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.TagList" c:type="GstTagList*"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="get_supported_output_modes"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="load_from_buffer"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="source_data" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="initial_subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="initial_subsong_mode" transfer-ownership="none"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="initial_output_mode" transfer-ownership="none"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode*"/> + </parameter> + <parameter name="initial_num_loops" transfer-ownership="none"> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="load_from_custom"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="initial_subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="initial_subsong_mode" transfer-ownership="none"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="initial_output_mode" transfer-ownership="none"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode*"/> + </parameter> + <parameter name="initial_num_loops" transfer-ownership="none"> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="negotiate"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="propose_allocation"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="seek"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="new_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="set_current_subsong"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="set_num_loops"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="num_loops" transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="set_output_mode"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="mode" transfer-ownership="none"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode"/> + </parameter> + <parameter name="current_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="set_subsong_mode"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="mode" transfer-ownership="none"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="tell"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <method name="allocate_output_buffer" c:identifier="gst_nonstream_audio_decoder_allocate_output_buffer"> + <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, +@load_from_custom, and @decode.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Newly allocated output buffer, or NULL +if allocation failed</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Decoder instance</doc> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="size" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Size of the output buffer, in bytes</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="get_downstream_info" c:identifier="gst_nonstream_audio_decoder_get_downstream_info"> + <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, +and the downstream caps are: + +"audio/x-raw, format={S16LE,S32LE}, rate=[1,32000], channels=[1,MAX]" + +Then @format and @channels stay the same, while @sample_rate is set to 32000 Hz. +This way, the initial values the the variables pointed to by the arguments +are set to can be used as default output values. Note that if no downstream +caps can be retrieved, then this function does nothing, therefore it is +necessary to ensure that @format, @sample_rate, and @channels have valid +initial values. + +Decoder lock is not held by this function, so it can be called from within +any of the class vfuncs.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">a #GstNonstreamAudioDecoder</doc> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">#GstAudioFormat value to fill with a sample format</doc> + <type name="GstAudio.AudioFormat" c:type="GstAudioFormat*"/> + </parameter> + <parameter name="sample_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Integer to fill with a sample rate</doc> + <type name="gint" c:type="gint*"/> + </parameter> + <parameter name="num_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Integer to fill with a channel count</doc> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </method> + <method name="handle_loop" c:identifier="gst_nonstream_audio_decoder_handle_loop"> + <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 +the lock already held).</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">a #GstNonstreamAudioDecoder +@new_position New position the next loop starts with</doc> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="new_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_output_format" c:identifier="gst_nonstream_audio_decoder_set_output_format"> + <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 +are called with the lock already held).</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <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> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">a #GstNonstreamAudioDecoder</doc> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="audio_info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Valid audio info structure containing the output format</doc> + <type name="GstAudio.AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + </parameters> + </method> + <method name="set_output_format_simple" c:identifier="gst_nonstream_audio_decoder_set_output_format_simple"> + <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 +gst_nonstream_audio_decoder_set_output_format().</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <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> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">a #GstNonstreamAudioDecoder</doc> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </instance-parameter> + <parameter name="sample_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Output sample rate to use, in Hz</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="sample_format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Output sample format to use</doc> + <type name="GstAudio.AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="num_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.c">Number of output channels to use</doc> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </method> + <property name="current-subsong" writable="1" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </property> + <property name="num-loops" writable="1" transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </property> + <property name="output-mode" introspectable="0" writable="1" transfer-ownership="none"> + <type/> + </property> + <property name="subsong-mode" introspectable="0" writable="1" transfer-ownership="none"> + <type/> + </property> + <field name="element"> + <type name="Gst.Element" c:type="GstElement"/> + </field> + <field name="sinkpad"> + <type name="Gst.Pad" c:type="GstPad*"/> + </field> + <field name="srcpad"> + <type name="Gst.Pad" c:type="GstPad*"/> + </field> + <field name="upstream_size"> + <type name="gint64" c:type="gint64"/> + </field> + <field name="loaded_mode"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="input_data_adapter"> + <type name="GstBase.Adapter" c:type="GstAdapter*"/> + </field> + <field name="current_subsong"> + <type name="guint" c:type="guint"/> + </field> + <field name="subsong_mode"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </field> + <field name="subsong_duration"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </field> + <field name="output_mode"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode"/> + </field> + <field name="num_loops"> + <type name="gint" c:type="gint"/> + </field> + <field name="output_format_changed"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="output_audio_info"> + <type name="GstAudio.AudioInfo" c:type="GstAudioInfo"/> + </field> + <field name="cur_pos_in_samples"> + <type name="guint64" c:type="guint64"/> + </field> + <field name="num_decoded_samples"> + <type name="guint64" c:type="guint64"/> + </field> + <field name="cur_segment"> + <type name="Gst.Segment" c:type="GstSegment"/> + </field> + <field name="discont"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="toc"> + <type name="Gst.Toc" c:type="GstToc*"/> + </field> + <field name="allocator"> + <type name="Gst.Allocator" c:type="GstAllocator*"/> + </field> + <field name="allocation_params"> + <type name="Gst.AllocationParams" c:type="GstAllocationParams"/> + </field> + <field name="mutex"> + <type name="GLib.Mutex" c:type="GMutex"/> + </field> + </class> + <record name="NonstreamAudioDecoderClass" c:type="GstNonstreamAudioDecoderClass" glib:is-gtype-struct-for="NonstreamAudioDecoder"> + <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 +loads_from_sinkpad is TRUE.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <field name="element_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">The parent class structure</doc> + <type name="Gst.ElementClass" c:type="GstElementClass"/> + </field> + <field name="loads_from_sinkpad"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="seek"> + <callback name="seek"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="new_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="tell"> + <callback name="tell"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="load_from_buffer"> + <callback name="load_from_buffer"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="source_data" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="initial_subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="initial_subsong_mode" transfer-ownership="none"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="initial_output_mode" transfer-ownership="none"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode*"/> + </parameter> + <parameter name="initial_num_loops" transfer-ownership="none"> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="load_from_custom"> + <callback name="load_from_custom"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="initial_subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="initial_subsong_mode" transfer-ownership="none"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="initial_output_mode" transfer-ownership="none"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode*"/> + </parameter> + <parameter name="initial_num_loops" transfer-ownership="none"> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_main_tags"> + <callback name="get_main_tags"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.TagList" c:type="GstTagList*"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="set_current_subsong"> + <callback name="set_current_subsong"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_current_subsong"> + <callback name="get_current_subsong"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_num_subsongs"> + <callback name="get_num_subsongs"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_subsong_duration"> + <callback name="get_subsong_duration"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_subsong_tags"> + <callback name="get_subsong_tags"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.TagList" c:type="GstTagList*"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="subsong" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="set_subsong_mode"> + <callback name="set_subsong_mode"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="mode" transfer-ownership="none"> + <type name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"/> + </parameter> + <parameter name="initial_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="set_num_loops"> + <callback name="set_num_loops"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="num_loops" transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_num_loops"> + <callback name="get_num_loops"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="get_supported_output_modes"> + <callback name="get_supported_output_modes"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="set_output_mode"> + <callback name="set_output_mode"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="mode" transfer-ownership="none"> + <type name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode"/> + </parameter> + <parameter name="current_position" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="decode"> + <callback name="decode"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer**"/> + </parameter> + <parameter name="num_samples" transfer-ownership="none"> + <type name="guint" c:type="guint*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="negotiate"> + <callback name="negotiate"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="decide_allocation"> + <callback name="decide_allocation"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="propose_allocation"> + <callback name="propose_allocation"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="NonstreamAudioDecoder" c:type="GstNonstreamAudioDecoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="20"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <enumeration name="NonstreamAudioOutputMode" c:type="GstNonstreamAudioOutputMode"> + <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 +loop were "unrolled".</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <member name="looping" value="0" c:identifier="GST_NONSTREAM_AUDIO_OUTPUT_MODE_LOOPING"> + <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> + </member> + <member name="steady" value="1" c:identifier="GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Playback position increases steadily, even when looping</doc> + </member> + </enumeration> + <enumeration name="NonstreamAudioSubsongMode" c:type="GstNonstreamAudioSubsongMode"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">The subsong mode defines how the decoder shall handle subsongs.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h"/> + <member name="single" value="0" c:identifier="GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Only the current subsong is played</doc> + </member> + <member name="all" value="1" c:identifier="GST_NONSTREAM_AUDIO_SUBSONG_MODE_ALL"> + <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> + </member> + <member name="decoder_default" value="2" c:identifier="GST_NONSTREAM_AUDIO_SUBSONG_MODE_DECODER_DEFAULT"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstnonstreamaudiodecoder.h">Use decoder specific default behavior</doc> + </member> + </enumeration> + <function-macro name="PLANAR_AUDIO_ADAPTER" c:identifier="GST_PLANAR_AUDIO_ADAPTER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="PLANAR_AUDIO_ADAPTER_CLASS" c:identifier="GST_PLANAR_AUDIO_ADAPTER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="PLANAR_AUDIO_ADAPTER_GET_CLASS" c:identifier="GST_PLANAR_AUDIO_ADAPTER_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <class name="PlanarAudioAdapter" c:symbol-prefix="planar_audio_adapter" c:type="GstPlanarAudioAdapter" parent="GObject.Object" glib:type-name="GstPlanarAudioAdapter" glib:get-type="gst_planar_audio_adapter_get_type" glib:type-struct="PlanarAudioAdapterClass"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <constructor name="new" c:identifier="gst_planar_audio_adapter_new"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a new #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </return-value> + </constructor> + <method name="available" c:identifier="gst_planar_audio_adapter_available"> + <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 +that function returning %NULL.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">number of samples available in @adapter</doc> + <type name="gsize" c:type="gsize"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + </parameters> + </method> + <method name="clear" c:identifier="gst_planar_audio_adapter_clear"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">Removes all buffers from @adapter.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + </parameters> + </method> + <method name="configure" c:identifier="gst_planar_audio_adapter_configure"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="info" transfer-ownership="none"> + <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> + <type name="GstAudio.AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + </parameters> + </method> + <method name="distance_from_discont" c:identifier="gst_planar_audio_adapter_distance_from_discont"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <type name="guint64" c:type="guint64"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + </parameters> + </method> + <method name="dts_at_discont" c:identifier="gst_planar_audio_adapter_dts_at_discont"> + <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 +flag, or GST_CLOCK_TIME_NONE.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <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> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + </parameters> + </method> + <method name="flush" c:identifier="gst_planar_audio_adapter_flush"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="to_flush" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">the number of samples to flush</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="get_buffer" c:identifier="gst_planar_audio_adapter_get_buffer"> + <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 +usage. + +Free-function: gst_buffer_unref</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstBuffer containing the first + @nsamples of the adapter, or %NULL if @nsamples samples are not + available. gst_buffer_unref() when no longer needed.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="nsamples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">the number of samples to get</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">hint the intended use of the returned buffer</doc> + <type name="Gst.MapFlags" c:type="GstMapFlags"/> + </parameter> + </parameters> + </method> + <method name="offset_at_discont" c:identifier="gst_planar_audio_adapter_offset_at_discont"> + <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 +flag, or GST_BUFFER_OFFSET_NONE.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <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> + <type name="guint64" c:type="guint64"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + </parameters> + </method> + <method name="prev_dts" c:identifier="gst_planar_audio_adapter_prev_dts"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">The previously seen dts.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="distance" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">pointer to location for distance, or %NULL</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + </parameters> + </method> + <method name="prev_offset" c:identifier="gst_planar_audio_adapter_prev_offset"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">The previous seen offset.</doc> + <type name="guint64" c:type="guint64"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="distance" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">pointer to a location for distance, or %NULL</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + </parameters> + </method> + <method name="prev_pts" c:identifier="gst_planar_audio_adapter_prev_pts"> + <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> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">The previously seen pts.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="distance" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">pointer to location for distance, or %NULL</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + </parameters> + </method> + <method name="pts_at_discont" c:identifier="gst_planar_audio_adapter_pts_at_discont"> + <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 +flag, or GST_CLOCK_TIME_NONE.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <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> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + </parameters> + </method> + <method name="push" c:identifier="gst_planar_audio_adapter_push"> + <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 +ownership of the buffer.</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="buf" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstBuffer to queue in the adapter</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </method> + <method name="take_buffer" c:identifier="gst_planar_audio_adapter_take_buffer"> + <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 +usage. + +Free-function: gst_buffer_unref</doc> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstBuffer containing the first + @nsamples of the adapter, or %NULL if @nsamples samples are not + available. gst_buffer_unref() when no longer needed.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="adapter" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">a #GstPlanarAudioAdapter</doc> + <type name="PlanarAudioAdapter" c:type="GstPlanarAudioAdapter*"/> + </instance-parameter> + <parameter name="nsamples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">the number of samples to take</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.c">hint the intended use of the returned buffer</doc> + <type name="Gst.MapFlags" c:type="GstMapFlags"/> + </parameter> + </parameters> + </method> + </class> + <record name="PlanarAudioAdapterClass" c:type="GstPlanarAudioAdapterClass" disguised="1" glib:is-gtype-struct-for="PlanarAudioAdapter"> + <source-position filename="../subprojects/gst-plugins-bad/gst-libs/gst/audio/gstplanaraudioadapter.h"/> + </record> + </namespace> +</repository> |