mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 09:55:36 +00:00
audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder
API: gst_gst_audio_decoder_finish_frame() API: gst_gst_audio_decoder_get_audio_info() API: gst_gst_audio_decoder_get_byte_time() API: gst_gst_audio_decoder_get_delay() API: gst_gst_audio_decoder_get_latency() API: gst_gst_audio_decoder_get_max_errors() API: gst_gst_audio_decoder_get_min_latenc()y API: gst_gst_audio_decoder_get_parse_state() API: gst_gst_audio_decoder_get_plc() API: gst_gst_audio_decoder_get_plc_aware() API: gst_gst_audio_decoder_get_tolerance() API: gst_gst_audio_decoder_get_type() API: gst_gst_audio_decoder_set_byte_time() API: gst_gst_audio_decoder_set_latency() API: gst_gst_audio_decoder_set_max_errors() API: gst_gst_audio_decoder_set_min_latency() API: gst_gst_audio_decoder_set_plc() API: gst_gst_audio_decoder_set_plc_aware() API: gst_gst_audio_decoder_set_tolerance() API: gst_gst_audio_encoder_finish_frame() API: gst_gst_audio_encoder_get_audio_info() API: gst_gst_audio_encoder_get_frame_max() API: gst_gst_audio_encoder_get_frame_samples() API: gst_gst_audio_encoder_get_hard_resync() API: gst_gst_audio_encoder_get_latency() API: gst_gst_audio_encoder_get_lookahead() API: gst_gst_audio_encoder_get_mark_granule() API: gst_gst_audio_encoder_get_perfect_timestamp() API: gst_gst_audio_encoder_get_tolerance() API: gst_gst_audio_encoder_get_type() API: gst_gst_audio_encoder_proxy_getcaps() API: gst_gst_audio_encoder_set_frame_max() API: gst_gst_audio_encoder_set_frame_samples() API: gst_gst_audio_encoder_set_hard_resync() API: gst_gst_audio_encoder_set_latency() API: gst_gst_audio_encoder_set_lookahead() API: gst_gst_audio_encoder_set_mark_granule() API: gst_gst_audio_encoder_set_perfect_timestamp() API: gst_gst_audio_encoder_set_tolerance() https://bugzilla.gnome.org/show_bug.cgi?id=642690
This commit is contained in:
parent
2768ed75e0
commit
86e6343759
7 changed files with 845 additions and 856 deletions
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@ -16,14 +16,15 @@ lib_LTLIBRARIES = \
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CLEANFILES = $(BUILT_SOURCES)
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# FIXME 0.11: rename GstBaseAudioSink to GstAudioBaseSink or merge with GstAudioSink
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libgstaudio_@GST_MAJORMINOR@_la_SOURCES = \
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audio.c \
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gstringbuffer.c \
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gstaudioclock.c \
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mixerutils.c \
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multichannel.c \
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gstbaseaudiodecoder.c \
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gstbaseaudioencoder.c \
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gstaudiodecoder.c \
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gstaudioencoder.c \
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gstbaseaudiosink.c \
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gstbaseaudiosrc.c \
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gstaudiofilter.c \
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@ -38,8 +39,8 @@ libgstaudio_@GST_MAJORMINOR@include_HEADERS = \
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gstringbuffer.h \
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gstaudioclock.h \
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gstaudiofilter.h \
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gstbaseaudiodecoder.h \
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gstbaseaudioencoder.h \
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gstaudiodecoder.h \
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gstaudioencoder.h \
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gstbaseaudiosink.h \
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gstbaseaudiosrc.h \
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gstaudiosink.h \
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@ -21,11 +21,11 @@
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef _GST_BASE_AUDIO_DECODER_H_
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#define _GST_BASE_AUDIO_DECODER_H_
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#ifndef _GST_AUDIO_DECODER_H_
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#define _GST_AUDIO_DECODER_H_
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#ifndef GST_USE_UNSTABLE_API
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#warning "GstBaseAudioDecoder is unstable API and may change in future."
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#warning "GstAudioDecoder is unstable API and may change in future."
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#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
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#endif
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@ -35,70 +35,70 @@
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G_BEGIN_DECLS
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#define GST_TYPE_BASE_AUDIO_DECODER \
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(gst_base_audio_decoder_get_type())
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#define GST_BASE_AUDIO_DECODER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoder))
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#define GST_BASE_AUDIO_DECODER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
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#define GST_BASE_AUDIO_DECODER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
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#define GST_IS_BASE_AUDIO_DECODER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_DECODER))
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#define GST_IS_BASE_AUDIO_DECODER_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_DECODER))
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#define GST_TYPE_AUDIO_DECODER \
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(gst_audio_decoder_get_type())
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#define GST_AUDIO_DECODER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoder))
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#define GST_AUDIO_DECODER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass))
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#define GST_AUDIO_DECODER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass))
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#define GST_IS_AUDIO_DECODER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_DECODER))
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#define GST_IS_AUDIO_DECODER_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_DECODER))
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/**
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* GST_BASE_AUDIO_DECODER_SINK_NAME:
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* GST_AUDIO_DECODER_SINK_NAME:
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*
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* The name of the templates for the sink pad.
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*
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* Since: 0.10.36
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*/
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#define GST_BASE_AUDIO_DECODER_SINK_NAME "sink"
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#define GST_AUDIO_DECODER_SINK_NAME "sink"
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/**
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* GST_BASE_AUDIO_DECODER_SRC_NAME:
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* GST_AUDIO_DECODER_SRC_NAME:
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*
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* The name of the templates for the source pad.
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*
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* Since: 0.10.36
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*/
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#define GST_BASE_AUDIO_DECODER_SRC_NAME "src"
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#define GST_AUDIO_DECODER_SRC_NAME "src"
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/**
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* GST_BASE_AUDIO_DECODER_SRC_PAD:
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* GST_AUDIO_DECODER_SRC_PAD:
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* @obj: base audio codec instance
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*
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* Gives the pointer to the source #GstPad object of the element.
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*
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* Since: 0.10.36
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*/
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#define GST_BASE_AUDIO_DECODER_SRC_PAD(obj) (((GstBaseAudioDecoder *) (obj))->srcpad)
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#define GST_AUDIO_DECODER_SRC_PAD(obj) (((GstAudioDecoder *) (obj))->srcpad)
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/**
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* GST_BASE_AUDIO_DECODER_SINK_PAD:
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* GST_AUDIO_DECODER_SINK_PAD:
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* @obj: base audio codec instance
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*
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* Gives the pointer to the sink #GstPad object of the element.
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*
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* Since: 0.10.36
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*/
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#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
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#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
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typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
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typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
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typedef struct _GstAudioDecoder GstAudioDecoder;
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typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
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typedef struct _GstBaseAudioDecoderPrivate GstBaseAudioDecoderPrivate;
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typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate;
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/* do not use this one, use macro below */
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GstFlowReturn _gst_base_audio_decoder_error (GstBaseAudioDecoder *dec, gint weight,
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GQuark domain, gint code,
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gchar *txt, gchar *debug,
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const gchar *file, const gchar *function,
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gint line);
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GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight,
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GQuark domain, gint code,
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gchar *txt, gchar *debug,
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const gchar *file, const gchar *function,
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gint line);
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/**
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* GST_BASE_AUDIO_DECODER_ERROR:
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* GST_AUDIO_DECODER_ERROR:
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* @el: the base audio decoder element that generates the error
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* @weight: element defined weight of the error, added to error count
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* @domain: like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)
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@ -120,24 +120,24 @@ GstFlowReturn _gst_base_audio_decoder_error (GstBaseAudioDecoder *dec, gint weig
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*
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* Since: 0.10.36
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*/
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#define GST_BASE_AUDIO_DECODER_ERROR(el, w, domain, code, text, debug, ret) \
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#define GST_AUDIO_DECODER_ERROR(el, w, domain, code, text, debug, ret) \
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G_STMT_START { \
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gchar *__txt = _gst_element_error_printf text; \
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gchar *__dbg = _gst_element_error_printf debug; \
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GstBaseAudioDecoder *dec = GST_BASE_AUDIO_DECODER (el); \
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ret = _gst_base_audio_decoder_error (dec, w, GST_ ## domain ## _ERROR, \
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GstAudioDecoder *dec = GST_AUDIO_DECODER (el); \
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ret = _gst_audio_decoder_error (dec, w, GST_ ## domain ## _ERROR, \
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GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \
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GST_FUNCTION, __LINE__); \
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} G_STMT_END
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/**
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* GstBaseAudioDecoder:
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* GstAudioDecoder:
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*
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* The opaque #GstBaseAudioDecoder data structure.
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* The opaque #GstAudioDecoder data structure.
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*
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* Since: 0.10.36
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*/
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struct _GstBaseAudioDecoder
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struct _GstAudioDecoder
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{
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GstElement element;
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GstSegment segment;
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/*< private >*/
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GstBaseAudioDecoderPrivate *priv;
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GstAudioDecoderPrivate *priv;
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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/**
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* GstBaseAudioDecoderClass:
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* GstAudioDecoderClass:
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* @start: Optional.
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* Called when the element starts processing.
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* Allows opening external resources.
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@ -191,85 +191,93 @@ struct _GstBaseAudioDecoder
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*
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* Since: 0.10.36
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*/
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struct _GstBaseAudioDecoderClass
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struct _GstAudioDecoderClass
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{
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GstElementClass parent_class;
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/*< public >*/
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/* virtual methods for subclasses */
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gboolean (*start) (GstBaseAudioDecoder *dec);
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gboolean (*start) (GstAudioDecoder *dec);
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gboolean (*stop) (GstBaseAudioDecoder *dec);
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gboolean (*stop) (GstAudioDecoder *dec);
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gboolean (*set_format) (GstBaseAudioDecoder *dec,
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gboolean (*set_format) (GstAudioDecoder *dec,
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GstCaps *caps);
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GstFlowReturn (*parse) (GstBaseAudioDecoder *dec,
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GstFlowReturn (*parse) (GstAudioDecoder *dec,
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GstAdapter *adapter,
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gint *offset, gint *length);
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GstFlowReturn (*handle_frame) (GstBaseAudioDecoder *dec,
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GstFlowReturn (*handle_frame) (GstAudioDecoder *dec,
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GstBuffer *buffer);
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void (*flush) (GstBaseAudioDecoder *dec, gboolean hard);
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void (*flush) (GstAudioDecoder *dec, gboolean hard);
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GstFlowReturn (*pre_push) (GstBaseAudioDecoder *dec,
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GstFlowReturn (*pre_push) (GstAudioDecoder *dec,
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GstBuffer **buffer);
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gboolean (*event) (GstBaseAudioDecoder *dec,
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gboolean (*event) (GstAudioDecoder *dec,
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GstEvent *event);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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GstFlowReturn gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec,
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GType gst_audio_decoder_get_type (void);
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GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
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GstBuffer * buf, gint frames);
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/* context parameters */
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GstAudioInfo * gst_base_audio_decoder_get_audio_info (GstBaseAudioDecoder * dec);
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GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
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void gst_base_audio_decoder_set_plc_aware (GstBaseAudioDecoder * dec,
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gboolean plc);
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gint gst_base_audio_decoder_get_plc_aware (GstBaseAudioDecoder * dec);
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void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec,
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gboolean plc);
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void gst_base_audio_decoder_set_byte_time (GstBaseAudioDecoder * dec,
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gboolean enabled);
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gint gst_base_audio_decoder_get_byte_time (GstBaseAudioDecoder * dec);
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gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec);
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gint gst_base_audio_decoder_get_delay (GstBaseAudioDecoder * dec);
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void gst_audio_decoder_set_byte_time (GstAudioDecoder * dec,
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gboolean enabled);
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void gst_base_audio_decoder_set_max_errors (GstBaseAudioDecoder * enc,
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gint num);
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gint gst_base_audio_decoder_get_max_errors (GstBaseAudioDecoder * dec);
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gint gst_audio_decoder_get_byte_time (GstAudioDecoder * dec);
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void gst_base_audio_decoder_set_latency (GstBaseAudioDecoder * dec,
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GstClockTime min, GstClockTime max);
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void gst_base_audio_decoder_get_latency (GstBaseAudioDecoder * dec,
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GstClockTime * min, GstClockTime * max);
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gint gst_audio_decoder_get_delay (GstAudioDecoder * dec);
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void gst_base_audio_decoder_get_parse_state (GstBaseAudioDecoder * dec,
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gboolean * sync, gboolean * eos);
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void gst_audio_decoder_set_max_errors (GstAudioDecoder * enc,
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gint num);
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gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
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void gst_audio_decoder_set_latency (GstAudioDecoder * dec,
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GstClockTime min,
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GstClockTime max);
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void gst_audio_decoder_get_latency (GstAudioDecoder * dec,
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GstClockTime * min,
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GstClockTime * max);
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void gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
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gboolean * sync,
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gboolean * eos);
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/* object properties */
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void gst_base_audio_decoder_set_plc (GstBaseAudioDecoder * dec,
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gboolean enabled);
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gboolean gst_base_audio_decoder_get_plc (GstBaseAudioDecoder * dec);
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void gst_audio_decoder_set_plc (GstAudioDecoder * dec,
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gboolean enabled);
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void gst_base_audio_decoder_set_min_latency (GstBaseAudioDecoder * dec,
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gint64 num);
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gint64 gst_base_audio_decoder_get_min_latency (GstBaseAudioDecoder * dec);
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gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec);
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void gst_base_audio_decoder_set_tolerance (GstBaseAudioDecoder * dec,
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gint64 tolerance);
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void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
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gint64 num);
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gint64 gst_base_audio_decoder_get_tolerance (GstBaseAudioDecoder * dec);
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gint64 gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
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GType gst_base_audio_decoder_get_type (void);
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void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec,
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gint64 tolerance);
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gint64 gst_audio_decoder_get_tolerance (GstAudioDecoder * dec);
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G_END_DECLS
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#endif
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#endif /* _GST_AUDIO_DECODER_H_ */
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Load diff
244
gst-libs/gst/audio/gstaudioencoder.h
Normal file
244
gst-libs/gst/audio/gstaudioencoder.h
Normal file
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@ -0,0 +1,244 @@
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/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_AUDIO_ENCODER_H__
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#define __GST_AUDIO_ENCODER_H__
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#ifndef GST_USE_UNSTABLE_API
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#warning "GstAudioEncoder is unstable API and may change in future."
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#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
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#endif
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_ENCODER (gst_audio_encoder_get_type())
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#define GST_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoder))
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#define GST_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
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#define GST_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
|
||||
#define GST_IS_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ENCODER))
|
||||
#define GST_IS_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_ENCODER))
|
||||
#define GST_AUDIO_ENCODER_CAST(obj) ((GstAudioEncoder *)(obj))
|
||||
|
||||
/**
|
||||
* GST_AUDIO_ENCODER_SINK_NAME:
|
||||
*
|
||||
* the name of the templates for the sink pad
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_AUDIO_ENCODER_SINK_NAME "sink"
|
||||
/**
|
||||
* GST_AUDIO_ENCODER_SRC_NAME:
|
||||
*
|
||||
* the name of the templates for the source pad
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_AUDIO_ENCODER_SRC_NAME "src"
|
||||
|
||||
/**
|
||||
* GST_AUDIO_ENCODER_SRC_PAD:
|
||||
* @obj: base parse instance
|
||||
*
|
||||
* Gives the pointer to the source #GstPad object of the element.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad)
|
||||
|
||||
/**
|
||||
* GST_AUDIO_ENCODER_SINK_PAD:
|
||||
* @obj: base parse instance
|
||||
*
|
||||
* Gives the pointer to the sink #GstPad object of the element.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
|
||||
|
||||
/**
|
||||
* GST_AUDIO_ENCODER_SEGMENT:
|
||||
* @obj: base parse instance
|
||||
*
|
||||
* Gives the segment of the element.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_AUDIO_ENCODER_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->segment)
|
||||
|
||||
|
||||
typedef struct _GstAudioEncoder GstAudioEncoder;
|
||||
typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
|
||||
|
||||
typedef struct _GstAudioEncoderPrivate GstAudioEncoderPrivate;
|
||||
|
||||
/**
|
||||
* GstAudioEncoder:
|
||||
* @element: the parent element.
|
||||
*
|
||||
* The opaque #GstAudioEncoder data structure.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
struct _GstAudioEncoder {
|
||||
GstElement element;
|
||||
|
||||
/*< protected >*/
|
||||
/* source and sink pads */
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
|
||||
/* MT-protected (with STREAM_LOCK) */
|
||||
GstSegment segment;
|
||||
|
||||
/*< private >*/
|
||||
GstAudioEncoderPrivate *priv;
|
||||
gpointer _gst_reserved[GST_PADDING_LARGE];
|
||||
};
|
||||
|
||||
/**
|
||||
* GstAudioEncoderClass:
|
||||
* @start: Optional.
|
||||
* Called when the element starts processing.
|
||||
* Allows opening external resources.
|
||||
* @stop: Optional.
|
||||
* Called when the element stops processing.
|
||||
* Allows closing external resources.
|
||||
* @set_format: Notifies subclass of incoming data format.
|
||||
* GstAudioInfo contains the format according to provided caps.
|
||||
* @handle_frame: Provides input samples (or NULL to clear any remaining data)
|
||||
* according to directions as provided by subclass in the
|
||||
* #GstAudioEncoderContext. Input data ref management
|
||||
* is performed by base class, subclass should not care or
|
||||
* intervene.
|
||||
* @flush: Optional.
|
||||
* Instructs subclass to clear any codec caches and discard
|
||||
* any pending samples and not yet returned encoded data.
|
||||
* @event: Optional.
|
||||
* Event handler on the sink pad. This function should return
|
||||
* TRUE if the event was handled and should be discarded
|
||||
* (i.e. not unref'ed).
|
||||
* @pre_push: Optional.
|
||||
* Called just prior to pushing (encoded data) buffer downstream.
|
||||
* Subclass has full discretionary access to buffer,
|
||||
* and a not OK flow return will abort downstream pushing.
|
||||
* @getcaps: Optional.
|
||||
* Allows for a custom sink getcaps implementation (e.g.
|
||||
* for multichannel input specification). If not implemented,
|
||||
* default returns gst_audio_encoder_proxy_getcaps
|
||||
* applied to sink template caps.
|
||||
*
|
||||
* Subclasses can override any of the available virtual methods or not, as
|
||||
* needed. At minimum @set_format and @handle_frame needs to be overridden.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
struct _GstAudioEncoderClass {
|
||||
GstElementClass parent_class;
|
||||
|
||||
/*< public >*/
|
||||
/* virtual methods for subclasses */
|
||||
|
||||
gboolean (*start) (GstAudioEncoder *enc);
|
||||
|
||||
gboolean (*stop) (GstAudioEncoder *enc);
|
||||
|
||||
gboolean (*set_format) (GstAudioEncoder *enc,
|
||||
GstAudioInfo *info);
|
||||
|
||||
GstFlowReturn (*handle_frame) (GstAudioEncoder *enc,
|
||||
GstBuffer *buffer);
|
||||
|
||||
void (*flush) (GstAudioEncoder *enc);
|
||||
|
||||
GstFlowReturn (*pre_push) (GstAudioEncoder *enc,
|
||||
GstBuffer **buffer);
|
||||
|
||||
gboolean (*event) (GstAudioEncoder *enc,
|
||||
GstEvent *event);
|
||||
|
||||
GstCaps * (*getcaps) (GstAudioEncoder *enc);
|
||||
|
||||
/*< private >*/
|
||||
gpointer _gst_reserved[GST_PADDING_LARGE];
|
||||
};
|
||||
|
||||
GType gst_audio_encoder_get_type (void);
|
||||
|
||||
GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
|
||||
GstBuffer * buffer,
|
||||
gint samples);
|
||||
|
||||
GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
|
||||
GstCaps * caps);
|
||||
|
||||
|
||||
/* context parameters */
|
||||
GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
|
||||
|
||||
gint gst_audio_encoder_get_frame_samples (GstAudioEncoder * enc);
|
||||
|
||||
void gst_audio_encoder_set_frame_samples (GstAudioEncoder * enc, gint num);
|
||||
|
||||
gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
|
||||
|
||||
void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
|
||||
|
||||
gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
|
||||
|
||||
void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
|
||||
|
||||
void gst_audio_encoder_get_latency (GstAudioEncoder * enc,
|
||||
GstClockTime * min,
|
||||
GstClockTime * max);
|
||||
|
||||
void gst_audio_encoder_set_latency (GstAudioEncoder * enc,
|
||||
GstClockTime min,
|
||||
GstClockTime max);
|
||||
|
||||
/* object properties */
|
||||
|
||||
void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
|
||||
gboolean enabled);
|
||||
|
||||
gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
|
||||
|
||||
void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
|
||||
gboolean enabled);
|
||||
|
||||
gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
|
||||
|
||||
void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
|
||||
gboolean enabled);
|
||||
|
||||
gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
|
||||
|
||||
void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
|
||||
gint64 tolerance);
|
||||
|
||||
gint64 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __GST_AUDIO_ENCODER_H__ */
|
|
@ -1,235 +0,0 @@
|
|||
/* GStreamer
|
||||
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
|
||||
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
|
||||
* Contact: Stefan Kost <stefan.kost@nokia.com>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or (at your option) any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#ifndef __GST_BASE_AUDIO_ENCODER_H__
|
||||
#define __GST_BASE_AUDIO_ENCODER_H__
|
||||
|
||||
#ifndef GST_USE_UNSTABLE_API
|
||||
#warning "GstBaseAudioEncoder is unstable API and may change in future."
|
||||
#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
|
||||
#endif
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/audio.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
#define GST_TYPE_BASE_AUDIO_ENCODER (gst_base_audio_encoder_get_type())
|
||||
#define GST_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoder))
|
||||
#define GST_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
|
||||
#define GST_BASE_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
|
||||
#define GST_IS_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_ENCODER))
|
||||
#define GST_IS_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_ENCODER))
|
||||
#define GST_BASE_AUDIO_ENCODER_CAST(obj) ((GstBaseAudioEncoder *)(obj))
|
||||
|
||||
/**
|
||||
* GST_BASE_AUDIO_ENCODER_SINK_NAME:
|
||||
*
|
||||
* the name of the templates for the sink pad
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_BASE_AUDIO_ENCODER_SINK_NAME "sink"
|
||||
/**
|
||||
* GST_BASE_AUDIO_ENCODER_SRC_NAME:
|
||||
*
|
||||
* the name of the templates for the source pad
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_BASE_AUDIO_ENCODER_SRC_NAME "src"
|
||||
|
||||
/**
|
||||
* GST_BASE_AUDIO_ENCODER_SRC_PAD:
|
||||
* @obj: base parse instance
|
||||
*
|
||||
* Gives the pointer to the source #GstPad object of the element.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_BASE_AUDIO_ENCODER_SRC_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->srcpad)
|
||||
|
||||
/**
|
||||
* GST_BASE_AUDIO_ENCODER_SINK_PAD:
|
||||
* @obj: base parse instance
|
||||
*
|
||||
* Gives the pointer to the sink #GstPad object of the element.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_BASE_AUDIO_ENCODER_SINK_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->sinkpad)
|
||||
|
||||
/**
|
||||
* GST_BASE_AUDIO_ENCODER_SEGMENT:
|
||||
* @obj: base parse instance
|
||||
*
|
||||
* Gives the segment of the element.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
#define GST_BASE_AUDIO_ENCODER_SEGMENT(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->segment)
|
||||
|
||||
|
||||
typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder;
|
||||
typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
|
||||
|
||||
typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate;
|
||||
|
||||
/**
|
||||
* GstBaseAudioEncoder:
|
||||
* @element: the parent element.
|
||||
*
|
||||
* The opaque #GstBaseAudioEncoder data structure.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
struct _GstBaseAudioEncoder {
|
||||
GstElement element;
|
||||
|
||||
/*< protected >*/
|
||||
/* source and sink pads */
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
|
||||
/* MT-protected (with STREAM_LOCK) */
|
||||
GstSegment segment;
|
||||
|
||||
/*< private >*/
|
||||
GstBaseAudioEncoderPrivate *priv;
|
||||
gpointer _gst_reserved[GST_PADDING_LARGE];
|
||||
};
|
||||
|
||||
/**
|
||||
* GstBaseAudioEncoderClass:
|
||||
* @start: Optional.
|
||||
* Called when the element starts processing.
|
||||
* Allows opening external resources.
|
||||
* @stop: Optional.
|
||||
* Called when the element stops processing.
|
||||
* Allows closing external resources.
|
||||
* @set_format: Notifies subclass of incoming data format.
|
||||
* GstAudioInfo contains the format according to provided caps.
|
||||
* @handle_frame: Provides input samples (or NULL to clear any remaining data)
|
||||
* according to directions as provided by subclass in the
|
||||
* #GstBaseAudioEncoderContext. Input data ref management
|
||||
* is performed by base class, subclass should not care or
|
||||
* intervene.
|
||||
* @flush: Optional.
|
||||
* Instructs subclass to clear any codec caches and discard
|
||||
* any pending samples and not yet returned encoded data.
|
||||
* @event: Optional.
|
||||
* Event handler on the sink pad. This function should return
|
||||
* TRUE if the event was handled and should be discarded
|
||||
* (i.e. not unref'ed).
|
||||
* @pre_push: Optional.
|
||||
* Called just prior to pushing (encoded data) buffer downstream.
|
||||
* Subclass has full discretionary access to buffer,
|
||||
* and a not OK flow return will abort downstream pushing.
|
||||
* @getcaps: Optional.
|
||||
* Allows for a custom sink getcaps implementation (e.g.
|
||||
* for multichannel input specification). If not implemented,
|
||||
* default returns gst_base_audio_encoder_proxy_getcaps
|
||||
* applied to sink template caps.
|
||||
*
|
||||
* Subclasses can override any of the available virtual methods or not, as
|
||||
* needed. At minimum @set_format and @handle_frame needs to be overridden.
|
||||
*
|
||||
* Since: 0.10.36
|
||||
*/
|
||||
struct _GstBaseAudioEncoderClass {
|
||||
GstElementClass parent_class;
|
||||
|
||||
/*< public >*/
|
||||
/* virtual methods for subclasses */
|
||||
|
||||
gboolean (*start) (GstBaseAudioEncoder *enc);
|
||||
|
||||
gboolean (*stop) (GstBaseAudioEncoder *enc);
|
||||
|
||||
gboolean (*set_format) (GstBaseAudioEncoder *enc,
|
||||
GstAudioInfo *info);
|
||||
|
||||
GstFlowReturn (*handle_frame) (GstBaseAudioEncoder *enc,
|
||||
GstBuffer *buffer);
|
||||
|
||||
void (*flush) (GstBaseAudioEncoder *enc);
|
||||
|
||||
GstFlowReturn (*pre_push) (GstBaseAudioEncoder *enc,
|
||||
GstBuffer **buffer);
|
||||
|
||||
gboolean (*event) (GstBaseAudioEncoder *enc,
|
||||
GstEvent *event);
|
||||
|
||||
GstCaps * (*getcaps) (GstBaseAudioEncoder *enc);
|
||||
|
||||
/*< private >*/
|
||||
gpointer _gst_reserved[GST_PADDING_LARGE];
|
||||
};
|
||||
|
||||
GType gst_base_audio_encoder_get_type (void);
|
||||
|
||||
GstFlowReturn gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc,
|
||||
GstBuffer *buffer, gint samples);
|
||||
|
||||
GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
|
||||
GstCaps * caps);
|
||||
|
||||
|
||||
/* context parameters */
|
||||
GstAudioInfo * gst_base_audio_encoder_get_audio_info (GstBaseAudioEncoder * enc);
|
||||
|
||||
gint gst_base_audio_encoder_get_frame_samples (GstBaseAudioEncoder * enc);
|
||||
void gst_base_audio_encoder_set_frame_samples (GstBaseAudioEncoder * enc,
|
||||
gint num);
|
||||
|
||||
gint gst_base_audio_encoder_get_frame_max (GstBaseAudioEncoder * enc);
|
||||
void gst_base_audio_encoder_set_frame_max (GstBaseAudioEncoder * enc,
|
||||
gint num);
|
||||
|
||||
gint gst_base_audio_encoder_get_lookahead (GstBaseAudioEncoder * enc);
|
||||
void gst_base_audio_encoder_set_lookahead (GstBaseAudioEncoder * enc,
|
||||
gint num);
|
||||
|
||||
void gst_base_audio_encoder_get_latency (GstBaseAudioEncoder * enc,
|
||||
GstClockTime * min, GstClockTime * max);
|
||||
void gst_base_audio_encoder_set_latency (GstBaseAudioEncoder * enc,
|
||||
GstClockTime min, GstClockTime max);
|
||||
|
||||
/* object properties */
|
||||
void gst_base_audio_encoder_set_mark_granule (GstBaseAudioEncoder * enc,
|
||||
gboolean enabled);
|
||||
gboolean gst_base_audio_encoder_get_mark_granule (GstBaseAudioEncoder * enc);
|
||||
|
||||
void gst_base_audio_encoder_set_perfect_timestamp (GstBaseAudioEncoder * enc,
|
||||
gboolean enabled);
|
||||
gboolean gst_base_audio_encoder_get_perfect_timestamp (GstBaseAudioEncoder * enc);
|
||||
|
||||
void gst_base_audio_encoder_set_hard_resync (GstBaseAudioEncoder * enc,
|
||||
gboolean enabled);
|
||||
gboolean gst_base_audio_encoder_get_hard_resync (GstBaseAudioEncoder * enc);
|
||||
|
||||
void gst_base_audio_encoder_set_tolerance (GstBaseAudioEncoder * enc,
|
||||
gint64 tolerance);
|
||||
gint64 gst_base_audio_encoder_get_tolerance (GstBaseAudioEncoder * enc);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __GST_BASE_AUDIO_ENCODER_H__ */
|
|
@ -1,5 +1,5 @@
|
|||
EXPORTS
|
||||
_gst_base_audio_decoder_error
|
||||
_gst_audio_decoder_error
|
||||
gst_audio_buffer_clip
|
||||
gst_audio_channel_position_get_type
|
||||
gst_audio_check_channel_positions
|
||||
|
@ -10,8 +10,47 @@ EXPORTS
|
|||
gst_audio_clock_new
|
||||
gst_audio_clock_new_full
|
||||
gst_audio_clock_reset
|
||||
gst_audio_decoder_finish_frame
|
||||
gst_audio_decoder_get_audio_info
|
||||
gst_audio_decoder_get_byte_time
|
||||
gst_audio_decoder_get_delay
|
||||
gst_audio_decoder_get_latency
|
||||
gst_audio_decoder_get_max_errors
|
||||
gst_audio_decoder_get_min_latency
|
||||
gst_audio_decoder_get_parse_state
|
||||
gst_audio_decoder_get_plc
|
||||
gst_audio_decoder_get_plc_aware
|
||||
gst_audio_decoder_get_tolerance
|
||||
gst_audio_decoder_get_type
|
||||
gst_audio_decoder_set_byte_time
|
||||
gst_audio_decoder_set_latency
|
||||
gst_audio_decoder_set_max_errors
|
||||
gst_audio_decoder_set_min_latency
|
||||
gst_audio_decoder_set_plc
|
||||
gst_audio_decoder_set_plc_aware
|
||||
gst_audio_decoder_set_tolerance
|
||||
gst_audio_default_registry_mixer_filter
|
||||
gst_audio_duration_from_pad_buffer
|
||||
gst_audio_encoder_finish_frame
|
||||
gst_audio_encoder_get_audio_info
|
||||
gst_audio_encoder_get_frame_max
|
||||
gst_audio_encoder_get_frame_samples
|
||||
gst_audio_encoder_get_hard_resync
|
||||
gst_audio_encoder_get_latency
|
||||
gst_audio_encoder_get_lookahead
|
||||
gst_audio_encoder_get_mark_granule
|
||||
gst_audio_encoder_get_perfect_timestamp
|
||||
gst_audio_encoder_get_tolerance
|
||||
gst_audio_encoder_get_type
|
||||
gst_audio_encoder_proxy_getcaps
|
||||
gst_audio_encoder_set_frame_max
|
||||
gst_audio_encoder_set_frame_samples
|
||||
gst_audio_encoder_set_hard_resync
|
||||
gst_audio_encoder_set_latency
|
||||
gst_audio_encoder_set_lookahead
|
||||
gst_audio_encoder_set_mark_granule
|
||||
gst_audio_encoder_set_perfect_timestamp
|
||||
gst_audio_encoder_set_tolerance
|
||||
gst_audio_filter_class_add_pad_templates
|
||||
gst_audio_filter_get_type
|
||||
gst_audio_fixate_channel_positions
|
||||
|
@ -34,45 +73,6 @@ EXPORTS
|
|||
gst_audio_sink_get_type
|
||||
gst_audio_src_get_type
|
||||
gst_audio_structure_set_int
|
||||
gst_base_audio_decoder_finish_frame
|
||||
gst_base_audio_decoder_get_audio_info
|
||||
gst_base_audio_decoder_get_byte_time
|
||||
gst_base_audio_decoder_get_delay
|
||||
gst_base_audio_decoder_get_latency
|
||||
gst_base_audio_decoder_get_max_errors
|
||||
gst_base_audio_decoder_get_min_latency
|
||||
gst_base_audio_decoder_get_parse_state
|
||||
gst_base_audio_decoder_get_plc
|
||||
gst_base_audio_decoder_get_plc_aware
|
||||
gst_base_audio_decoder_get_tolerance
|
||||
gst_base_audio_decoder_get_type
|
||||
gst_base_audio_decoder_set_byte_time
|
||||
gst_base_audio_decoder_set_latency
|
||||
gst_base_audio_decoder_set_max_errors
|
||||
gst_base_audio_decoder_set_min_latency
|
||||
gst_base_audio_decoder_set_plc
|
||||
gst_base_audio_decoder_set_plc_aware
|
||||
gst_base_audio_decoder_set_tolerance
|
||||
gst_base_audio_encoder_finish_frame
|
||||
gst_base_audio_encoder_get_audio_info
|
||||
gst_base_audio_encoder_get_frame_max
|
||||
gst_base_audio_encoder_get_frame_samples
|
||||
gst_base_audio_encoder_get_hard_resync
|
||||
gst_base_audio_encoder_get_latency
|
||||
gst_base_audio_encoder_get_lookahead
|
||||
gst_base_audio_encoder_get_mark_granule
|
||||
gst_base_audio_encoder_get_perfect_timestamp
|
||||
gst_base_audio_encoder_get_tolerance
|
||||
gst_base_audio_encoder_get_type
|
||||
gst_base_audio_encoder_proxy_getcaps
|
||||
gst_base_audio_encoder_set_frame_max
|
||||
gst_base_audio_encoder_set_frame_samples
|
||||
gst_base_audio_encoder_set_hard_resync
|
||||
gst_base_audio_encoder_set_latency
|
||||
gst_base_audio_encoder_set_lookahead
|
||||
gst_base_audio_encoder_set_mark_granule
|
||||
gst_base_audio_encoder_set_perfect_timestamp
|
||||
gst_base_audio_encoder_set_tolerance
|
||||
gst_base_audio_sink_create_ringbuffer
|
||||
gst_base_audio_sink_get_drift_tolerance
|
||||
gst_base_audio_sink_get_provide_clock
|
||||
|
|
Loading…
Reference in a new issue