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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-26 17:18:15 +00:00
rename baseaudio* -> audiobase*
This commit is contained in:
parent
ee7072fe7e
commit
a3416bc11f
11 changed files with 52 additions and 52 deletions
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@ -286,8 +286,8 @@ GST_AUDIO_SRC_GET_CLASS
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</SECTION>
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<SECTION>
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<FILE>gstbaseaudiosink</FILE>
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<INCLUDE>gst/audio/gstbaseaudiosink.h</INCLUDE>
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<FILE>gstaudiobasesink</FILE>
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<INCLUDE>gst/audio/gstaudiobasesink.h</INCLUDE>
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GstAudioBaseSink
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GstAudioBaseSinkClass
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GstAudioBaseSinkSlaveMethod
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@ -315,8 +315,8 @@ GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD
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</SECTION>
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<SECTION>
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<FILE>gstbaseaudiosrc</FILE>
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<INCLUDE>gst/audio/gstbaseaudiosrc.h</INCLUDE>
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<FILE>gstaudiobasesrc</FILE>
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<INCLUDE>gst/audio/gstaudiobasesrc.h</INCLUDE>
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GstAudioBaseSrc
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GstAudioBaseSrcClass
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GstAudioBaseSrcSlaveMethod
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@ -13,9 +13,9 @@ gst_audio_filter_get_type
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gst_audio_sink_get_type
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#include <gst/audio/gstaudiosrc.h>
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gst_audio_src_get_type
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#include <gst/audio/gstbaseaudiosink.h>
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#include <gst/audio/gstaudiobasesink.h>
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gst_audio_base_sink_get_type
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#include <gst/audio/gstbaseaudiosrc.h>
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#include <gst/audio/gstaudiobasesrc.h>
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gst_audio_base_src_get_type
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#include <gst/audio/gstaudioringbuffer.h>
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gst_audio_ring_buffer_get_type
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@ -25,8 +25,8 @@ libgstaudio_@GST_MAJORMINOR@_la_SOURCES = \
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multichannel.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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gstaudiobasesink.c \
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gstaudiobasesrc.c \
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gstaudiofilter.c \
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gstaudiosink.c \
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gstaudiosrc.c \
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@ -41,8 +41,8 @@ libgstaudio_@GST_MAJORMINOR@include_HEADERS = \
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gstaudiofilter.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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gstaudiobasesink.h \
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gstaudiobasesrc.h \
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gstaudiosink.h \
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gstaudiosrc.h \
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mixerutils.h \
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@ -2,7 +2,7 @@
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbaseaudiosink.c:
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* gstaudiobasesink.c:
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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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@ -21,7 +21,7 @@
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*/
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/**
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* SECTION:gstbaseaudiosink
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* SECTION:gstaudiobasesink
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* @short_description: Base class for audio sinks
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* @see_also: #GstAudioSink, #GstAudioRingBuffer.
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*
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@ -34,7 +34,7 @@
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#include <string.h>
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#include "gstbaseaudiosink.h"
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#include "gstaudiobasesink.h"
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GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug);
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#define GST_CAT_DEFAULT gst_audio_base_sink_debug
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@ -139,7 +139,7 @@ gst_audio_base_sink_slave_method_get_type (void)
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#define _do_init \
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GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
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GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "audiobasesink", 0, "audiobasesink element");
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#define gst_audio_base_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSink, gst_audio_base_sink,
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GST_TYPE_BASE_SINK, _do_init);
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@ -297,25 +297,25 @@ gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
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}
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static void
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gst_audio_base_sink_init (GstAudioBaseSink * baseaudiosink)
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gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
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{
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GstBaseSink *basesink;
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baseaudiosink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (baseaudiosink);
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audiobasesink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (audiobasesink);
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baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
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baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
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baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
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baseaudiosink->priv->slave_method = DEFAULT_SLAVE_METHOD;
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baseaudiosink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
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baseaudiosink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
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baseaudiosink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
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audiobasesink->buffer_time = DEFAULT_BUFFER_TIME;
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audiobasesink->latency_time = DEFAULT_LATENCY_TIME;
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audiobasesink->provide_clock = DEFAULT_PROVIDE_CLOCK;
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audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD;
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audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
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audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
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audiobasesink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
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baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
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(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, baseaudiosink,
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audiobasesink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
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(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, audiobasesink,
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NULL);
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basesink = GST_BASE_SINK_CAST (baseaudiosink);
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basesink = GST_BASE_SINK_CAST (audiobasesink);
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basesink->can_activate_push = TRUE;
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basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
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@ -2,7 +2,7 @@
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbaseaudiosink.h:
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* gstaudiobasesink.h:
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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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@ -2,7 +2,7 @@
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbaseaudiosrc.c:
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* gstaudiobasesrc.c:
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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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@ -21,7 +21,7 @@
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*/
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/**
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* SECTION:gstbaseaudiosrc
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* SECTION:gstaudiobasesrc
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* @short_description: Base class for audio sources
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* @see_also: #GstAudioSrc, #GstAudioRingBuffer.
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*
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@ -38,7 +38,7 @@
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#include <string.h>
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#include "gstbaseaudiosrc.h"
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#include "gstaudiobasesrc.h"
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#include "gst/gst-i18n-plugin.h"
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@ -108,8 +108,8 @@ enum
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static void
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_do_init (GType type)
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{
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GST_DEBUG_CATEGORY_INIT (gst_audio_base_src_debug, "baseaudiosrc", 0,
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"baseaudiosrc element");
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GST_DEBUG_CATEGORY_INIT (gst_audio_base_src_debug, "audiobasesrc", 0,
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"audiobasesrc element");
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#ifdef ENABLE_NLS
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GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
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@ -234,26 +234,26 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
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}
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static void
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gst_audio_base_src_init (GstAudioBaseSrc * baseaudiosrc)
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gst_audio_base_src_init (GstAudioBaseSrc * audiobasesrc)
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{
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baseaudiosrc->priv = GST_AUDIO_BASE_SRC_GET_PRIVATE (baseaudiosrc);
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audiobasesrc->priv = GST_AUDIO_BASE_SRC_GET_PRIVATE (audiobasesrc);
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baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
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baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
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baseaudiosrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK;
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baseaudiosrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
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audiobasesrc->buffer_time = DEFAULT_BUFFER_TIME;
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audiobasesrc->latency_time = DEFAULT_LATENCY_TIME;
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audiobasesrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK;
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audiobasesrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
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/* reset blocksize we use latency time to calculate a more useful
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* value based on negotiated format. */
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GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
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GST_BASE_SRC (audiobasesrc)->blocksize = 0;
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baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
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(GstAudioClockGetTimeFunc) gst_audio_base_src_get_time, baseaudiosrc,
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audiobasesrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
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(GstAudioClockGetTimeFunc) gst_audio_base_src_get_time, audiobasesrc,
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NULL);
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/* we are always a live source */
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gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
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gst_base_src_set_live (GST_BASE_SRC (audiobasesrc), TRUE);
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/* we operate in time */
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gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
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gst_base_src_set_format (GST_BASE_SRC (audiobasesrc), GST_FORMAT_TIME);
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}
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static void
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@ -2,7 +2,7 @@
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbaseaudiosrc.h:
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* gstaudiobasesrc.h:
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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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@ -600,11 +600,11 @@ static GstAudioRingBuffer *gst_audio_sink_create_ringbuffer (GstAudioBaseSink *
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static void
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gst_audio_sink_class_init (GstAudioSinkClass * klass)
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{
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GstAudioBaseSinkClass *gstbaseaudiosink_class;
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GstAudioBaseSinkClass *gstaudiobasesink_class;
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gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
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gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
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gstbaseaudiosink_class->create_ringbuffer =
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gstaudiobasesink_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
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g_type_class_ref (GST_TYPE_AUDIO_SINK_RING_BUFFER);
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@ -24,7 +24,7 @@
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#define __GST_AUDIO_SINK_H__
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#include <gst/gst.h>
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#include <gst/audio/gstbaseaudiosink.h>
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#include <gst/audio/gstaudiobasesink.h>
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G_BEGIN_DECLS
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@ -513,11 +513,11 @@ static GstAudioRingBuffer *gst_audio_src_create_ringbuffer (GstAudioBaseSrc *
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static void
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gst_audio_src_class_init (GstAudioSrcClass * klass)
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{
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GstAudioBaseSrcClass *gstbaseaudiosrc_class;
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GstAudioBaseSrcClass *gstaudiobasesrc_class;
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gstbaseaudiosrc_class = (GstAudioBaseSrcClass *) klass;
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gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
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gstbaseaudiosrc_class->create_ringbuffer =
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gstaudiobasesrc_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_audio_src_create_ringbuffer);
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g_type_class_ref (GST_TYPE_AUDIO_SRC_RING_BUFFER);
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@ -24,7 +24,7 @@
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#define __GST_AUDIO_SRC_H__
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#include <gst/gst.h>
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#include <gst/audio/gstbaseaudiosrc.h>
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#include <gst/audio/gstaudiobasesrc.h>
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G_BEGIN_DECLS
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