audio: GST_EXPORT -> GST_AUDIO_API

We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.

26 files changed, 257 insertions, 221 deletions

diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am
index 358adbc32..2922245ca 100644
--- a/gst-libs/gst/audio/Makefile.am
+++ b/gst-libs/gst/audio/Makefile.am
@@ -19,7 +19,8 @@ glib_enum_headers= 		\
 glib_enum_define = GST_AUDIO
 glib_gen_prefix = gst_audio
 glib_gen_basename = audio
-glib_gen_decl_banner=GST_EXPORT
+glib_gen_decl_banner=GST_AUDIO_API
+glib_gen_decl_include=\#include <gst/audio/audio-prelude.h>
 
 built_sources = audio-enumtypes.c
 built_headers = audio-enumtypes.h
@@ -63,6 +64,7 @@ nodist_libgstaudio_@GST_API_VERSION@_la_SOURCES = $(BUILT_SOURCES)
 libgstaudio_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/audio
 libgstaudio_@GST_API_VERSION@include_HEADERS = \
 	audio.h \
+	audio-prelude.h	\
 	audio-format.h \
 	audio-channels.h \
 	audio-channel-mixer.h \
diff --git a/gst-libs/gst/audio/audio-channel-mixer.h b/gst-libs/gst/audio/audio-channel-mixer.h
index 8c2b8365a..35c2f6d83 100644
--- a/gst-libs/gst/audio/audio-channel-mixer.h
+++ b/gst-libs/gst/audio/audio-channel-mixer.h
@@ -46,7 +46,7 @@ typedef enum {
   GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT    = (1 << 3)
 } GstAudioChannelMixerFlags;
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioChannelMixer * gst_audio_channel_mixer_new   (GstAudioChannelMixerFlags flags,
                                                       GstAudioFormat format,
                                                       gint in_channels,
@@ -54,28 +54,28 @@ GstAudioChannelMixer * gst_audio_channel_mixer_new   (GstAudioChannelMixerFlags
                                                       gint out_channels,
                                                       GstAudioChannelPosition *out_position);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioChannelMixer * gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags,
                                                                 GstAudioFormat format,
                                                                 gint in_channels,
                                                                 gint out_channels,
                                                                 gfloat **matrix);
 
-GST_EXPORT
+GST_AUDIO_API
 void                   gst_audio_channel_mixer_free  (GstAudioChannelMixer *mix);
 
 /*
  * Checks for passthrough (= identity matrix).
  */
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_channel_mixer_is_passthrough  (GstAudioChannelMixer *mix);
 
 /*
  * Do actual mixing.
  */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_channel_mixer_samples   (GstAudioChannelMixer * mix,
                                                    const gpointer         in[],
                                                    gpointer               out[],
diff --git a/gst-libs/gst/audio/audio-channels.h b/gst-libs/gst/audio/audio-channels.h
index 5836633a6..18b6f88c2 100644
--- a/gst-libs/gst/audio/audio-channels.h
+++ b/gst-libs/gst/audio/audio-channels.h
@@ -131,47 +131,47 @@ typedef enum {
 
 #define GST_AUDIO_CHANNEL_POSITION_MASK(pos) (G_GUINT64_CONSTANT(1)<< GST_AUDIO_CHANNEL_POSITION_ ## pos)
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_buffer_reorder_channels (GstBuffer * buffer,
                                                   GstAudioFormat format,
                                                   gint channels,
                                                   const GstAudioChannelPosition * from,
                                                   const GstAudioChannelPosition * to);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_reorder_channels        (gpointer data, gsize size,
                                                   GstAudioFormat format,
                                                   gint channels,
                                                   const GstAudioChannelPosition * from,
                                                   const GstAudioChannelPosition * to);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_channel_positions_to_valid_order (GstAudioChannelPosition *position,
                                                            gint channels);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_check_valid_channel_positions (const GstAudioChannelPosition *position,
                                                         gint channels, gboolean force_order);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_channel_positions_to_mask  (const GstAudioChannelPosition *position,
                                                      gint channels, gboolean force_order,
                                                      guint64 *channel_mask);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_channel_positions_from_mask (gint channels, guint64 channel_mask,
                                                       GstAudioChannelPosition * position);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_get_channel_reorder_map (gint channels,
                                                   const GstAudioChannelPosition * from,
                                                   const GstAudioChannelPosition * to,
                                                   gint *reorder_map);
 
-GST_EXPORT
+GST_AUDIO_API
 guint64        gst_audio_channel_get_fallback_mask (gint channels);
 
-GST_EXPORT
+GST_AUDIO_API
 gchar*         gst_audio_channel_positions_to_string (const GstAudioChannelPosition * position,
                                                       gint channels);
 
diff --git a/gst-libs/gst/audio/audio-converter.h b/gst-libs/gst/audio/audio-converter.h
index e04bcad06..9e858980d 100644
--- a/gst-libs/gst/audio/audio-converter.h
+++ b/gst-libs/gst/audio/audio-converter.h
@@ -115,51 +115,51 @@ typedef enum {
   GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE   = (1 << 1)
 } GstAudioConverterFlags;
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioConverter *  gst_audio_converter_new             (GstAudioConverterFlags flags,
                                                           GstAudioInfo *in_info,
                                                           GstAudioInfo *out_info,
                                                           GstStructure *config);
 
-GST_EXPORT
+GST_AUDIO_API
 GType                gst_audio_converter_get_type        (void);
 
-GST_EXPORT
+GST_AUDIO_API
 void                 gst_audio_converter_free            (GstAudioConverter * convert);
 
-GST_EXPORT
+GST_AUDIO_API
 void                 gst_audio_converter_reset           (GstAudioConverter * convert);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean             gst_audio_converter_update_config   (GstAudioConverter * convert,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *config);
 
-GST_EXPORT
+GST_AUDIO_API
 const GstStructure * gst_audio_converter_get_config      (GstAudioConverter * convert,
                                                           gint *in_rate, gint *out_rate);
 
-GST_EXPORT
+GST_AUDIO_API
 gsize                gst_audio_converter_get_out_frames  (GstAudioConverter *convert,
                                                           gsize in_frames);
 
-GST_EXPORT
+GST_AUDIO_API
 gsize                gst_audio_converter_get_in_frames   (GstAudioConverter *convert,
                                                           gsize out_frames);
 
-GST_EXPORT
+GST_AUDIO_API
 gsize                gst_audio_converter_get_max_latency (GstAudioConverter *convert);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean             gst_audio_converter_samples         (GstAudioConverter * convert,
                                                           GstAudioConverterFlags flags,
                                                           gpointer in[], gsize in_frames,
                                                           gpointer out[], gsize out_frames);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean             gst_audio_converter_supports_inplace (GstAudioConverter *convert);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean             gst_audio_converter_convert          (GstAudioConverter * convert,
                                                            GstAudioConverterFlags flags,
                                                            gpointer in, gsize in_size,
diff --git a/gst-libs/gst/audio/audio-format.h b/gst-libs/gst/audio/audio-format.h
index b6aeccdca..4f830c4a8 100644
--- a/gst-libs/gst/audio/audio-format.h
+++ b/gst-libs/gst/audio/audio-format.h
@@ -250,7 +250,7 @@ struct _GstAudioFormatInfo {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_format_info_get_type (void);
 
 #define GST_AUDIO_FORMAT_INFO_FORMAT(info)           ((info)->format)
@@ -268,21 +268,21 @@ GType gst_audio_format_info_get_type (void);
 #define GST_AUDIO_FORMAT_INFO_DEPTH(info)            ((info)->depth)
 
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioFormat gst_audio_format_build_integer    (gboolean sign, gint endianness,
                                                   gint width, gint depth) G_GNUC_CONST;
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioFormat gst_audio_format_from_string      (const gchar *format) G_GNUC_CONST;
 
-GST_EXPORT
+GST_AUDIO_API
 const gchar *  gst_audio_format_to_string        (GstAudioFormat format) G_GNUC_CONST;
 
-GST_EXPORT
+GST_AUDIO_API
 const GstAudioFormatInfo *
                gst_audio_format_get_info         (GstAudioFormat format) G_GNUC_CONST;
 
-GST_EXPORT
+GST_AUDIO_API
 void           gst_audio_format_fill_silence     (const GstAudioFormatInfo *info,
                                                   gpointer dest, gsize length);
 
diff --git a/gst-libs/gst/audio/audio-info.h b/gst-libs/gst/audio/audio-info.h
index 2541ca77d..6d5fcd33e 100644
--- a/gst-libs/gst/audio/audio-info.h
+++ b/gst-libs/gst/audio/audio-info.h
@@ -85,7 +85,7 @@ struct _GstAudioInfo {
 };
 
 #define GST_TYPE_AUDIO_INFO                  (gst_audio_info_get_type ())
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_info_get_type                (void);
 
 #define GST_AUDIO_INFO_IS_VALID(i)           ((i)->finfo != NULL && (i)->rate > 0 && (i)->channels > 0 && (i)->bpf > 0)
@@ -113,35 +113,35 @@ GType gst_audio_info_get_type                (void);
 #define GST_AUDIO_INFO_BPF(info)             ((info)->bpf)
 #define GST_AUDIO_INFO_POSITION(info,c)      ((info)->position[c])
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioInfo * gst_audio_info_new         (void);
 
-GST_EXPORT
+GST_AUDIO_API
 void           gst_audio_info_init        (GstAudioInfo *info);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioInfo * gst_audio_info_copy        (const GstAudioInfo *info);
 
-GST_EXPORT
+GST_AUDIO_API
 void           gst_audio_info_free        (GstAudioInfo *info);
 
-GST_EXPORT
+GST_AUDIO_API
 void           gst_audio_info_set_format  (GstAudioInfo *info, GstAudioFormat format,
                                            gint rate, gint channels,
                                            const GstAudioChannelPosition *position);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_info_from_caps   (GstAudioInfo *info, const GstCaps *caps);
 
-GST_EXPORT
+GST_AUDIO_API
 GstCaps *      gst_audio_info_to_caps     (const GstAudioInfo *info);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_info_convert     (const GstAudioInfo * info,
                                            GstFormat src_fmt, gint64 src_val,
                                            GstFormat dest_fmt, gint64 * dest_val);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean       gst_audio_info_is_equal    (const GstAudioInfo *info,
                                            const GstAudioInfo *other);
 
diff --git a/gst-libs/gst/audio/audio-prelude.h b/gst-libs/gst/audio/audio-prelude.h
new file mode 100644
index 000000000..300fb1e89
--- /dev/null
+++ b/gst-libs/gst/audio/audio-prelude.h
@@ -0,0 +1,31 @@
+/* GStreamer Audio Library
+ * Copyright (C) 2018 GStreamer developers
+ *
+ * audio-prelude.h: prelude include header for gst-audio library
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_AUDIO_PRELUDE_H__
+#define __GST_AUDIO_PRELUDE_H__
+
+#include <gst/gst.h>
+
+#ifndef GST_AUDIO_API
+#define GST_AUDIO_API GST_EXPORT
+#endif
+
+#endif /* __GST_AUDIO_PRELUDE_H__ */
diff --git a/gst-libs/gst/audio/audio-quantize.h b/gst-libs/gst/audio/audio-quantize.h
index 8a0bd7926..2944b4b91 100644
--- a/gst-libs/gst/audio/audio-quantize.h
+++ b/gst-libs/gst/audio/audio-quantize.h
@@ -81,7 +81,7 @@ typedef enum
 
 typedef struct _GstAudioQuantize GstAudioQuantize;
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioQuantize *  gst_audio_quantize_new      (GstAudioDitherMethod dither,
                                                  GstAudioNoiseShapingMethod ns,
                                                  GstAudioQuantizeFlags flags,
@@ -89,13 +89,13 @@ GstAudioQuantize *  gst_audio_quantize_new      (GstAudioDitherMethod dither,
                                                  guint channels,
                                                  guint quantizer);
 
-GST_EXPORT
+GST_AUDIO_API
 void                gst_audio_quantize_free     (GstAudioQuantize * quant);
 
-GST_EXPORT
+GST_AUDIO_API
 void                gst_audio_quantize_reset    (GstAudioQuantize * quant);
 
-GST_EXPORT
+GST_AUDIO_API
 void                gst_audio_quantize_samples  (GstAudioQuantize * quant,
                                                  const gpointer in[],
                                                  gpointer out[], guint samples);
diff --git a/gst-libs/gst/audio/audio-resampler.h b/gst-libs/gst/audio/audio-resampler.h
index cecda5795..1f3045d54 100644
--- a/gst-libs/gst/audio/audio-resampler.h
+++ b/gst-libs/gst/audio/audio-resampler.h
@@ -201,42 +201,42 @@ typedef enum {
 #define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
 #define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
 
-GST_EXPORT
+GST_AUDIO_API
 void           gst_audio_resampler_options_set_quality   (GstAudioResamplerMethod method,
                                                           guint quality,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *options);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioResampler * gst_audio_resampler_new              (GstAudioResamplerMethod method,
                                                           GstAudioResamplerFlags flags,
                                                           GstAudioFormat format, gint channels,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *options);
 
-GST_EXPORT
+GST_AUDIO_API
 void                gst_audio_resampler_free             (GstAudioResampler *resampler);
 
-GST_EXPORT
+GST_AUDIO_API
 void                gst_audio_resampler_reset            (GstAudioResampler *resampler);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean            gst_audio_resampler_update           (GstAudioResampler *resampler,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *options);
 
-GST_EXPORT
+GST_AUDIO_API
 gsize               gst_audio_resampler_get_out_frames   (GstAudioResampler *resampler,
                                                           gsize in_frames);
 
-GST_EXPORT
+GST_AUDIO_API
 gsize               gst_audio_resampler_get_in_frames    (GstAudioResampler *resampler,
                                                           gsize out_frames);
 
-GST_EXPORT
+GST_AUDIO_API
 gsize               gst_audio_resampler_get_max_latency  (GstAudioResampler *resampler);
 
-GST_EXPORT
+GST_AUDIO_API
 void                gst_audio_resampler_resample         (GstAudioResampler * resampler,
                                                           gpointer in[], gsize in_frames,
                                                           gpointer out[], gsize out_frames);
diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
index 90d22a087..dad01c0ef 100644
--- a/gst-libs/gst/audio/audio.h
+++ b/gst-libs/gst/audio/audio.h
@@ -23,6 +23,7 @@
 #define __GST_AUDIO_AUDIO_H__
 
 #include <gst/gst.h>
+#include <gst/audio/audio-prelude.h>
 #include <gst/audio/audio-enumtypes.h>
 #include <gst/audio/audio-format.h>
 #include <gst/audio/audio-channels.h>
@@ -90,7 +91,7 @@ G_BEGIN_DECLS
  * handling
  */
 
-GST_EXPORT
+GST_AUDIO_API
 GstBuffer *    gst_audio_buffer_clip     (GstBuffer *buffer,
                                           const GstSegment *segment,
                                           gint rate, gint bpf);
diff --git a/gst-libs/gst/audio/audio_mkenum.py b/gst-libs/gst/audio/audio_mkenum.py
index 2f51aab5d..af1b0a0e2 100755
--- a/gst-libs/gst/audio/audio_mkenum.py
+++ b/gst-libs/gst/audio/audio_mkenum.py
@@ -8,11 +8,11 @@
 import sys, os, shutil, subprocess
 
 h_array = ['--fhead',
-           "#ifndef __GST_AUDIO_ENUM_TYPES_H__\n#define __GST_AUDIO_ENUM_TYPES_H__\n\n#include <gst/gst.h>\n\nG_BEGIN_DECLS\n",
+           "#ifndef __GST_AUDIO_ENUM_TYPES_H__\n#define __GST_AUDIO_ENUM_TYPES_H__\n\n#include <gst/gst.h>\n#include <gst/audio/audio-prelude.h>\nG_BEGIN_DECLS\n",
            '--fprod',
            "\n/* enumerations from \"@filename@\" */\n",
            '--vhead',
-           'GST_EXPORT GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n',
+           'GST_AUDIO_API GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n',
            '--ftail',
            'G_END_DECLS\n\n#endif /* __GST_AUDIO_ENUM_TYPES_H__ */',
            ]
diff --git a/gst-libs/gst/audio/gstaudioaggregator.h b/gst-libs/gst/audio/gstaudioaggregator.h
index d72638c23..ccc46dd27 100644
--- a/gst-libs/gst/audio/gstaudioaggregator.h
+++ b/gst-libs/gst/audio/gstaudioaggregator.h
@@ -98,7 +98,7 @@ struct _GstAudioAggregatorPadClass
   gpointer      _gst_reserved[GST_PADDING_LARGE];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_aggregator_pad_get_type           (void);
 
 #define GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD            (gst_audio_aggregator_convert_pad_get_type())
@@ -146,7 +146,7 @@ struct _GstAudioAggregatorConvertPadClass
   gpointer      _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_aggregator_convert_pad_get_type           (void);
 
 /**************************
@@ -205,10 +205,10 @@ struct _GstAudioAggregatorClass {
  * GstAggregator methods *
  ************************/
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_aggregator_get_type(void);
 
-GST_EXPORT
+GST_AUDIO_API
 void  gst_audio_aggregator_set_sink_caps (GstAudioAggregator    * aagg,
                                           GstAudioAggregatorPad * pad,
                                           GstCaps               * caps);
diff --git a/gst-libs/gst/audio/gstaudiobasesink.h b/gst-libs/gst/audio/gstaudiobasesink.h
index 35da8643f..b0fdd5e9f 100644
--- a/gst-libs/gst/audio/gstaudiobasesink.h
+++ b/gst-libs/gst/audio/gstaudiobasesink.h
@@ -220,54 +220,54 @@ struct _GstAudioBaseSinkClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_base_sink_get_type(void);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioRingBuffer *
            gst_audio_base_sink_create_ringbuffer       (GstAudioBaseSink *sink);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_sink_set_provide_clock       (GstAudioBaseSink *sink, gboolean provide);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean   gst_audio_base_sink_get_provide_clock       (GstAudioBaseSink *sink);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_sink_set_slave_method        (GstAudioBaseSink *sink,
                                                         GstAudioBaseSinkSlaveMethod method);
-GST_EXPORT
+GST_AUDIO_API
 GstAudioBaseSinkSlaveMethod
            gst_audio_base_sink_get_slave_method        (GstAudioBaseSink *sink);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_sink_set_drift_tolerance     (GstAudioBaseSink *sink,
                                                         gint64 drift_tolerance);
-GST_EXPORT
+GST_AUDIO_API
 gint64     gst_audio_base_sink_get_drift_tolerance     (GstAudioBaseSink *sink);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
                                                         GstClockTime alignment_threshold);
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime
            gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_sink_set_discont_wait        (GstAudioBaseSink * sink,
                                                         GstClockTime discont_wait);
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime
            gst_audio_base_sink_get_discont_wait        (GstAudioBaseSink * sink);
 
-GST_EXPORT
+GST_AUDIO_API
 void
 gst_audio_base_sink_set_custom_slaving_callback        (GstAudioBaseSink * sink,
                                                         GstAudioBaseSinkCustomSlavingCallback callback,
                                                         gpointer user_data,
                                                         GDestroyNotify notify);
 
-GST_EXPORT
+GST_AUDIO_API
 void gst_audio_base_sink_report_device_failure         (GstAudioBaseSink * sink);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
diff --git a/gst-libs/gst/audio/gstaudiobasesrc.h b/gst-libs/gst/audio/gstaudiobasesrc.h
index 27b7c4628..fcf6abe8f 100644
--- a/gst-libs/gst/audio/gstaudiobasesrc.h
+++ b/gst-libs/gst/audio/gstaudiobasesrc.h
@@ -132,23 +132,23 @@ struct _GstAudioBaseSrcClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType      gst_audio_base_src_get_type(void);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioRingBuffer *
            gst_audio_base_src_create_ringbuffer        (GstAudioBaseSrc *src);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_src_set_provide_clock        (GstAudioBaseSrc *src, gboolean provide);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean   gst_audio_base_src_get_provide_clock        (GstAudioBaseSrc *src);
 
-GST_EXPORT
+GST_AUDIO_API
 void       gst_audio_base_src_set_slave_method         (GstAudioBaseSrc *src,
                                                         GstAudioBaseSrcSlaveMethod method);
-GST_EXPORT
+GST_AUDIO_API
 GstAudioBaseSrcSlaveMethod
            gst_audio_base_src_get_slave_method         (GstAudioBaseSrc *src);
 
diff --git a/gst-libs/gst/audio/gstaudiocdsrc.h b/gst-libs/gst/audio/gstaudiocdsrc.h
index 84d648347..de3413fb5 100644
--- a/gst-libs/gst/audio/gstaudiocdsrc.h
+++ b/gst-libs/gst/audio/gstaudiocdsrc.h
@@ -131,10 +131,10 @@ struct _GstAudioCdSrcClass {
   gpointer       _gst_reserved[GST_PADDING_LARGE];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType    gst_audio_cd_src_get_type (void);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean gst_audio_cd_src_add_track (GstAudioCdSrc      * src,
                                      GstAudioCdSrcTrack * track);
 
diff --git a/gst-libs/gst/audio/gstaudioclock.h b/gst-libs/gst/audio/gstaudioclock.h
index 80d8ce3c8..4870dd989 100644
--- a/gst-libs/gst/audio/gstaudioclock.h
+++ b/gst-libs/gst/audio/gstaudioclock.h
@@ -89,23 +89,23 @@ struct _GstAudioClockClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType           gst_audio_clock_get_type        (void);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClock*       gst_audio_clock_new             (const gchar *name, GstAudioClockGetTimeFunc func,
                                                  gpointer user_data, GDestroyNotify destroy_notify);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_clock_reset           (GstAudioClock *clock, GstClockTime time);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime    gst_audio_clock_get_time        (GstAudioClock * clock);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime    gst_audio_clock_adjust          (GstAudioClock * clock, GstClockTime time);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_clock_invalidate      (GstAudioClock * clock);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
diff --git a/gst-libs/gst/audio/gstaudiodecoder.h b/gst-libs/gst/audio/gstaudiodecoder.h
index bd231dd1c..f03d9ac7d 100644
--- a/gst-libs/gst/audio/gstaudiodecoder.h
+++ b/gst-libs/gst/audio/gstaudiodecoder.h
@@ -103,7 +103,7 @@ typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate;
 
 /* do not use this one, use macro below */
 
-GST_EXPORT
+GST_AUDIO_API
 GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight,
                                         GQuark domain, gint code,
                                         gchar *txt, gchar *debug,
@@ -314,124 +314,124 @@ struct _GstAudioDecoderClass
   gpointer       _gst_reserved[GST_PADDING_LARGE - 4];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType             gst_audio_decoder_get_type (void);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean          gst_audio_decoder_set_output_format  (GstAudioDecoder    * dec,
                                                         const GstAudioInfo * info);
 
-GST_EXPORT
+GST_AUDIO_API
 GstCaps *         gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder,
                                                    GstCaps         * caps,
                                                    GstCaps         * filter);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean          gst_audio_decoder_negotiate (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 GstFlowReturn     gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
                                                   GstBuffer * buf, gint frames);
 
-GST_EXPORT
+GST_AUDIO_API
 GstBuffer *       gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec,
                                                             gsize              size);
 
 /* context parameters */
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioInfo    * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_plc_aware  (GstAudioDecoder * dec,
                                                     gboolean          plc);
 
-GST_EXPORT
+GST_AUDIO_API
 gint              gst_audio_decoder_get_plc_aware  (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_estimate_rate  (GstAudioDecoder * dec,
                                                     gboolean          enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gint              gst_audio_decoder_get_estimate_rate  (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 gint              gst_audio_decoder_get_delay      (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_max_errors (GstAudioDecoder * dec,
                                                    gint               num);
 
-GST_EXPORT
+GST_AUDIO_API
 gint              gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_latency (GstAudioDecoder * dec,
                                                  GstClockTime      min,
                                                  GstClockTime      max);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_get_latency (GstAudioDecoder * dec,
                                                  GstClockTime    * min,
                                                  GstClockTime    * max);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
                                                      gboolean        * sync,
                                                      gboolean        * eos);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec,
                                                          GstCaps         * allocation_caps);
 
 /* object properties */
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_plc (GstAudioDecoder * dec,
                                              gboolean          enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean          gst_audio_decoder_get_plc (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
                                                      GstClockTime      num);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime      gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_tolerance   (GstAudioDecoder * dec,
                                                      GstClockTime      tolerance);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime      gst_audio_decoder_get_tolerance   (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
                                                    gboolean enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean          gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
                                                       gboolean enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean          gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_get_allocator (GstAudioDecoder * dec,
                                                    GstAllocator ** allocator,
                                                    GstAllocationParams * params);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_merge_tags (GstAudioDecoder * dec,
                                                 const GstTagList * tags, GstTagMergeMode mode);
 
-GST_EXPORT
+GST_AUDIO_API
 void              gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder,
                                                                    gboolean use);
 
diff --git a/gst-libs/gst/audio/gstaudioencoder.h b/gst-libs/gst/audio/gstaudioencoder.h
index a72224afa..348b70e50 100644
--- a/gst-libs/gst/audio/gstaudioencoder.h
+++ b/gst-libs/gst/audio/gstaudioencoder.h
@@ -247,127 +247,127 @@ struct _GstAudioEncoderClass {
   gpointer       _gst_reserved[GST_PADDING_LARGE-3];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType           gst_audio_encoder_get_type         (void);
 
-GST_EXPORT
+GST_AUDIO_API
 GstFlowReturn   gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
                                                 GstBuffer       * buffer,
                                                 gint              samples);
 
-GST_EXPORT
+GST_AUDIO_API
 GstCaps *       gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
                                                  GstCaps         * caps,
                                                  GstCaps         * filter);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_set_output_format  (GstAudioEncoder    * enc,
                                                       GstCaps            * caps);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_negotiate          (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 GstBuffer *     gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc,
                                                           gsize             size);
 
 /* context parameters */
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioInfo  * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 gint            gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
 
-GST_EXPORT
+GST_AUDIO_API
 gint            gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
 
-GST_EXPORT
+GST_AUDIO_API
 gint            gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
 
-GST_EXPORT
+GST_AUDIO_API
 gint            gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_get_latency (GstAudioEncoder * enc,
                                                GstClockTime    * min,
                                                GstClockTime    * max);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_latency (GstAudioEncoder * enc,
                                                GstClockTime      min,
                                                GstClockTime      max);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_headers (GstAudioEncoder * enc,
                                                GList           * headers);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc,
                                                        GstCaps         * allocation_caps);
 
 /* object properties */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
                                                     gboolean enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
                                                          gboolean          enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
                                                    gboolean          enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
                                                  GstClockTime      tolerance);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime    gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
                                                 gboolean enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
                                                  gboolean enabled);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_get_allocator (GstAudioEncoder * enc,
                                                  GstAllocator ** allocator,
                                                  GstAllocationParams * params);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
                                               const GstTagList * tags, GstTagMergeMode mode);
 
diff --git a/gst-libs/gst/audio/gstaudiofilter.h b/gst-libs/gst/audio/gstaudiofilter.h
index ee8cc3ee5..ac1b5704e 100644
--- a/gst-libs/gst/audio/gstaudiofilter.h
+++ b/gst-libs/gst/audio/gstaudiofilter.h
@@ -92,10 +92,10 @@ struct _GstAudioFilterClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType   gst_audio_filter_get_type (void);
 
-GST_EXPORT
+GST_AUDIO_API
 void    gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,
                                                   GstCaps             * allowed_caps);
 
diff --git a/gst-libs/gst/audio/gstaudioiec61937.h b/gst-libs/gst/audio/gstaudioiec61937.h
index fcd1b0879..1885a921d 100644
--- a/gst-libs/gst/audio/gstaudioiec61937.h
+++ b/gst-libs/gst/audio/gstaudioiec61937.h
@@ -24,10 +24,10 @@
 
 #include <gst/audio/gstaudioringbuffer.h>
 
-GST_EXPORT
+GST_AUDIO_API
 guint       gst_audio_iec61937_frame_size  (const GstAudioRingBufferSpec * spec);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean    gst_audio_iec61937_payload     (const guint8 * src, guint src_n,
                                             guint8 * dst, guint dst_n,
                                             const GstAudioRingBufferSpec * spec,
diff --git a/gst-libs/gst/audio/gstaudiometa.h b/gst-libs/gst/audio/gstaudiometa.h
index 4cd5cbe69..2f6c72f7f 100644
--- a/gst-libs/gst/audio/gstaudiometa.h
+++ b/gst-libs/gst/audio/gstaudiometa.h
@@ -56,19 +56,19 @@ struct _GstAudioDownmixMeta {
   gfloat       **matrix;
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_downmix_meta_api_get_type (void);
 
-GST_EXPORT
+GST_AUDIO_API
 const GstMetaInfo * gst_audio_downmix_meta_get_info (void);
 
 #define gst_buffer_get_audio_downmix_meta(b) ((GstAudioDownmixMeta*)gst_buffer_get_meta((b), GST_AUDIO_DOWNMIX_META_API_TYPE))
-GST_EXPORT
+GST_AUDIO_API
 GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels    (GstBuffer *buffer,
                                                                          const GstAudioChannelPosition *to_position,
                                                                          gint                           to_channels);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer    *buffer,
                                                          const GstAudioChannelPosition *from_position,
                                                          gint                           from_channels,
@@ -111,15 +111,15 @@ struct _GstAudioClippingMeta {
   guint64   end;
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_clipping_meta_api_get_type (void);
 
-GST_EXPORT
+GST_AUDIO_API
 const GstMetaInfo * gst_audio_clipping_meta_get_info (void);
 
 #define gst_buffer_get_audio_clipping_meta(b) ((GstAudioClippingMeta*)gst_buffer_get_meta((b), GST_AUDIO_CLIPPING_META_API_TYPE))
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer *buffer,
                                                            GstFormat  format,
                                                            guint64    start,
diff --git a/gst-libs/gst/audio/gstaudioringbuffer.h b/gst-libs/gst/audio/gstaudioringbuffer.h
index c447c0ee3..64c602699 100644
--- a/gst-libs/gst/audio/gstaudioringbuffer.h
+++ b/gst-libs/gst/audio/gstaudioringbuffer.h
@@ -265,122 +265,122 @@ struct _GstAudioRingBufferClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_ring_buffer_get_type(void);
 
 /* callback stuff */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_set_callback      (GstAudioRingBuffer *buf,
                                                          GstAudioRingBufferCallback cb,
                                                          gpointer user_data);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf,
                                                          GstAudioRingBufferCallback cb,
                                                          gpointer user_data,
                                                          GDestroyNotify notify);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_parse_caps      (GstAudioRingBufferSpec *spec, GstCaps *caps);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_convert         (GstAudioRingBuffer * buf, GstFormat src_fmt,
                                                        gint64 src_val, GstFormat dest_fmt,
                                                        gint64 * dest_val);
 
 /* device state */
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_open_device     (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_close_device    (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_device_is_open  (GstAudioRingBuffer *buf);
 
 /* allocate resources */
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_acquire         (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_release         (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_is_acquired     (GstAudioRingBuffer *buf);
 
 /* set the device channel positions */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position);
 
 /* activating */
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_activate        (GstAudioRingBuffer *buf, gboolean active);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_is_active       (GstAudioRingBuffer *buf);
 
 /* flushing */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_set_flushing    (GstAudioRingBuffer *buf, gboolean flushing);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_is_flushing     (GstAudioRingBuffer *buf);
 
 /* playback/pause */
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_start           (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_pause           (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_stop            (GstAudioRingBuffer *buf);
 
 /* get status */
 
-GST_EXPORT
+GST_AUDIO_API
 guint           gst_audio_ring_buffer_delay           (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 guint64         gst_audio_ring_buffer_samples_done    (GstAudioRingBuffer *buf);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_set_sample      (GstAudioRingBuffer *buf, guint64 sample);
 
 /* clear all segments */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_clear_all       (GstAudioRingBuffer *buf);
 
 /* commit samples */
 
-GST_EXPORT
+GST_AUDIO_API
 guint           gst_audio_ring_buffer_commit          (GstAudioRingBuffer * buf, guint64 *sample,
                                                        guint8 * data, gint in_samples,
                                                        gint out_samples, gint * accum);
 
 /* read samples */
 
-GST_EXPORT
+GST_AUDIO_API
 guint           gst_audio_ring_buffer_read            (GstAudioRingBuffer *buf, guint64 sample,
                                                        guint8 *data, guint len, GstClockTime *timestamp);
 
 /* Set timestamp on buffer */
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_set_timestamp   (GstAudioRingBuffer * buf, gint readseg, GstClockTime 
                                                        timestamp);
 
@@ -389,17 +389,17 @@ void            gst_audio_ring_buffer_set_timestamp   (GstAudioRingBuffer * buf,
 gboolean        gst_audio_ring_buffer_prepare_write   (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len);
 */
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_audio_ring_buffer_prepare_read    (GstAudioRingBuffer *buf, gint *segment,
                                                        guint8 **readptr, gint *len);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_clear           (GstAudioRingBuffer *buf, gint segment);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_advance         (GstAudioRingBuffer *buf, guint advance);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_audio_ring_buffer_may_start       (GstAudioRingBuffer *buf, gboolean allowed);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
diff --git a/gst-libs/gst/audio/gstaudiosink.h b/gst-libs/gst/audio/gstaudiosink.h
index 04d429912..b3c482e3b 100644
--- a/gst-libs/gst/audio/gstaudiosink.h
+++ b/gst-libs/gst/audio/gstaudiosink.h
@@ -97,7 +97,7 @@ struct _GstAudioSinkClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_sink_get_type(void);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
diff --git a/gst-libs/gst/audio/gstaudiosrc.h b/gst-libs/gst/audio/gstaudiosrc.h
index 04a673128..8fd0afba5 100644
--- a/gst-libs/gst/audio/gstaudiosrc.h
+++ b/gst-libs/gst/audio/gstaudiosrc.h
@@ -96,7 +96,7 @@ struct _GstAudioSrcClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GST_EXPORT
+GST_AUDIO_API
 GType gst_audio_src_get_type(void);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
diff --git a/gst-libs/gst/audio/gstaudiostreamalign.h b/gst-libs/gst/audio/gstaudiostreamalign.h
index 080088136..02cedf34a 100644
--- a/gst-libs/gst/audio/gstaudiostreamalign.h
+++ b/gst-libs/gst/audio/gstaudiostreamalign.h
@@ -23,6 +23,7 @@
 #define __GST_AUDIO_STREAM_ALIGN_H__
 
 #include <gst/gst.h>
+#include <gst/audio/audio-prelude.h>
 
 G_BEGIN_DECLS
 
@@ -30,47 +31,47 @@ G_BEGIN_DECLS
 
 typedef struct _GstAudioStreamAlign GstAudioStreamAlign;
 
-GST_EXPORT
+GST_AUDIO_API
 GType                   gst_audio_stream_align_get_type                  (void);
 
-GST_EXPORT
+GST_AUDIO_API
 GstAudioStreamAlign *   gst_audio_stream_align_new                       (gint rate,
                                                                           GstClockTime alignment_threshold,
                                                                           GstClockTime discont_wait);
-GST_EXPORT
+GST_AUDIO_API
 GstAudioStreamAlign *   gst_audio_stream_align_copy                      (const GstAudioStreamAlign * align);
-GST_EXPORT
+GST_AUDIO_API
 void                    gst_audio_stream_align_free                      (GstAudioStreamAlign * align);
 
-GST_EXPORT
+GST_AUDIO_API
 void                    gst_audio_stream_align_set_rate                  (GstAudioStreamAlign * align,
                                                                           gint rate);
-GST_EXPORT
+GST_AUDIO_API
 gint                    gst_audio_stream_align_get_rate                  (GstAudioStreamAlign * align);
 
-GST_EXPORT
+GST_AUDIO_API
 void                    gst_audio_stream_align_set_alignment_threshold   (GstAudioStreamAlign * align,
                                                                           GstClockTime alignment_threshold);
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime            gst_audio_stream_align_get_alignment_threshold   (GstAudioStreamAlign * align);
 
-GST_EXPORT
+GST_AUDIO_API
 void                    gst_audio_stream_align_set_discont_wait          (GstAudioStreamAlign * align,
                                                                           GstClockTime discont_wait);
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime            gst_audio_stream_align_get_discont_wait          (GstAudioStreamAlign * align);
 
 
-GST_EXPORT
+GST_AUDIO_API
 void                    gst_audio_stream_align_mark_discont              (GstAudioStreamAlign * align);
 
-GST_EXPORT
+GST_AUDIO_API
 GstClockTime            gst_audio_stream_align_get_timestamp_at_discont  (GstAudioStreamAlign * align);
 
-GST_EXPORT
+GST_AUDIO_API
 guint64                 gst_audio_stream_align_get_samples_since_discont (GstAudioStreamAlign * align);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean                gst_audio_stream_align_process                   (GstAudioStreamAlign * align,
                                                                           gboolean discont,
                                                                           GstClockTime timestamp,
diff --git a/gst-libs/gst/audio/streamvolume.h b/gst-libs/gst/audio/streamvolume.h
index 6d602c003..4802d04b0 100644
--- a/gst-libs/gst/audio/streamvolume.h
+++ b/gst-libs/gst/audio/streamvolume.h
@@ -21,6 +21,7 @@
 #define __GST_STREAM_VOLUME_H__
 
 #include <gst/gst.h>
+#include <gst/audio/audio-prelude.h>
 
 G_BEGIN_DECLS
 
@@ -58,26 +59,26 @@ typedef enum {
   GST_STREAM_VOLUME_FORMAT_DB
 } GstStreamVolumeFormat;
 
-GST_EXPORT
+GST_AUDIO_API
 GType           gst_stream_volume_get_type        (void);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_stream_volume_set_volume      (GstStreamVolume *volume,
                                                    GstStreamVolumeFormat format,
                                                    gdouble val);
 
-GST_EXPORT
+GST_AUDIO_API
 gdouble         gst_stream_volume_get_volume      (GstStreamVolume *volume,
                                                    GstStreamVolumeFormat format);
 
-GST_EXPORT
+GST_AUDIO_API
 void            gst_stream_volume_set_mute        (GstStreamVolume *volume,
                                                    gboolean mute);
 
-GST_EXPORT
+GST_AUDIO_API
 gboolean        gst_stream_volume_get_mute        (GstStreamVolume *volume);
 
-GST_EXPORT
+GST_AUDIO_API
 gdouble         gst_stream_volume_convert_volume  (GstStreamVolumeFormat from,
                                                    GstStreamVolumeFormat to,
                                                    gdouble val) G_GNUC_CONST;