audio: mark symbols explicitly for export with GST_EXPORT

diff --git a/common b/common
index 48a5d85ebf4a0bad1c997c83100f710fe2154fbf..29046b89d80bbca22eb222c18820fb40a4ac5bde 160000 (submodule)
--- a/common
+++ b/common
@@ -1 +1 @@
-Subproject commit 48a5d85ebf4a0bad1c997c83100f710fe2154fbf
+Subproject commit 29046b89d80bbca22eb222c18820fb40a4ac5bde

diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am
index 7878f363db8e7eebc8de25139379ae044af4bf52..e6fcde15c65f2fb709702e2b54b637e1cbe5e9ac 100644 (file)
--- a/gst-libs/gst/audio/Makefile.am
+++ b/gst-libs/gst/audio/Makefile.am
@@ -19,6 +19,7 @@ glib_enum_headers=            \
 glib_enum_define = GST_AUDIO
 glib_gen_prefix = gst_audio
 glib_gen_basename = audio
+glib_gen_decl_banner=GST_EXPORT
 
 built_sources = audio-enumtypes.c
 built_headers = audio-enumtypes.h

diff --git a/gst-libs/gst/audio/audio-channel-mixer.h b/gst-libs/gst/audio/audio-channel-mixer.h
index 813ef3522acde42656c6ec6a9e2fb6a24437a318..a9ce535b343774fcf1f18f9dbf28b48b5a120668 100644 (file)
--- a/gst-libs/gst/audio/audio-channel-mixer.h
+++ b/gst-libs/gst/audio/audio-channel-mixer.h
@@ -46,22 +46,29 @@ typedef enum {
   GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT    = (1 << 3)
 } GstAudioChannelMixerFlags;
 
+GST_EXPORT
 GstAudioChannelMixer * gst_audio_channel_mixer_new   (GstAudioChannelMixerFlags flags,
                                                       GstAudioFormat format,
                                                       gint in_channels,
                                                       GstAudioChannelPosition *in_position,
                                                       gint out_channels,
                                                       GstAudioChannelPosition *out_position);
+
+GST_EXPORT
 void                   gst_audio_channel_mixer_free  (GstAudioChannelMixer *mix);
 
 /*
  * Checks for passthrough (= identity matrix).
  */
+
+GST_EXPORT
 gboolean        gst_audio_channel_mixer_is_passthrough  (GstAudioChannelMixer *mix);
 
 /*
  * Do actual mixing.
  */
+
+GST_EXPORT
 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 cd57898e786443f15ec06517067ff0029b33d4c6..5836633a6bcafe3a13bfb0a3c10bd1f5c27a22cc 100644 (file)
--- a/gst-libs/gst/audio/audio-channels.h
+++ b/gst-libs/gst/audio/audio-channels.h
@@ -131,37 +131,47 @@ typedef enum {
 
 #define GST_AUDIO_CHANNEL_POSITION_MASK(pos) (G_GUINT64_CONSTANT(1)<< GST_AUDIO_CHANNEL_POSITION_ ## pos)
 
+GST_EXPORT
 gboolean       gst_audio_buffer_reorder_channels (GstBuffer * buffer,
                                                   GstAudioFormat format,
                                                   gint channels,
                                                   const GstAudioChannelPosition * from,
                                                   const GstAudioChannelPosition * to);
 
+GST_EXPORT
 gboolean       gst_audio_reorder_channels        (gpointer data, gsize size,
                                                   GstAudioFormat format,
                                                   gint channels,
                                                   const GstAudioChannelPosition * from,
                                                   const GstAudioChannelPosition * to);
 
+GST_EXPORT
 gboolean       gst_audio_channel_positions_to_valid_order (GstAudioChannelPosition *position,
                                                            gint channels);
 
+GST_EXPORT
 gboolean       gst_audio_check_valid_channel_positions (const GstAudioChannelPosition *position,
                                                         gint channels, gboolean force_order);
 
+GST_EXPORT
 gboolean       gst_audio_channel_positions_to_mask  (const GstAudioChannelPosition *position,
                                                      gint channels, gboolean force_order,
                                                      guint64 *channel_mask);
 
+GST_EXPORT
 gboolean       gst_audio_channel_positions_from_mask (gint channels, guint64 channel_mask,
                                                       GstAudioChannelPosition * position);
 
+GST_EXPORT
 gboolean       gst_audio_get_channel_reorder_map (gint channels,
                                                   const GstAudioChannelPosition * from,
                                                   const GstAudioChannelPosition * to,
                                                   gint *reorder_map);
+
+GST_EXPORT
 guint64        gst_audio_channel_get_fallback_mask (gint channels);
 
+GST_EXPORT
 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 672d30c33c0d3bf95021ba14448a6a2c781f9c0a..e49f8a90fe87d0529700f836384d83334f65412f 100644 (file)
--- a/gst-libs/gst/audio/audio-converter.h
+++ b/gst-libs/gst/audio/audio-converter.h
@@ -81,33 +81,45 @@ typedef enum {
   GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE   = (1 << 1)
 } GstAudioConverterFlags;
 
+GST_EXPORT
 GstAudioConverter *  gst_audio_converter_new             (GstAudioConverterFlags flags,
                                                           GstAudioInfo *in_info,
                                                           GstAudioInfo *out_info,
                                                           GstStructure *config);
 
+GST_EXPORT
 void                 gst_audio_converter_free            (GstAudioConverter * convert);
 
+GST_EXPORT
 void                 gst_audio_converter_reset           (GstAudioConverter * convert);
 
+GST_EXPORT
 gboolean             gst_audio_converter_update_config   (GstAudioConverter * convert,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *config);
+
+GST_EXPORT
 const GstStructure * gst_audio_converter_get_config      (GstAudioConverter * convert,
                                                           gint *in_rate, gint *out_rate);
 
+GST_EXPORT
 gsize                gst_audio_converter_get_out_frames  (GstAudioConverter *convert,
                                                           gsize in_frames);
+
+GST_EXPORT
 gsize                gst_audio_converter_get_in_frames   (GstAudioConverter *convert,
                                                           gsize out_frames);
 
+GST_EXPORT
 gsize                gst_audio_converter_get_max_latency (GstAudioConverter *convert);
 
+GST_EXPORT
 gboolean             gst_audio_converter_samples         (GstAudioConverter * convert,
                                                           GstAudioConverterFlags flags,
                                                           gpointer in[], gsize in_frames,
                                                           gpointer out[], gsize out_frames);
 
+GST_EXPORT
 gboolean             gst_audio_converter_supports_inplace (GstAudioConverter *convert);
 
 #endif /* __GST_AUDIO_CONVERTER_H__ */

diff --git a/gst-libs/gst/audio/audio-format.h b/gst-libs/gst/audio/audio-format.h
index 0fbc4151a36099b0adbac3096ee1337da1a9ece2..44398914d3535180564e58e0adea915a5b7c7dbf 100644 (file)
--- a/gst-libs/gst/audio/audio-format.h
+++ b/gst-libs/gst/audio/audio-format.h
@@ -250,6 +250,7 @@ struct _GstAudioFormatInfo {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 GType gst_audio_format_info_get_type (void);
 
 #define GST_AUDIO_FORMAT_INFO_FORMAT(info)           ((info)->format)
@@ -267,15 +268,20 @@ GType gst_audio_format_info_get_type (void);
 #define GST_AUDIO_FORMAT_INFO_DEPTH(info)            ((info)->depth)
 
 
+GST_EXPORT
 GstAudioFormat gst_audio_format_build_integer    (gboolean sign, gint endianness,
                                                   gint width, gint depth) G_GNUC_CONST;
 
+GST_EXPORT
 GstAudioFormat gst_audio_format_from_string      (const gchar *format) G_GNUC_CONST;
+
+GST_EXPORT
 const gchar *  gst_audio_format_to_string        (GstAudioFormat format) G_GNUC_CONST;
 
 const GstAudioFormatInfo *
                gst_audio_format_get_info         (GstAudioFormat format) G_GNUC_CONST;
 
+GST_EXPORT
 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 0f7c7d8bd9645298fae57ada15675b18e5d8d10f..2541ca77d526df849e261ca5b3e9907cad189dda 100644 (file)
--- a/gst-libs/gst/audio/audio-info.h
+++ b/gst-libs/gst/audio/audio-info.h
@@ -85,6 +85,7 @@ struct _GstAudioInfo {
 };
 
 #define GST_TYPE_AUDIO_INFO                  (gst_audio_info_get_type ())
+GST_EXPORT
 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)
@@ -112,22 +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
 GstAudioInfo * gst_audio_info_new         (void);
+
+GST_EXPORT
 void           gst_audio_info_init        (GstAudioInfo *info);
+
+GST_EXPORT
 GstAudioInfo * gst_audio_info_copy        (const GstAudioInfo *info);
+
+GST_EXPORT
 void           gst_audio_info_free        (GstAudioInfo *info);
 
+GST_EXPORT
 void           gst_audio_info_set_format  (GstAudioInfo *info, GstAudioFormat format,
                                            gint rate, gint channels,
                                            const GstAudioChannelPosition *position);
 
+GST_EXPORT
 gboolean       gst_audio_info_from_caps   (GstAudioInfo *info, const GstCaps *caps);
+
+GST_EXPORT
 GstCaps *      gst_audio_info_to_caps     (const GstAudioInfo *info);
 
+GST_EXPORT
 gboolean       gst_audio_info_convert     (const GstAudioInfo * info,
                                            GstFormat src_fmt, gint64 src_val,
                                            GstFormat dest_fmt, gint64 * dest_val);
 
+GST_EXPORT
 gboolean       gst_audio_info_is_equal    (const GstAudioInfo *info,
                                            const GstAudioInfo *other);
 

diff --git a/gst-libs/gst/audio/audio-quantize.h b/gst-libs/gst/audio/audio-quantize.h
index e60c51682c5444d585930b7f55fa831912502402..8a0bd7926cd7c1c47670105ee0703238390fc872 100644 (file)
--- a/gst-libs/gst/audio/audio-quantize.h
+++ b/gst-libs/gst/audio/audio-quantize.h
@@ -81,6 +81,7 @@ typedef enum
 
 typedef struct _GstAudioQuantize GstAudioQuantize;
 
+GST_EXPORT
 GstAudioQuantize *  gst_audio_quantize_new      (GstAudioDitherMethod dither,
                                                  GstAudioNoiseShapingMethod ns,
                                                  GstAudioQuantizeFlags flags,
@@ -88,10 +89,13 @@ GstAudioQuantize *  gst_audio_quantize_new      (GstAudioDitherMethod dither,
                                                  guint channels,
                                                  guint quantizer);
 
+GST_EXPORT
 void                gst_audio_quantize_free     (GstAudioQuantize * quant);
 
+GST_EXPORT
 void                gst_audio_quantize_reset    (GstAudioQuantize * quant);
 
+GST_EXPORT
 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 fe4c30090a762556d46be99078bcaf5e98fc039d..cecda5795eb16bcfc2e05963ddce58d455247cc1 100644 (file)
--- a/gst-libs/gst/audio/audio-resampler.h
+++ b/gst-libs/gst/audio/audio-resampler.h
@@ -201,31 +201,42 @@ typedef enum {
 #define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
 #define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
 
+GST_EXPORT
 void           gst_audio_resampler_options_set_quality   (GstAudioResamplerMethod method,
                                                           guint quality,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *options);
 
+GST_EXPORT
 GstAudioResampler * gst_audio_resampler_new              (GstAudioResamplerMethod method,
                                                           GstAudioResamplerFlags flags,
                                                           GstAudioFormat format, gint channels,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *options);
+
+GST_EXPORT
 void                gst_audio_resampler_free             (GstAudioResampler *resampler);
 
+GST_EXPORT
 void                gst_audio_resampler_reset            (GstAudioResampler *resampler);
 
+GST_EXPORT
 gboolean            gst_audio_resampler_update           (GstAudioResampler *resampler,
                                                           gint in_rate, gint out_rate,
                                                           GstStructure *options);
 
+GST_EXPORT
 gsize               gst_audio_resampler_get_out_frames   (GstAudioResampler *resampler,
                                                           gsize in_frames);
+
+GST_EXPORT
 gsize               gst_audio_resampler_get_in_frames    (GstAudioResampler *resampler,
                                                           gsize out_frames);
 
+GST_EXPORT
 gsize               gst_audio_resampler_get_max_latency  (GstAudioResampler *resampler);
 
+GST_EXPORT
 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 ac45b2e2be508916665a9f8d35cb263fef37ef13..edb81368531c970576aab984117ee69fbfa722bd 100644 (file)
--- a/gst-libs/gst/audio/audio.h
+++ b/gst-libs/gst/audio/audio.h
@@ -88,6 +88,7 @@ G_BEGIN_DECLS
  * handling
  */
 
+GST_EXPORT
 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 b3572de3019192c4492721d8e22667a760dd3b3f..2f51aab5d03c25f77f04d9ee252fa4ecaae85965 100755 (executable)
--- 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 <glib-object.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\nG_BEGIN_DECLS\n",
            '--fprod',
            "\n/* enumerations from \"@filename@\" */\n",
            '--vhead',
-           'GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n',
+           'GST_EXPORT 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/gstaudiobasesink.h b/gst-libs/gst/audio/gstaudiobasesink.h
index af491a6d26d64fbd496fd9fd49225f59dbffefc3..35da8643ff706b4c896d84f4d873833e18344e71 100644 (file)
--- a/gst-libs/gst/audio/gstaudiobasesink.h
+++ b/gst-libs/gst/audio/gstaudiobasesink.h
@@ -220,39 +220,54 @@ struct _GstAudioBaseSinkClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 GType gst_audio_base_sink_get_type(void);
 
+GST_EXPORT
 GstAudioRingBuffer *
            gst_audio_base_sink_create_ringbuffer       (GstAudioBaseSink *sink);
 
+GST_EXPORT
 void       gst_audio_base_sink_set_provide_clock       (GstAudioBaseSink *sink, gboolean provide);
+
+GST_EXPORT
 gboolean   gst_audio_base_sink_get_provide_clock       (GstAudioBaseSink *sink);
 
+GST_EXPORT
 void       gst_audio_base_sink_set_slave_method        (GstAudioBaseSink *sink,
                                                         GstAudioBaseSinkSlaveMethod method);
+GST_EXPORT
 GstAudioBaseSinkSlaveMethod
            gst_audio_base_sink_get_slave_method        (GstAudioBaseSink *sink);
 
+GST_EXPORT
 void       gst_audio_base_sink_set_drift_tolerance     (GstAudioBaseSink *sink,
                                                         gint64 drift_tolerance);
+GST_EXPORT
 gint64     gst_audio_base_sink_get_drift_tolerance     (GstAudioBaseSink *sink);
 
+GST_EXPORT
 void       gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
                                                         GstClockTime alignment_threshold);
+GST_EXPORT
 GstClockTime
            gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
 
+GST_EXPORT
 void       gst_audio_base_sink_set_discont_wait        (GstAudioBaseSink * sink,
                                                         GstClockTime discont_wait);
+GST_EXPORT
 GstClockTime
            gst_audio_base_sink_get_discont_wait        (GstAudioBaseSink * sink);
 
+GST_EXPORT
 void
 gst_audio_base_sink_set_custom_slaving_callback        (GstAudioBaseSink * sink,
                                                         GstAudioBaseSinkCustomSlavingCallback callback,
                                                         gpointer user_data,
                                                         GDestroyNotify notify);
 
+GST_EXPORT
 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 2247bad5e30a59ca4dca3cba2352ef8a2ff53bc2..27b7c46285cd6e645911e4291caf4fb729fa5410 100644 (file)
--- a/gst-libs/gst/audio/gstaudiobasesrc.h
+++ b/gst-libs/gst/audio/gstaudiobasesrc.h
@@ -132,16 +132,23 @@ struct _GstAudioBaseSrcClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GType gst_audio_base_src_get_type(void);
+GST_EXPORT
+GType      gst_audio_base_src_get_type(void);
 
+GST_EXPORT
 GstAudioRingBuffer *
            gst_audio_base_src_create_ringbuffer        (GstAudioBaseSrc *src);
 
+GST_EXPORT
 void       gst_audio_base_src_set_provide_clock        (GstAudioBaseSrc *src, gboolean provide);
+
+GST_EXPORT
 gboolean   gst_audio_base_src_get_provide_clock        (GstAudioBaseSrc *src);
 
+GST_EXPORT
 void       gst_audio_base_src_set_slave_method         (GstAudioBaseSrc *src,
                                                         GstAudioBaseSrcSlaveMethod method);
+GST_EXPORT
 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 5044d32792a360fb8382df41d88c3d4ecc8df739..84d648347ccc39ac0a7811a0966650ee4665379e 100644 (file)
--- a/gst-libs/gst/audio/gstaudiocdsrc.h
+++ b/gst-libs/gst/audio/gstaudiocdsrc.h
@@ -131,8 +131,10 @@ struct _GstAudioCdSrcClass {
   gpointer       _gst_reserved[GST_PADDING_LARGE];
 };
 
+GST_EXPORT
 GType    gst_audio_cd_src_get_type (void);
 
+GST_EXPORT
 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 9b8f51c57584e10c70de3178a72a762b7c6294f0..80d8ce3c8677aebd62b4dd50f8a23ec7c74e3128 100644 (file)
--- a/gst-libs/gst/audio/gstaudioclock.h
+++ b/gst-libs/gst/audio/gstaudioclock.h
@@ -89,14 +89,23 @@ struct _GstAudioClockClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 GType           gst_audio_clock_get_type        (void);
+
+GST_EXPORT
 GstClock*       gst_audio_clock_new             (const gchar *name, GstAudioClockGetTimeFunc func,
                                                  gpointer user_data, GDestroyNotify destroy_notify);
+
+GST_EXPORT
 void            gst_audio_clock_reset           (GstAudioClock *clock, GstClockTime time);
 
+GST_EXPORT
 GstClockTime    gst_audio_clock_get_time        (GstAudioClock * clock);
+
+GST_EXPORT
 GstClockTime    gst_audio_clock_adjust          (GstAudioClock * clock, GstClockTime time);
 
+GST_EXPORT
 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 71e0ce79214e2f678841892872c05606525ec587..bd231dd1cfb070260b5da6f81d4d809da2dfb553 100644 (file)
--- a/gst-libs/gst/audio/gstaudiodecoder.h
+++ b/gst-libs/gst/audio/gstaudiodecoder.h
@@ -102,6 +102,8 @@ typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
 typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate;
 
 /* do not use this one, use macro below */
+
+GST_EXPORT
 GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight,
                                         GQuark domain, gint code,
                                         gchar *txt, gchar *debug,
@@ -312,90 +314,124 @@ struct _GstAudioDecoderClass
   gpointer       _gst_reserved[GST_PADDING_LARGE - 4];
 };
 
+GST_EXPORT
 GType             gst_audio_decoder_get_type (void);
 
+GST_EXPORT
 gboolean          gst_audio_decoder_set_output_format  (GstAudioDecoder    * dec,
                                                         const GstAudioInfo * info);
 
+GST_EXPORT
 GstCaps *         gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder,
                                                    GstCaps         * caps,
                                                    GstCaps         * filter);
+
+GST_EXPORT
 gboolean          gst_audio_decoder_negotiate (GstAudioDecoder * dec);
 
+GST_EXPORT
 GstFlowReturn     gst_audio_decoder_finish_frame (GstAudioDecoder * dec,
                                                   GstBuffer * buf, gint frames);
 
+GST_EXPORT
 GstBuffer *       gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec,
                                                             gsize              size);
 
 /* context parameters */
+
+GST_EXPORT
 GstAudioInfo    * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_plc_aware  (GstAudioDecoder * dec,
                                                     gboolean          plc);
 
+GST_EXPORT
 gint              gst_audio_decoder_get_plc_aware  (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_estimate_rate  (GstAudioDecoder * dec,
                                                     gboolean          enabled);
 
+GST_EXPORT
 gint              gst_audio_decoder_get_estimate_rate  (GstAudioDecoder * dec);
 
+GST_EXPORT
 gint              gst_audio_decoder_get_delay      (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_max_errors (GstAudioDecoder * dec,
                                                    gint               num);
 
+GST_EXPORT
 gint              gst_audio_decoder_get_max_errors (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_latency (GstAudioDecoder * dec,
                                                  GstClockTime      min,
                                                  GstClockTime      max);
 
+GST_EXPORT
 void              gst_audio_decoder_get_latency (GstAudioDecoder * dec,
                                                  GstClockTime    * min,
                                                  GstClockTime    * max);
 
+GST_EXPORT
 void              gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
                                                      gboolean        * sync,
                                                      gboolean        * eos);
 
+GST_EXPORT
 void              gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec,
                                                          GstCaps         * allocation_caps);
 
 /* object properties */
+
+GST_EXPORT
 void              gst_audio_decoder_set_plc (GstAudioDecoder * dec,
                                              gboolean          enabled);
 
+GST_EXPORT
 gboolean          gst_audio_decoder_get_plc (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_min_latency (GstAudioDecoder * dec,
                                                      GstClockTime      num);
 
+GST_EXPORT
 GstClockTime      gst_audio_decoder_get_min_latency (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_tolerance   (GstAudioDecoder * dec,
                                                      GstClockTime      tolerance);
 
+GST_EXPORT
 GstClockTime      gst_audio_decoder_get_tolerance   (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
                                                    gboolean enabled);
 
+GST_EXPORT
 gboolean          gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
                                                       gboolean enabled);
 
+GST_EXPORT
 gboolean          gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
 
+GST_EXPORT
 void              gst_audio_decoder_get_allocator (GstAudioDecoder * dec,
                                                    GstAllocator ** allocator,
                                                    GstAllocationParams * params);
 
+GST_EXPORT
 void              gst_audio_decoder_merge_tags (GstAudioDecoder * dec,
                                                 const GstTagList * tags, GstTagMergeMode mode);
 
+GST_EXPORT
 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 a186bd582b230e005e62fdd62ee0705dd7188639..a72224afab6274d4c1da3c86d7ae4ba0e63be24f 100644 (file)
--- a/gst-libs/gst/audio/gstaudioencoder.h
+++ b/gst-libs/gst/audio/gstaudioencoder.h
@@ -247,93 +247,127 @@ struct _GstAudioEncoderClass {
   gpointer       _gst_reserved[GST_PADDING_LARGE-3];
 };
 
+GST_EXPORT
 GType           gst_audio_encoder_get_type         (void);
 
+GST_EXPORT
 GstFlowReturn   gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
                                                 GstBuffer       * buffer,
                                                 gint              samples);
 
+GST_EXPORT
 GstCaps *       gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
                                                  GstCaps         * caps,
                                                  GstCaps         * filter);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_set_output_format  (GstAudioEncoder    * enc,
                                                       GstCaps            * caps);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_negotiate          (GstAudioEncoder * enc);
 
+GST_EXPORT
 GstBuffer *     gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc,
                                                           gsize             size);
 
 /* context parameters */
+
+GST_EXPORT
 GstAudioInfo  * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
 
+GST_EXPORT
 gint            gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
 
+GST_EXPORT
 gint            gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
 
+GST_EXPORT
 gint            gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
 
+GST_EXPORT
 gint            gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
 
+GST_EXPORT
 void            gst_audio_encoder_get_latency (GstAudioEncoder * enc,
                                                GstClockTime    * min,
                                                GstClockTime    * max);
 
+GST_EXPORT
 void            gst_audio_encoder_set_latency (GstAudioEncoder * enc,
                                                GstClockTime      min,
                                                GstClockTime      max);
 
+GST_EXPORT
 void            gst_audio_encoder_set_headers (GstAudioEncoder * enc,
                                                GList           * headers);
 
+GST_EXPORT
 void            gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc,
                                                        GstCaps         * allocation_caps);
 
 /* object properties */
 
+GST_EXPORT
 void            gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
                                                     gboolean enabled);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
                                                          gboolean          enabled);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
                                                    gboolean          enabled);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
                                                  GstClockTime      tolerance);
 
+GST_EXPORT
 GstClockTime    gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
                                                 gboolean enabled);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
                                                  gboolean enabled);
 
+GST_EXPORT
 gboolean        gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
 
+GST_EXPORT
 void            gst_audio_encoder_get_allocator (GstAudioEncoder * enc,
                                                  GstAllocator ** allocator,
                                                  GstAllocationParams * params);
 
+GST_EXPORT
 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 4610f3ae01b1f5ae2ad6654bdad656303a8c96f4..ee8cc3ee50b4f3692df2c587a69f8fbcb456d845 100644 (file)
--- a/gst-libs/gst/audio/gstaudiofilter.h
+++ b/gst-libs/gst/audio/gstaudiofilter.h
@@ -92,8 +92,10 @@ struct _GstAudioFilterClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 GType   gst_audio_filter_get_type (void);
 
+GST_EXPORT
 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 7c766fd4e06b91352b30a58bf16afd4b043c8c9e..fcd1b087949199f4eb380273b0658f80361b821a 100644 (file)
--- a/gst-libs/gst/audio/gstaudioiec61937.h
+++ b/gst-libs/gst/audio/gstaudioiec61937.h
@@ -24,7 +24,10 @@
 
 #include <gst/audio/gstaudioringbuffer.h>
 
+GST_EXPORT
 guint       gst_audio_iec61937_frame_size  (const GstAudioRingBufferSpec * spec);
+
+GST_EXPORT
 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 ca58fb1bb397ca6d5f38b82bb69cbbd3e15b6c72..4cd5cbe69abbf27ebc1816590b70ade2bf49a77e 100644 (file)
--- a/gst-libs/gst/audio/gstaudiometa.h
+++ b/gst-libs/gst/audio/gstaudiometa.h
@@ -56,14 +56,19 @@ struct _GstAudioDownmixMeta {
   gfloat       **matrix;
 };
 
+GST_EXPORT
 GType gst_audio_downmix_meta_api_get_type (void);
+
+GST_EXPORT
 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
 GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels    (GstBuffer *buffer,
                                                                          const GstAudioChannelPosition *to_position,
                                                                          gint                           to_channels);
 
+GST_EXPORT
 GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer    *buffer,
                                                          const GstAudioChannelPosition *from_position,
                                                          gint                           from_channels,
@@ -106,11 +111,15 @@ struct _GstAudioClippingMeta {
   guint64   end;
 };
 
+GST_EXPORT
 GType gst_audio_clipping_meta_api_get_type (void);
+
+GST_EXPORT
 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
 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 9524721cb211136e2bcf4bcdbfbbdf68292982ec..c447c0ee3cd1f33fb881023236bd0105f158d93b 100644 (file)
--- a/gst-libs/gst/audio/gstaudioringbuffer.h
+++ b/gst-libs/gst/audio/gstaudioringbuffer.h
@@ -265,72 +265,122 @@ struct _GstAudioRingBufferClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 GType gst_audio_ring_buffer_get_type(void);
 
 /* callback stuff */
+
+GST_EXPORT
 void            gst_audio_ring_buffer_set_callback      (GstAudioRingBuffer *buf,
                                                          GstAudioRingBufferCallback cb,
                                                          gpointer user_data);
+
+GST_EXPORT
 void            gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf,
                                                          GstAudioRingBufferCallback cb,
                                                          gpointer user_data,
                                                          GDestroyNotify notify);
 
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_parse_caps      (GstAudioRingBufferSpec *spec, GstCaps *caps);
+
+GST_EXPORT
 void            gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec);
+
+GST_EXPORT
 void            gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec);
 
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_convert         (GstAudioRingBuffer * buf, GstFormat src_fmt,
                                                        gint64 src_val, GstFormat dest_fmt,
                                                        gint64 * dest_val);
 
 /* device state */
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_open_device     (GstAudioRingBuffer *buf);
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_close_device    (GstAudioRingBuffer *buf);
 
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_device_is_open  (GstAudioRingBuffer *buf);
 
 /* allocate resources */
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_acquire         (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_release         (GstAudioRingBuffer *buf);
 
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_is_acquired     (GstAudioRingBuffer *buf);
 
 /* set the device channel positions */
+
+GST_EXPORT
 void            gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position);
 
 /* activating */
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_activate        (GstAudioRingBuffer *buf, gboolean active);
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_is_active       (GstAudioRingBuffer *buf);
 
 /* flushing */
+
+GST_EXPORT
 void            gst_audio_ring_buffer_set_flushing    (GstAudioRingBuffer *buf, gboolean flushing);
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_is_flushing     (GstAudioRingBuffer *buf);
 
 /* playback/pause */
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_start           (GstAudioRingBuffer *buf);
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_pause           (GstAudioRingBuffer *buf);
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_stop            (GstAudioRingBuffer *buf);
 
 /* get status */
+
+GST_EXPORT
 guint           gst_audio_ring_buffer_delay           (GstAudioRingBuffer *buf);
+
+GST_EXPORT
 guint64         gst_audio_ring_buffer_samples_done    (GstAudioRingBuffer *buf);
 
+GST_EXPORT
 void            gst_audio_ring_buffer_set_sample      (GstAudioRingBuffer *buf, guint64 sample);
 
 /* clear all segments */
+
+GST_EXPORT
 void            gst_audio_ring_buffer_clear_all       (GstAudioRingBuffer *buf);
 
 /* commit samples */
+
+GST_EXPORT
 guint           gst_audio_ring_buffer_commit          (GstAudioRingBuffer * buf, guint64 *sample,
                                                        guint8 * data, gint in_samples,
                                                        gint out_samples, gint * accum);
 
 /* read samples */
+
+GST_EXPORT
 guint           gst_audio_ring_buffer_read            (GstAudioRingBuffer *buf, guint64 sample,
                                                        guint8 *data, guint len, GstClockTime *timestamp);
 
 /* Set timestamp on buffer */
+
+GST_EXPORT
 void            gst_audio_ring_buffer_set_timestamp   (GstAudioRingBuffer * buf, gint readseg, GstClockTime 
                                                        timestamp);
 
@@ -338,11 +388,18 @@ void            gst_audio_ring_buffer_set_timestamp   (GstAudioRingBuffer * buf,
 /* not yet implemented
 gboolean        gst_audio_ring_buffer_prepare_write   (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len);
 */
+
+GST_EXPORT
 gboolean        gst_audio_ring_buffer_prepare_read    (GstAudioRingBuffer *buf, gint *segment,
                                                        guint8 **readptr, gint *len);
+
+GST_EXPORT
 void            gst_audio_ring_buffer_clear           (GstAudioRingBuffer *buf, gint segment);
+
+GST_EXPORT
 void            gst_audio_ring_buffer_advance         (GstAudioRingBuffer *buf, guint advance);
 
+GST_EXPORT
 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 78b5bae842a16b70961587e51b8b18f4c2ba8562..04d429912b8810cf3dbb914835560141c903cdfe 100644 (file)
--- a/gst-libs/gst/audio/gstaudiosink.h
+++ b/gst-libs/gst/audio/gstaudiosink.h
@@ -97,6 +97,7 @@ struct _GstAudioSinkClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 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 db2392ac537e40e5b5f5aae0a22f600e53a1845d..04a673128ebd85bea24c91e5d2d87a6cf46c36f4 100644 (file)
--- a/gst-libs/gst/audio/gstaudiosrc.h
+++ b/gst-libs/gst/audio/gstaudiosrc.h
@@ -96,6 +96,7 @@ struct _GstAudioSrcClass {
   gpointer _gst_reserved[GST_PADDING];
 };
 
+GST_EXPORT
 GType gst_audio_src_get_type(void);
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

diff --git a/gst-libs/gst/audio/streamvolume.h b/gst-libs/gst/audio/streamvolume.h
index c8d16e9bc3a314052588294a8a7f3ab2895c9173..6d602c003553ffd8c5ec85a134cb0827a8d185eb 100644 (file)
--- a/gst-libs/gst/audio/streamvolume.h
+++ b/gst-libs/gst/audio/streamvolume.h
@@ -58,17 +58,26 @@ typedef enum {
   GST_STREAM_VOLUME_FORMAT_DB
 } GstStreamVolumeFormat;
 
+GST_EXPORT
 GType           gst_stream_volume_get_type        (void);
+
+GST_EXPORT
 void            gst_stream_volume_set_volume      (GstStreamVolume *volume,
                                                    GstStreamVolumeFormat format,
                                                    gdouble val);
+
+GST_EXPORT
 gdouble         gst_stream_volume_get_volume      (GstStreamVolume *volume,
                                                    GstStreamVolumeFormat format);
 
+GST_EXPORT
 void            gst_stream_volume_set_mute        (GstStreamVolume *volume,
                                                    gboolean mute);
+
+GST_EXPORT
 gboolean        gst_stream_volume_get_mute        (GstStreamVolume *volume);
 
+GST_EXPORT
 gdouble         gst_stream_volume_convert_volume  (GstStreamVolumeFormat from,
                                                    GstStreamVolumeFormat to,
                                                    gdouble val) G_GNUC_CONST;