gst-plugins-base: Fix several annotations

Add annotations for virtual methods when possible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1965>

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h
index b2f3664..678c1f2 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h
@@ -272,6 +272,11 @@ struct _GstAudioDecoderClass
   gboolean      (*set_format)         (GstAudioDecoder *dec,
                                        GstCaps *caps);
 
+  /**
+   * GstAudioDecoderClass::parse:
+   * @offset: (out):
+   * @length: (out):
+   */
   GstFlowReturn (*parse)              (GstAudioDecoder *dec,
                                        GstAdapter *adapter,
                                        gint *offset, gint *length);

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
index 439a2a0..ac71dd0 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
@@ -1675,7 +1675,7 @@ not_started:
 /**
  * gst_audio_ring_buffer_commit:
  * @buf: the #GstAudioRingBuffer to commit
- * @sample: the sample position of the data
+ * @sample: (inout): the sample position of the data
  * @data: (array length=in_samples): the data to commit
  * @in_samples: the number of samples in the data to commit
  * @out_samples: the number of samples to write to the ringbuffer

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h
index 06f65ff..0fce413 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h
@@ -98,7 +98,12 @@ struct _GstAudioSinkClass {
   gboolean (*unprepare) (GstAudioSink *sink);
   /* close the device */
   gboolean (*close)     (GstAudioSink *sink);
-  /* write samples to the device */
+  /**
+   * GstAudioSinkClass::write:
+   * @data: (type guint8) (array length=length): the sample data
+   *
+   * Write samples to the device.
+   */
   gint     (*write)     (GstAudioSink *sink, gpointer data, guint length);
   /* get number of frames queued in the device */
   guint    (*delay)     (GstAudioSink *sink);

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h
index 93cb5d0..41c7b38 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h
@@ -84,7 +84,13 @@ struct _GstAudioSrcClass {
   gboolean (*unprepare) (GstAudioSrc *src);
   /* close the device */
   gboolean (*close)     (GstAudioSrc *src);
-  /* read samples from the device */
+  /**
+   * GstAudioSrcClass::read:
+   * @data: (type guint8) (array length=length): the sample data
+   * @timestamp: (out): a #GstClockTime
+   *
+   * Read samples from the device.
+   */
   guint    (*read)      (GstAudioSrc *src, gpointer data, guint length,
       GstClockTime *timestamp);
   /* get number of frames queued in the device */

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h b/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h
index 368c723..4863c34 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h
+++ b/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.h
@@ -310,6 +310,10 @@ struct _GstVideoAggregatorClass
                                                    GstBuffer          *  outbuffer);
   GstFlowReturn      (*create_output_buffer)      (GstVideoAggregator *  videoaggregator,
                                                    GstBuffer          ** outbuffer);
+  /**
+   * GstVideoAggregatorClass::find_best_format:
+   * @at_least_one_alpha: (out):
+   */
   void               (*find_best_format)          (GstVideoAggregator *  vagg,
                                                    GstCaps            *  downstream_caps,
                                                    GstVideoInfo       *  best_info,

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/video/video-chroma.c b/subprojects/gst-plugins-base/gst-libs/gst/video/video-chroma.c
index 3d7f07c..c538bba 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/video/video-chroma.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/video/video-chroma.c
@@ -1111,8 +1111,8 @@ gst_video_chroma_resample_new (GstVideoChromaMethod method,
 /**
  * gst_video_chroma_resample_get_info:
  * @resample: a #GstVideoChromaResample
- * @n_lines: the number of input lines
- * @offset: the first line
+ * @n_lines: (out) (optional): the number of input lines
+ * @offset: (out) (optional): the first line
  *
  * The resampler must be fed @n_lines at a time. The first line should be
  * at @offset.

diff --git a/subprojects/gst-plugins-base/gst-libs/gst/video/video-scaler.c b/subprojects/gst-plugins-base/gst-libs/gst/video/video-scaler.c
index eb1effd..a4d8fb8 100644 (file)
--- a/subprojects/gst-plugins-base/gst-libs/gst/video/video-scaler.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/video/video-scaler.c
@@ -302,8 +302,8 @@ gst_video_scaler_get_max_taps (GstVideoScaler * scale)
  * gst_video_scaler_get_coeff:
  * @scale: a #GstVideoScaler
  * @out_offset: an output offset
- * @in_offset: result input offset
- * @n_taps: result n_taps
+ * @in_offset: (out) (optional): result input offset
+ * @n_taps: (out) (optional): result n_taps
  *
  * For a given pixel at @out_offset, get the first required input pixel at
  * @in_offset and the @n_taps filter coefficients.