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audio: Add/fix various annotations

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
This commit is contained in:
Sebastian Dröge 2022-10-14 22:04:00 +03:00 committed by GStreamer Marge Bot
parent 2e5c73fff7
commit e0b06df223
9 changed files with 28 additions and 31 deletions
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5
subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c
View file

@ -817,8 +817,7 @@ DEFINE_FLOAT_MIX_FUNC (double, planar, planar);
*
* Create a new channel mixer object for the given parameters.
*
* Returns: a new #GstAudioChannelMixer object, or %NULL if @format isn't supported,
* @matrix is invalid, or @matrix is %NULL and @in_channels != @out_channels.
* Returns: a new #GstAudioChannelMixer object.
* Free with gst_audio_channel_mixer_free() after usage.
*
* Since: 1.14
@ -980,7 +979,7 @@ gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags,
*
* Create a new channel mixer object for the given parameters.
*
* Returns: a new #GstAudioChannelMixer object, or %NULL if @format isn't supported.
* Returns: a new #GstAudioChannelMixer object.
* Free with gst_audio_channel_mixer_free() after usage.
*/
GstAudioChannelMixer *

2
subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c
View file

@ -1320,7 +1320,7 @@ converter_resample (GstAudioConverter * convert,
* @config contains extra configuration options, see `GST_AUDIO_CONVERTER_OPT_*`
* parameters for details about the options and values.
*
* Returns: a #GstAudioConverter or %NULL if conversion is not possible.
* Returns: (nullable): a #GstAudioConverter or %NULL if conversion is not possible.
*/
GstAudioConverter *
gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,

2
subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c
View file

@ -326,7 +326,7 @@ invalid_channel_mask:
*
* Parse @caps to generate a #GstAudioInfo.
*
* Returns: A #GstAudioInfo, or %NULL if @caps couldn't be parsed
* Returns: (nullable): A #GstAudioInfo, or %NULL if @caps couldn't be parsed
* Since: 1.20
*/
GstAudioInfo *

3
subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c
View file

@ -1342,8 +1342,7 @@ gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
*
* Make a new resampler.
*
* Returns: (skip) (transfer full): The new #GstAudioResampler, or
* %NULL on failure.
* Returns: (skip) (transfer full): The new #GstAudioResampler.
*/
GstAudioResampler *
gst_audio_resampler_new (GstAudioResamplerMethod method,

5
subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c
View file

@ -69,7 +69,7 @@ ensure_debug_category (void)
* After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: (transfer full): %NULL if the buffer is completely outside the configured segment,
* Returns: (transfer full) (nullable): %NULL if the buffer is completely outside the configured segment,
* otherwise the clipped buffer is returned.
*
* If the buffer has no timestamp, it is assumed to be inside the segment and
@ -267,8 +267,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
* After calling this function the caller does not own a reference to
* @buffer anymore.
*
* Returns: (transfer full): the truncated buffer or %NULL if the arguments
* were invalid
* Returns: (transfer full): the truncated buffer
*
* Since: 1.16
*/

2
subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c
View file

@ -2256,7 +2256,7 @@ sync_latency_failed:
* call the ::create_ringbuffer vmethod and will set @sink as the parent of
* the returned buffer (see gst_object_set_parent()).
*
* Returns: (transfer none): The new ringbuffer of @sink.
* Returns: (transfer none) (nullable): The new ringbuffer of @sink.
*/
GstAudioRingBuffer *
gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)

2
subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c
View file

@ -1086,7 +1086,7 @@ got_error:
* the ::create_ringbuffer vmethod and will set @src as the parent of the
* returned buffer (see gst_object_set_parent()).
*
* Returns: (transfer none): The new ringbuffer of @src.
* Returns: (transfer none) (nullable): The new ringbuffer of @src.
*/
GstAudioRingBuffer *
gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc * src)

20
subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c
View file

@ -1272,7 +1272,7 @@ foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
/**
* gst_audio_decoder_finish_subframe:
* @dec: a #GstAudioDecoder
* @buf: (transfer full) (allow-none): decoded data
* @buf: (transfer full) (nullable): decoded data
*
* Collects decoded data and pushes it downstream. This function may be called
* multiple times for a given input frame.
@ -1306,7 +1306,7 @@ gst_audio_decoder_finish_subframe (GstAudioDecoder * dec, GstBuffer * buf)
/**
* gst_audio_decoder_finish_frame:
* @dec: a #GstAudioDecoder
* @buf: (transfer full) (allow-none): decoded data
* @buf: (transfer full) (nullable): decoded data
* @frames: number of decoded frames represented by decoded data
*
* Collects decoded data and pushes it downstream.
@ -2762,8 +2762,8 @@ gst_audio_decoder_propose_allocation_default (GstAudioDecoder * dec,
/**
* gst_audio_decoder_proxy_getcaps:
* @decoder: a #GstAudioDecoder
* @caps: (allow-none): initial caps
* @filter: (allow-none): filter caps
* @caps: (nullable): initial caps
* @filter: (nullable): filter caps
*
* Returns caps that express @caps (or sink template caps if @caps == NULL)
* restricted to rate/channels/... combinations supported by downstream
@ -3414,8 +3414,8 @@ gst_audio_decoder_set_latency (GstAudioDecoder * dec,
/**
* gst_audio_decoder_get_latency:
* @dec: a #GstAudioDecoder
* @min: (out) (allow-none): a pointer to storage to hold minimum latency
* @max: (out) (allow-none): a pointer to storage to hold maximum latency
* @min: (out) (optional): a pointer to storage to hold minimum latency
* @max: (out) (optional): a pointer to storage to hold maximum latency
*
* Sets the variables pointed to by @min and @max to the currently configured
* latency.
@ -3457,7 +3457,7 @@ gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
/**
* gst_audio_decoder_set_allocation_caps:
* @dec: a #GstAudioDecoder
* @allocation_caps: (allow-none): a #GstCaps or %NULL
* @allocation_caps: (nullable): a #GstCaps or %NULL
*
* Sets a caps in allocation query which are different from the set
* pad's caps. Use this function before calling
@ -3706,7 +3706,7 @@ gst_audio_decoder_get_needs_format (GstAudioDecoder * dec)
/**
* gst_audio_decoder_merge_tags:
* @dec: a #GstAudioDecoder
* @tags: (allow-none): a #GstTagList to merge, or NULL
* @tags: (nullable): a #GstTagList to merge, or NULL
* @mode: the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
*
* Sets the audio decoder tags and how they should be merged with any
@ -3796,9 +3796,9 @@ fallback:
/**
* gst_audio_decoder_get_allocator:
* @dec: a #GstAudioDecoder
* @allocator: (out) (allow-none) (transfer full): the #GstAllocator
* @allocator: (out) (optional) (nullable) (transfer full): the #GstAllocator
* used
* @params: (out) (allow-none) (transfer full): the
* @params: (out) (optional) (transfer full): the
* #GstAllocationParams of @allocator
*
* Lets #GstAudioDecoder sub-classes to know the memory @allocator

18
subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c
View file

@ -762,7 +762,7 @@ foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
/**
* gst_audio_encoder_finish_frame:
* @enc: a #GstAudioEncoder
* @buffer: (transfer full) (allow-none): encoded data
* @buffer: (transfer full) (nullable): encoded data
* @samples: number of samples (per channel) represented by encoded data
*
* Collects encoded data and pushes encoded data downstream.
@ -1531,8 +1531,8 @@ refuse_caps:
/**
* gst_audio_encoder_proxy_getcaps:
* @enc: a #GstAudioEncoder
* @caps: (allow-none): initial caps
* @filter: (allow-none): filter caps
* @caps: (nullable): initial caps
* @filter: (nullable): filter caps
*
* Returns caps that express @caps (or sink template caps if @caps == NULL)
* restricted to channel/rate combinations supported by downstream elements
@ -2372,8 +2372,8 @@ gst_audio_encoder_set_latency (GstAudioEncoder * enc,
/**
* gst_audio_encoder_get_latency:
* @enc: a #GstAudioEncoder
* @min: (out) (allow-none): a pointer to storage to hold minimum latency
* @max: (out) (allow-none): a pointer to storage to hold maximum latency
* @min: (out) (optional): a pointer to storage to hold minimum latency
* @max: (out) (optional): a pointer to storage to hold maximum latency
*
* Sets the variables pointed to by @min and @max to the currently configured
* latency.
@ -2416,7 +2416,7 @@ gst_audio_encoder_set_headers (GstAudioEncoder * enc, GList * headers)
/**
* gst_audio_encoder_set_allocation_caps:
* @enc: a #GstAudioEncoder
* @allocation_caps: (allow-none): a #GstCaps or %NULL
* @allocation_caps: (nullable): a #GstCaps or %NULL
*
* Sets a caps in allocation query which are different from the set
* pad's caps. Use this function before calling
@ -2711,7 +2711,7 @@ gst_audio_encoder_get_drainable (GstAudioEncoder * enc)
/**
* gst_audio_encoder_merge_tags:
* @enc: a #GstAudioEncoder
* @tags: (allow-none): a #GstTagList to merge, or NULL to unset
* @tags: (nullable): a #GstTagList to merge, or NULL to unset
* previously-set tags
* @mode: the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
*
@ -2990,9 +2990,9 @@ fallback:
/**
* gst_audio_encoder_get_allocator:
* @enc: a #GstAudioEncoder
* @allocator: (out) (allow-none) (transfer full): the #GstAllocator
* @allocator: (out) (optional) (nullable) (transfer full): the #GstAllocator
* used
* @params: (out) (allow-none) (transfer full): the
* @params: (out) (optional) (transfer full): the
* #GstAllocationParams of @allocator
*
* Lets #GstAudioEncoder sub-classes to know the memory @allocator