docs: add new audio API

diff --git a/docs/libs/gst-plugins-base-libs-docs.sgml b/docs/libs/gst-plugins-base-libs-docs.sgml
index 640fba0..57e2407 100644 (file)
--- a/docs/libs/gst-plugins-base-libs-docs.sgml
+++ b/docs/libs/gst-plugins-base-libs-docs.sgml
@@ -65,6 +65,8 @@
       <xi:include href="xml/gstaudiobasesink.xml" />
       <xi:include href="xml/gstaudiobasesrc.xml" />
       <xi:include href="xml/gstaudiochannels.xml" />
+      <xi:include href="xml/gstaudioconverter.xml" />
+      <xi:include href="xml/gstaudioquantize.xml" />
       <xi:include href="xml/gstaudioringbuffer.xml" />
       <xi:include href="xml/gstaudioiec61937.xml" />
       <xi:include href="xml/gststreamvolume.xml" />

diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt
index 410c2cc..661280d 100644 (file)
--- a/docs/libs/gst-plugins-base-libs-sections.txt
+++ b/docs/libs/gst-plugins-base-libs-sections.txt
@@ -501,22 +501,66 @@ GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD
 
 <SECTION>
 <FILE>gstaudiochannels</FILE>
-<INCLUDE>gst/audio/audio-channels.h</INCLUDE>
+<INCLUDE>gst/audio/audio.h</INCLUDE>
 GstAudioChannelPosition
 GST_AUDIO_CHANNEL_POSITION_MASK
 gst_audio_channel_positions_to_mask
 gst_audio_channel_positions_from_mask
 gst_audio_channel_positions_to_valid_order
+gst_audio_channel_get_fallback_mask
 gst_audio_check_valid_channel_positions
 gst_audio_buffer_reorder_channels
 gst_audio_reorder_channels
 gst_audio_get_channel_reorder_map
+GstAudioChannelMix
+GstAudioChannelMixFlags
+gst_audio_channel_mix_new
+gst_audio_channel_mix_free
+gst_audio_channel_mix_is_passthrough
+gst_audio_channel_mix_samples
 <SUBSECTION Standard>
 GST_TYPE_AUDIO_CHANNEL_POSITION
 gst_audio_channel_position_get_type
 </SECTION>
 
 <SECTION>
+<FILE>gstaudioconverter</FILE>
+<TITLE>GstAudioConverter</TITLE>
+<INCLUDE>gst/audio/audio.h</INCLUDE>
+GstAudioConverter
+GstAudioConverterFlags
+gst_audio_converter_new
+gst_audio_converter_free
+gst_audio_converter_samples
+gst_audio_converter_get_in_frames
+gst_audio_converter_get_max_latency
+gst_audio_converter_get_out_frames
+GST_AUDIO_CONVERTER_OPT_DITHER_METHOD
+GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD
+GST_AUDIO_CONVERTER_OPT_QUANTIZATION
+gst_audio_converter_set_config
+gst_audio_converter_get_config
+<SUBSECTION Standard>
+gst_audio_converter_flags_get_type
+</SECTION>
+
+<SECTION>
+<FILE>gstaudioquantize</FILE>
+<TITLE>GstAudioQuantize</TITLE>
+<INCLUDE>gst/audio/audio.h</INCLUDE>
+GstAudioQuantize
+GstAudioQuantizeFlags
+GstAudioDitherMethod
+GstAudioNoiseShapingMethod
+gst_audio_quantize_free
+gst_audio_quantize_new
+gst_audio_quantize_samples
+<SUBSECTION Standard>
+gst_audio_quantize_flags_get_type
+GST_TYPE_AUDIO_QUANTIZE_FLAGS
+</SECTION>
+
+<SECTION>
 <FILE>gstaudioringbuffer</FILE>
 <INCLUDE>gst/audio/gstaudioringbuffer.h</INCLUDE>
 GstAudioRingBuffer