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| author | Tim-Philipp Müller <tim@centricular.com> | 2016-01-05 22:52:34 +0000 |
|---|---|---|
| committer | Tim-Philipp Müller <tim@centricular.com> | 2016-01-05 22:52:34 +0000 |
| commit | 63d7a2a89a304f1e69e434d375af89a220df1104 (patch) | |
| tree | 34d5365afb794095068d4a30cf06dde8e54e63a7 | |
| parent | 7ffd9ce2914b386e0f8deac5ab71c3cf6876841f (diff) | |
docs: add new audio API
| -rw-r--r-- | docs/libs/gst-plugins-base-libs-docs.sgml | 2 | ||||
| -rw-r--r-- | docs/libs/gst-plugins-base-libs-sections.txt | 46 |
2 files changed, 47 insertions, 1 deletions
diff --git a/docs/libs/gst-plugins-base-libs-docs.sgml b/docs/libs/gst-plugins-base-libs-docs.sgml index 640fba007..57e240780 100644 --- 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 410c2ccc6..661280dca 100644 --- 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 |