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audioconvert: cleanups and add some docs
Add docs for the internal audioconvert object before moving it to the audio library. Remove get_sizes and implement the trivial logic in the element. Remove some unused orc functions
This commit is contained in:
Wim Taymans
parent
4ae24dcb25
commit
1635bc0a45
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@ -1,7 +1,8 @@
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/* GStreamer
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* Copyright (C) 2005 Wim Taymans <wim at fluendo dot com>
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* (C) 2015 Wim Taymans <wim.taymans@gmail.com>
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*
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* audioconvert.c: Convert audio to different audio formats automatically
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* audioconverter.c: Convert audio to different audio formats automatically
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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@ -30,6 +31,28 @@
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#include "audioconvert.h"
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#include "gstaudioconvertorc.h"
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/**
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* SECTION:audioconverter
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* @short_description: Generic audio conversion
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*
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* <refsect2>
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* <para>
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* This object is used to convert audio samples from one format to another.
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* The object can perform conversion of:
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* <itemizedlist>
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* <listitem><para>
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* audio format with optional dithering and noise shaping
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* </para></listitem>
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* <listitem><para>
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* audio samplerate
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* </para></listitem>
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* <listitem><para>
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* audio channels and channel layout
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* </para></listitem>
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* </para>
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* </refsect2>
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*/
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typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
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/**
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@ -160,9 +183,19 @@ gst_audio_converter_get_config (GstAudioConverter * convert)
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return convert->config;
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}
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/**
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* gst_audio_converter_new: (skip)
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* @in: a source #GstAudioInfo
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* @out: a destination #GstAudioInfo
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* @config: (transfer full): a #GstStructure with configuration options
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*
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* Create a new #GstAudioConverter that is able to convert between @in and @out
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* audio formats.
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*
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* @config contains extra configuration options, see #GST_VIDEO_CONVERTER_OPT_*
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* parameters for details about the options and values.
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*
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* Returns: a #GstAudioConverter or %NULL if conversion is not possible.
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*/
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GstAudioConverter *
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gst_audio_converter_new (GstAudioInfo * in, GstAudioInfo * out,
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@ -289,6 +322,12 @@ unpositioned:
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}
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}
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/**
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* gst_audio_converter_free:
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* @convert: a #GstAudioConverter
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*
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* Free a previously allocated @convert instance.
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*/
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void
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gst_audio_converter_free (GstAudioConverter * convert)
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{
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@ -308,20 +347,18 @@ gst_audio_converter_free (GstAudioConverter * convert)
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g_slice_free (GstAudioConverter, convert);
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}
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gboolean
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gst_audio_converter_get_sizes (GstAudioConverter * convert, gint samples,
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gint * srcsize, gint * dstsize)
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{
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g_return_val_if_fail (convert != NULL, FALSE);
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if (srcsize)
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*srcsize = samples * convert->in.bpf;
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if (dstsize)
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*dstsize = samples * convert->out.bpf;
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return TRUE;
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}
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/**
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* gst_audio_converter_samples:
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* @convert: a #GstAudioConverter
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* @flags: extra #GstAudioConverterFlags
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* @src: source samples
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* @dst: output samples
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* @samples: number of samples
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*
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* Perform the conversion @src to @dst using @convert.
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*
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* Returns: %TRUE is the conversion could be performed.
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*/
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gboolean
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gst_audio_converter_samples (GstAudioConverter * convert,
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GstAudioConverterFlags flags, gpointer src, gpointer dst, gint samples)
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@ -80,10 +80,6 @@ gboolean gst_audio_converter_set_config (GstAudioConverter * con
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const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert);
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gboolean gst_audio_converter_get_sizes (GstAudioConverter * convert,
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gint samples,
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gint * srcsize, gint * dstsize);
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gboolean gst_audio_converter_samples (GstAudioConverter * convert,
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GstAudioConverterFlags flags,
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gpointer src, gpointer dst,
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@ -198,7 +198,8 @@ gst_audio_convert_dispose (GObject * obj)
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{
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GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
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gst_audio_converter_free (this->convert);
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if (this->convert)
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gst_audio_converter_free (this->convert);
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G_OBJECT_CLASS (parent_class)->dispose (obj);
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}
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@ -708,9 +709,8 @@ gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
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/* get in/output sizes, to see if the buffers we got are of correct
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* sizes */
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if (!gst_audio_converter_get_sizes (this->convert, samples, &insize,
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&outsize))
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goto error;
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insize = samples * this->in_info.bpf;
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outsize = samples * this->out_info.bpf;
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if (insize == 0 || outsize == 0)
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return GST_FLOW_OK;
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@ -752,12 +752,6 @@ done:
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return ret;
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/* ERRORS */
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error:
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{
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GST_ELEMENT_ERROR (this, STREAM, FORMAT,
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(NULL), ("cannot get input/output sizes for %d samples", samples));
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return GST_FLOW_ERROR;
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}
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wrong_size:
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{
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GST_ELEMENT_ERROR (this, STREAM, FORMAT,
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@ -13,24 +13,3 @@ divd d1, t1, 2147483648.0L
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muld t1, s1, 2147483648.0L
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convdl d1, t1
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.function audio_convert_orc_int_bias
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.dest 4 d1 gint32
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.source 4 s1 gint32
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.param 4 bias gint32
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.param 4 mask gint32
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.temp 4 t1
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addssl t1, s1, bias
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andl d1, t1, mask
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.function audio_convert_orc_int_dither
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.dest 4 d1 gint32
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.source 4 s1 gint32
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.source 4 dither gint32
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.param 4 mask gint32
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.temp 4 t1
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addssl t1, s1, dither
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andl d1, t1, mask
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@ -818,14 +818,15 @@ gst_channel_mix_mix_double (GstChannelMix * mix,
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/**
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* gst_channel_mix_mix:
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* @mix:
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* @format:
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* @layout:
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* @in_data:
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* @out_data:
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* @samples:
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* @mix: a #GstChannelMix
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* @format: a #GstAudioFormat
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* @layout: a #GstAudioLayout
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* @in_data: input samples
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* @out_data: output samples
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* @samples: number of samples
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*
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* Perform channel mixing
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* Perform channel mixing on @in_data and write the result to @out_data.
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* @in_data and @out_data need to be in @format and @layout.
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*/
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void
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gst_channel_mix_mix (GstChannelMix * mix, GstAudioFormat format,
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