gstreamer/gst/audioconvert/gstaudioconvert.c

778 lines
24 KiB
C
Raw Normal View History

/* GStreamer
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) 2011 Wim Taymans <wim.taymans at gmail dot com>
*
* gstaudioconvert.c: Convert audio to different audio formats automatically
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-audioconvert
*
* Audioconvert converts raw audio buffers between various possible formats.
* It supports integer to float conversion, width/depth conversion,
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* signedness and endianness conversion and channel transformations.
*
* <refsect2>
* <title>Example launch line</title>
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* |[
* gst-launch -v -m audiotestsrc ! audioconvert ! audio/x-raw,format=S8,channels=2 ! level ! fakesink silent=TRUE
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* ]| This pipeline converts audio to 8-bit. The level element shows that
* the output levels still match the one for a sine wave.
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* |[
* gst-launch -v -m audiotestsrc ! audioconvert ! vorbisenc ! fakesink silent=TRUE
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* ]| The vorbis encoder takes float audio data instead of the integer data
* generated by audiotestsrc.
* </refsect2>
*
* Last reviewed on 2006-03-02 (0.10.4)
*/
/*
* design decisions:
* - audioconvert converts buffers in a set of supported caps. If it supports
* a caps, it supports conversion from these caps to any other caps it
* supports. (example: if it does A=>B and A=>C, it also does B=>C)
* - audioconvert does not save state between buffers. Every incoming buffer is
* converted and the converted buffer is pushed out.
* conclusion:
* audioconvert is not supposed to be a one-element-does-anything solution for
* audio conversions.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstaudioconvert.h"
#include "gstchannelmix.h"
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
#include "gstaudioquantize.h"
#include "plugin.h"
GST_DEBUG_CATEGORY (audio_convert_debug);
GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
/*** DEFINITIONS **************************************************************/
/* type functions */
2004-11-25 20:36:29 +00:00
static void gst_audio_convert_dispose (GObject * obj);
/* gstreamer functions */
static gboolean gst_audio_convert_get_unit_size (GstBaseTransform * base,
GstCaps * caps, gsize * size);
static GstCaps *gst_audio_convert_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter);
2012-02-22 11:27:49 +00:00
static GstCaps *gst_audio_convert_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
static gboolean gst_audio_convert_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_audio_convert_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static GstFlowReturn gst_audio_convert_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
static void gst_audio_convert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* AudioConvert signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
ARG_DITHERING,
ARG_NOISE_SHAPING,
};
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element"); \
GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
#define gst_audio_convert_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioConvert, gst_audio_convert,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
/*** GSTREAMER PROTOTYPES *****************************************************/
#define STATIC_CAPS \
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
", layout = (string) interleaved")
static GstStaticPadTemplate gst_audio_convert_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
STATIC_CAPS);
static GstStaticPadTemplate gst_audio_convert_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
STATIC_CAPS);
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
#define GST_TYPE_AUDIO_CONVERT_DITHERING (gst_audio_convert_dithering_get_type ())
static GType
gst_audio_convert_dithering_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{DITHER_NONE, "No dithering",
"none"},
{DITHER_RPDF, "Rectangular dithering", "rpdf"},
{DITHER_TPDF, "Triangular dithering (default)", "tpdf"},
{DITHER_TPDF_HF, "High frequency triangular dithering", "tpdf-hf"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioConvertDithering", values);
}
return gtype;
}
#define GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING (gst_audio_convert_ns_get_type ())
static GType
gst_audio_convert_ns_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{NOISE_SHAPING_NONE, "No noise shaping (default)",
"none"},
{NOISE_SHAPING_ERROR_FEEDBACK, "Error feedback", "error-feedback"},
{NOISE_SHAPING_SIMPLE, "Simple 2-pole noise shaping", "simple"},
{NOISE_SHAPING_MEDIUM, "Medium 5-pole noise shaping", "medium"},
{NOISE_SHAPING_HIGH, "High 8-pole noise shaping", "high"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioConvertNoiseShaping", values);
}
return gtype;
}
/*** TYPE FUNCTIONS ***********************************************************/
static void
gst_audio_convert_class_init (GstAudioConvertClass * klass)
{
2004-11-25 20:36:29 +00:00
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *basetransform_class = GST_BASE_TRANSFORM_CLASS (klass);
2004-11-25 20:36:29 +00:00
gobject_class->dispose = gst_audio_convert_dispose;
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
gobject_class->set_property = gst_audio_convert_set_property;
gobject_class->get_property = gst_audio_convert_get_property;
2004-11-25 20:36:29 +00:00
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
g_object_class_install_property (gobject_class, ARG_DITHERING,
g_param_spec_enum ("dithering", "Dithering",
"Selects between different dithering methods.",
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
GST_TYPE_AUDIO_CONVERT_DITHERING, DITHER_TPDF,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
g_object_class_install_property (gobject_class, ARG_NOISE_SHAPING,
g_param_spec_enum ("noise-shaping", "Noise shaping",
"Selects between different noise shaping methods.",
GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING, NOISE_SHAPING_NONE,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_convert_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_convert_sink_template));
gst_element_class_set_details_simple (element_class,
"Audio converter", "Filter/Converter/Audio",
"Convert audio to different formats", "Benjamin Otte <otte@gnome.org>");
basetransform_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
basetransform_class->transform_caps =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform_caps);
basetransform_class->fixate_caps =
GST_DEBUG_FUNCPTR (gst_audio_convert_fixate_caps);
basetransform_class->set_caps =
GST_DEBUG_FUNCPTR (gst_audio_convert_set_caps);
basetransform_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform_ip);
basetransform_class->transform =
GST_DEBUG_FUNCPTR (gst_audio_convert_transform);
basetransform_class->passthrough_on_same_caps = TRUE;
}
static void
gst_audio_convert_init (GstAudioConvert * this)
{
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
this->dither = DITHER_TPDF;
this->ns = NOISE_SHAPING_NONE;
memset (&this->ctx, 0, sizeof (AudioConvertCtx));
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
2004-11-25 20:36:29 +00:00
}
static void
gst_audio_convert_dispose (GObject * obj)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
audio_convert_clean_context (&this->ctx);
G_OBJECT_CLASS (parent_class)->dispose (obj);
}
/*** GSTREAMER FUNCTIONS ******************************************************/
/* BaseTransform vmethods */
static gboolean
gst_audio_convert_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
GstAudioInfo info;
g_assert (size);
if (!gst_audio_info_from_caps (&info, caps))
goto parse_error;
*size = info.bpf;
GST_INFO_OBJECT (base, "unit_size = %" G_GSIZE_FORMAT, *size);
return TRUE;
parse_error:
{
GST_INFO_OBJECT (base, "failed to parse caps to get unit_size");
return FALSE;
}
}
/* copies the given caps */
static GstCaps *
gst_audio_convert_caps_remove_format_info (GstCaps * caps)
{
GstStructure *st;
gint i, n;
GstCaps *res;
guint64 channel_mask;
res = gst_caps_new_empty ();
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
st = gst_caps_get_structure (caps, i);
/* If this is already expressed by the existing caps
* skip this structure */
if (i > 0 && gst_caps_is_subset_structure (res, st))
continue;
st = gst_structure_copy (st);
gst_structure_remove_field (st, "format");
/* Only remove the channels and channel-mask for non-NONE layouts */
if (gst_structure_get (st, "channel-mask", GST_TYPE_BITMASK, &channel_mask,
NULL)) {
if (channel_mask != 0)
gst_structure_remove_fields (st, "channel-mask", "channels", NULL);
} else {
gst_structure_remove_fields (st, "channel-mask", "channels", NULL);
}
gst_caps_append_structure (res, st);
}
return res;
}
/* The caps can be transformed into any other caps with format info removed.
* However, we should prefer passthrough, so if passthrough is possible,
* put it first in the list. */
static GstCaps *
gst_audio_convert_transform_caps (GstBaseTransform * btrans,
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
GstCaps *tmp, *tmp2;
GstCaps *result;
/* Get all possible caps that we can transform to */
tmp = gst_audio_convert_caps_remove_format_info (caps);
if (filter) {
tmp2 = gst_caps_intersect_full (filter, tmp, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (tmp);
tmp = tmp2;
}
result = tmp;
GST_DEBUG_OBJECT (btrans, "transformed %" GST_PTR_FORMAT " into %"
GST_PTR_FORMAT, caps, result);
return result;
}
static const GstAudioChannelPosition default_positions[8][8] = {
/* 1 channel */
{
GST_AUDIO_CHANNEL_POSITION_MONO,
},
/* 2 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
},
/* 3 channels (2.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE1,
},
/* 4 channels (4.0) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
/* 5 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
},
/* 6 channels (5.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
},
/* 7 channels (6.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
},
/* 8 channels (7.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
}
};
static gint
n_bits_set (guint64 x)
{
gint i;
gint c = 0;
guint64 y = 1;
for (i = 0; i < 64; i++) {
if (x & y)
c++;
y <<= 1;
}
return c;
}
static guint64
find_suitable_mask (guint64 mask, gint n_chans)
{
guint64 intersection;
gint i;
i = 0;
g_assert (n_bits_set (mask) >= n_chans);
intersection = mask;
do {
intersection = intersection & ((~G_GUINT64_CONSTANT (0)) >> i);
i++;
} while (n_bits_set (intersection) > n_chans && i < 64);
if (i < 64)
return intersection;
return 0;
}
static void
gst_audio_convert_fixate_channels (GstBaseTransform * base, GstStructure * ins,
GstStructure * outs)
{
gint in_chans, out_chans;
guint64 in_mask = 0, out_mask = 0;
gboolean has_in_mask = FALSE, has_out_mask = FALSE;
if (!gst_structure_get_int (ins, "channels", &in_chans))
return; /* this shouldn't really happen, should it? */
if (!gst_structure_has_field (outs, "channels")) {
/* we could try to get the implied number of channels from the layout,
* but that seems overdoing it for a somewhat exotic corner case */
gst_structure_remove_field (outs, "channel-mask");
return;
}
/* ok, let's fixate the channels if they are not fixated yet */
gst_structure_fixate_field_nearest_int (outs, "channels", in_chans);
if (!gst_structure_get_int (outs, "channels", &out_chans)) {
/* shouldn't really happen ... */
gst_structure_remove_field (outs, "channel-mask");
return;
}
/* get the channel layout of the output if any */
has_out_mask = gst_structure_has_field (outs, "channel-mask");
if (has_out_mask) {
gst_structure_get (outs, "channel-mask", GST_TYPE_BITMASK, &out_mask, NULL);
} else {
/* channels == 1 => MONO */
if (out_chans == 2) {
out_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
has_out_mask = TRUE;
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask,
NULL);
}
}
/* get the channel layout of the input if any */
has_in_mask = gst_structure_has_field (ins, "channel-mask");
if (has_in_mask) {
gst_structure_get (ins, "channel-mask", GST_TYPE_BITMASK, &in_mask, NULL);
} else {
/* channels == 1 => MONO */
if (in_chans == 2) {
in_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
has_in_mask = TRUE;
} else if (in_chans > 2)
g_warning ("%s: Upstream caps contain no channel mask",
GST_ELEMENT_NAME (base));
}
if (!has_out_mask && out_chans == 1 && (in_chans != out_chans
|| !has_in_mask))
return; /* nothing to do, default layout will be assumed */
if (in_chans == out_chans && (has_in_mask || in_chans == 1)) {
/* same number of channels and no output layout: just use input layout */
if (!has_out_mask) {
/* in_chans == 1 handled above already */
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask, NULL);
return;
}
/* If both masks are the same we're done, this includes the NONE layout case */
if (in_mask == out_mask)
return;
/* if output layout is fixed already and looks sane, we're done */
if (n_bits_set (out_mask) == out_chans)
return;
if (n_bits_set (out_mask) < in_chans) {
/* Not much we can do here, this shouldn't just happen */
g_warning ("%s: Invalid downstream channel-mask with too few bits set",
GST_ELEMENT_NAME (base));
} else {
guint64 intersection;
/* if the output layout is not fixed, check if the output layout contains
* the input layout */
intersection = in_mask & out_mask;
if (n_bits_set (intersection) >= in_chans) {
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, in_mask,
NULL);
return;
}
/* output layout is not fixed and does not contain the input layout, so
* just pick the first possibility */
intersection = find_suitable_mask (out_mask, out_chans);
if (intersection) {
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,
NULL);
return;
}
}
/* ... else fall back to default layout (NB: out_layout is NULL here) */
GST_WARNING_OBJECT (base, "unexpected output channel layout");
} else {
guint64 intersection;
/* number of input channels != number of output channels:
* if this value contains a list of channel layouts (or even worse: a list
* with another list), just pick the first value and repeat until we find a
* channel position array or something else that's not a list; we assume
* the input if half-way sane and don't try to fall back on other list items
* if the first one is something unexpected or non-channel-pos-array-y */
if (n_bits_set (out_mask) >= out_chans) {
intersection = find_suitable_mask (out_mask, out_chans);
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, intersection,
NULL);
return;
}
/* what now?! Just ignore what we're given and use default positions */
GST_WARNING_OBJECT (base, "invalid or unexpected channel-positions");
}
/* missing or invalid output layout and we can't use the input layout for
* one reason or another, so just pick a default layout (we could be smarter
* and try to add/remove channels from the input layout, or pick a default
* layout based on LFE-presence in input layout, but let's save that for
* another day) */
if (out_chans > 0 && out_chans <= G_N_ELEMENTS (default_positions[0])) {
gint i;
GST_DEBUG_OBJECT (base, "using default channel layout as fallback");
out_mask = 0;
for (i = 0; i < out_chans; i++)
out_mask |= G_GUINT64_CONSTANT (1) << default_positions[out_chans - 1][i];
gst_structure_set (outs, "channel-mask", GST_TYPE_BITMASK, out_mask, NULL);
} else {
GST_ERROR_OBJECT (base, "Have no default layout for %d channels",
out_chans);
}
}
/* try to keep as many of the structure members the same by fixating the
* possible ranges; this way we convert the least amount of things as possible
*/
2012-02-22 11:27:49 +00:00
static GstCaps *
gst_audio_convert_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *ins, *outs;
2012-02-27 11:52:07 +00:00
GstCaps *result;
GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT
" based on caps %" GST_PTR_FORMAT, othercaps, caps);
2012-02-27 11:52:07 +00:00
result = gst_caps_intersect (othercaps, caps);
if (gst_caps_is_empty (result)) {
result = othercaps;
} else {
gst_caps_unref (othercaps);
}
2012-02-27 11:52:07 +00:00
/* fixate remaining fields */
result = gst_caps_make_writable (result);
2012-02-27 11:52:07 +00:00
ins = gst_caps_get_structure (caps, 0);
outs = gst_caps_get_structure (result, 0);
2012-02-27 11:52:07 +00:00
gst_audio_convert_fixate_channels (base, ins, outs);
2012-03-11 18:04:41 +00:00
result = gst_caps_fixate (result);
2012-02-27 11:52:07 +00:00
GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, result);
2012-02-22 11:27:49 +00:00
2012-02-27 11:52:07 +00:00
return result;
}
static gboolean
gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
GstAudioInfo in_info;
GstAudioInfo out_info;
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
if (!gst_audio_info_from_caps (&in_info, incaps))
goto invalid_in;
if (!gst_audio_info_from_caps (&out_info, outcaps))
goto invalid_out;
if (!audio_convert_prepare_context (&this->ctx, &in_info, &out_info,
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
this->dither, this->ns))
goto no_converter;
return TRUE;
/* ERRORS */
invalid_in:
{
GST_ERROR_OBJECT (base, "invalid input caps");
return FALSE;
}
invalid_out:
{
GST_ERROR_OBJECT (base, "invalid output caps");
return FALSE;
}
no_converter:
{
GST_ERROR_OBJECT (base, "could not find converter");
return FALSE;
}
}
static GstFlowReturn
gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
/* nothing to do here */
return GST_FLOW_OK;
}
static GstFlowReturn
gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFlowReturn ret;
GstAudioConvert *this = GST_AUDIO_CONVERT (base);
2012-01-20 15:11:54 +00:00
GstMapInfo srcmap, dstmap;
gint insize, outsize;
2012-01-20 15:11:54 +00:00
gint samples;
/* get amount of samples to convert. */
samples = gst_buffer_get_size (inbuf) / this->ctx.in.bpf;
/* get in/output sizes, to see if the buffers we got are of correct
* sizes */
if (!audio_convert_get_sizes (&this->ctx, samples, &insize, &outsize))
goto error;
if (insize == 0 || outsize == 0)
return GST_FLOW_OK;
/* get src and dst data */
2012-01-20 15:11:54 +00:00
gst_buffer_map (inbuf, &srcmap, GST_MAP_READ);
gst_buffer_map (outbuf, &dstmap, GST_MAP_WRITE);
/* check in and outsize */
2012-01-20 15:11:54 +00:00
if (srcmap.size < insize)
goto wrong_size;
2012-01-20 15:11:54 +00:00
if (dstmap.size < outsize)
goto wrong_size;
/* and convert the samples */
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
2012-01-20 15:11:54 +00:00
if (!audio_convert_convert (&this->ctx, srcmap.data, dstmap.data,
samples, gst_buffer_is_writable (inbuf)))
goto convert_error;
} else {
/* Create silence buffer */
2012-01-20 15:11:54 +00:00
gst_audio_format_fill_silence (this->ctx.out.finfo, dstmap.data, outsize);
}
ret = GST_FLOW_OK;
done:
2012-01-20 15:11:54 +00:00
gst_buffer_unmap (outbuf, &dstmap);
gst_buffer_unmap (inbuf, &srcmap);
return ret;
/* ERRORS */
error:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL), ("cannot get input/output sizes for %d samples", samples));
return GST_FLOW_ERROR;
}
wrong_size:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL),
("input/output buffers are of wrong size in: %" G_GSIZE_FORMAT " < %d"
" or out: %" G_GSIZE_FORMAT " < %d",
2012-01-20 15:11:54 +00:00
srcmap.size, insize, dstmap.size, outsize));
ret = GST_FLOW_ERROR;
goto done;
}
convert_error:
{
GST_ELEMENT_ERROR (this, STREAM, FORMAT,
(NULL), ("error while converting"));
ret = GST_FLOW_ERROR;
goto done;
}
}
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
static void
gst_audio_convert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
switch (prop_id) {
case ARG_DITHERING:
this->dither = g_value_get_enum (value);
break;
case ARG_NOISE_SHAPING:
this->ns = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
switch (prop_id) {
case ARG_DITHERING:
g_value_set_enum (value, this->dither);
break;
case ARG_NOISE_SHAPING:
g_value_set_enum (value, this->ns);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}