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Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567...
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-spectrum.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init), (gst_audio_reverb_class_init), (gst_audio_reverb_init), (gst_audio_reverb_finalize), (gst_audio_reverb_set_property), (gst_audio_reverb_get_property), (gst_audio_reverb_setup), (gst_audio_reverb_stop), (gst_audio_reverb_transform_ip): * gst/audiofx/audioreverb.h: * tests/check/Makefile.am: * tests/check/elements/audioreverb.c: (setup_reverb), (cleanup_reverb), (GST_START_TEST), (audioreverb_suite): Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567874.
This commit is contained in:
parent
d912a42065
commit
344a9f4229
14 changed files with 798 additions and 3 deletions
23
ChangeLog
23
ChangeLog
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@ -1,3 +1,26 @@
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2009-01-19 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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* docs/plugins/inspect/plugin-audiofx.xml:
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* docs/plugins/inspect/plugin-spectrum.xml:
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* gst/audiofx/Makefile.am:
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* gst/audiofx/audiofx.c: (plugin_init):
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* gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init),
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(gst_audio_reverb_class_init), (gst_audio_reverb_init),
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(gst_audio_reverb_finalize), (gst_audio_reverb_set_property),
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(gst_audio_reverb_get_property), (gst_audio_reverb_setup),
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(gst_audio_reverb_stop), (gst_audio_reverb_transform_ip):
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* gst/audiofx/audioreverb.h:
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* tests/check/Makefile.am:
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* tests/check/elements/audioreverb.c: (setup_reverb),
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(cleanup_reverb), (GST_START_TEST), (audioreverb_suite):
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Add an echo/reverb filter to the audiofx plugin, with configurable
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echo delay, intensity and feedback. Fixes bug #567874.
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2009-01-19 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
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@ -106,6 +106,7 @@ EXTRA_HFILES = \
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$(top_srcdir)/gst/alpha/gstalphacolor.h \
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$(top_srcdir)/gst/apetag/gstapedemux.h \
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$(top_srcdir)/gst/audiofx/audioamplify.h \
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$(top_srcdir)/gst/audiofx/audioreverb.h \
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$(top_srcdir)/gst/audiofx/audiodynamic.h \
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$(top_srcdir)/gst/audiofx/audioinvert.h \
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$(top_srcdir)/gst/audiofx/audiokaraoke.h \
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@ -23,6 +23,7 @@
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<xi:include href="xml/element-audiowsincband.xml" />
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<xi:include href="xml/element-audiowsinclimit.xml" />
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<xi:include href="xml/element-audiofirfilter.xml" />
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<xi:include href="xml/element-audioreverb.xml" />
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<xi:include href="xml/element-audiodynamic.xml" />
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<xi:include href="xml/element-audioinvert.xml" />
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<xi:include href="xml/element-audiopanorama.xml" />
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@ -116,6 +116,22 @@ GST_TYPE_AUDIO_IIR_FILTER
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gst_audio_iir_filter_get_type
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</SECTION>
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<SECTION>
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<FILE>element-audioreverb</FILE>
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<TITLE>audioreverb</TITLE>
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GstAudioReverb
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<SUBSECTION Standard>
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GstAudioReverbClass
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GstAudioReverbProcessFunc
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GST_AUDIO_REVERB
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GST_AUDIO_REVERB_CLASS
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GST_AUDIO_REVERB_GET_CLASS
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GST_IS_AUDIO_REVERB
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GST_IS_AUDIO_REVERB_CLASS
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GST_TYPE_AUDIO_REVERB
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gst_audio_reverb_get_type
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</SECTION>
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<SECTION>
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<FILE>element-audiodynamic</FILE>
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<TITLE>audiodynamic</TITLE>
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@ -19708,3 +19708,63 @@
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<DEFAULT></DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioDelay::delay</NAME>
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<TYPE>guint64</TYPE>
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<RANGE>>= 1</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Delay</NICK>
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<BLURB>Delay in nanoseconds.</BLURB>
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<DEFAULT>1</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioDelay::feedback</NAME>
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<TYPE>gfloat</TYPE>
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<RANGE>[0,1]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Feedback</NICK>
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<BLURB>Amount of feedback.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioDelay::intensity</NAME>
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<TYPE>gfloat</TYPE>
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<RANGE>[0,1]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Intensity</NICK>
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<BLURB>Intensity of the echo.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioReverb::delay</NAME>
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<TYPE>guint64</TYPE>
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<RANGE>>= 1</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Delay</NICK>
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<BLURB>Delay of the echo in nanoseconds.</BLURB>
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<DEFAULT>1</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioReverb::feedback</NAME>
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<TYPE>gfloat</TYPE>
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<RANGE>[0,1]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Feedback</NICK>
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<BLURB>Amount of feedback.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioReverb::intensity</NAME>
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<TYPE>gfloat</TYPE>
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<RANGE>[0,1]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Intensity</NICK>
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<BLURB>Intensity of the echo.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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@ -64,6 +64,7 @@ GObject
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GstAudioWSincLimit
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GstAudioWSincBand
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GstAudioFIRFilter
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GstAudioReverb
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GstIirEqualizer
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GstIirEqualizerNBands
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GstIirEqualizer3Bands
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@ -221,6 +222,8 @@ GObject
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GstRegistry
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GstRingBuffer
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GstSignalObject
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GstMixerTrack
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GstMixerOptions
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GstCmmlTagStream
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GstCmmlTagHead
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GstCmmlTagClip
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@ -198,6 +198,27 @@
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</caps>
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</pads>
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</element>
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<element>
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<name>audioreverb</name>
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<longname>Audio reverb</longname>
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<class>Filter/Effect/Audio</class>
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<description>Adds an echo or reverb effect to an audio stream</description>
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<author>Sebastian Dröge <sebastian.droege@collabora.co.uk></author>
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<pads>
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<caps>
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<name>sink</name>
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<direction>sink</direction>
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<presence>always</presence>
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<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
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</caps>
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<caps>
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<name>src</name>
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<direction>source</direction>
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<presence>always</presence>
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<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
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</caps>
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</pads>
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</element>
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<element>
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<name>audiowsincband</name>
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<longname>Band pass & band reject filter</longname>
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@ -14,7 +14,7 @@
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<longname>Spectrum analyzer</longname>
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<class>Filter/Analyzer/Audio</class>
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<description>Run an FFT on the audio signal, output spectrum data</description>
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<author>Erik Walthinsen <omega@cse.ogi.edu>, Stefan Kost <ensonic@users.sf.net>, Sebastian Dröge <slomo@circular-chaos.org></author>
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<author>Erik Walthinsen <omega@cse.ogi.edu>, Stefan Kost <ensonic@users.sf.net>, Sebastian Dröge <sebastian.droege@collabora.co.uk></author>
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<pads>
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<caps>
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<name>sink</name>
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@ -16,7 +16,8 @@ libgstaudiofx_la_SOURCES = audiofx.c\
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audiofxbasefirfilter.c \
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audiowsincband.c \
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audiowsinclimit.c \
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audiofirfilter.c
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audiofirfilter.c \
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audioreverb.c
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# flags used to compile this plugin
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libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
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@ -46,5 +47,6 @@ noinst_HEADERS = audiopanorama.h \
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audiowsincband.h \
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audiowsinclimit.h \
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audiofirfilter.h \
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audioreverb.h \
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math_compat.h
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@ -36,6 +36,7 @@
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#include "audiowsincband.h"
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#include "audiowsinclimit.h"
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#include "audiofirfilter.h"
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#include "audioreverb.h"
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/* entry point to initialize the plug-in
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* initialize the plug-in itself
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@ -69,7 +70,9 @@ plugin_init (GstPlugin * plugin)
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gst_element_register (plugin, "audiowsincband", GST_RANK_NONE,
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GST_TYPE_AUDIO_WSINC_BAND) &&
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gst_element_register (plugin, "audiofirfilter", GST_RANK_NONE,
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GST_TYPE_AUDIO_FIR_FILTER));
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GST_TYPE_AUDIO_FIR_FILTER) &&
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gst_element_register (plugin, "audioreverb", GST_RANK_NONE,
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GST_TYPE_AUDIO_REVERB));
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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367
gst/audiofx/audioreverb.c
Normal file
367
gst/audiofx/audioreverb.c
Normal file
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@ -0,0 +1,367 @@
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/*
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* GStreamer
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* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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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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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audioreverb
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*
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* <refsect2>
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* audioreverb adds an echo or revert effect to an audio stream. The echo
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* reverb, intensity and the percentage of feedback can be configured.
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* <para>
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* <programlisting>
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* gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioreverb reverb=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
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* gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioreverb reverb=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
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* </programlisting>
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* </para>
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* </refsect2>
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*
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* Since: 0.10.12
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <gst/controller/gstcontroller.h>
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#include "audioreverb.h"
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#define GST_CAT_DEFAULT gst_audio_reverb_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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enum
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{
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PROP_0,
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PROP_DELAY,
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PROP_INTENSITY,
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PROP_FEEDBACK
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};
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#define ALLOWED_CAPS \
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"audio/x-raw-float," \
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" width=(int) { 32, 64 }, " \
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" endianness=(int)BYTE_ORDER," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]"
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_audio_reverb_debug, "audioreverb", 0, "audioreverb element");
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GST_BOILERPLATE_FULL (GstAudioReverb, gst_audio_reverb, GstAudioFilter,
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GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
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static void gst_audio_reverb_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_reverb_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_audio_reverb_finalize (GObject * object);
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static gboolean gst_audio_reverb_setup (GstAudioFilter * self,
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GstRingBufferSpec * format);
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static gboolean gst_audio_reverb_stop (GstBaseTransform * base);
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static GstFlowReturn gst_audio_reverb_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static void gst_audio_reverb_transform_float (GstAudioReverb * self,
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gfloat * data, guint num_samples);
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static void gst_audio_reverb_transform_double (GstAudioReverb * self,
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gdouble * data, guint num_samples);
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/* GObject vmethod implementations */
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static void
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gst_audio_reverb_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstCaps *caps;
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gst_element_class_set_details_simple (element_class, "Audio reverb",
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"Filter/Effect/Audio",
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"Adds an echo or reverb effect to an audio stream",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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}
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static void
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gst_audio_reverb_class_init (GstAudioReverbClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
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GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
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gobject_class->set_property = gst_audio_reverb_set_property;
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gobject_class->get_property = gst_audio_reverb_get_property;
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gobject_class->finalize = gst_audio_reverb_finalize;
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g_object_class_install_property (gobject_class, PROP_DELAY,
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g_param_spec_uint64 ("delay", "Delay",
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"Delay of the echo in nanoseconds", 1, G_MAXUINT64,
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1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
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| GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_INTENSITY,
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g_param_spec_float ("intensity", "Intensity",
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"Intensity of the echo", 0.0, 1.0,
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0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
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| GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_FEEDBACK,
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g_param_spec_float ("feedback", "Feedback",
|
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"Amount of feedback", 0.0, 1.0,
|
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0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
|
||||
| GST_PARAM_CONTROLLABLE));
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||||
|
||||
audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_reverb_setup);
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basetransform_class->transform_ip =
|
||||
GST_DEBUG_FUNCPTR (gst_audio_reverb_transform_ip);
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basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_reverb_stop);
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||||
}
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|
||||
static void
|
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gst_audio_reverb_init (GstAudioReverb * self, GstAudioReverbClass * klass)
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{
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self->delay = 0;
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self->intensity = 0.0;
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self->feedback = 0.0;
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||||
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);
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}
|
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|
||||
static void
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gst_audio_reverb_finalize (GObject * object)
|
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{
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GstAudioReverb *self = GST_AUDIO_REVERB (object);
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g_free (self->buffer);
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self->buffer = NULL;
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|
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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|
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static void
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gst_audio_reverb_set_property (GObject * object, guint prop_id,
|
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const GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
GstAudioReverb *self = GST_AUDIO_REVERB (object);
|
||||
|
||||
switch (prop_id) {
|
||||
case PROP_DELAY:{
|
||||
guint rate, width, channels;
|
||||
|
||||
GST_BASE_TRANSFORM_LOCK (self);
|
||||
self->delay = g_value_get_uint64 (value);
|
||||
|
||||
rate = GST_AUDIO_FILTER (self)->format.rate;
|
||||
width = GST_AUDIO_FILTER (self)->format.width / 8;
|
||||
channels = GST_AUDIO_FILTER (self)->format.channels;
|
||||
|
||||
if (self->buffer && rate > 0) {
|
||||
guint new_reverb =
|
||||
MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
|
||||
guint new_size = new_reverb * width * channels;
|
||||
|
||||
if (new_size > self->buffer_size) {
|
||||
guint i;
|
||||
guint8 *old_buffer = self->buffer;
|
||||
|
||||
self->buffer_size = new_size;
|
||||
self->buffer = g_malloc0 (new_size);
|
||||
|
||||
for (i = 0; i < self->buffer_size_frames; i++) {
|
||||
memcpy (&self->buffer[i * width * channels],
|
||||
&old_buffer[((i +
|
||||
self->buffer_pos) % self->buffer_size_frames) *
|
||||
width * channels], channels * width);
|
||||
}
|
||||
self->buffer_size_frames = self->delay_frames = new_reverb;
|
||||
self->buffer_pos = 0;
|
||||
}
|
||||
} else if (self->buffer) {
|
||||
g_free (self->buffer);
|
||||
self->buffer = NULL;
|
||||
}
|
||||
|
||||
GST_BASE_TRANSFORM_UNLOCK (self);
|
||||
}
|
||||
break;
|
||||
case PROP_INTENSITY:{
|
||||
GST_BASE_TRANSFORM_LOCK (self);
|
||||
self->intensity = g_value_get_float (value);
|
||||
GST_BASE_TRANSFORM_UNLOCK (self);
|
||||
}
|
||||
break;
|
||||
case PROP_FEEDBACK:{
|
||||
GST_BASE_TRANSFORM_LOCK (self);
|
||||
self->feedback = g_value_get_float (value);
|
||||
GST_BASE_TRANSFORM_UNLOCK (self);
|
||||
}
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
gst_audio_reverb_get_property (GObject * object, guint prop_id,
|
||||
GValue * value, GParamSpec * pspec)
|
||||
{
|
||||
GstAudioReverb *self = GST_AUDIO_REVERB (object);
|
||||
|
||||
switch (prop_id) {
|
||||
case PROP_DELAY:
|
||||
GST_BASE_TRANSFORM_LOCK (self);
|
||||
g_value_set_uint64 (value, self->delay);
|
||||
GST_BASE_TRANSFORM_UNLOCK (self);
|
||||
break;
|
||||
case PROP_INTENSITY:
|
||||
GST_BASE_TRANSFORM_LOCK (self);
|
||||
g_value_set_float (value, self->intensity);
|
||||
GST_BASE_TRANSFORM_UNLOCK (self);
|
||||
break;
|
||||
case PROP_FEEDBACK:
|
||||
GST_BASE_TRANSFORM_LOCK (self);
|
||||
g_value_set_float (value, self->feedback);
|
||||
GST_BASE_TRANSFORM_UNLOCK (self);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/* GstAudioFilter vmethod implementations */
|
||||
|
||||
static gboolean
|
||||
gst_audio_reverb_setup (GstAudioFilter * base, GstRingBufferSpec * format)
|
||||
{
|
||||
GstAudioReverb *self = GST_AUDIO_REVERB (base);
|
||||
gboolean ret = TRUE;
|
||||
|
||||
if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
|
||||
self->process = (GstAudioReverbProcessFunc)
|
||||
gst_audio_reverb_transform_float;
|
||||
else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64)
|
||||
self->process = (GstAudioReverbProcessFunc)
|
||||
gst_audio_reverb_transform_double;
|
||||
else
|
||||
ret = FALSE;
|
||||
|
||||
g_free (self->buffer);
|
||||
self->buffer = NULL;
|
||||
self->buffer_pos = 0;
|
||||
self->buffer_size = 0;
|
||||
self->buffer_size_frames = 0;
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_audio_reverb_stop (GstBaseTransform * base)
|
||||
{
|
||||
GstAudioReverb *self = GST_AUDIO_REVERB (base);
|
||||
|
||||
g_free (self->buffer);
|
||||
self->buffer = NULL;
|
||||
self->buffer_pos = 0;
|
||||
self->buffer_size = 0;
|
||||
self->buffer_size_frames = 0;
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
#define TRANSFORM_FUNC(name, type) \
|
||||
static void \
|
||||
gst_audio_reverb_transform_##name (GstAudioReverb * self, \
|
||||
type * data, guint num_samples) \
|
||||
{ \
|
||||
type *buffer = (type *) self->buffer; \
|
||||
guint channels = GST_AUDIO_FILTER (self)->format.channels; \
|
||||
guint rate = GST_AUDIO_FILTER (self)->format.rate; \
|
||||
guint i, j; \
|
||||
guint reverb_index = self->buffer_size_frames - self->delay_frames; \
|
||||
gdouble reverb_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \
|
||||
\
|
||||
if (reverb_off < 0.0) \
|
||||
reverb_off = 0.0; \
|
||||
\
|
||||
num_samples /= channels; \
|
||||
\
|
||||
for (i = 0; i < num_samples; i++) { \
|
||||
guint echo0_index = ((reverb_index + self->buffer_pos) % self->buffer_size_frames) * channels; \
|
||||
guint echo1_index = ((reverb_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \
|
||||
guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \
|
||||
for (j = 0; j < channels; j++) { \
|
||||
gdouble in = data[i*channels + j]; \
|
||||
gdouble echo0 = buffer[echo0_index + j]; \
|
||||
gdouble echo1 = buffer[echo1_index + j]; \
|
||||
gdouble echo = echo0 + (echo1-echo0)*reverb_off; \
|
||||
type out = in + self->intensity * echo; \
|
||||
\
|
||||
data[i*channels + j] = out; \
|
||||
\
|
||||
buffer[rbout_index + j] = in + self->feedback * echo; \
|
||||
} \
|
||||
self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \
|
||||
} \
|
||||
}
|
||||
|
||||
TRANSFORM_FUNC (float, gfloat);
|
||||
TRANSFORM_FUNC (double, gdouble);
|
||||
|
||||
/* GstBaseTransform vmethod implementations */
|
||||
static GstFlowReturn
|
||||
gst_audio_reverb_transform_ip (GstBaseTransform * base, GstBuffer * buf)
|
||||
{
|
||||
GstAudioReverb *self = GST_AUDIO_REVERB (base);
|
||||
guint num_samples =
|
||||
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
|
||||
|
||||
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
|
||||
gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf));
|
||||
|
||||
if (self->buffer == NULL) {
|
||||
guint width, rate, channels;
|
||||
|
||||
width = GST_AUDIO_FILTER (self)->format.width / 8;
|
||||
rate = GST_AUDIO_FILTER (self)->format.rate;
|
||||
channels = GST_AUDIO_FILTER (self)->format.channels;
|
||||
|
||||
self->delay_frames =
|
||||
MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);
|
||||
|
||||
self->buffer_size_frames = MAX (self->delay_frames, 1000);
|
||||
self->buffer_size = self->buffer_size_frames * width * channels;
|
||||
self->buffer = g_malloc0 (self->buffer_size);
|
||||
self->buffer_pos = 0;
|
||||
}
|
||||
|
||||
self->process (self, GST_BUFFER_DATA (buf), num_samples);
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
68
gst/audiofx/audioreverb.h
Normal file
68
gst/audiofx/audioreverb.h
Normal file
|
@ -0,0 +1,68 @@
|
|||
/*
|
||||
* GStreamer
|
||||
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef __GST_AUDIO_REVERB_H__
|
||||
#define __GST_AUDIO_REVERB_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstbasetransform.h>
|
||||
#include <gst/audio/audio.h>
|
||||
#include <gst/audio/gstaudiofilter.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
#define GST_TYPE_AUDIO_REVERB (gst_audio_reverb_get_type())
|
||||
#define GST_AUDIO_REVERB(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_REVERB,GstAudioReverb))
|
||||
#define GST_IS_AUDIO_REVERB(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_REVERB))
|
||||
#define GST_AUDIO_REVERB_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_REVERB,GstAudioReverbClass))
|
||||
#define GST_IS_AUDIO_REVERB_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_REVERB))
|
||||
#define GST_AUDIO_REVERB_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_REVERB,GstAudioReverbClass))
|
||||
typedef struct _GstAudioReverb GstAudioReverb;
|
||||
typedef struct _GstAudioReverbClass GstAudioReverbClass;
|
||||
|
||||
typedef void (*GstAudioReverbProcessFunc) (GstAudioReverb *, guint8 *, guint);
|
||||
|
||||
struct _GstAudioReverb
|
||||
{
|
||||
GstAudioFilter audiofilter;
|
||||
|
||||
guint64 delay;
|
||||
gfloat intensity;
|
||||
gfloat feedback;
|
||||
|
||||
/* < private > */
|
||||
GstAudioReverbProcessFunc process;
|
||||
guint delay_frames;
|
||||
guint8 *buffer;
|
||||
guint buffer_pos;
|
||||
guint buffer_size;
|
||||
guint buffer_size_frames;
|
||||
};
|
||||
|
||||
struct _GstAudioReverbClass
|
||||
{
|
||||
GstAudioFilterClass parent;
|
||||
};
|
||||
|
||||
GType gst_audio_reverb_get_type (void);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __GST_AUDIO_REVERB_H__ */
|
|
@ -74,6 +74,7 @@ check_PROGRAMS = \
|
|||
elements/audiocheblimit \
|
||||
elements/audioiirfilter \
|
||||
elements/audioamplify \
|
||||
elements/audioreverb \
|
||||
elements/audiodynamic \
|
||||
elements/audiowsincband \
|
||||
elements/audiowsinclimit \
|
||||
|
|
229
tests/check/elements/audioreverb.c
Normal file
229
tests/check/elements/audioreverb.c
Normal file
|
@ -0,0 +1,229 @@
|
|||
/* GStreamer
|
||||
*
|
||||
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#include <gst/check/gstcheck.h>
|
||||
|
||||
gboolean have_eos = FALSE;
|
||||
|
||||
/* For ease of programming we use globals to keep refs for our floating
|
||||
* src and sink pads we create; otherwise we always have to do get_pad,
|
||||
* get_peer, and then remove references in every test function */
|
||||
GstPad *mysrcpad, *mysinkpad;
|
||||
|
||||
#define REVERB_CAPS_STRING \
|
||||
"audio/x-raw-float, " \
|
||||
"channels = (int) 2, " \
|
||||
"rate = (int) 100000, " \
|
||||
"endianness = (int) BYTE_ORDER, " \
|
||||
"width = (int) 64"
|
||||
|
||||
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
|
||||
GST_PAD_SINK,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS ("audio/x-raw-float, "
|
||||
"channels = (int) [ 1, 2 ], "
|
||||
"rate = (int) [ 1, MAX ], "
|
||||
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 }")
|
||||
);
|
||||
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
|
||||
GST_PAD_SRC,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS ("audio/x-raw-float, "
|
||||
"channels = (int) [ 1, 2 ], "
|
||||
"rate = (int) [ 1, MAX ], "
|
||||
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 }")
|
||||
);
|
||||
|
||||
GstElement *
|
||||
setup_reverb ()
|
||||
{
|
||||
GstElement *reverb;
|
||||
|
||||
GST_DEBUG ("setup_reverb");
|
||||
reverb = gst_check_setup_element ("audioreverb");
|
||||
mysrcpad = gst_check_setup_src_pad (reverb, &srctemplate, NULL);
|
||||
mysinkpad = gst_check_setup_sink_pad (reverb, &sinktemplate, NULL);
|
||||
gst_pad_set_active (mysrcpad, TRUE);
|
||||
gst_pad_set_active (mysinkpad, TRUE);
|
||||
|
||||
return reverb;
|
||||
}
|
||||
|
||||
void
|
||||
cleanup_reverb (GstElement * reverb)
|
||||
{
|
||||
GST_DEBUG ("cleanup_reverb");
|
||||
|
||||
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
|
||||
g_list_free (buffers);
|
||||
buffers = NULL;
|
||||
|
||||
gst_pad_set_active (mysrcpad, FALSE);
|
||||
gst_pad_set_active (mysinkpad, FALSE);
|
||||
gst_check_teardown_src_pad (reverb);
|
||||
gst_check_teardown_sink_pad (reverb);
|
||||
gst_check_teardown_element (reverb);
|
||||
}
|
||||
|
||||
GST_START_TEST (test_passthrough)
|
||||
{
|
||||
GstElement *reverb;
|
||||
GstBuffer *inbuffer, *outbuffer;
|
||||
GstCaps *caps;
|
||||
gdouble in[] = { 1.0, -1.0, 0.0, 0.5, -0.5, 0.0 };
|
||||
gdouble *res;
|
||||
|
||||
reverb = setup_reverb ();
|
||||
g_object_set (G_OBJECT (reverb), "delay", 1, "intensity", 0.0, "feedback",
|
||||
0.0, NULL);
|
||||
fail_unless (gst_element_set_state (reverb,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (sizeof (in));
|
||||
memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
|
||||
fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
|
||||
caps = gst_caps_from_string (REVERB_CAPS_STRING);
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
gst_caps_unref (caps);
|
||||
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
||||
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
/* ... but it ends up being collected on the global buffer list */
|
||||
fail_unless_equals_int (g_list_length (buffers), 1);
|
||||
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
|
||||
|
||||
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
|
||||
GST_INFO
|
||||
("expected %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf",
|
||||
in[0], in[1], in[2], in[3], in[4], in[5], res[0], res[1], res[2], res[3],
|
||||
res[4], res[5]);
|
||||
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), in, sizeof (in)) == 0);
|
||||
|
||||
/* cleanup */
|
||||
cleanup_reverb (reverb);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_reverb)
|
||||
{
|
||||
GstElement *reverb;
|
||||
GstBuffer *inbuffer, *outbuffer;
|
||||
GstCaps *caps;
|
||||
gdouble in[] = { 1.0, -1.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, };
|
||||
gdouble out[] = { 1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 0.0, 0.0, 0.0 };
|
||||
gdouble *res;
|
||||
|
||||
reverb = setup_reverb ();
|
||||
g_object_set (G_OBJECT (reverb), "delay", 20000, "intensity", 1.0, "feedback",
|
||||
0.0, NULL);
|
||||
fail_unless (gst_element_set_state (reverb,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (sizeof (in));
|
||||
memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
|
||||
fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
|
||||
caps = gst_caps_from_string (REVERB_CAPS_STRING);
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
gst_caps_unref (caps);
|
||||
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
||||
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
/* ... but it ends up being collected on the global buffer list */
|
||||
fail_unless_equals_int (g_list_length (buffers), 1);
|
||||
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
|
||||
|
||||
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
|
||||
GST_INFO
|
||||
("expected %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf",
|
||||
out[0], out[1], out[2], out[3], out[4], out[5], out[6], out[7], out[8],
|
||||
out[9], res[0], res[1], res[2], res[3], res[4], res[5], res[6], res[7],
|
||||
res[8], res[9]);
|
||||
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, sizeof (out)) == 0);
|
||||
|
||||
/* cleanup */
|
||||
cleanup_reverb (reverb);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_feedback)
|
||||
{
|
||||
GstElement *reverb;
|
||||
GstBuffer *inbuffer, *outbuffer;
|
||||
GstCaps *caps;
|
||||
gdouble in[] = { 1.0, -1.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, };
|
||||
gdouble out[] = { 1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 0.0, 1.0, -1.0 };
|
||||
gdouble *res;
|
||||
|
||||
reverb = setup_reverb ();
|
||||
g_object_set (G_OBJECT (reverb), "delay", 20000, "intensity", 1.0, "feedback",
|
||||
1.0, NULL);
|
||||
fail_unless (gst_element_set_state (reverb,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (sizeof (in));
|
||||
memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
|
||||
fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
|
||||
caps = gst_caps_from_string (REVERB_CAPS_STRING);
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
gst_caps_unref (caps);
|
||||
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
||||
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
/* ... but it ends up being collected on the global buffer list */
|
||||
fail_unless_equals_int (g_list_length (buffers), 1);
|
||||
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
|
||||
|
||||
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
|
||||
GST_INFO
|
||||
("expected %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf",
|
||||
out[0], out[1], out[2], out[3], out[4], out[5], out[6], out[7], out[8],
|
||||
out[9], res[0], res[1], res[2], res[3], res[4], res[5], res[6], res[7],
|
||||
res[8], res[9]);
|
||||
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, sizeof (out)) == 0);
|
||||
|
||||
/* cleanup */
|
||||
cleanup_reverb (reverb);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
static Suite *
|
||||
audioreverb_suite (void)
|
||||
{
|
||||
Suite *s = suite_create ("audioreverb");
|
||||
TCase *tc_chain = tcase_create ("general");
|
||||
|
||||
suite_add_tcase (s, tc_chain);
|
||||
tcase_add_test (tc_chain, test_passthrough);
|
||||
tcase_add_test (tc_chain, test_reverb);
|
||||
tcase_add_test (tc_chain, test_feedback);
|
||||
|
||||
return s;
|
||||
}
|
||||
|
||||
GST_CHECK_MAIN (audioreverb);
|
Loading…
Reference in a new issue