diff --git a/ChangeLog b/ChangeLog index ca5dbf1f44..bb24114c21 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,66 @@ +2007-08-16 Sebastian Dröge + + reviewed by: Stefan Kost + + * gst/audiofx/Makefile.am: + * gst/audiofx/audiochebyshevfreqband.c: + (gst_audio_chebyshev_freq_band_mode_get_type), + (gst_audio_chebyshev_freq_band_base_init), + (gst_audio_chebyshev_freq_band_dispose), + (gst_audio_chebyshev_freq_band_class_init), + (gst_audio_chebyshev_freq_band_init), + (generate_biquad_coefficients), (calculate_gain), + (generate_coefficients), + (gst_audio_chebyshev_freq_band_set_property), + (gst_audio_chebyshev_freq_band_get_property), + (gst_audio_chebyshev_freq_band_setup), (process), (process_64), + (process_32), (gst_audio_chebyshev_freq_band_transform_ip), + (gst_audio_chebyshev_freq_band_start): + * gst/audiofx/audiochebyshevfreqband.h: + * gst/audiofx/audiochebyshevfreqlimit.c: + (gst_audio_chebyshev_freq_limit_mode_get_type), + (gst_audio_chebyshev_freq_limit_base_init), + (gst_audio_chebyshev_freq_limit_dispose), + (gst_audio_chebyshev_freq_limit_class_init), + (gst_audio_chebyshev_freq_limit_init), + (generate_biquad_coefficients), (calculate_gain), + (generate_coefficients), + (gst_audio_chebyshev_freq_limit_set_property), + (gst_audio_chebyshev_freq_limit_get_property), + (gst_audio_chebyshev_freq_limit_setup), (process), (process_64), + (process_32), (gst_audio_chebyshev_freq_limit_transform_ip), + (gst_audio_chebyshev_freq_limit_start): + * gst/audiofx/audiochebyshevfreqlimit.h: + * gst/audiofx/audiofx.c: (plugin_init): + Add Chebyshev lowpass/highpass and bandpass/bandreject elements. + Fixes #464800. + + * tests/check/Makefile.am: + * tests/check/elements/.cvsignore: + * tests/check/elements/audiochebyshevfreqband.c: + (setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband), + (GST_START_TEST), (audiochebyshevfreqband_suite), (main): + * tests/check/elements/audiochebyshevfreqlimit.c: + (setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit), + (GST_START_TEST), (audiochebyshevfreqlimit_suite), (main): + Add unit tests for the chebyshev filters. + + * 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/inspect/plugin-1394.xml: + * docs/plugins/inspect/plugin-audiofx.xml: + * docs/plugins/inspect/plugin-dv.xml: + * docs/plugins/inspect/plugin-flac.xml: + * docs/plugins/inspect/plugin-jpeg.xml: + * docs/plugins/inspect/plugin-png.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * docs/plugins/inspect/plugin-shout2send.xml: + * docs/plugins/inspect/plugin-wavpack.xml: + And add docs for the chebyshev filters. While doing + that also run make update in docs/plugins. + 2007-08-16 Stefan Kost * ext/annodex/gstcmmltag.c: diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index fb59456ee5..bcd985a182 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -100,6 +100,8 @@ EXTRA_HFILES = \ $(top_srcdir)/gst/audiofx/audiodynamic.h \ $(top_srcdir)/gst/audiofx/audioinvert.h \ $(top_srcdir)/gst/audiofx/audiopanorama.h \ + $(top_srcdir)/gst/audiofx/audiochebyshevfreqlimit.h \ + $(top_srcdir)/gst/audiofx/audiochebyshevfreqband.h \ $(top_srcdir)/gst/autodetect/gstautoaudiosink.h \ $(top_srcdir)/gst/autodetect/gstautovideosink.h \ $(top_srcdir)/gst/avi/gstavidemux.h \ diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index 57b948c336..c90312a86c 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -19,6 +19,8 @@ + + diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index bc431f13d9..d8f6b7d8c6 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -81,6 +81,26 @@ GST_AUDIO_DYNAMIC GST_AUDIO_DYNAMIC_CLASS +
+element-audiochebyshevfreqlimit +audiochebyshevfreqlimit +GstAudioChebyshevFreqLimit + +GstAudioChebyshevFreqLimitClass +GST_AUDIO_CHEBYSHEV_FREQ_LIMIT +GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS +
+ +
+element-audiochebyshevfreqband +audiochebyshevfreqband +GstAudioChebyshevFreqBand + +GstAudioChebyshevFreqBandClass +GST_AUDIO_CHEBYSHEV_FREQ_BAND +GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS +
+
element-autoaudiosink autoaudiosink diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index bdb97130f5..935e27badd 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -401,21 +401,21 @@ GstVertigoTV::speed gfloat -[0,01,100] +[0.01,100] rw Speed Control the speed of movement. -0,02 +0.02 GstVertigoTV::zoom-speed gfloat -[1,01,1,1] +[1.01,1.1] rw Zoom Speed Control the rate of zooming. -1,01 +1.01 @@ -1141,7 +1141,7 @@ GstDV1394Src::port gint -[G_MAXULONG,16] +[-1,16] rw Port Port number (-1 automatic). @@ -17241,7 +17241,7 @@ GstGamma::gamma gdouble -[0,01,10] +[0.01,10] rw Gamma gamma. @@ -17328,3 +17328,113 @@ 2 + +GstAudioChebyshevFreqBand::lower-frequency +gfloat +>= 0 +rw +Lower frequency +Start frequency of the band (Hz). +0 + + + +GstAudioChebyshevFreqBand::mode +GstAudioChebyshevFreqBandMode + +rw +Mode +Low pass or high pass mode. +Band pass (default) + + + +GstAudioChebyshevFreqBand::poles +gint +[4,32] +rw +Poles +Number of poles to use, will be rounded up to the next multiply of four. +4 + + + +GstAudioChebyshevFreqBand::ripple +gfloat +>= 0 +rw +Ripple +Amount of ripple (dB). +0.25 + + + +GstAudioChebyshevFreqBand::type +gint +[1,2] +rw +Type +Type of the chebychev filter. +1 + + + +GstAudioChebyshevFreqBand::upper-frequency +gfloat +>= 0 +rw +Upper frequency +Stop frequency of the band (Hz). +0 + + + +GstAudioChebyshevFreqLimit::cutoff +gfloat +>= 0 +rw +Cutoff +Cut off frequency (Hz). +0 + + + +GstAudioChebyshevFreqLimit::mode +GstAudioChebyshevFreqLimitMode + +rw +Mode +Low pass or high pass mode. +Low pass (default) + + + +GstAudioChebyshevFreqLimit::poles +gint +[2,32] +rw +Poles +Number of poles to use, will be rounded up to the next even number. +4 + + + +GstAudioChebyshevFreqLimit::ripple +gfloat +>= 0 +rw +Ripple +Amount of ripple (dB). +0.25 + + + +GstAudioChebyshevFreqLimit::type +gint +[1,2] +rw +Type +Type of the chebychev filter. +1 + + diff --git a/docs/plugins/inspect/plugin-1394.xml b/docs/plugins/inspect/plugin-1394.xml index 29078d7536..8e095aa0eb 100644 --- a/docs/plugins/inspect/plugin-1394.xml +++ b/docs/plugins/inspect/plugin-1394.xml @@ -3,10 +3,10 @@ Source for DV data via IEEE1394 interface ../../ext/raw1394/.libs/libgst1394.so libgst1394.so - 0.10.6 + 0.10.6.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins CVS/prerelease Unknown package origin @@ -18,6 +18,14 @@ Daniel Fischer <dan@f3c.com> Wim Taymans <wim@fluendo.com> Zaheer Abbas Merali <zaheerabbas at merali dot org> + + + src + source + always +
video/x-dv, format=(string){ NTSC, PAL }, systemstream=(boolean)true
+ + \ No newline at end of file diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml index 825b1dc581..3828fe727c 100644 --- a/docs/plugins/inspect/plugin-audiofx.xml +++ b/docs/plugins/inspect/plugin-audiofx.xml @@ -30,6 +30,48 @@ + + audiochebyshevfreqband + AudioChebyshevFreqBand + Filter/Effect/Audio + Chebyshev band pass and band reject filter + Sebastian Dröge <slomo@circular-chaos.org> + + + src + source + always +
audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+ + + sink + sink + always +
audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+ + + + + audiochebyshevfreqlimit + AudioChebyshevFreqLimit + Filter/Effect/Audio + Chebyshev low pass and high pass filter + Sebastian Dröge <slomo@circular-chaos.org> + + + src + source + always +
audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+ + + sink + sink + always +
audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+ + + audiodynamic AudioDynamic diff --git a/docs/plugins/inspect/plugin-dv.xml b/docs/plugins/inspect/plugin-dv.xml index a62f5f21ca..951488dcec 100644 --- a/docs/plugins/inspect/plugin-dv.xml +++ b/docs/plugins/inspect/plugin-dv.xml @@ -3,10 +3,10 @@ DV demuxer and decoder based on libdv (libdv.sf.net) ../../ext/dv/.libs/libgstdv.so libgstdv.so - 0.10.6 + 0.10.6.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins CVS/prerelease Unknown package origin @@ -15,6 +15,20 @@ Codec/Decoder/Video Uses libdv to decode DV video (smpte314) (libdv.sourceforge.net) Erik Walthinsen <omega@cse.ogi.edu>,Wim Taymans <wim@fluendo.com> + + + sink + sink + always +
video/x-dv, systemstream=(boolean)false
+ + + src + source + always +
video/x-raw-yuv, format=(fourcc)YUY2, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)24, endianness=(int)4321, red_mask=(int)65280, green_mask=(int)16711680, blue_mask=(int)-16777216, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]
+ + dvdemux @@ -22,6 +36,26 @@ Codec/Demuxer Uses libdv to separate DV audio from DV video (libdv.sourceforge.net) Erik Walthinsen <omega@cse.ogi.edu>, Wim Taymans <wim@fluendo.com> + + + sink + sink + always +
video/x-dv, systemstream=(boolean)true
+ + + video + source + sometimes +
video/x-dv, systemstream=(boolean)false
+ + + audio + source + sometimes +
audio/x-raw-int, depth=(int)16, width=(int)16, signed=(boolean)true, channels=(int){ 2, 4 }, endianness=(int)1234, rate=(int){ 32000, 44100, 48000 }
+ + \ No newline at end of file diff --git a/docs/plugins/inspect/plugin-flac.xml b/docs/plugins/inspect/plugin-flac.xml index 83fa4d39d6..818917cb0d 100644 --- a/docs/plugins/inspect/plugin-flac.xml +++ b/docs/plugins/inspect/plugin-flac.xml @@ -47,7 +47,7 @@ sink sink always -
audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ]
+
audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]
 diff --git a/docs/plugins/inspect/plugin-jpeg.xml b/docs/plugins/inspect/plugin-jpeg.xml index e59bc371f4..bf9aaca088 100644 --- a/docs/plugins/inspect/plugin-jpeg.xml +++ b/docs/plugins/inspect/plugin-jpeg.xml @@ -16,18 +16,18 @@ Decode images from JPEG format Wim Taymans <wim@fluendo.com> - - sink - sink - always -
image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 8, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
- src source always
video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 + + sink + sink + always +
image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 8, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ @@ -37,18 +37,18 @@ Encode images in JPEG format Wim Taymans <wim.taymans@tvd.be> - - src - source - always -
image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
- sink sink always
video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 + + src + source + always +
image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ @@ -58,18 +58,18 @@ Decode video from Smoke format Wim Taymans <wim@fluendo.com> - - sink - sink - always -
video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
- src source always
video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 + + sink + sink + always +
video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ @@ -79,18 +79,18 @@ Encode images into the Smoke format Wim Taymans <wim@fluendo.com> - - src - source - always -
video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
- sink sink always
video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 + + src + source + always +
video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ diff --git a/docs/plugins/inspect/plugin-png.xml b/docs/plugins/inspect/plugin-png.xml index f668bfbb7e..dc23545173 100644 --- a/docs/plugins/inspect/plugin-png.xml +++ b/docs/plugins/inspect/plugin-png.xml @@ -16,18 +16,18 @@ Decode a png video frame to a raw image Wim Taymans <wim@fluendo.com> - - sink - sink - always -
image/png
- src source always
video/x-raw-rgb, bpp=(int)32, depth=(int)32, endianness=(int)4321, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, alpha_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 + + sink + sink + always +
image/png
+ @@ -37,18 +37,18 @@ Encode a video frame to a .png image Jeremy SIMON <jsimon13@yahoo.fr> - - src - source - always -
image/png, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
- sink sink always
video/x-raw-rgb, bpp=(int)32, depth=(int)32, endianness=(int)4321, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, alpha_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 + + src + source + always +
image/png, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index 2a7d8236c8..b934b1a0e7 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -137,7 +137,7 @@ rtpdepay - RTP payloader + RTP depayloader Codec/Depayloader/Network Accepts raw RTP and RTCP packets and sends them forward Wim Taymans <wim@fluendo.com> @@ -332,7 +332,7 @@ sink sink always -
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(int){ 20, 30 }
+
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }
 @@ -353,7 +353,7 @@ src source always -
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(int){ 20, 30 }
+
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }
@@ -558,7 +558,7 @@ src source always -
video/mpeg, systemstream=(boolean)false
+
video/mpeg, mpegversion=(int)2, systemstream=(boolean)false
sink diff --git a/docs/plugins/inspect/plugin-shout2send.xml b/docs/plugins/inspect/plugin-shout2send.xml index 414b10b44f..75e67d7e5a 100644 --- a/docs/plugins/inspect/plugin-shout2send.xml +++ b/docs/plugins/inspect/plugin-shout2send.xml @@ -3,7 +3,7 @@ Sends data to an icecast server using libshout2 ../../ext/shout2/.libs/libgstshout2.so libgstshout2.so - 0.10.6 + 0.10.6.1 LGPL gst-plugins-good libshout2 @@ -17,6 +17,14 @@ Wim Taymans <wim.taymans@chello.be> Pedro Corte-Real <typo@netcabo.pt> Zaheer Abbas Merali <zaheerabbas at merali dot org> + + + sink + sink + always +
application/ogg; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]
+ + \ No newline at end of file diff --git a/docs/plugins/inspect/plugin-wavpack.xml b/docs/plugins/inspect/plugin-wavpack.xml index 67f4264173..b0af798524 100644 --- a/docs/plugins/inspect/plugin-wavpack.xml +++ b/docs/plugins/inspect/plugin-wavpack.xml @@ -3,10 +3,10 @@ Wavpack lossless/lossy audio format handling ../../ext/wavpack/.libs/libgstwavpack.so libgstwavpack.so - 0.10.6 + 0.10.6.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins CVS/prerelease Unknown package origin @@ -15,6 +15,20 @@ Codec/Decoder/Audio Decodes Wavpack audio data Arwed v. Merkatz <v.merkatz@gmx.net>, Sebastian Dröge <slomo@circular-chaos.org> + + + src + source + always +
audio/x-raw-int, width=(int)32, depth=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], endianness=(int)1234, signed=(boolean)true
+ + + sink + sink + always +
audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true
+ + wavpackenc @@ -22,6 +36,26 @@ Codec/Encoder/Audio Encodes audio with the Wavpack lossless/lossy audio codec Sebastian Dröge <slomo@circular-chaos.org> + + + sink + sink + always +
audio/x-raw-int, width=(int)32, depth=(int)[ 1, 32 ], endianness=(int)1234, channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], signed=(boolean)true
+ + + src + source + always +
audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true
+ + + wvcsrc + source + sometimes +
audio/x-wavpack-correction, framed=(boolean)true
+ + wavpackparse @@ -29,6 +63,26 @@ Codec/Demuxer/Audio Parses Wavpack files Arwed v. Merkatz <v.merkatz@gmx.net>, Sebastian Dröge <slomo@circular-chaos.org> + + + src + source + sometimes +
audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true
+ + + wvcsrc + source + sometimes +
audio/x-wavpack-correction, framed=(boolean)true
+ + + sink + sink + always +
audio/x-wavpack, framed=(boolean)false; audio/x-wavpack-correction, framed=(boolean)false
+ + \ No newline at end of file diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index b5bf0bf930..61a8dcc898 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -7,7 +7,9 @@ libgstaudiofx_la_SOURCES = audiofx.c\ audiopanorama.c \ audioinvert.c \ audioamplify.c \ - audiodynamic.c + audiodynamic.c \ + audiochebyshevfreqlimit.c \ + audiochebyshevfreqband.c # flags used to compile this plugin libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \ @@ -18,12 +20,15 @@ libgstaudiofx_la_LIBADD = $(GST_LIBS) \ $(GST_BASE_LIBS) \ $(GST_CONTROLLER_LIBS) \ $(GST_PLUGINS_BASE_LIBS) \ - -lgstaudio-$(GST_MAJORMINOR) + -lgstaudio-$(GST_MAJORMINOR) \ + $(LIBM) libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) # headers we need but don't want installed noinst_HEADERS = audiopanorama.h \ audioinvert.h \ audioamplify.h \ - audiodynamic.h + audiodynamic.h \ + audiochebyshevfreqlimit.h \ + audiochebyshevfreqband.c diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c new file mode 100644 index 0000000000..d4730607af --- /dev/null +++ b/gst/audiofx/audiochebband.c @@ -0,0 +1,916 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + * Transformation from lowpass to bandpass/bandreject: + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm + * + */ + +/** + * SECTION:element-audiochebyshevfreqband + * @short_description: Chebyshev band pass and band reject filter + * + * + * + * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency + * band. The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqband.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand", + "Filter/Effect/Audio", + "Chebyshev band pass and band reject filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_LOWER_FREQUENCY, + PROP_UPPER_FREQUENCY, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band, + GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +static void gst_audio_chebyshev_freq_band_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_band_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_BAND_PASS = 0, + MODE_BAND_REJECT +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_band_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_BAND_PASS, "Band pass (default)", + "band-pass"}, + {MODE_BAND_REJECT, "Band reject", + "band-reject"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_band_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_band_dispose (GObject * object) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE, + MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", + "Type of the chebychev filter", 1, 2, + 1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY, + g_param_spec_float ("lower-frequency", "Lower frequency", + "Start frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, + g_param_spec_float ("upper-frequency", "Upper frequency", + "Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", + "Amount of ripple (dB)", 0.0, G_MAXFLOAT, + 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next multiply of four", + 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start); +} + +static void +gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandClass * klass) +{ + filter->lower_frequency = filter->upper_frequency = 0.0; + filter->mode = MODE_BAND_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3, + gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4) +{ + gint np = filter->poles / 2; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to move from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either bandpass + * or band reject. + * + * For bandpass substitute z^(-1) with: + * + * -2 -1 + * -z + alpha * z - beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a*b)/(1+b) + * beta = (b-1)/(b+1) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * cot((w1 - w0)/2) + * + * For bandreject substitute z^(-1) with: + * + * -2 -1 + * z - alpha * z + beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a)/(1+b) + * beta = (1-b)/(1+b) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * tan((w1 - w0)/2) + * + */ + { + gdouble a, b, d; + gdouble alpha, beta; + gdouble w0 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w1 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_BAND_PASS) { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a * b) / (1.0 + b); + beta = (b - 1.0) / (b + 1.0); + + d = 1.0 + beta * (y1 - beta * y2); + + *a0 = (x0 + beta * (-x1 + beta * x2)) / d; + *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) + + alpha * alpha * (x0 - x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d; + *a4 = (beta * (beta * x0 - x1) + x2) / d; + *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d; + *b2 = + (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) + + 2.0 * beta * (-1.0 + y2)) / d; + *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d; + *b4 = (-beta * beta - beta * y1 + y2) / d; + } else { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a) / (1.0 + b); + beta = (1.0 - b) / (1.0 + b); + + d = -1.0 + beta * (beta * y2 + y1); + + *a0 = (-x0 - beta * x1 - beta * beta * x2) / d; + *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) - + alpha * alpha * (x0 + x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d; + *a4 = (-beta * beta * x0 - beta * x1 - x2) / d; + *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d; + *b2 = + -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) + + alpha * alpha * (-1.0 + y1 + y2)) / d; + *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d; + *b4 = -(-beta * beta + beta * y1 + y2) / d; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqBand * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->upper_frequency <= filter->lower_frequency) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "frequency band had no or negative dimension"); + return; + } + + if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) { + filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2; + GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency"); + } + + if (filter->lower_frequency < 0.0) { + filter->lower_frequency = 0.0; + GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0"); + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 5); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 5); + + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[4] = 1.0; + b[4] = 1.0; + + for (p = 1; p <= np / 4; p++) { + gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4; + gdouble *ta = g_new0 (gdouble, np + 5); + gdouble *tb = g_new0 (gdouble, np + 5); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1, + &b2, &b3, &b4); + + memcpy (ta, a, sizeof (gdouble) * (np + 5)); + memcpy (tb, b, sizeof (gdouble) * (np + 5)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 4; i < np + 5; i++) { + a[i] = + a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] + + a4 * ta[i - 4]; + b[i] = + tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] - + b4 * tb[i - 4]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[4] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 4]; + b[i] = -b[i + 4]; + } + + /* Normalize to unity gain at frequency 0 and frequency + * 0.5 for bandreject and unity gain at band center frequency + * for bandpass */ + if (filter->mode == MODE_BAND_REJECT) { + /* gain is sqrt(H(0)*H(0.5)) */ + + gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0); + gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0); + + gain1 = sqrt (gain1 * gain2); + + for (i = 0; i <= np; i++) { + a[i] /= gain1; + } + } else { + /* gain is H(wc), wc = center frequency */ + + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr = cos (w0), zi = sin (w0); + gdouble gain = calculate_gain (a, b, np, np, zr, zi); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject", + filter->type, filter->poles, filter->lower_frequency, + filter->upper_frequency, filter->ripple); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr, zi; + + zr = cos (w1); + zi = sin (w1); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->lower_frequency); + zr = cos (w0); + zi = sin (w0); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0)); + zr = cos (w2); + zi = sin (w2); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->upper_frequency); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_LOWER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->lower_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_UPPER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->upper_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_4 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_LOWER_FREQUENCY: + g_value_set_float (value, filter->lower_frequency); + break; + case PROP_UPPER_FREQUENCY: + g_value_set_float (value, filter->upper_frequency); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + +static gboolean +gst_audio_chebyshev_freq_band_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiochebband.h b/gst/audiofx/audiochebband.h new file mode 100644 index 0000000000..e8c58074cf --- /dev/null +++ b/gst/audiofx/audiochebband.h @@ -0,0 +1,79 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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_CHEBYSHEV_FREQ_BAND_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand; +typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass; + +typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqBandChannelCtx; + +struct _GstAudioChebyshevFreqBand +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat lower_frequency; + gfloat upper_frequency; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqBandProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqBandChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqBandClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_band_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */ diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c new file mode 100644 index 0000000000..872b277dec --- /dev/null +++ b/gst/audiofx/audiocheblimit.c @@ -0,0 +1,816 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + */ + +/** + * SECTION:element-audiochebyshevfreqlimit + * @short_description: Chebyshev low pass and high pass filter + * + * + * + * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the + * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqlimit.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit", + "Filter/Effect/Audio", + "Chebyshev low pass and high pass filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_CUTOFF, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit, + gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, + DEBUG_INIT); + +static void gst_audio_chebyshev_freq_limit_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_limit_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_LOW_PASS = 0, + MODE_HIGH_PASS +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_limit_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_LOW_PASS, "Low pass (default)", + "low-pass"}, + {MODE_HIGH_PASS, "High pass", + "high-pass"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_limit_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_limit_dispose (GObject * object) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_CUTOFF, + g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0, + G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0, + G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next even number", + 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start); +} + +static void +gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitClass * klass) +{ + filter->cutoff = 0.0; + filter->mode = MODE_LOW_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, + gdouble * b1, gdouble * b2) +{ + gint np = filter->poles; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to convert from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either lowpass + * or highpass. + * + * For lowpass substitute z^(-1) with: + * -1 + * z - k + * ------------ + * -1 + * 1 - k * z + * + * k = sin((1-w)/2) / sin((1+w)/2) + * + * For highpass substitute z^(-1) with: + * + * -1 + * -z - k + * ------------ + * -1 + * 1 + k * z + * + * k = -cos((1+w)/2) / cos((1-w)/2) + * + */ + { + gdouble k, d; + gdouble omega = + 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_LOW_PASS) + k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0); + else + k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0); + + d = 1.0 + y1 * k - y2 * k * k; + *a0 = (x0 + k * (-x1 + k * x2)) / d; + *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d; + *a2 = (x0 * k * k - x1 * k + x2) / d; + *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d; + *b2 = (-k * k - y1 * k + y2) / d; + + if (filter->mode == MODE_HIGH_PASS) { + *a1 = -*a1; + *b1 = -*b1; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqLimit * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency"); + return; + } else if (filter->cutoff <= 0.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff is lower than zero"); + return; + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 3); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 3); + + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[2] = 1.0; + b[2] = 1.0; + + for (p = 1; p <= np / 2; p++) { + gdouble a0, a1, a2, b1, b2; + gdouble *ta = g_new0 (gdouble, np + 3); + gdouble *tb = g_new0 (gdouble, np + 3); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2); + + memcpy (ta, a, sizeof (gdouble) * (np + 3)); + memcpy (tb, b, sizeof (gdouble) * (np + 3)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 2; i < np + 3; i++) { + a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2]; + b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[2] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 2]; + b[i] = -b[i + 2]; + } + + /* Normalize to unity gain at frequency 0 for lowpass + * and frequency 0.5 for highpass */ + { + gdouble gain; + + if (filter->mode == MODE_LOW_PASS) + gain = calculate_gain (a, b, np, np, 1.0, 0.0); + else + gain = calculate_gain (a, b, np, np, -1.0, 0.0); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass", + filter->type, filter->poles, filter->cutoff, filter->ripple); + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble wc = + 2.0 * M_PI * (filter->cutoff / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble zr = cos (wc), zi = sin (wc); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->cutoff); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_CUTOFF: + GST_BASE_TRANSFORM_LOCK (filter); + filter->cutoff = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_2 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_CUTOFF: + g_value_set_float (value, filter->cutoff); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + + +static gboolean +gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiocheblimit.h b/gst/audiofx/audiocheblimit.h new file mode 100644 index 0000000000..4c87ba8eab --- /dev/null +++ b/gst/audiofx/audiocheblimit.h @@ -0,0 +1,78 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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_CHEBYSHEV_FREQ_LIMIT_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit; +typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass; + +typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqLimitChannelCtx; + +struct _GstAudioChebyshevFreqLimit +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat cutoff; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqLimitProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqLimitChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqLimitClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_limit_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */ diff --git a/gst/audiofx/audiochebyshevfreqband.c b/gst/audiofx/audiochebyshevfreqband.c new file mode 100644 index 0000000000..d4730607af --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqband.c @@ -0,0 +1,916 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + * Transformation from lowpass to bandpass/bandreject: + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm + * + */ + +/** + * SECTION:element-audiochebyshevfreqband + * @short_description: Chebyshev band pass and band reject filter + * + * + * + * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency + * band. The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqband.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand", + "Filter/Effect/Audio", + "Chebyshev band pass and band reject filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_LOWER_FREQUENCY, + PROP_UPPER_FREQUENCY, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band, + GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +static void gst_audio_chebyshev_freq_band_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_band_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_BAND_PASS = 0, + MODE_BAND_REJECT +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_band_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_BAND_PASS, "Band pass (default)", + "band-pass"}, + {MODE_BAND_REJECT, "Band reject", + "band-reject"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_band_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_band_dispose (GObject * object) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE, + MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", + "Type of the chebychev filter", 1, 2, + 1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY, + g_param_spec_float ("lower-frequency", "Lower frequency", + "Start frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, + g_param_spec_float ("upper-frequency", "Upper frequency", + "Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", + "Amount of ripple (dB)", 0.0, G_MAXFLOAT, + 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next multiply of four", + 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start); +} + +static void +gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandClass * klass) +{ + filter->lower_frequency = filter->upper_frequency = 0.0; + filter->mode = MODE_BAND_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3, + gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4) +{ + gint np = filter->poles / 2; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to move from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either bandpass + * or band reject. + * + * For bandpass substitute z^(-1) with: + * + * -2 -1 + * -z + alpha * z - beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a*b)/(1+b) + * beta = (b-1)/(b+1) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * cot((w1 - w0)/2) + * + * For bandreject substitute z^(-1) with: + * + * -2 -1 + * z - alpha * z + beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a)/(1+b) + * beta = (1-b)/(1+b) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * tan((w1 - w0)/2) + * + */ + { + gdouble a, b, d; + gdouble alpha, beta; + gdouble w0 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w1 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_BAND_PASS) { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a * b) / (1.0 + b); + beta = (b - 1.0) / (b + 1.0); + + d = 1.0 + beta * (y1 - beta * y2); + + *a0 = (x0 + beta * (-x1 + beta * x2)) / d; + *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) + + alpha * alpha * (x0 - x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d; + *a4 = (beta * (beta * x0 - x1) + x2) / d; + *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d; + *b2 = + (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) + + 2.0 * beta * (-1.0 + y2)) / d; + *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d; + *b4 = (-beta * beta - beta * y1 + y2) / d; + } else { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a) / (1.0 + b); + beta = (1.0 - b) / (1.0 + b); + + d = -1.0 + beta * (beta * y2 + y1); + + *a0 = (-x0 - beta * x1 - beta * beta * x2) / d; + *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) - + alpha * alpha * (x0 + x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d; + *a4 = (-beta * beta * x0 - beta * x1 - x2) / d; + *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d; + *b2 = + -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) + + alpha * alpha * (-1.0 + y1 + y2)) / d; + *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d; + *b4 = -(-beta * beta + beta * y1 + y2) / d; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqBand * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->upper_frequency <= filter->lower_frequency) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "frequency band had no or negative dimension"); + return; + } + + if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) { + filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2; + GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency"); + } + + if (filter->lower_frequency < 0.0) { + filter->lower_frequency = 0.0; + GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0"); + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 5); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 5); + + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[4] = 1.0; + b[4] = 1.0; + + for (p = 1; p <= np / 4; p++) { + gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4; + gdouble *ta = g_new0 (gdouble, np + 5); + gdouble *tb = g_new0 (gdouble, np + 5); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1, + &b2, &b3, &b4); + + memcpy (ta, a, sizeof (gdouble) * (np + 5)); + memcpy (tb, b, sizeof (gdouble) * (np + 5)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 4; i < np + 5; i++) { + a[i] = + a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] + + a4 * ta[i - 4]; + b[i] = + tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] - + b4 * tb[i - 4]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[4] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 4]; + b[i] = -b[i + 4]; + } + + /* Normalize to unity gain at frequency 0 and frequency + * 0.5 for bandreject and unity gain at band center frequency + * for bandpass */ + if (filter->mode == MODE_BAND_REJECT) { + /* gain is sqrt(H(0)*H(0.5)) */ + + gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0); + gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0); + + gain1 = sqrt (gain1 * gain2); + + for (i = 0; i <= np; i++) { + a[i] /= gain1; + } + } else { + /* gain is H(wc), wc = center frequency */ + + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr = cos (w0), zi = sin (w0); + gdouble gain = calculate_gain (a, b, np, np, zr, zi); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject", + filter->type, filter->poles, filter->lower_frequency, + filter->upper_frequency, filter->ripple); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr, zi; + + zr = cos (w1); + zi = sin (w1); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->lower_frequency); + zr = cos (w0); + zi = sin (w0); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0)); + zr = cos (w2); + zi = sin (w2); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->upper_frequency); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_LOWER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->lower_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_UPPER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->upper_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_4 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_LOWER_FREQUENCY: + g_value_set_float (value, filter->lower_frequency); + break; + case PROP_UPPER_FREQUENCY: + g_value_set_float (value, filter->upper_frequency); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + +static gboolean +gst_audio_chebyshev_freq_band_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiochebyshevfreqband.h b/gst/audiofx/audiochebyshevfreqband.h new file mode 100644 index 0000000000..e8c58074cf --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqband.h @@ -0,0 +1,79 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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_CHEBYSHEV_FREQ_BAND_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand; +typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass; + +typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqBandChannelCtx; + +struct _GstAudioChebyshevFreqBand +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat lower_frequency; + gfloat upper_frequency; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqBandProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqBandChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqBandClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_band_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */ diff --git a/gst/audiofx/audiochebyshevfreqlimit.c b/gst/audiofx/audiochebyshevfreqlimit.c new file mode 100644 index 0000000000..872b277dec --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqlimit.c @@ -0,0 +1,816 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + */ + +/** + * SECTION:element-audiochebyshevfreqlimit + * @short_description: Chebyshev low pass and high pass filter + * + * + * + * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the + * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqlimit.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit", + "Filter/Effect/Audio", + "Chebyshev low pass and high pass filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_CUTOFF, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit, + gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, + DEBUG_INIT); + +static void gst_audio_chebyshev_freq_limit_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_limit_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_LOW_PASS = 0, + MODE_HIGH_PASS +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_limit_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_LOW_PASS, "Low pass (default)", + "low-pass"}, + {MODE_HIGH_PASS, "High pass", + "high-pass"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_limit_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_limit_dispose (GObject * object) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_CUTOFF, + g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0, + G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0, + G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next even number", + 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start); +} + +static void +gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitClass * klass) +{ + filter->cutoff = 0.0; + filter->mode = MODE_LOW_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, + gdouble * b1, gdouble * b2) +{ + gint np = filter->poles; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to convert from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either lowpass + * or highpass. + * + * For lowpass substitute z^(-1) with: + * -1 + * z - k + * ------------ + * -1 + * 1 - k * z + * + * k = sin((1-w)/2) / sin((1+w)/2) + * + * For highpass substitute z^(-1) with: + * + * -1 + * -z - k + * ------------ + * -1 + * 1 + k * z + * + * k = -cos((1+w)/2) / cos((1-w)/2) + * + */ + { + gdouble k, d; + gdouble omega = + 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_LOW_PASS) + k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0); + else + k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0); + + d = 1.0 + y1 * k - y2 * k * k; + *a0 = (x0 + k * (-x1 + k * x2)) / d; + *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d; + *a2 = (x0 * k * k - x1 * k + x2) / d; + *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d; + *b2 = (-k * k - y1 * k + y2) / d; + + if (filter->mode == MODE_HIGH_PASS) { + *a1 = -*a1; + *b1 = -*b1; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqLimit * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency"); + return; + } else if (filter->cutoff <= 0.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff is lower than zero"); + return; + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 3); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 3); + + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[2] = 1.0; + b[2] = 1.0; + + for (p = 1; p <= np / 2; p++) { + gdouble a0, a1, a2, b1, b2; + gdouble *ta = g_new0 (gdouble, np + 3); + gdouble *tb = g_new0 (gdouble, np + 3); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2); + + memcpy (ta, a, sizeof (gdouble) * (np + 3)); + memcpy (tb, b, sizeof (gdouble) * (np + 3)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 2; i < np + 3; i++) { + a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2]; + b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[2] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 2]; + b[i] = -b[i + 2]; + } + + /* Normalize to unity gain at frequency 0 for lowpass + * and frequency 0.5 for highpass */ + { + gdouble gain; + + if (filter->mode == MODE_LOW_PASS) + gain = calculate_gain (a, b, np, np, 1.0, 0.0); + else + gain = calculate_gain (a, b, np, np, -1.0, 0.0); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass", + filter->type, filter->poles, filter->cutoff, filter->ripple); + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble wc = + 2.0 * M_PI * (filter->cutoff / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble zr = cos (wc), zi = sin (wc); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->cutoff); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_CUTOFF: + GST_BASE_TRANSFORM_LOCK (filter); + filter->cutoff = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_2 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_CUTOFF: + g_value_set_float (value, filter->cutoff); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + + +static gboolean +gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiochebyshevfreqlimit.h b/gst/audiofx/audiochebyshevfreqlimit.h new file mode 100644 index 0000000000..4c87ba8eab --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqlimit.h @@ -0,0 +1,78 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * 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_CHEBYSHEV_FREQ_LIMIT_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit; +typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass; + +typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqLimitChannelCtx; + +struct _GstAudioChebyshevFreqLimit +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat cutoff; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqLimitProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqLimitChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqLimitClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_limit_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */ diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c index e6d84d243e..2c198f32b6 100644 --- a/gst/audiofx/audiofx.c +++ b/gst/audiofx/audiofx.c @@ -29,6 +29,8 @@ #include "audioinvert.h" #include "audioamplify.h" #include "audiodynamic.h" +#include "audiochebyshevfreqlimit.h" +#include "audiochebyshevfreqband.h" /* entry point to initialize the plug-in * initialize the plug-in itself @@ -48,7 +50,11 @@ plugin_init (GstPlugin * plugin) gst_element_register (plugin, "audioamplify", GST_RANK_NONE, GST_TYPE_AUDIO_AMPLIFY) && gst_element_register (plugin, "audiodynamic", GST_RANK_NONE, - GST_TYPE_AUDIO_DYNAMIC)); + GST_TYPE_AUDIO_DYNAMIC) && + gst_element_register (plugin, "audiochebyshevfreqlimit", GST_RANK_NONE, + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT) && + gst_element_register (plugin, "audiochebyshevfreqband", GST_RANK_NONE, + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 83fa465bc1..b882500e36 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -55,6 +55,8 @@ check_PROGRAMS = \ elements/alphacolor \ elements/audiopanorama \ elements/audioinvert \ + elements/audiochebyshevfreqband \ + elements/audiochebyshevfreqlimit \ elements/audioamplify \ elements/audiodynamic \ elements/avimux \ diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore index 58930486a6..7e670bb4d9 100644 --- a/tests/check/elements/.gitignore +++ b/tests/check/elements/.gitignore @@ -3,6 +3,8 @@ alphacolor audioamplify audiodynamic audioinvert +audiochebyshevfreqband +audiochebyshevfreqlimit level matroskamux cmmldec diff --git a/tests/check/elements/audiochebband.c b/tests/check/elements/audiochebband.c new file mode 100644 index 0000000000..ecacbd2b63 --- /dev/null +++ b/tests/check/elements/audiochebband.c @@ -0,0 +1,471 @@ +/* GStreamer + * + * Copyright (C) 2007 Sebastian Dröge + * + * audiochebyshevfreqband.c: Unit test for the audiochebyshevfreqband element + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public License + * as published by the Free Software Foundation; either version 2.1 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include +#include +#include + +#include + +/* 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 CAPS_STRING \ + "audio/x-raw-float, " \ + "channels = (int) 1, " \ + "rate = (int) 44100, " \ + "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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 64") + ); + +GstElement * +setup_audiochebyshevfreqband () +{ + GstElement *audiochebyshevfreqband; + + GST_DEBUG ("setup_audiochebyshevfreqband"); + audiochebyshevfreqband = gst_check_setup_element ("audiochebyshevfreqband"); + mysrcpad = + gst_check_setup_src_pad (audiochebyshevfreqband, &srctemplate, NULL); + mysinkpad = + gst_check_setup_sink_pad (audiochebyshevfreqband, &sinktemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); + + return audiochebyshevfreqband; +} + +void +cleanup_audiochebyshevfreqband (GstElement * audiochebyshevfreqband) +{ + GST_DEBUG ("cleanup_audiochebyshevfreqband"); + + 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 (audiochebyshevfreqband); + gst_check_teardown_sink_pad (audiochebyshevfreqband); + gst_check_teardown_element (audiochebyshevfreqband); +} + +/* Test if data containing only one frequency component + * at 0 is erased with bandpass mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_bp_0hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandpass */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at band center is preserved with bandpass mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_bp_11025hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandpass */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 4) { + in[i] = 0.0; + in[i + 1] = 1.0; + in[i + 2] = 0.0; + in[i + 3] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms >= 0.6); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is erased with bandpass mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_bp_22050hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandpass */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at 0 is preserved with bandreject mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_br_0hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandreject */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at band center is erased with bandreject mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_br_11025hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandreject */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 4) { + in[i] = 0.0; + in[i + 1] = 1.0; + in[i + 2] = 0.0; + in[i + 3] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is preserved with bandreject mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_br_22050hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandreject */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +Suite * +audiochebyshevfreqband_suite (void) +{ + Suite *s = suite_create ("audiochebyshevfreqband"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + tcase_add_test (tc_chain, test_bp_0hz); + tcase_add_test (tc_chain, test_bp_11025hz); + tcase_add_test (tc_chain, test_bp_22050hz); + tcase_add_test (tc_chain, test_br_0hz); + tcase_add_test (tc_chain, test_br_11025hz); + tcase_add_test (tc_chain, test_br_22050hz); + + return s; +} + +int +main (int argc, char **argv) +{ + int nf; + + Suite *s = audiochebyshevfreqband_suite (); + SRunner *sr = srunner_create (s); + + gst_check_init (&argc, &argv); + + srunner_run_all (sr, CK_NORMAL); + nf = srunner_ntests_failed (sr); + srunner_free (sr); + + return nf; +} diff --git a/tests/check/elements/audiocheblimit.c b/tests/check/elements/audiocheblimit.c new file mode 100644 index 0000000000..35a21e51bf --- /dev/null +++ b/tests/check/elements/audiocheblimit.c @@ -0,0 +1,341 @@ +/* GStreamer + * + * Copyright (C) 2007 Sebastian Dröge + * + * audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public License + * as published by the Free Software Foundation; either version 2.1 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include +#include +#include + +#include + +/* 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 CAPS_STRING \ + "audio/x-raw-float, " \ + "channels = (int) 1, " \ + "rate = (int) 44100, " \ + "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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 64") + ); + +GstElement * +setup_audiochebyshevfreqlimit () +{ + GstElement *audiochebyshevfreqlimit; + + GST_DEBUG ("setup_audiochebyshevfreqlimit"); + audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit"); + mysrcpad = + gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL); + mysinkpad = + gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); + + return audiochebyshevfreqlimit; +} + +void +cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit) +{ + GST_DEBUG ("cleanup_audiochebyshevfreqlimit"); + + 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 (audiochebyshevfreqlimit); + gst_check_teardown_sink_pad (audiochebyshevfreqlimit); + gst_check_teardown_element (audiochebyshevfreqlimit); +} + +/* Test if data containing only one frequency component + * at 0 is preserved with lowpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_lp_0hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to lowpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is erased with lowpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_lp_22050hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to lowpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at 0 is erased with highpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_hp_0hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to highpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is preserved with highpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_hp_22050hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to highpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +Suite * +audiochebyshevfreqlimit_suite (void) +{ + Suite *s = suite_create ("audiochebyshevfreqlimit"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + tcase_add_test (tc_chain, test_lp_0hz); + tcase_add_test (tc_chain, test_lp_22050hz); + tcase_add_test (tc_chain, test_hp_0hz); + tcase_add_test (tc_chain, test_hp_22050hz); + + return s; +} + +int +main (int argc, char **argv) +{ + int nf; + + Suite *s = audiochebyshevfreqlimit_suite (); + SRunner *sr = srunner_create (s); + + gst_check_init (&argc, &argv); + + srunner_run_all (sr, CK_NORMAL); + nf = srunner_ntests_failed (sr); + srunner_free (sr); + + return nf; +} diff --git a/tests/check/elements/audiochebyshevfreqband.c b/tests/check/elements/audiochebyshevfreqband.c new file mode 100644 index 0000000000..ecacbd2b63 --- /dev/null +++ b/tests/check/elements/audiochebyshevfreqband.c @@ -0,0 +1,471 @@ +/* GStreamer + * + * Copyright (C) 2007 Sebastian Dröge + * + * audiochebyshevfreqband.c: Unit test for the audiochebyshevfreqband element + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public License + * as published by the Free Software Foundation; either version 2.1 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include +#include +#include + +#include + +/* 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 CAPS_STRING \ + "audio/x-raw-float, " \ + "channels = (int) 1, " \ + "rate = (int) 44100, " \ + "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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 64") + ); + +GstElement * +setup_audiochebyshevfreqband () +{ + GstElement *audiochebyshevfreqband; + + GST_DEBUG ("setup_audiochebyshevfreqband"); + audiochebyshevfreqband = gst_check_setup_element ("audiochebyshevfreqband"); + mysrcpad = + gst_check_setup_src_pad (audiochebyshevfreqband, &srctemplate, NULL); + mysinkpad = + gst_check_setup_sink_pad (audiochebyshevfreqband, &sinktemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); + + return audiochebyshevfreqband; +} + +void +cleanup_audiochebyshevfreqband (GstElement * audiochebyshevfreqband) +{ + GST_DEBUG ("cleanup_audiochebyshevfreqband"); + + 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 (audiochebyshevfreqband); + gst_check_teardown_sink_pad (audiochebyshevfreqband); + gst_check_teardown_element (audiochebyshevfreqband); +} + +/* Test if data containing only one frequency component + * at 0 is erased with bandpass mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_bp_0hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandpass */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at band center is preserved with bandpass mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_bp_11025hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandpass */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 4) { + in[i] = 0.0; + in[i + 1] = 1.0; + in[i + 2] = 0.0; + in[i + 3] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms >= 0.6); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is erased with bandpass mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_bp_22050hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandpass */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at 0 is preserved with bandreject mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_br_0hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandreject */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at band center is erased with bandreject mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_br_11025hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandreject */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 4) { + in[i] = 0.0; + in[i + 1] = 1.0; + in[i + 2] = 0.0; + in[i + 3] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is preserved with bandreject mode and a + * 2000Hz frequency band around rate/4 */ +GST_START_TEST (test_br_22050hz) +{ + GstElement *audiochebyshevfreqband; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqband = setup_audiochebyshevfreqband (); + /* Set to bandreject */ + g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqband, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency", + 44100 / 4.0 - 1000, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency", + 44100 / 4.0 + 1000, NULL); + inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 1024; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 1024; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 1024.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqband (audiochebyshevfreqband); +} + +GST_END_TEST; + +Suite * +audiochebyshevfreqband_suite (void) +{ + Suite *s = suite_create ("audiochebyshevfreqband"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + tcase_add_test (tc_chain, test_bp_0hz); + tcase_add_test (tc_chain, test_bp_11025hz); + tcase_add_test (tc_chain, test_bp_22050hz); + tcase_add_test (tc_chain, test_br_0hz); + tcase_add_test (tc_chain, test_br_11025hz); + tcase_add_test (tc_chain, test_br_22050hz); + + return s; +} + +int +main (int argc, char **argv) +{ + int nf; + + Suite *s = audiochebyshevfreqband_suite (); + SRunner *sr = srunner_create (s); + + gst_check_init (&argc, &argv); + + srunner_run_all (sr, CK_NORMAL); + nf = srunner_ntests_failed (sr); + srunner_free (sr); + + return nf; +} diff --git a/tests/check/elements/audiochebyshevfreqlimit.c b/tests/check/elements/audiochebyshevfreqlimit.c new file mode 100644 index 0000000000..35a21e51bf --- /dev/null +++ b/tests/check/elements/audiochebyshevfreqlimit.c @@ -0,0 +1,341 @@ +/* GStreamer + * + * Copyright (C) 2007 Sebastian Dröge + * + * audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public License + * as published by the Free Software Foundation; either version 2.1 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include +#include +#include + +#include + +/* 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 CAPS_STRING \ + "audio/x-raw-float, " \ + "channels = (int) 1, " \ + "rate = (int) 44100, " \ + "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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 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, " + "rate = (int) 44100, " + "endianness = (int) BYTE_ORDER, " "width = (int) 64") + ); + +GstElement * +setup_audiochebyshevfreqlimit () +{ + GstElement *audiochebyshevfreqlimit; + + GST_DEBUG ("setup_audiochebyshevfreqlimit"); + audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit"); + mysrcpad = + gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL); + mysinkpad = + gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); + + return audiochebyshevfreqlimit; +} + +void +cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit) +{ + GST_DEBUG ("cleanup_audiochebyshevfreqlimit"); + + 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 (audiochebyshevfreqlimit); + gst_check_teardown_sink_pad (audiochebyshevfreqlimit); + gst_check_teardown_element (audiochebyshevfreqlimit); +} + +/* Test if data containing only one frequency component + * at 0 is preserved with lowpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_lp_0hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to lowpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is erased with lowpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_lp_22050hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to lowpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at 0 is erased with highpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_hp_0hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to highpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i++) + in[i] = 1.0; + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms <= 0.1); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +/* Test if data containing only one frequency component + * at rate/2 is preserved with highpass mode and a cutoff + * at rate/4 */ +GST_START_TEST (test_hp_22050hz) +{ + GstElement *audiochebyshevfreqlimit; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gdouble *in, *res, rms; + gint i; + + audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); + /* Set to highpass */ + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); + + fail_unless (gst_element_set_state (audiochebyshevfreqlimit, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, + NULL); + inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); + in = (gdouble *) GST_BUFFER_DATA (inbuffer); + for (i = 0; i < 128; i += 2) { + in[i] = 1.0; + in[i + 1] = -1.0; + } + + caps = gst_caps_from_string (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); + /* ... and puts a new buffer on the global list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gdouble *) GST_BUFFER_DATA (outbuffer); + + rms = 0.0; + for (i = 0; i < 128; i++) + rms += res[i] * res[i]; + rms = sqrt (rms / 128.0); + fail_unless (rms >= 0.9); + + /* cleanup */ + cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); +} + +GST_END_TEST; + +Suite * +audiochebyshevfreqlimit_suite (void) +{ + Suite *s = suite_create ("audiochebyshevfreqlimit"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + tcase_add_test (tc_chain, test_lp_0hz); + tcase_add_test (tc_chain, test_lp_22050hz); + tcase_add_test (tc_chain, test_hp_0hz); + tcase_add_test (tc_chain, test_hp_22050hz); + + return s; +} + +int +main (int argc, char **argv) +{ + int nf; + + Suite *s = audiochebyshevfreqlimit_suite (); + SRunner *sr = srunner_create (s); + + gst_check_init (&argc, &argv); + + srunner_run_all (sr, CK_NORMAL); + nf = srunner_ntests_failed (sr); + srunner_free (sr); + + return nf; +}